Adding upstream version 115.8.0esr.upstream/115.8.0esr

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

684 files changed, 62289 insertions, 0 deletions

diff --git a/third_party/libwebrtc/test/BUILD.gn b/third_party/libwebrtc/test/BUILD.gn
new file mode 100644
index 0000000000..04a718c411
--- /dev/null
+++ b/third_party/libwebrtc/test/BUILD.gn
@@ -0,0 +1,1263 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//build/config/ui.gni")
+import("//third_party/google_benchmark/buildconfig.gni")
+import("../webrtc.gni")
+if (is_android) {
+  import("//build/config/android/rules.gni")
+}
+
+if (!build_with_chromium) {
+  group("test") {
+    testonly = true
+
+    deps = [
+      ":copy_to_file_audio_capturer",
+      ":rtp_test_utils",
+      ":test_common",
+      ":test_renderer",
+      ":test_support",
+      ":video_test_common",
+    ]
+
+    if (rtc_include_tests) {
+      deps += [
+        ":test_main",
+        ":test_support_unittests",
+        "pc/e2e",
+      ]
+    }
+  }
+}
+
+rtc_library("frame_generator_impl") {
+  visibility = [
+    ":*",
+    "../api:create_frame_generator",
+  ]
+  testonly = true
+  sources = [
+    "frame_generator.cc",
+    "frame_generator.h",
+    "testsupport/ivf_video_frame_generator.cc",
+    "testsupport/ivf_video_frame_generator.h",
+  ]
+  deps = [
+    ":frame_utils",
+    "../api:frame_generator_api",
+    "../api:scoped_refptr",
+    "../api:sequence_checker",
+    "../api/video:encoded_image",
+    "../api/video:video_frame",
+    "../api/video:video_frame_i010",
+    "../api/video:video_rtp_headers",
+    "../api/video_codecs:video_codecs_api",
+    "../common_video",
+    "../media:media_constants",
+    "../media:rtc_media_base",
+    "../modules/video_coding:video_codec_interface",
+    "../modules/video_coding:video_coding_utility",
+    "../modules/video_coding:webrtc_h264",
+    "../modules/video_coding:webrtc_vp8",
+    "../modules/video_coding:webrtc_vp9",
+    "../rtc_base:checks",
+    "../rtc_base:criticalsection",
+    "../rtc_base:logging",
+    "../rtc_base:random",
+    "../rtc_base:rtc_event",
+    "../rtc_base/synchronization:mutex",
+    "../rtc_base/system:file_wrapper",
+    "../system_wrappers",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("frame_utils") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "frame_utils.cc",
+    "frame_utils.h",
+  ]
+  deps = [
+    "../api:scoped_refptr",
+    "../api/video:video_frame",
+  ]
+}
+
+rtc_library("video_test_common") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "fake_texture_frame.cc",
+    "fake_texture_frame.h",
+    "fake_videorenderer.h",
+    "frame_forwarder.cc",
+    "frame_forwarder.h",
+    "frame_generator_capturer.cc",
+    "frame_generator_capturer.h",
+    "mappable_native_buffer.cc",
+    "mappable_native_buffer.h",
+    "test_video_capturer.cc",
+    "test_video_capturer.h",
+    "video_codec_settings.h",
+    "video_decoder_proxy_factory.h",
+    "video_encoder_nullable_proxy_factory.h",
+    "video_encoder_proxy_factory.h",
+  ]
+
+  deps = [
+    ":fileutils",
+    ":frame_utils",
+    ":scoped_key_value_config",
+    "../api:array_view",
+    "../api:create_frame_generator",
+    "../api:frame_generator_api",
+    "../api:scoped_refptr",
+    "../api/task_queue",
+    "../api/video:video_frame",
+    "../api/video:video_rtp_headers",
+    "../api/video_codecs:video_codecs_api",
+    "../common_video",
+    "../media:rtc_media_base",
+    "../rtc_base:checks",
+    "../rtc_base:criticalsection",
+    "../rtc_base:logging",
+    "../rtc_base:refcount",
+    "../rtc_base:rtc_task_queue",
+    "../rtc_base:timeutils",
+    "../rtc_base/synchronization:mutex",
+    "../rtc_base/task_utils:repeating_task",
+    "../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/strings",
+  ]
+}
+
+rtc_library("audio_test_common") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "audio_decoder_proxy_factory.h",
+    "function_audio_decoder_factory.h",
+  ]
+  deps = [
+    "../api/audio_codecs:audio_codecs_api",
+    "../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+}
+
+if (!build_with_mozilla) {
+if (!build_with_chromium) {
+  if (is_mac || is_ios) {
+    rtc_library("video_test_mac") {
+      testonly = true
+      sources = [
+        "mac_capturer.h",
+        "mac_capturer.mm",
+      ]
+      deps = [
+        ":video_test_common",
+        "../api:libjingle_peerconnection_api",
+        "../api:media_stream_interface",
+        "../api:scoped_refptr",
+        "../modules/video_capture:video_capture_module",
+        "../rtc_base:threading",
+        "../sdk:base_objc",
+        "../sdk:native_api",
+        "../sdk:native_video",
+        "../sdk:videocapture_objc",
+      ]
+    }
+  }
+
+  rtc_library("platform_video_capturer") {
+    testonly = true
+    sources = [
+      "platform_video_capturer.cc",
+      "platform_video_capturer.h",
+    ]
+    deps = [ ":video_test_common" ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+    if (is_mac || is_ios) {
+      deps += [ ":video_test_mac" ]
+    } else {
+      sources += [
+        "vcm_capturer.cc",
+        "vcm_capturer.h",
+      ]
+      deps += [
+        ":scoped_key_value_config",
+        "../api:scoped_refptr",
+        "../modules/video_capture:video_capture_module",
+        "../rtc_base:checks",
+        "../rtc_base:logging",
+      ]
+    }
+  }
+}
+}
+
+rtc_library("rtp_test_utils") {
+  if (build_with_mozilla) {
+  sources = []
+  } else {
+  testonly = true
+  sources = [
+    "rtcp_packet_parser.cc",
+    "rtcp_packet_parser.h",
+    "rtp_file_reader.cc",
+    "rtp_file_reader.h",
+    "rtp_file_writer.cc",
+    "rtp_file_writer.h",
+  ]
+  }
+
+  deps = [
+    "../api:array_view",
+    "../api:rtp_parameters",
+    "../modules/rtp_rtcp",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
+    "../rtc_base:checks",
+    "../rtc_base:criticalsection",
+    "../rtc_base:logging",
+    "../rtc_base:macromagic",
+    "../rtc_base/synchronization:mutex",
+    "../rtc_base/system:arch",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("field_trial") {
+  testonly = true
+  visibility = [ "*" ]
+  sources = [
+    "field_trial.cc",
+    "field_trial.h",
+  ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  deps = [
+    "../rtc_base:checks",
+    "../system_wrappers:field_trial",
+  ]
+}
+
+rtc_library("explicit_key_value_config") {
+  sources = [
+    "explicit_key_value_config.cc",
+    "explicit_key_value_config.h",
+  ]
+
+  deps = [
+    "../api:field_trials_registry",
+    "../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+}
+
+rtc_library("scoped_key_value_config") {
+  testonly = true
+  visibility = [ "*" ]
+  sources = [
+    "scoped_key_value_config.cc",
+    "scoped_key_value_config.h",
+  ]
+
+  deps = [
+    ":field_trial",
+    "../api:field_trials_registry",
+    "../rtc_base:checks",
+    "../system_wrappers:field_trial",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+}
+
+rtc_library("perf_test") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "testsupport/perf_test.cc",
+    "testsupport/perf_test.h",
+    "testsupport/perf_test_histogram_writer.h",
+    "testsupport/perf_test_result_writer.h",
+  ]
+  deps = [
+    "../api:array_view",
+    "../api/numerics",
+    "../rtc_base:checks",
+    "../rtc_base:criticalsection",
+    "../rtc_base:logging",
+    "../rtc_base:rtc_numerics",
+    "../rtc_base:stringutils",
+    "../rtc_base/synchronization:mutex",
+    "../test:fileutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  if (rtc_enable_protobuf) {
+    sources += [ "testsupport/perf_test_histogram_writer.cc" ]
+    deps += [
+      "//third_party/catapult/tracing/tracing:histogram",
+      "//third_party/catapult/tracing/tracing:reserved_infos",
+    ]
+  } else {
+    sources += [ "testsupport/perf_test_histogram_writer_no_protobuf.cc" ]
+  }
+}
+
+if (is_ios) {
+  rtc_library("test_support_objc") {
+    testonly = true
+    visibility = [
+      ":google_test_runner_objc",
+      ":test_support",
+    ]
+    sources = [
+      "ios/coverage_util_ios.h",
+      "ios/coverage_util_ios.mm",
+      "ios/google_test_runner_delegate.h",
+      "ios/test_support.h",
+      "ios/test_support.mm",
+    ]
+    deps = [
+      ":perf_test",
+      "../api/test/metrics:chrome_perf_dashboard_metrics_exporter",
+      "../api/test/metrics:global_metrics_logger_and_exporter",
+      "../api/test/metrics:metrics_exporter",
+      "../api/test/metrics:metrics_set_proto_file_exporter",
+      "../api/test/metrics:print_result_proxy_metrics_exporter",
+      "../api/test/metrics:stdout_metrics_exporter",
+      "../sdk:helpers_objc",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    configs += [ ":test_support_objc_config" ]
+  }
+
+  rtc_library("google_test_runner_objc") {
+    testonly = true
+    visibility = [ "*" ]
+    sources = [ "ios/google_test_runner.mm" ]
+    deps = [ ":test_support_objc" ]
+    configs += [ "//build/config/ios:xctest_config" ]
+    frameworks = [ "UIKit.framework" ]
+  }
+
+  config("test_support_objc_config") {
+    defines = []
+
+    if (use_clang_coverage) {
+      defines += [ "WEBRTC_IOS_ENABLE_COVERAGE" ]
+    }
+  }
+}
+
+config("suppress_warning_4373") {
+  if (is_win) {
+    cflags = [
+      # MSVC has a bug which generates this warning when using mocks; see the
+      # section on warning 4373 in he googlemock FAQ. This warning isn't the
+      # least relevant for us, anyway.
+      "/wd4373",
+    ]
+  }
+}
+
+config("test_main_direct_config") {
+  visibility = [ ":*" ]
+  defines = [ "WEBRTC_UNIT_TEST" ]
+}
+rtc_source_set("test_support") {
+  visibility = [ "*" ]
+  testonly = true
+
+  all_dependent_configs = [
+    ":suppress_warning_4373",
+    "//third_party/googletest:gmock_config",
+    "//third_party/googletest:gtest_config",
+  ]
+
+  sources = [
+    "gmock.h",
+    "gtest.h",
+  ]
+
+  public_deps = []  # no-presubmit-check TODO(webrtc:8603)
+  if (is_ios) {
+    public_deps +=  # no-presubmit-check TODO(webrtc:8603)
+        [ ":test_support_objc" ]
+  }
+
+  public_configs = [ ":test_main_direct_config" ]
+  deps = [
+    "../rtc_base:ignore_wundef",
+    "//testing/gmock",
+    "//testing/gtest",
+  ]
+}
+
+rtc_library("fixed_fps_video_frame_writer_adapter") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "testsupport/fixed_fps_video_frame_writer_adapter.cc",
+    "testsupport/fixed_fps_video_frame_writer_adapter.h",
+  ]
+  deps = [
+    ":video_test_support",
+    "../api/test/video:video_frame_writer",
+    "../api/units:time_delta",
+    "../api/video:video_frame",
+    "../rtc_base:checks",
+    "../system_wrappers",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("video_test_support") {
+  testonly = true
+
+  sources = [
+    "testsupport/frame_reader.h",
+    "testsupport/frame_writer.h",
+    "testsupport/mock/mock_frame_reader.h",
+    "testsupport/video_frame_writer.cc",
+    "testsupport/video_frame_writer.h",
+    "testsupport/y4m_frame_reader.cc",
+    "testsupport/y4m_frame_writer.cc",
+    "testsupport/yuv_frame_reader.cc",
+    "testsupport/yuv_frame_writer.cc",
+  ]
+
+  deps = [
+    ":fileutils",
+    ":frame_utils",
+    ":test_support",
+    ":video_test_common",
+    "../api:scoped_refptr",
+    "../api:sequence_checker",
+    "../api/test/video:video_frame_writer",
+    "../api/video:encoded_image",
+    "../api/video:resolution",
+    "../api/video:video_frame",
+    "../api/video_codecs:video_codecs_api",
+    "../common_video",
+    "../media:rtc_media_base",
+    "../modules/video_coding:video_codec_interface",
+    "../modules/video_coding:video_coding_utility",
+    "../modules/video_coding:webrtc_h264",
+    "../modules/video_coding:webrtc_vp8",
+    "../modules/video_coding:webrtc_vp9",
+    "../rtc_base:buffer",
+    "../rtc_base:checks",
+    "../rtc_base:criticalsection",
+    "../rtc_base:logging",
+    "../rtc_base:rtc_event",
+    "../rtc_base:stringutils",
+    "../rtc_base/system:file_wrapper",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+
+  if (!is_ios) {
+    if (!build_with_mozilla) {
+    deps += [ "//third_party:jpeg" ]
+    }
+    sources += [ "testsupport/jpeg_frame_writer.cc" ]
+  } else {
+    sources += [ "testsupport/jpeg_frame_writer_ios.cc" ]
+  }
+}
+
+if (rtc_include_tests && enable_google_benchmarks) {
+  rtc_library("benchmark_main") {
+    testonly = true
+    sources = [ "benchmark_main.cc" ]
+
+    deps = [ "//third_party/google_benchmark" ]
+  }
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  rtc_library("resources_dir_flag") {
+    testonly = true
+    visibility = [ "*" ]
+    sources = [
+      "testsupport/resources_dir_flag.cc",
+      "testsupport/resources_dir_flag.h",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/flags:flag" ]
+  }
+
+  rtc_library("test_flags") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "test_flags.cc",
+      "test_flags.h",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/flags:flag" ]
+  }
+
+  rtc_library("test_main_lib") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "test_main_lib.cc",
+      "test_main_lib.h",
+    ]
+
+    deps = [
+      ":field_trial",
+      ":perf_test",
+      ":resources_dir_flag",
+      ":test_flags",
+      ":test_support",
+      "../api/test/metrics:chrome_perf_dashboard_metrics_exporter",
+      "../api/test/metrics:global_metrics_logger_and_exporter",
+      "../api/test/metrics:metrics_exporter",
+      "../api/test/metrics:metrics_set_proto_file_exporter",
+      "../api/test/metrics:print_result_proxy_metrics_exporter",
+      "../api/test/metrics:stdout_metrics_exporter",
+      "../rtc_base:checks",
+      "../rtc_base:event_tracer",
+      "../rtc_base:logging",
+      "../rtc_base:ssl",
+      "../rtc_base:threading",
+      "../system_wrappers:field_trial",
+      "../system_wrappers:metrics",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/strings:strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+
+    if (is_win) {
+      deps += [ "../rtc_base:win32_socket_init" ]
+    }
+  }
+
+  rtc_library("test_main") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "test_main.cc" ]
+
+    deps = [
+      ":test_main_lib",
+      ":test_support",
+    ]
+
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/debugging:failure_signal_handler",
+      "//third_party/abseil-cpp/absl/debugging:symbolize",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+
+  rtc_library("test_support_test_artifacts") {
+    testonly = true
+    sources = [
+      "testsupport/test_artifacts.cc",
+      "testsupport/test_artifacts.h",
+    ]
+    deps = [
+      ":fileutils",
+      "../rtc_base:logging",
+      "../rtc_base/system:file_wrapper",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+    ]
+  }
+
+  test_support_unittests_resources = [
+    "../resources/foreman_cif_short.yuv",
+    "../resources/video_coding/frame-ethernet-ii.pcap",
+    "../resources/video_coding/frame-loopback.pcap",
+    "../resources/video_coding/pltype103.rtp",
+    "../resources/video_coding/pltype103_header_only.rtp",
+    "../resources/video_coding/ssrcs-2.pcap",
+    "../resources/video_coding/ssrcs-3.pcap",
+  ]
+
+  if (is_ios) {
+    bundle_data("test_support_unittests_bundle_data") {
+      testonly = true
+      sources = test_support_unittests_resources
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+  }
+
+  rtc_library("fixed_fps_video_frame_writer_adapter_test") {
+    testonly = true
+    sources = [ "testsupport/fixed_fps_video_frame_writer_adapter_test.cc" ]
+    deps = [
+      ":fixed_fps_video_frame_writer_adapter",
+      ":test_support",
+      ":video_test_support",
+      "../api/units:time_delta",
+      "../api/units:timestamp",
+      "../api/video:video_frame",
+      "../rtc_base/synchronization:mutex",
+      "time_controller",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  rtc_test("test_support_unittests") {
+    deps = [
+      ":call_config_utils",
+      ":copy_to_file_audio_capturer_unittest",
+      ":direct_transport",
+      ":fake_video_codecs",
+      ":fileutils",
+      ":fileutils_unittests",
+      ":fixed_fps_video_frame_writer_adapter_test",
+      ":frame_generator_impl",
+      ":perf_test",
+      ":rtc_expect_death",
+      ":rtp_test_utils",
+      ":run_loop",
+      ":scoped_key_value_config",
+      ":test_main",
+      ":test_support",
+      ":test_support_test_artifacts",
+      ":video_test_common",
+      ":video_test_support",
+      "../api:array_view",
+      "../api:create_frame_generator",
+      "../api:create_simulcast_test_fixture_api",
+      "../api:frame_generator_api",
+      "../api:scoped_refptr",
+      "../api:simulcast_test_fixture_api",
+      "../api/task_queue:task_queue_test",
+      "../api/test/video:function_video_factory",
+      "../api/test/video:video_frame_writer",
+      "../api/units:time_delta",
+      "../api/video:encoded_image",
+      "../api/video:video_frame",
+      "../api/video_codecs:video_codecs_api",
+      "../call:video_stream_api",
+      "../common_video",
+      "../media:codec",
+      "../media:media_constants",
+      "../media:rtc_media_base",
+      "../modules/rtp_rtcp:rtp_rtcp_format",
+      "../modules/video_coding:simulcast_test_fixture_impl",
+      "../modules/video_coding:video_codec_interface",
+      "../modules/video_coding:video_coding_utility",
+      "../modules/video_coding:webrtc_h264",
+      "../modules/video_coding:webrtc_vp8",
+      "../modules/video_coding:webrtc_vp9",
+      "../rtc_base:criticalsection",
+      "../rtc_base:rtc_event",
+      "../rtc_base:rtc_task_queue",
+      "../rtc_base/synchronization:mutex",
+      "../rtc_base/system:file_wrapper",
+      "pc/e2e:e2e_unittests",
+      "pc/e2e/analyzer/video:video_analyzer_unittests",
+      "peer_scenario/tests",
+      "scenario:scenario_unittests",
+      "time_controller:time_controller",
+      "time_controller:time_controller_unittests",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+    sources = [
+      "call_config_utils_unittest.cc",
+      "direct_transport_unittest.cc",
+      "fake_vp8_encoder_unittest.cc",
+      "frame_generator_capturer_unittest.cc",
+      "frame_generator_unittest.cc",
+      "rtp_file_reader_unittest.cc",
+      "rtp_file_writer_unittest.cc",
+      "run_loop_unittest.cc",
+      "testsupport/ivf_video_frame_generator_unittest.cc",
+      "testsupport/perf_test_unittest.cc",
+      "testsupport/test_artifacts_unittest.cc",
+      "testsupport/video_frame_writer_unittest.cc",
+      "testsupport/y4m_frame_reader_unittest.cc",
+      "testsupport/y4m_frame_writer_unittest.cc",
+      "testsupport/yuv_frame_reader_unittest.cc",
+      "testsupport/yuv_frame_writer_unittest.cc",
+    ]
+
+    if (rtc_enable_protobuf) {
+      sources += [ "testsupport/perf_test_histogram_writer_unittest.cc" ]
+      deps += [ "//third_party/catapult/tracing/tracing:histogram" ]
+    }
+
+    data = test_support_unittests_resources
+    if (is_android) {
+      deps += [ "//testing/android/native_test:native_test_support" ]
+      shard_timeout = 900
+    }
+
+    if (is_ios) {
+      deps += [ ":test_support_unittests_bundle_data" ]
+    }
+
+    if (!is_android) {
+      # This is needed in order to avoid:
+      # undefined symbol: webrtc::videocapturemodule::VideoCaptureImpl::Create
+      deps += [ "../modules/video_capture:video_capture_internal_impl" ]
+    }
+  }
+}
+
+if (is_ios) {
+  rtc_library("fileutils_ios_objc") {
+    visibility = [
+      ":fileutils",
+      ":fileutils_override_impl",
+    ]
+    sources = [
+      "testsupport/ios_file_utils.h",
+      "testsupport/ios_file_utils.mm",
+    ]
+    deps = [
+      "../rtc_base:checks",
+      "../sdk:helpers_objc",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+  }
+}
+
+if (is_mac) {
+  rtc_library("fileutils_mac_objc") {
+    visibility = [
+      ":fileutils",
+      ":fileutils_override_impl",
+    ]
+    sources = [
+      "testsupport/mac_file_utils.h",
+      "testsupport/mac_file_utils.mm",
+    ]
+    deps = [ "../rtc_base:checks" ]
+  }
+}
+
+rtc_library("fileutils") {
+  testonly = true
+  visibility = [ "*" ]
+  sources = [
+    "testsupport/file_utils.cc",
+    "testsupport/file_utils.h",
+  ]
+  deps = [
+    ":fileutils_override_api",
+    ":fileutils_override_impl",
+    "../rtc_base:checks",
+    "../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  if (is_ios) {
+    deps += [ ":fileutils_ios_objc" ]
+  }
+  if (is_mac) {
+    deps += [ ":fileutils_mac_objc" ]
+  }
+  if (is_win) {
+    deps += [ "../rtc_base:win32" ]
+  }
+}
+
+# We separate header into own target to make it possible for downstream
+# projects to override implementation.
+rtc_source_set("fileutils_override_api") {
+  testonly = true
+  sources = [ "testsupport/file_utils_override.h" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+}
+
+rtc_library("fileutils_override_impl") {
+  testonly = true
+  visibility = [ ":fileutils" ]
+  sources = [ "testsupport/file_utils_override.cc" ]
+  deps = [
+    ":fileutils_override_api",
+    "../rtc_base:checks",
+    "../rtc_base:macromagic",
+    "../rtc_base:stringutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+  if (is_ios) {
+    deps += [ ":fileutils_ios_objc" ]
+  }
+  if (is_mac) {
+    deps += [ ":fileutils_mac_objc" ]
+  }
+  if (is_win) {
+    deps += [ "../rtc_base:win32" ]
+  }
+}
+
+rtc_source_set("run_test") {
+  testonly = true
+  if (is_mac) {
+    public_deps = [ ":run_test_objc" ]  # no-presubmit-check TODO(webrtc:8603)
+  } else {
+    public_deps =  # no-presubmit-check TODO(webrtc:8603)
+        [ ":run_test_generic" ]
+  }
+}
+
+rtc_source_set("run_test_interface") {
+  sources = [ "run_test.h" ]
+}
+
+if (is_mac) {
+  rtc_library("run_test_objc") {
+    testonly = true
+    visibility = [ ":run_test" ]
+    sources = [ "mac/run_test.mm" ]
+    deps = [ ":run_test_interface" ]
+  }
+}
+
+rtc_library("run_test_generic") {
+  testonly = true
+  visibility = [ ":run_test" ]
+  sources = [ "run_test.cc" ]
+  deps = [ ":run_test_interface" ]
+}
+
+rtc_library("fileutils_unittests") {
+  testonly = true
+  visibility = [ ":*" ]  # Only targets in this file can depend on this.
+  sources = [ "testsupport/file_utils_unittest.cc" ]
+  deps = [
+    ":fileutils",
+    ":test_support",
+    "../rtc_base:checks",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("direct_transport") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "direct_transport.cc",
+    "direct_transport.h",
+  ]
+  deps = [
+    "../api:rtp_parameters",
+    "../api:sequence_checker",
+    "../api:simulated_network_api",
+    "../api:transport_api",
+    "../api/task_queue",
+    "../api/units:time_delta",
+    "../call:call_interfaces",
+    "../call:simulated_packet_receiver",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
+    "../rtc_base:checks",
+    "../rtc_base:macromagic",
+    "../rtc_base:timeutils",
+    "../rtc_base/synchronization:mutex",
+    "../rtc_base/task_utils:repeating_task",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+  public_deps =  # no-presubmit-check TODO(webrtc:8603)
+      [ "../call:fake_network" ]
+}
+
+rtc_library("fake_video_codecs") {
+  allow_poison = [ "software_video_codecs" ]
+  visibility = [ "*" ]
+  sources = [
+    "configurable_frame_size_encoder.cc",
+    "configurable_frame_size_encoder.h",
+    "fake_decoder.cc",
+    "fake_decoder.h",
+    "fake_encoder.cc",
+    "fake_encoder.h",
+    "fake_vp8_decoder.cc",
+    "fake_vp8_decoder.h",
+    "fake_vp8_encoder.cc",
+    "fake_vp8_encoder.h",
+  ]
+  deps = [
+    "../api:fec_controller_api",
+    "../api:scoped_refptr",
+    "../api:sequence_checker",
+    "../api/task_queue",
+    "../api/video:encoded_image",
+    "../api/video:video_bitrate_allocation",
+    "../api/video:video_frame",
+    "../api/video:video_rtp_headers",
+    "../api/video_codecs:video_codecs_api",
+    "../api/video_codecs:vp8_temporal_layers_factory",
+    "../modules/video_coding:codec_globals_headers",
+    "../modules/video_coding:video_codec_interface",
+    "../modules/video_coding:video_coding_utility",
+    "../rtc_base:checks",
+    "../rtc_base:criticalsection",
+    "../rtc_base:macromagic",
+    "../rtc_base:rtc_task_queue",
+    "../rtc_base:timeutils",
+    "../rtc_base/synchronization:mutex",
+    "../system_wrappers",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("null_transport") {
+  testonly = true
+  sources = [
+    "null_transport.cc",
+    "null_transport.h",
+  ]
+  deps = [ "../api:transport_api" ]
+}
+
+rtc_library("encoder_settings") {
+  testonly = true
+  sources = [
+    "encoder_settings.cc",
+    "encoder_settings.h",
+  ]
+  deps = [
+    "../api:scoped_refptr",
+    "../api/video_codecs:video_codecs_api",
+    "../call:rtp_interfaces",
+    "../call:video_stream_api",
+    "../rtc_base:checks",
+    "../rtc_base:refcount",
+    "../video/config:encoder_config",
+  ]
+}
+
+rtc_library("rtc_expect_death") {
+  testonly = true
+  sources = [ "testsupport/rtc_expect_death.h" ]
+  deps = [ ":test_support" ]
+}
+
+rtc_library("run_loop") {
+  testonly = true
+  sources = [
+    "run_loop.cc",
+    "run_loop.h",
+  ]
+  deps = [
+    "../api/task_queue",
+    "../rtc_base:threading",
+    "../rtc_base:timeutils",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/functional:any_invocable" ]
+}
+
+rtc_library("test_common") {
+  testonly = true
+  sources = [
+    "call_test.cc",
+    "call_test.h",
+    "drifting_clock.cc",
+    "drifting_clock.h",
+    "layer_filtering_transport.cc",
+    "layer_filtering_transport.h",
+    "rtp_rtcp_observer.h",
+  ]
+
+  deps = [
+    ":direct_transport",
+    ":encoder_settings",
+    ":fake_video_codecs",
+    ":fileutils",
+    ":mock_transport",
+    ":run_loop",
+    ":scoped_key_value_config",
+    ":test_support",
+    ":video_test_common",
+    "../api:array_view",
+    "../api:create_frame_generator",
+    "../api:frame_generator_api",
+    "../api:rtp_headers",
+    "../api:rtp_parameters",
+    "../api:simulated_network_api",
+    "../api:transport_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
+    "../api/audio_codecs:builtin_audio_encoder_factory",
+    "../api/rtc_event_log",
+    "../api/task_queue",
+    "../api/task_queue:default_task_queue_factory",
+    "../api/test/video:function_video_factory",
+    "../api/transport:field_trial_based_config",
+    "../api/units:time_delta",
+    "../api/video:builtin_video_bitrate_allocator_factory",
+    "../api/video:video_bitrate_allocator_factory",
+    "../api/video:video_frame",
+    "../api/video_codecs:video_codecs_api",
+    "../call",
+    "../call:call_interfaces",
+    "../call:fake_network",
+    "../call:simulated_network",
+    "../call:simulated_packet_receiver",
+    "../call:video_stream_api",
+    "../modules/audio_device:audio_device_impl",
+    "../modules/audio_mixer:audio_mixer_impl",
+    "../modules/rtp_rtcp",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
+    "../modules/rtp_rtcp:rtp_video_header",
+    "../modules/video_coding:codec_globals_headers",
+    "../rtc_base:checks",
+    "../rtc_base:criticalsection",
+    "../rtc_base:rtc_event",
+    "../rtc_base:task_queue_for_test",
+    "../rtc_base:threading",
+    "../rtc_base:timeutils",
+    "../system_wrappers",
+    "../system_wrappers:field_trial",
+    "../video/config:encoder_config",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  if (!is_android && !build_with_chromium) {
+    deps += [ "../modules/video_capture:video_capture_internal_impl" ]
+  }
+  # This, or some form of it should be upstreamed.
+  if (!rtc_include_tests) {
+    deps -= [ "../rtc_base:task_queue_for_test" ]
+  }
+}
+
+rtc_library("mock_transport") {
+  testonly = true
+  sources = [
+    "mock_transport.cc",
+    "mock_transport.h",
+  ]
+  deps = [
+    ":test_support",
+    "../api:transport_api",
+  ]
+}
+
+rtc_source_set("test_renderer") {
+  public_deps =  # no-presubmit-check TODO(webrtc:8603)
+      [ ":test_renderer_generic" ]
+  testonly = true
+  if (is_mac) {
+    public_deps +=  # no-presubmit-check TODO(webrtc:8603)
+        [ ":test_renderer_objc" ]
+  }
+}
+
+rtc_library("mock_frame_transformer") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [ "mock_frame_transformer.h" ]
+  deps = [
+    "../api:frame_transformer_interface",
+    "../test:test_support",
+  ]
+}
+
+rtc_library("mock_transformable_frame") {
+  visibility = [ "*" ]
+
+  testonly = true
+  sources = [ "mock_transformable_frame.h" ]
+
+  deps = [
+    "../api:frame_transformer_interface",
+    "../test:test_support",
+  ]
+}
+
+if (is_mac) {
+  rtc_library("test_renderer_objc") {
+    testonly = true
+    visibility = [ ":test_renderer" ]
+    sources = [
+      "mac/video_renderer_mac.h",
+      "mac/video_renderer_mac.mm",
+    ]
+    deps = [ ":test_renderer_generic" ]
+    frameworks = [
+      "Cocoa.framework",
+      "OpenGL.framework",
+      "CoreVideo.framework",
+    ]
+    defines = [ "GL_SILENCE_DEPRECATION" ]
+  }
+}
+
+rtc_library("test_renderer_generic") {
+  testonly = true
+  visibility = [
+    ":test_renderer",
+    ":test_renderer_objc",
+  ]
+  libs = []
+  sources = [
+    "video_renderer.cc",
+    "video_renderer.h",
+  ]
+  deps = [
+    "../api/video:video_frame",
+    "../common_video",
+    "../rtc_base:checks",
+  ]
+  if (is_win) {
+    sources += [
+      "win/d3d_renderer.cc",
+      "win/d3d_renderer.h",
+    ]
+    deps += [ "../api:scoped_refptr" ]
+  }
+  if (!((is_linux || is_chromeos) && rtc_use_x11) && !is_mac && !is_win) {
+    sources += [ "null_platform_renderer.cc" ]
+  }
+  if (((is_linux || is_chromeos) && rtc_use_x11) || is_mac) {
+    sources += [
+      "gl/gl_renderer.cc",
+      "gl/gl_renderer.h",
+    ]
+  }
+  if (is_mac) {
+    defines = [ "GL_SILENCE_DEPRECATION" ]
+  }
+
+  if ((is_linux || is_chromeos) && rtc_use_x11) {
+    sources += [
+      "linux/glx_renderer.cc",
+      "linux/glx_renderer.h",
+      "linux/video_renderer_linux.cc",
+    ]
+    libs += [
+      "Xext",
+      "X11",
+      "GL",
+    ]
+  }
+  if (is_android) {
+    libs += [
+      "GLESv2",
+      "log",
+    ]
+  }
+}
+
+rtc_library("audio_codec_mocks") {
+  testonly = true
+  sources = [
+    "mock_audio_decoder.cc",
+    "mock_audio_decoder.h",
+    "mock_audio_decoder_factory.h",
+    "mock_audio_encoder.cc",
+    "mock_audio_encoder.h",
+    "mock_audio_encoder_factory.h",
+  ]
+
+  deps = [
+    ":test_support",
+    "../api:array_view",
+    "../api:make_ref_counted",
+    "../api:scoped_refptr",
+    "../api/audio_codecs:audio_codecs_api",
+    "../api/audio_codecs:builtin_audio_decoder_factory",
+  ]
+}
+
+rtc_library("copy_to_file_audio_capturer") {
+  testonly = true
+  sources = [
+    "testsupport/copy_to_file_audio_capturer.cc",
+    "testsupport/copy_to_file_audio_capturer.h",
+  ]
+  deps = [
+    "../api:array_view",
+    "../common_audio",
+    "../modules/audio_device:audio_device_impl",
+    "../rtc_base:buffer",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("copy_to_file_audio_capturer_unittest") {
+  testonly = true
+  sources = [ "testsupport/copy_to_file_audio_capturer_unittest.cc" ]
+  deps = [
+    ":copy_to_file_audio_capturer",
+    ":fileutils",
+    ":test_support",
+    "../modules/audio_device:audio_device_impl",
+  ]
+}
+
+if (!build_with_mozilla) {
+if (!build_with_chromium && is_android) {
+  rtc_android_library("native_test_java") {
+    testonly = true
+    sources = [
+      "android/org/webrtc/native_test/RTCNativeUnitTest.java",
+      "android/org/webrtc/native_test/RTCNativeUnitTestActivity.java",
+    ]
+    deps = [
+      "../rtc_base:base_java",
+      "//testing/android/native_test:native_test_java",
+    ]
+  }
+}
+}
+
+rtc_library("call_config_utils") {
+  # TODO(bugs.webrtc.org/10814): Remove rtc_json_suppressions as soon as it
+  # gets removed upstream.
+  public_configs = [ "../rtc_base:rtc_json_suppressions" ]
+  sources = [
+    "call_config_utils.cc",
+    "call_config_utils.h",
+  ]
+  deps = [
+    "../call:video_stream_api",
+    "../rtc_base:rtc_json",
+  ]
+}
+
+rtc_library("fake_encoded_frame") {
+  testonly = true
+  sources = [
+    "fake_encoded_frame.cc",
+    "fake_encoded_frame.h",
+  ]
+  deps = [
+    ":test_support",
+    "../api:rtp_packet_info",
+    "../api/video:encoded_frame",
+    "../api/video:encoded_image",
+    "../api/video:video_frame_type",
+    "../api/video:video_rtp_headers",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
diff --git a/third_party/libwebrtc/test/DEPS b/third_party/libwebrtc/test/DEPS
new file mode 100644
index 0000000000..a9e9a7b5f1
--- /dev/null
+++ b/third_party/libwebrtc/test/DEPS
@@ -0,0 +1,88 @@
+include_rules = [
+  "+third_party/libjpeg",
+  "+third_party/libjpeg_turbo",
+  "+call",
+  "+common_audio",
+  "+common_video",
+  "+logging/rtc_event_log",
+  "+media/base",
+  "+media/sctp",
+  "+media/engine",
+  "+modules/audio_coding",
+  "+modules/congestion_controller",
+  "+modules/audio_device",
+  "+modules/audio_mixer",
+  "+modules/audio_processing",
+  "+modules/congestion_controller/bbr",
+  "+modules/rtp_rtcp",
+  "+modules/utility",
+  "+modules/video_capture",
+  "+modules/video_coding",
+  "+p2p/base/basic_packet_socket_factory.h",
+  "+sdk",
+  "+system_wrappers",
+  "+third_party/libyuv",
+  "+video/config",
+]
+
+specific_include_rules = {
+  "gmock\.h": [
+    "+testing/gmock/include/gmock",
+  ],
+  "gtest\.h": [
+    "+testing/gtest/include/gtest",
+  ],
+  ".*congestion_controller_feedback_fuzzer\.cc": [
+    "+modules/congestion_controller/include/receive_side_congestion_controller.h",
+    "+modules/pacing/packet_router.h",
+    "+modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h",
+  ],
+  ".*mdns_parser_fuzzer\.cc": [
+    "+p2p/base/mdns_message.h",
+  ],
+  ".*pseudotcp_parser_fuzzer\.cc": [
+    "+p2p/base/pseudo_tcp.h",
+  ],
+  ".*stun_parser_fuzzer\.cc": [
+    "+p2p/base/stun.h",
+  ],
+  ".*stun_validator_fuzzer\.cc": [
+    "+p2p/base/stun.h",
+  ],
+  ".*test_main\.cc": [
+    "+absl/debugging/failure_signal_handler.h",
+    "+absl/debugging/symbolize.h",
+  ],
+  ".*test_peer\.(h|cc)": [
+    "+pc",
+    "+p2p",
+  ],
+  ".*test_peer_factory\.(h|cc)": [
+    "+pc",
+    "+p2p",
+  ],
+  ".*peer_connection_quality_test_params\.h": [
+    "+p2p/base/port_allocator.h",
+  ],
+  ".*network_emulation_pc_unittest\.cc": [
+    "+pc/peer_connection_wrapper.h",
+    "+pc/test/mock_peer_connection_observers.h",
+    "+p2p/client/basic_port_allocator.h",
+  ],
+  ".*peer_connection_quality_test\.(h|cc)": [
+    "+pc",
+  ],
+  ".*sdp_changer\.(h|cc)": [
+    "+pc",
+    "+p2p",
+  ],
+  ".*test_video_capturer_video_track_source.h": [
+    "+pc",
+  ],
+  "benchmark_main\.cc": [
+    "+benchmark",
+  ],
+  "emulated_turn_server\.h": [
+     "+p2p/base/turn_server.h",
+  ]
+}
diff --git a/third_party/libwebrtc/test/OWNERS b/third_party/libwebrtc/test/OWNERS
new file mode 100644
index 0000000000..a1bd812244
--- /dev/null
+++ b/third_party/libwebrtc/test/OWNERS
@@ -0,0 +1,7 @@
+sprang@webrtc.org
+srte@webrtc.org
+stefan@webrtc.org
+titovartem@webrtc.org
+landrey@webrtc.org
+mbonadei@webrtc.org
+jleconte@webrtc.org
diff --git a/third_party/libwebrtc/test/android/AndroidManifest.xml b/third_party/libwebrtc/test/android/AndroidManifest.xml
new file mode 100644
index 0000000000..ad3f434b4f
--- /dev/null
+++ b/third_party/libwebrtc/test/android/AndroidManifest.xml
@@ -0,0 +1,47 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!--
+Copyright 2017 The WebRTC project authors. All Rights Reserved.
+
+Use of this source code is governed by a BSD-style license
+that can be found in the LICENSE file in the root of the source
+tree. An additional intellectual property rights grant can be found
+in the file PATENTS.  All contributing project authors may
+be found in the AUTHORS file in the root of the source tree.
+-->
+
+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
+      package="org.webrtc.native_test"
+      android:versionCode="1"
+      android:versionName="1.0">
+
+    <uses-sdk android:minSdkVersion="21" android:targetSdkVersion="23" />
+    <uses-permission android:name="android.permission.ACCESS_NETWORK_STATE"/>
+    <uses-permission android:name="android.permission.BLUETOOTH"/>
+    <uses-permission android:name="android.permission.BLUETOOTH_ADMIN"/>
+    <uses-permission android:name="android.permission.CAMERA" />
+    <uses-permission android:name="android.permission.INTERNET"/>
+    <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS"/>
+    <uses-permission android:name="android.permission.RECORD_AUDIO"/>
+    <uses-permission android:name="android.permission.WAKE_LOCK"/>
+    <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
+
+    <application android:label="NativeTests"
+            android:name="org.chromium.native_test.NativeTestApplication">
+        <uses-library android:name="android.test.runner"/>
+        <activity android:name=".RTCNativeUnitTestActivity"
+                android:label="NativeTest"
+                android:configChanges="orientation|keyboardHidden"
+                android:process=":test_process">
+            <intent-filter>
+                <action android:name="android.intent.action.MAIN" />
+                <category android:name="android.intent.category.LAUNCHER" />
+            </intent-filter>
+        </activity>
+    </application>
+
+    <instrumentation android:name="org.chromium.build.gtest_apk.NativeTestInstrumentationTestRunner"
+            android:targetPackage="org.webrtc.native_test"
+            android:label="Instrumentation entry point for org.webrtc.native_test"
+            chromium-junit3="true"/>
+
+</manifest>
diff --git a/third_party/libwebrtc/test/android/org/webrtc/native_test/RTCNativeUnitTest.java b/third_party/libwebrtc/test/android/org/webrtc/native_test/RTCNativeUnitTest.java
new file mode 100644
index 0000000000..dede7edd1f
--- /dev/null
+++ b/third_party/libwebrtc/test/android/org/webrtc/native_test/RTCNativeUnitTest.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.native_test;
+
+import android.app.Activity;
+import org.chromium.native_test.NativeUnitTest;
+import org.webrtc.ContextUtils;
+
+/**
+ * Native unit test that calls ContextUtils.initialize for WebRTC.
+ */
+public class RTCNativeUnitTest extends NativeUnitTest {
+  @Override
+  public void preCreate(Activity activity) {
+    super.preCreate(activity);
+    ContextUtils.initialize(activity.getApplicationContext());
+  }
+}
diff --git a/third_party/libwebrtc/test/android/org/webrtc/native_test/RTCNativeUnitTestActivity.java b/third_party/libwebrtc/test/android/org/webrtc/native_test/RTCNativeUnitTestActivity.java
new file mode 100644
index 0000000000..2a413682fe
--- /dev/null
+++ b/third_party/libwebrtc/test/android/org/webrtc/native_test/RTCNativeUnitTestActivity.java
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.native_test;
+
+import android.app.Activity;
+import android.os.Bundle;
+
+/**
+ * Activity that uses RTCNativeUnitTest to run the tests.
+ */
+public class RTCNativeUnitTestActivity extends Activity {
+  private RTCNativeUnitTest mTest = new RTCNativeUnitTest();
+
+  @Override
+  public void onCreate(Bundle savedInstanceState) {
+    mTest.preCreate(this);
+    super.onCreate(savedInstanceState);
+    mTest.postCreate(this);
+  }
+
+  @Override
+  public void onStart() {
+    super.onStart();
+    mTest.postStart(this, false);
+  }
+}
diff --git a/third_party/libwebrtc/test/audio_decoder_proxy_factory.h b/third_party/libwebrtc/test/audio_decoder_proxy_factory.h
new file mode 100644
index 0000000000..95606d6ff7
--- /dev/null
+++ b/third_party/libwebrtc/test/audio_decoder_proxy_factory.h
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_AUDIO_DECODER_PROXY_FACTORY_H_
+#define TEST_AUDIO_DECODER_PROXY_FACTORY_H_
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+
+namespace webrtc {
+namespace test {
+
+// A decoder factory with a single underlying AudioDecoder object, intended for
+// test purposes. Each call to MakeAudioDecoder returns a proxy for the same
+// decoder, typically a mock or fake decoder.
+class AudioDecoderProxyFactory : public AudioDecoderFactory {
+ public:
+  explicit AudioDecoderProxyFactory(AudioDecoder* decoder)
+      : decoder_(decoder) {}
+
+  // Unused by tests.
+  std::vector<AudioCodecSpec> GetSupportedDecoders() override {
+    RTC_DCHECK_NOTREACHED();
+    return {};
+  }
+
+  bool IsSupportedDecoder(const SdpAudioFormat& format) override {
+    return true;
+  }
+
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& /* format */,
+      absl::optional<AudioCodecPairId> /* codec_pair_id */) override {
+    return std::make_unique<DecoderProxy>(decoder_);
+  }
+
+ private:
+  // Wrapper class, since CreateAudioDecoder needs to surrender
+  // ownership to the object it returns.
+  class DecoderProxy final : public AudioDecoder {
+   public:
+    explicit DecoderProxy(AudioDecoder* decoder) : decoder_(decoder) {}
+
+   private:
+    std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
+                                          uint32_t timestamp) override {
+      return decoder_->ParsePayload(std::move(payload), timestamp);
+    }
+
+    bool HasDecodePlc() const override { return decoder_->HasDecodePlc(); }
+
+    int ErrorCode() override { return decoder_->ErrorCode(); }
+
+    void Reset() override { decoder_->Reset(); }
+
+    int SampleRateHz() const override { return decoder_->SampleRateHz(); }
+
+    size_t Channels() const override { return decoder_->Channels(); }
+
+    int DecodeInternal(const uint8_t* encoded,
+                       size_t encoded_len,
+                       int sample_rate_hz,
+                       int16_t* decoded,
+                       SpeechType* speech_type) override {
+      // Needed for tests of NetEqImpl::DecodeCng, which calls the deprecated
+      // Decode method.
+      size_t max_decoded_bytes =
+          decoder_->PacketDuration(encoded, encoded_len) *
+          decoder_->Channels() * sizeof(int16_t);
+      return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
+                              max_decoded_bytes, decoded, speech_type);
+    }
+
+    void GeneratePlc(size_t requested_samples_per_channel,
+                     rtc::BufferT<int16_t>* concealment_audio) override {
+      decoder_->GeneratePlc(requested_samples_per_channel, concealment_audio);
+    }
+
+    AudioDecoder* const decoder_;
+  };
+
+  AudioDecoder* const decoder_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_AUDIO_DECODER_PROXY_FACTORY_H_
diff --git a/third_party/libwebrtc/test/benchmark_main.cc b/third_party/libwebrtc/test/benchmark_main.cc
new file mode 100644
index 0000000000..1a79c24913
--- /dev/null
+++ b/third_party/libwebrtc/test/benchmark_main.cc
@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "benchmark/benchmark.h"
+
+int main(int argc, char* argv[]) {
+  benchmark::Initialize(&argc, argv);
+  benchmark::RunSpecifiedBenchmarks();
+  return 0;
+}
diff --git a/third_party/libwebrtc/test/call_config_utils.cc b/third_party/libwebrtc/test/call_config_utils.cc
new file mode 100644
index 0000000000..da3d76c689
--- /dev/null
+++ b/third_party/libwebrtc/test/call_config_utils.cc
@@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/call_config_utils.h"
+
+#include <string>
+#include <vector>
+
+namespace webrtc {
+namespace test {
+
+// Deserializes a JSON representation of the VideoReceiveStreamInterface::Config
+// back into a valid object. This will not initialize the decoders or the
+// renderer.
+VideoReceiveStreamInterface::Config ParseVideoReceiveStreamJsonConfig(
+    webrtc::Transport* transport,
+    const Json::Value& json) {
+  auto receive_config = VideoReceiveStreamInterface::Config(transport);
+  for (const auto& decoder_json : json["decoders"]) {
+    VideoReceiveStreamInterface::Decoder decoder;
+    decoder.video_format =
+        SdpVideoFormat(decoder_json["payload_name"].asString());
+    decoder.payload_type = decoder_json["payload_type"].asInt64();
+    for (const auto& params_json : decoder_json["codec_params"]) {
+      std::vector<std::string> members = params_json.getMemberNames();
+      RTC_CHECK_EQ(members.size(), 1);
+      decoder.video_format.parameters[members[0]] =
+          params_json[members[0]].asString();
+    }
+    receive_config.decoders.push_back(decoder);
+  }
+  receive_config.render_delay_ms = json["render_delay_ms"].asInt64();
+  receive_config.rtp.remote_ssrc = json["rtp"]["remote_ssrc"].asInt64();
+  receive_config.rtp.local_ssrc = json["rtp"]["local_ssrc"].asInt64();
+  receive_config.rtp.rtcp_mode =
+      json["rtp"]["rtcp_mode"].asString() == "RtcpMode::kCompound"
+          ? RtcpMode::kCompound
+          : RtcpMode::kReducedSize;
+  receive_config.rtp.lntf.enabled = json["rtp"]["lntf"]["enabled"].asInt64();
+  receive_config.rtp.nack.rtp_history_ms =
+      json["rtp"]["nack"]["rtp_history_ms"].asInt64();
+  receive_config.rtp.ulpfec_payload_type =
+      json["rtp"]["ulpfec_payload_type"].asInt64();
+  receive_config.rtp.red_payload_type =
+      json["rtp"]["red_payload_type"].asInt64();
+  receive_config.rtp.rtx_ssrc = json["rtp"]["rtx_ssrc"].asInt64();
+
+  for (const auto& pl_json : json["rtp"]["rtx_payload_types"]) {
+    std::vector<std::string> members = pl_json.getMemberNames();
+    RTC_CHECK_EQ(members.size(), 1);
+    Json::Value rtx_payload_type = pl_json[members[0]];
+    receive_config.rtp.rtx_associated_payload_types[std::stoi(members[0])] =
+        rtx_payload_type.asInt64();
+  }
+  for (const auto& ext_json : json["rtp"]["extensions"]) {
+    receive_config.rtp.extensions.emplace_back(ext_json["uri"].asString(),
+                                               ext_json["id"].asInt64(),
+                                               ext_json["encrypt"].asBool());
+  }
+  return receive_config;
+}
+
+Json::Value GenerateVideoReceiveStreamJsonConfig(
+    const VideoReceiveStreamInterface::Config& config) {
+  Json::Value root_json;
+
+  root_json["decoders"] = Json::Value(Json::arrayValue);
+  for (const auto& decoder : config.decoders) {
+    Json::Value decoder_json;
+    decoder_json["payload_type"] = decoder.payload_type;
+    decoder_json["payload_name"] = decoder.video_format.name;
+    decoder_json["codec_params"] = Json::Value(Json::arrayValue);
+    for (const auto& codec_param_entry : decoder.video_format.parameters) {
+      Json::Value codec_param_json;
+      codec_param_json[codec_param_entry.first] = codec_param_entry.second;
+      decoder_json["codec_params"].append(codec_param_json);
+    }
+    root_json["decoders"].append(decoder_json);
+  }
+
+  Json::Value rtp_json;
+  rtp_json["remote_ssrc"] = config.rtp.remote_ssrc;
+  rtp_json["local_ssrc"] = config.rtp.local_ssrc;
+  rtp_json["rtcp_mode"] = config.rtp.rtcp_mode == RtcpMode::kCompound
+                              ? "RtcpMode::kCompound"
+                              : "RtcpMode::kReducedSize";
+  rtp_json["lntf"]["enabled"] = config.rtp.lntf.enabled;
+  rtp_json["nack"]["rtp_history_ms"] = config.rtp.nack.rtp_history_ms;
+  rtp_json["ulpfec_payload_type"] = config.rtp.ulpfec_payload_type;
+  rtp_json["red_payload_type"] = config.rtp.red_payload_type;
+  rtp_json["rtx_ssrc"] = config.rtp.rtx_ssrc;
+  rtp_json["rtx_payload_types"] = Json::Value(Json::arrayValue);
+
+  for (auto& kv : config.rtp.rtx_associated_payload_types) {
+    Json::Value val;
+    val[std::to_string(kv.first)] = kv.second;
+    rtp_json["rtx_payload_types"].append(val);
+  }
+
+  rtp_json["extensions"] = Json::Value(Json::arrayValue);
+  for (auto& ext : config.rtp.extensions) {
+    Json::Value ext_json;
+    ext_json["uri"] = ext.uri;
+    ext_json["id"] = ext.id;
+    ext_json["encrypt"] = ext.encrypt;
+    rtp_json["extensions"].append(ext_json);
+  }
+  root_json["rtp"] = rtp_json;
+
+  root_json["render_delay_ms"] = config.render_delay_ms;
+
+  return root_json;
+}
+
+}  // namespace test.
+}  // namespace webrtc.
diff --git a/third_party/libwebrtc/test/call_config_utils.h b/third_party/libwebrtc/test/call_config_utils.h
new file mode 100644
index 0000000000..97cfdc3396
--- /dev/null
+++ b/third_party/libwebrtc/test/call_config_utils.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_CALL_CONFIG_UTILS_H_
+#define TEST_CALL_CONFIG_UTILS_H_
+
+#include "call/video_receive_stream.h"
+#include "rtc_base/strings/json.h"
+
+namespace webrtc {
+namespace test {
+
+// Deserializes a JSON representation of the VideoReceiveStreamInterface::Config
+// back into a valid object. This will not initialize the decoders or the
+// renderer.
+VideoReceiveStreamInterface::Config ParseVideoReceiveStreamJsonConfig(
+    webrtc::Transport* transport,
+    const Json::Value& json);
+
+// Serialize a VideoReceiveStreamInterface::Config into a Json object.
+Json::Value GenerateVideoReceiveStreamJsonConfig(
+    const VideoReceiveStreamInterface::Config& config);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_CALL_CONFIG_UTILS_H_
diff --git a/third_party/libwebrtc/test/call_config_utils_unittest.cc b/third_party/libwebrtc/test/call_config_utils_unittest.cc
new file mode 100644
index 0000000000..e010ab6707
--- /dev/null
+++ b/third_party/libwebrtc/test/call_config_utils_unittest.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/call_config_utils.h"
+
+#include "call/video_receive_stream.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(CallConfigUtils, MarshalUnmarshalProcessSameObject) {
+  VideoReceiveStreamInterface::Config recv_config(nullptr);
+
+  VideoReceiveStreamInterface::Decoder decoder;
+  decoder.payload_type = 10;
+  decoder.video_format.name = "test";
+  decoder.video_format.parameters["99"] = "b";
+  recv_config.decoders.push_back(decoder);
+  recv_config.render_delay_ms = 10;
+  recv_config.rtp.remote_ssrc = 100;
+  recv_config.rtp.local_ssrc = 101;
+  recv_config.rtp.rtcp_mode = RtcpMode::kCompound;
+  recv_config.rtp.lntf.enabled = false;
+  recv_config.rtp.nack.rtp_history_ms = 150;
+  recv_config.rtp.red_payload_type = 50;
+  recv_config.rtp.rtx_ssrc = 1000;
+  recv_config.rtp.rtx_associated_payload_types[10] = 10;
+  recv_config.rtp.extensions.emplace_back("uri", 128, true);
+
+  VideoReceiveStreamInterface::Config unmarshaled_config =
+      ParseVideoReceiveStreamJsonConfig(
+          nullptr, GenerateVideoReceiveStreamJsonConfig(recv_config));
+
+  EXPECT_EQ(recv_config.decoders[0].payload_type,
+            unmarshaled_config.decoders[0].payload_type);
+  EXPECT_EQ(recv_config.decoders[0].video_format.name,
+            unmarshaled_config.decoders[0].video_format.name);
+  EXPECT_EQ(recv_config.decoders[0].video_format.parameters,
+            unmarshaled_config.decoders[0].video_format.parameters);
+  EXPECT_EQ(recv_config.render_delay_ms, unmarshaled_config.render_delay_ms);
+  EXPECT_EQ(recv_config.rtp.remote_ssrc, unmarshaled_config.rtp.remote_ssrc);
+  EXPECT_EQ(recv_config.rtp.local_ssrc, unmarshaled_config.rtp.local_ssrc);
+  EXPECT_EQ(recv_config.rtp.rtcp_mode, unmarshaled_config.rtp.rtcp_mode);
+  EXPECT_EQ(recv_config.rtp.lntf.enabled, unmarshaled_config.rtp.lntf.enabled);
+  EXPECT_EQ(recv_config.rtp.nack.rtp_history_ms,
+            unmarshaled_config.rtp.nack.rtp_history_ms);
+  EXPECT_EQ(recv_config.rtp.red_payload_type,
+            unmarshaled_config.rtp.red_payload_type);
+  EXPECT_EQ(recv_config.rtp.rtx_ssrc, unmarshaled_config.rtp.rtx_ssrc);
+  EXPECT_EQ(recv_config.rtp.rtx_associated_payload_types,
+            unmarshaled_config.rtp.rtx_associated_payload_types);
+  EXPECT_EQ(recv_config.rtp.extensions, recv_config.rtp.extensions);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/call_test.cc b/third_party/libwebrtc/test/call_test.cc
new file mode 100644
index 0000000000..62d18394f9
--- /dev/null
+++ b/third_party/libwebrtc/test/call_test.cc
@@ -0,0 +1,861 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/call_test.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/create_frame_generator.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "call/fake_network_pipe.h"
+#include "call/packet_receiver.h"
+#include "call/simulated_network.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/fake_encoder.h"
+#include "test/rtp_rtcp_observer.h"
+#include "test/testsupport/file_utils.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+namespace test {
+
+CallTest::CallTest()
+    : clock_(Clock::GetRealTimeClock()),
+      task_queue_factory_(CreateDefaultTaskQueueFactory()),
+      send_event_log_(std::make_unique<RtcEventLogNull>()),
+      recv_event_log_(std::make_unique<RtcEventLogNull>()),
+      audio_send_config_(/*send_transport=*/nullptr),
+      audio_send_stream_(nullptr),
+      frame_generator_capturer_(nullptr),
+      fake_encoder_factory_([this]() {
+        std::unique_ptr<FakeEncoder> fake_encoder;
+        if (video_encoder_configs_[0].codec_type == kVideoCodecVP8) {
+          fake_encoder = std::make_unique<FakeVp8Encoder>(clock_);
+        } else {
+          fake_encoder = std::make_unique<FakeEncoder>(clock_);
+        }
+        fake_encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
+        return fake_encoder;
+      }),
+      fake_decoder_factory_([]() { return std::make_unique<FakeDecoder>(); }),
+      bitrate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()),
+      num_video_streams_(1),
+      num_audio_streams_(0),
+      num_flexfec_streams_(0),
+      audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
+      audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
+      task_queue_(task_queue_factory_->CreateTaskQueue(
+          "CallTestTaskQueue",
+          TaskQueueFactory::Priority::NORMAL)) {}
+
+CallTest::~CallTest() = default;
+
+void CallTest::RegisterRtpExtension(const RtpExtension& extension) {
+  for (const RtpExtension& registered_extension : rtp_extensions_) {
+    if (registered_extension.id == extension.id) {
+      ASSERT_EQ(registered_extension.uri, extension.uri)
+          << "Different URIs associated with ID " << extension.id << ".";
+      ASSERT_EQ(registered_extension.encrypt, extension.encrypt)
+          << "Encryption mismatch associated with ID " << extension.id << ".";
+      return;
+    } else {  // Different IDs.
+      // Different IDs referring to the same extension probably indicate
+      // a mistake in the test.
+      ASSERT_FALSE(registered_extension.uri == extension.uri &&
+                   registered_extension.encrypt == extension.encrypt)
+          << "URI " << extension.uri
+          << (extension.encrypt ? " with " : " without ")
+          << "encryption already registered with a different "
+             "ID ("
+          << extension.id << " vs. " << registered_extension.id << ").";
+    }
+  }
+  rtp_extensions_.push_back(extension);
+}
+
+void CallTest::RunBaseTest(BaseTest* test) {
+  SendTask(task_queue(), [this, test]() {
+    num_video_streams_ = test->GetNumVideoStreams();
+    num_audio_streams_ = test->GetNumAudioStreams();
+    num_flexfec_streams_ = test->GetNumFlexfecStreams();
+    RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
+    Call::Config send_config(send_event_log_.get());
+    test->ModifySenderBitrateConfig(&send_config.bitrate_config);
+    if (num_audio_streams_ > 0) {
+      CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
+      test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
+                                      fake_recv_audio_device_.get());
+      apm_send_ = AudioProcessingBuilder().Create();
+      apm_recv_ = AudioProcessingBuilder().Create();
+      EXPECT_EQ(0, fake_send_audio_device_->Init());
+      EXPECT_EQ(0, fake_recv_audio_device_->Init());
+      AudioState::Config audio_state_config;
+      audio_state_config.audio_mixer = AudioMixerImpl::Create();
+      audio_state_config.audio_processing = apm_send_;
+      audio_state_config.audio_device_module = fake_send_audio_device_;
+      send_config.audio_state = AudioState::Create(audio_state_config);
+      fake_send_audio_device_->RegisterAudioCallback(
+          send_config.audio_state->audio_transport());
+    }
+    CreateSenderCall(send_config);
+    if (test->ShouldCreateReceivers()) {
+      Call::Config recv_config(recv_event_log_.get());
+      test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
+      if (num_audio_streams_ > 0) {
+        AudioState::Config audio_state_config;
+        audio_state_config.audio_mixer = AudioMixerImpl::Create();
+        audio_state_config.audio_processing = apm_recv_;
+        audio_state_config.audio_device_module = fake_recv_audio_device_;
+        recv_config.audio_state = AudioState::Create(audio_state_config);
+        fake_recv_audio_device_->RegisterAudioCallback(
+            recv_config.audio_state->audio_transport());
+      }
+      CreateReceiverCall(recv_config);
+    }
+    test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
+    CreateReceiveTransport(test->GetReceiveTransportConfig(), test);
+    CreateSendTransport(test->GetSendTransportConfig(), test);
+    test->OnTransportCreated(send_transport_.get(), send_simulated_network_,
+                             receive_transport_.get(),
+                             receive_simulated_network_);
+    if (test->ShouldCreateReceivers()) {
+      if (num_video_streams_ > 0)
+        receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
+      if (num_audio_streams_ > 0)
+        receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+    } else {
+      // Sender-only call delivers to itself.
+      send_transport_->SetReceiver(sender_call_->Receiver());
+      receive_transport_->SetReceiver(nullptr);
+    }
+
+    CreateSendConfig(num_video_streams_, num_audio_streams_,
+                     num_flexfec_streams_, send_transport_.get());
+    if (test->ShouldCreateReceivers()) {
+      CreateMatchingReceiveConfigs();
+    }
+    if (num_video_streams_ > 0) {
+      test->ModifyVideoConfigs(GetVideoSendConfig(), &video_receive_configs_,
+                               GetVideoEncoderConfig());
+    }
+    if (num_audio_streams_ > 0) {
+      test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
+    }
+    if (num_flexfec_streams_ > 0) {
+      test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
+    }
+
+    if (num_flexfec_streams_ > 0) {
+      CreateFlexfecStreams();
+      test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
+    }
+    if (num_video_streams_ > 0) {
+      CreateVideoStreams();
+      test->OnVideoStreamsCreated(GetVideoSendStream(), video_receive_streams_);
+    }
+    if (num_audio_streams_ > 0) {
+      CreateAudioStreams();
+      test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
+    }
+
+    if (num_video_streams_ > 0) {
+      int width = kDefaultWidth;
+      int height = kDefaultHeight;
+      int frame_rate = kDefaultFramerate;
+      test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
+      test->ModifyVideoDegradationPreference(&degradation_preference_);
+      CreateFrameGeneratorCapturer(frame_rate, width, height);
+      test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_);
+    }
+
+    Start();
+  });
+
+  test->PerformTest();
+
+  SendTask(task_queue(), [this, test]() {
+    Stop();
+    test->OnStreamsStopped();
+    DestroyStreams();
+    send_transport_.reset();
+    receive_transport_.reset();
+
+    frame_generator_capturer_ = nullptr;
+    DestroyCalls();
+
+    fake_send_audio_device_ = nullptr;
+    fake_recv_audio_device_ = nullptr;
+  });
+}
+
+void CallTest::CreateCalls() {
+  CreateCalls(Call::Config(send_event_log_.get()),
+              Call::Config(recv_event_log_.get()));
+}
+
+void CallTest::CreateCalls(const Call::Config& sender_config,
+                           const Call::Config& receiver_config) {
+  CreateSenderCall(sender_config);
+  CreateReceiverCall(receiver_config);
+}
+
+void CallTest::CreateSenderCall() {
+  CreateSenderCall(Call::Config(send_event_log_.get()));
+}
+
+void CallTest::CreateSenderCall(const Call::Config& config) {
+  auto sender_config = config;
+  sender_config.task_queue_factory = task_queue_factory_.get();
+  sender_config.network_state_predictor_factory =
+      network_state_predictor_factory_.get();
+  sender_config.network_controller_factory = network_controller_factory_.get();
+  sender_config.trials = &field_trials_;
+  sender_call_.reset(Call::Create(sender_config));
+}
+
+void CallTest::CreateReceiverCall(const Call::Config& config) {
+  auto receiver_config = config;
+  receiver_config.task_queue_factory = task_queue_factory_.get();
+  receiver_config.trials = &field_trials_;
+  receiver_call_.reset(Call::Create(receiver_config));
+}
+
+void CallTest::DestroyCalls() {
+  send_transport_.reset();
+  receive_transport_.reset();
+  sender_call_.reset();
+  receiver_call_.reset();
+}
+
+void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
+                                     size_t num_video_streams,
+                                     size_t num_used_ssrcs,
+                                     Transport* send_transport) {
+  RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, kNumSsrcs);
+  *video_config = VideoSendStream::Config(send_transport);
+  video_config->encoder_settings.encoder_factory = &fake_encoder_factory_;
+  video_config->encoder_settings.bitrate_allocator_factory =
+      bitrate_allocator_factory_.get();
+  video_config->rtp.payload_name = "FAKE";
+  video_config->rtp.payload_type = kFakeVideoSendPayloadType;
+  video_config->rtp.extmap_allow_mixed = true;
+  AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kAbsSendTimeUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kTimestampOffsetUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri00,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kDependencyDescriptorUri,
+                       &video_config->rtp.extensions);
+  if (video_encoder_configs_.empty()) {
+    video_encoder_configs_.emplace_back();
+    FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
+                             &video_encoder_configs_.back());
+  }
+  for (size_t i = 0; i < num_video_streams; ++i)
+    video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
+  AddRtpExtensionByUri(RtpExtension::kVideoRotationUri,
+                       &video_config->rtp.extensions);
+  AddRtpExtensionByUri(RtpExtension::kColorSpaceUri,
+                       &video_config->rtp.extensions);
+}
+
+void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
+                                            size_t num_flexfec_streams,
+                                            Transport* send_transport) {
+  RTC_DCHECK_LE(num_audio_streams, 1);
+  RTC_DCHECK_LE(num_flexfec_streams, 1);
+  if (num_audio_streams > 0) {
+    AudioSendStream::Config audio_send_config(send_transport);
+    audio_send_config.rtp.ssrc = kAudioSendSsrc;
+    AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri,
+                         &audio_send_config.rtp.extensions);
+
+    audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+        kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
+    audio_send_config.encoder_factory = audio_encoder_factory_;
+    SetAudioConfig(audio_send_config);
+  }
+
+  // TODO(brandtr): Update this when we support multistream protection.
+  if (num_flexfec_streams > 0) {
+    SetSendFecConfig({kVideoSendSsrcs[0]});
+  }
+}
+
+void CallTest::SetAudioConfig(const AudioSendStream::Config& config) {
+  audio_send_config_ = config;
+}
+
+void CallTest::SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs) {
+  GetVideoSendConfig()->rtp.flexfec.payload_type = kFlexfecPayloadType;
+  GetVideoSendConfig()->rtp.flexfec.ssrc = kFlexfecSendSsrc;
+  GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs;
+}
+
+void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) {
+  send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
+  send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
+  send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
+}
+
+void CallTest::SetReceiveUlpFecConfig(
+    VideoReceiveStreamInterface::Config* receive_config) {
+  receive_config->rtp.red_payload_type = kRedPayloadType;
+  receive_config->rtp.ulpfec_payload_type = kUlpfecPayloadType;
+  receive_config->rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
+      kRedPayloadType;
+}
+
+void CallTest::CreateSendConfig(size_t num_video_streams,
+                                size_t num_audio_streams,
+                                size_t num_flexfec_streams,
+                                Transport* send_transport) {
+  if (num_video_streams > 0) {
+    video_send_configs_.clear();
+    video_send_configs_.emplace_back(nullptr);
+    CreateVideoSendConfig(&video_send_configs_.back(), num_video_streams, 0,
+                          send_transport);
+  }
+  CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
+                               send_transport);
+}
+
+void CallTest::CreateMatchingVideoReceiveConfigs(
+    const VideoSendStream::Config& video_send_config,
+    Transport* rtcp_send_transport) {
+  CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport,
+                                    &fake_decoder_factory_, absl::nullopt,
+                                    false, 0);
+}
+
+void CallTest::CreateMatchingVideoReceiveConfigs(
+    const VideoSendStream::Config& video_send_config,
+    Transport* rtcp_send_transport,
+    VideoDecoderFactory* decoder_factory,
+    absl::optional<size_t> decode_sub_stream,
+    bool receiver_reference_time_report,
+    int rtp_history_ms) {
+  AddMatchingVideoReceiveConfigs(
+      &video_receive_configs_, video_send_config, rtcp_send_transport,
+      decoder_factory, decode_sub_stream, receiver_reference_time_report,
+      rtp_history_ms);
+}
+
+void CallTest::AddMatchingVideoReceiveConfigs(
+    std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
+    const VideoSendStream::Config& video_send_config,
+    Transport* rtcp_send_transport,
+    VideoDecoderFactory* decoder_factory,
+    absl::optional<size_t> decode_sub_stream,
+    bool receiver_reference_time_report,
+    int rtp_history_ms) {
+  RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
+  VideoReceiveStreamInterface::Config default_config(rtcp_send_transport);
+  default_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+  for (const RtpExtension& extension : video_send_config.rtp.extensions)
+    default_config.rtp.extensions.push_back(extension);
+  default_config.rtp.nack.rtp_history_ms = rtp_history_ms;
+  // Enable RTT calculation so NTP time estimator will work.
+  default_config.rtp.rtcp_xr.receiver_reference_time_report =
+      receiver_reference_time_report;
+  default_config.renderer = &fake_renderer_;
+
+  for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
+    VideoReceiveStreamInterface::Config video_recv_config(
+        default_config.Copy());
+    video_recv_config.decoders.clear();
+    if (!video_send_config.rtp.rtx.ssrcs.empty()) {
+      video_recv_config.rtp.rtx_ssrc = video_send_config.rtp.rtx.ssrcs[i];
+      video_recv_config.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
+          video_send_config.rtp.payload_type;
+    }
+    video_recv_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
+    VideoReceiveStreamInterface::Decoder decoder;
+
+    decoder.payload_type = video_send_config.rtp.payload_type;
+    decoder.video_format = SdpVideoFormat(video_send_config.rtp.payload_name);
+    // Force fake decoders on non-selected simulcast streams.
+    if (!decode_sub_stream || i == *decode_sub_stream) {
+      video_recv_config.decoder_factory = decoder_factory;
+    } else {
+      video_recv_config.decoder_factory = &fake_decoder_factory_;
+    }
+    video_recv_config.decoders.push_back(decoder);
+    receive_configs->emplace_back(std::move(video_recv_config));
+  }
+}
+
+void CallTest::CreateMatchingAudioAndFecConfigs(
+    Transport* rtcp_send_transport) {
+  RTC_DCHECK_GE(1, num_audio_streams_);
+  if (num_audio_streams_ == 1) {
+    CreateMatchingAudioConfigs(rtcp_send_transport, "");
+  }
+
+  // TODO(brandtr): Update this when we support multistream protection.
+  RTC_DCHECK(num_flexfec_streams_ <= 1);
+  if (num_flexfec_streams_ == 1) {
+    CreateMatchingFecConfig(rtcp_send_transport, *GetVideoSendConfig());
+    for (const RtpExtension& extension : GetVideoSendConfig()->rtp.extensions)
+      GetFlexFecConfig()->rtp.extensions.push_back(extension);
+  }
+}
+
+void CallTest::CreateMatchingAudioConfigs(Transport* transport,
+                                          std::string sync_group) {
+  audio_receive_configs_.push_back(CreateMatchingAudioConfig(
+      audio_send_config_, audio_decoder_factory_, transport, sync_group));
+}
+
+AudioReceiveStreamInterface::Config CallTest::CreateMatchingAudioConfig(
+    const AudioSendStream::Config& send_config,
+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
+    Transport* transport,
+    std::string sync_group) {
+  AudioReceiveStreamInterface::Config audio_config;
+  audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
+  audio_config.rtcp_send_transport = transport;
+  audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
+  audio_config.rtp.extensions = send_config.rtp.extensions;
+  audio_config.decoder_factory = audio_decoder_factory;
+  audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
+  audio_config.sync_group = sync_group;
+  return audio_config;
+}
+
+void CallTest::CreateMatchingFecConfig(
+    Transport* transport,
+    const VideoSendStream::Config& send_config) {
+  FlexfecReceiveStream::Config config(transport);
+  config.payload_type = send_config.rtp.flexfec.payload_type;
+  config.rtp.remote_ssrc = send_config.rtp.flexfec.ssrc;
+  config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
+  config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
+  if (!video_receive_configs_.empty()) {
+    video_receive_configs_[0].rtp.protected_by_flexfec = true;
+    video_receive_configs_[0].rtp.packet_sink_ = this;
+  }
+  flexfec_receive_configs_.push_back(config);
+}
+
+void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
+  video_receive_configs_.clear();
+  for (VideoSendStream::Config& video_send_config : video_send_configs_) {
+    CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport);
+  }
+  CreateMatchingAudioAndFecConfigs(rtcp_send_transport);
+}
+
+void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
+                                                     float speed,
+                                                     int framerate,
+                                                     int width,
+                                                     int height) {
+  video_sources_.clear();
+  auto frame_generator_capturer =
+      std::make_unique<test::FrameGeneratorCapturer>(
+          clock,
+          test::CreateSquareFrameGenerator(width, height, absl::nullopt,
+                                           absl::nullopt),
+          framerate * speed, *task_queue_factory_);
+  frame_generator_capturer_ = frame_generator_capturer.get();
+  frame_generator_capturer->Init();
+  video_sources_.push_back(std::move(frame_generator_capturer));
+  ConnectVideoSourcesToStreams();
+}
+
+void CallTest::CreateFrameGeneratorCapturer(int framerate,
+                                            int width,
+                                            int height) {
+  video_sources_.clear();
+  auto frame_generator_capturer =
+      std::make_unique<test::FrameGeneratorCapturer>(
+          clock_,
+          test::CreateSquareFrameGenerator(width, height, absl::nullopt,
+                                           absl::nullopt),
+          framerate, *task_queue_factory_);
+  frame_generator_capturer_ = frame_generator_capturer.get();
+  frame_generator_capturer->Init();
+  video_sources_.push_back(std::move(frame_generator_capturer));
+  ConnectVideoSourcesToStreams();
+}
+
+void CallTest::CreateFakeAudioDevices(
+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+    std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
+  fake_send_audio_device_ = TestAudioDeviceModule::Create(
+      task_queue_factory_.get(), std::move(capturer), nullptr, 1.f);
+  fake_recv_audio_device_ = TestAudioDeviceModule::Create(
+      task_queue_factory_.get(), nullptr, std::move(renderer), 1.f);
+}
+
+void CallTest::CreateVideoStreams() {
+  RTC_DCHECK(video_receive_streams_.empty());
+  CreateVideoSendStreams();
+  for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
+    video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
+        video_receive_configs_[i].Copy()));
+  }
+}
+
+void CallTest::CreateVideoSendStreams() {
+  RTC_DCHECK(video_send_streams_.empty());
+
+  // We currently only support testing external fec controllers with a single
+  // VideoSendStream.
+  if (fec_controller_factory_.get()) {
+    RTC_DCHECK_LE(video_send_configs_.size(), 1);
+  }
+
+  // TODO(http://crbug/818127):
+  // Remove this workaround when ALR is not screenshare-specific.
+  std::list<size_t> streams_creation_order;
+  for (size_t i = 0; i < video_send_configs_.size(); ++i) {
+    // If dual streams are created, add the screenshare stream last.
+    if (video_encoder_configs_[i].content_type ==
+        VideoEncoderConfig::ContentType::kScreen) {
+      streams_creation_order.push_back(i);
+    } else {
+      streams_creation_order.push_front(i);
+    }
+  }
+
+  video_send_streams_.resize(video_send_configs_.size(), nullptr);
+
+  for (size_t i : streams_creation_order) {
+    if (fec_controller_factory_.get()) {
+      video_send_streams_[i] = sender_call_->CreateVideoSendStream(
+          video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
+          fec_controller_factory_->CreateFecController());
+    } else {
+      video_send_streams_[i] = sender_call_->CreateVideoSendStream(
+          video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy());
+    }
+  }
+}
+
+void CallTest::CreateVideoSendStream(const VideoEncoderConfig& encoder_config) {
+  RTC_DCHECK(video_send_streams_.empty());
+  video_send_streams_.push_back(sender_call_->CreateVideoSendStream(
+      GetVideoSendConfig()->Copy(), encoder_config.Copy()));
+}
+
+void CallTest::CreateAudioStreams() {
+  RTC_DCHECK(audio_send_stream_ == nullptr);
+  RTC_DCHECK(audio_receive_streams_.empty());
+  audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
+  for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
+    audio_receive_streams_.push_back(
+        receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i]));
+  }
+}
+
+void CallTest::CreateFlexfecStreams() {
+  for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) {
+    flexfec_receive_streams_.push_back(
+        receiver_call_->CreateFlexfecReceiveStream(
+            flexfec_receive_configs_[i]));
+  }
+}
+
+void CallTest::CreateSendTransport(const BuiltInNetworkBehaviorConfig& config,
+                                   RtpRtcpObserver* observer) {
+  PacketReceiver* receiver =
+      receiver_call_ ? receiver_call_->Receiver() : nullptr;
+
+  auto network = std::make_unique<SimulatedNetwork>(config);
+  send_simulated_network_ = network.get();
+  send_transport_ = std::make_unique<PacketTransport>(
+      task_queue(), sender_call_.get(), observer,
+      test::PacketTransport::kSender, payload_type_map_,
+      std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
+                                        std::move(network), receiver),
+      rtp_extensions_, rtp_extensions_);
+}
+
+void CallTest::CreateReceiveTransport(
+    const BuiltInNetworkBehaviorConfig& config,
+    RtpRtcpObserver* observer) {
+  auto network = std::make_unique<SimulatedNetwork>(config);
+  receive_simulated_network_ = network.get();
+  receive_transport_ = std::make_unique<PacketTransport>(
+      task_queue(), nullptr, observer, test::PacketTransport::kReceiver,
+      payload_type_map_,
+      std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
+                                        std::move(network),
+                                        sender_call_->Receiver()),
+      rtp_extensions_, rtp_extensions_);
+}
+
+void CallTest::ConnectVideoSourcesToStreams() {
+  for (size_t i = 0; i < video_sources_.size(); ++i)
+    video_send_streams_[i]->SetSource(video_sources_[i].get(),
+                                      degradation_preference_);
+}
+
+void CallTest::Start() {
+  StartVideoStreams();
+  if (audio_send_stream_) {
+    audio_send_stream_->Start();
+  }
+  for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
+    audio_recv_stream->Start();
+}
+
+void CallTest::StartVideoStreams() {
+  for (size_t i = 0; i < video_send_streams_.size(); ++i) {
+    std::vector<bool> active_rtp_streams(
+        video_send_configs_[i].rtp.ssrcs.size(), true);
+    video_send_streams_[i]->StartPerRtpStream(active_rtp_streams);
+  }
+  for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
+    video_recv_stream->Start();
+}
+
+void CallTest::Stop() {
+  for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
+    audio_recv_stream->Stop();
+  if (audio_send_stream_) {
+    audio_send_stream_->Stop();
+  }
+  StopVideoStreams();
+}
+
+void CallTest::StopVideoStreams() {
+  for (VideoSendStream* video_send_stream : video_send_streams_)
+    video_send_stream->Stop();
+  for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
+    video_recv_stream->Stop();
+}
+
+void CallTest::DestroyStreams() {
+  if (audio_send_stream_)
+    sender_call_->DestroyAudioSendStream(audio_send_stream_);
+  audio_send_stream_ = nullptr;
+  for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_)
+    receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
+
+  DestroyVideoSendStreams();
+
+  for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_)
+    receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
+
+  for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
+    receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
+
+  video_receive_streams_.clear();
+  video_sources_.clear();
+}
+
+void CallTest::DestroyVideoSendStreams() {
+  for (VideoSendStream* video_send_stream : video_send_streams_)
+    sender_call_->DestroyVideoSendStream(video_send_stream);
+  video_send_streams_.clear();
+}
+
+void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
+  frame_generator_capturer_->SetFakeRotation(rotation);
+}
+
+void CallTest::SetVideoDegradation(DegradationPreference preference) {
+  GetVideoSendStream()->SetSource(frame_generator_capturer_, preference);
+}
+
+VideoSendStream::Config* CallTest::GetVideoSendConfig() {
+  return &video_send_configs_[0];
+}
+
+void CallTest::SetVideoSendConfig(const VideoSendStream::Config& config) {
+  video_send_configs_.clear();
+  video_send_configs_.push_back(config.Copy());
+}
+
+VideoEncoderConfig* CallTest::GetVideoEncoderConfig() {
+  return &video_encoder_configs_[0];
+}
+
+void CallTest::SetVideoEncoderConfig(const VideoEncoderConfig& config) {
+  video_encoder_configs_.clear();
+  video_encoder_configs_.push_back(config.Copy());
+}
+
+VideoSendStream* CallTest::GetVideoSendStream() {
+  return video_send_streams_[0];
+}
+FlexfecReceiveStream::Config* CallTest::GetFlexFecConfig() {
+  return &flexfec_receive_configs_[0];
+}
+
+void CallTest::OnRtpPacket(const RtpPacketReceived& packet) {
+  // All FlexFEC streams protect all of the video streams.
+  for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
+    flexfec_recv_stream->OnRtpPacket(packet);
+}
+
+absl::optional<RtpExtension> CallTest::GetRtpExtensionByUri(
+    const std::string& uri) const {
+  for (const auto& extension : rtp_extensions_) {
+    if (extension.uri == uri) {
+      return extension;
+    }
+  }
+  return absl::nullopt;
+}
+
+void CallTest::AddRtpExtensionByUri(
+    const std::string& uri,
+    std::vector<RtpExtension>* extensions) const {
+  const absl::optional<RtpExtension> extension = GetRtpExtensionByUri(uri);
+  if (extension) {
+    extensions->push_back(*extension);
+  }
+}
+
+constexpr size_t CallTest::kNumSsrcs;
+const int CallTest::kDefaultWidth;
+const int CallTest::kDefaultHeight;
+const int CallTest::kDefaultFramerate;
+const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {
+    0xBADCAFD, 0xBADCAFE, 0xBADCAFF, 0xBADCB00, 0xBADCB01, 0xBADCB02};
+const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {
+    0xC0FFED, 0xC0FFEE, 0xC0FFEF, 0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
+const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
+const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF;
+const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
+const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567;
+const int CallTest::kNackRtpHistoryMs = 1000;
+
+const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
+    {CallTest::kVideoSendPayloadType, MediaType::VIDEO},
+    {CallTest::kFakeVideoSendPayloadType, MediaType::VIDEO},
+    {CallTest::kSendRtxPayloadType, MediaType::VIDEO},
+    {CallTest::kPayloadTypeVP8, MediaType::VIDEO},
+    {CallTest::kPayloadTypeVP9, MediaType::VIDEO},
+    {CallTest::kPayloadTypeH264, MediaType::VIDEO},
+    {CallTest::kPayloadTypeGeneric, MediaType::VIDEO},
+    {CallTest::kRedPayloadType, MediaType::VIDEO},
+    {CallTest::kRtxRedPayloadType, MediaType::VIDEO},
+    {CallTest::kUlpfecPayloadType, MediaType::VIDEO},
+    {CallTest::kFlexfecPayloadType, MediaType::VIDEO},
+    {CallTest::kAudioSendPayloadType, MediaType::AUDIO}};
+
+BaseTest::BaseTest() {}
+
+BaseTest::BaseTest(TimeDelta timeout) : RtpRtcpObserver(timeout) {}
+
+BaseTest::~BaseTest() {}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer> BaseTest::CreateCapturer() {
+  return TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() {
+  return TestAudioDeviceModule::CreateDiscardRenderer(48000);
+}
+
+void BaseTest::OnFakeAudioDevicesCreated(
+    TestAudioDeviceModule* send_audio_device,
+    TestAudioDeviceModule* recv_audio_device) {}
+
+void BaseTest::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {}
+
+void BaseTest::ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) {
+}
+
+void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
+
+void BaseTest::OnTransportCreated(PacketTransport* to_receiver,
+                                  SimulatedNetworkInterface* sender_network,
+                                  PacketTransport* to_sender,
+                                  SimulatedNetworkInterface* receiver_network) {
+}
+
+BuiltInNetworkBehaviorConfig BaseTest::GetSendTransportConfig() const {
+  return BuiltInNetworkBehaviorConfig();
+}
+BuiltInNetworkBehaviorConfig BaseTest::GetReceiveTransportConfig() const {
+  return BuiltInNetworkBehaviorConfig();
+}
+size_t BaseTest::GetNumVideoStreams() const {
+  return 1;
+}
+
+size_t BaseTest::GetNumAudioStreams() const {
+  return 0;
+}
+
+size_t BaseTest::GetNumFlexfecStreams() const {
+  return 0;
+}
+
+void BaseTest::ModifyVideoConfigs(
+    VideoSendStream::Config* send_config,
+    std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
+    VideoEncoderConfig* encoder_config) {}
+
+void BaseTest::ModifyVideoCaptureStartResolution(int* width,
+                                                 int* heigt,
+                                                 int* frame_rate) {}
+
+void BaseTest::ModifyVideoDegradationPreference(
+    DegradationPreference* degradation_preference) {}
+
+void BaseTest::OnVideoStreamsCreated(
+    VideoSendStream* send_stream,
+    const std::vector<VideoReceiveStreamInterface*>& receive_streams) {}
+
+void BaseTest::ModifyAudioConfigs(
+    AudioSendStream::Config* send_config,
+    std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {}
+
+void BaseTest::OnAudioStreamsCreated(
+    AudioSendStream* send_stream,
+    const std::vector<AudioReceiveStreamInterface*>& receive_streams) {}
+
+void BaseTest::ModifyFlexfecConfigs(
+    std::vector<FlexfecReceiveStream::Config>* receive_configs) {}
+
+void BaseTest::OnFlexfecStreamsCreated(
+    const std::vector<FlexfecReceiveStream*>& receive_streams) {}
+
+void BaseTest::OnFrameGeneratorCapturerCreated(
+    FrameGeneratorCapturer* frame_generator_capturer) {}
+
+void BaseTest::OnStreamsStopped() {}
+
+SendTest::SendTest(TimeDelta timeout) : BaseTest(timeout) {}
+
+bool SendTest::ShouldCreateReceivers() const {
+  return false;
+}
+
+EndToEndTest::EndToEndTest() {}
+
+EndToEndTest::EndToEndTest(TimeDelta timeout) : BaseTest(timeout) {}
+
+bool EndToEndTest::ShouldCreateReceivers() const {
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/call_test.h b/third_party/libwebrtc/test/call_test.h
new file mode 100644
index 0000000000..3324bc4b5e
--- /dev/null
+++ b/third_party/libwebrtc/test/call_test.h
@@ -0,0 +1,359 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_CALL_TEST_H_
+#define TEST_CALL_TEST_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/simulated_network.h"
+#include "api/test/video/function_video_decoder_factory.h"
+#include "api/test/video/function_video_encoder_factory.h"
+#include "api/units/time_delta.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "call/call.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "test/encoder_settings.h"
+#include "test/fake_decoder.h"
+#include "test/fake_videorenderer.h"
+#include "test/fake_vp8_encoder.h"
+#include "test/frame_generator_capturer.h"
+#include "test/rtp_rtcp_observer.h"
+#include "test/run_loop.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+namespace test {
+
+class BaseTest;
+
+class CallTest : public ::testing::Test, public RtpPacketSinkInterface {
+ public:
+  CallTest();
+  virtual ~CallTest();
+
+  static constexpr size_t kNumSsrcs = 6;
+  static const int kNumSimulcastStreams = 3;
+  static const int kDefaultWidth = 320;
+  static const int kDefaultHeight = 180;
+  static const int kDefaultFramerate = 30;
+  static constexpr TimeDelta kDefaultTimeout = TimeDelta::Seconds(30);
+  static constexpr TimeDelta kLongTimeout = TimeDelta::Seconds(120);
+  enum classPayloadTypes : uint8_t {
+    kSendRtxPayloadType = 98,
+    kRtxRedPayloadType = 99,
+    kVideoSendPayloadType = 100,
+    kAudioSendPayloadType = 103,
+    kRedPayloadType = 118,
+    kUlpfecPayloadType = 119,
+    kFlexfecPayloadType = 120,
+    kPayloadTypeH264 = 122,
+    kPayloadTypeVP8 = 123,
+    kPayloadTypeVP9 = 124,
+    kPayloadTypeGeneric = 125,
+    kFakeVideoSendPayloadType = 126,
+  };
+  static const uint32_t kSendRtxSsrcs[kNumSsrcs];
+  static const uint32_t kVideoSendSsrcs[kNumSsrcs];
+  static const uint32_t kAudioSendSsrc;
+  static const uint32_t kFlexfecSendSsrc;
+  static const uint32_t kReceiverLocalVideoSsrc;
+  static const uint32_t kReceiverLocalAudioSsrc;
+  static const int kNackRtpHistoryMs;
+  static const std::map<uint8_t, MediaType> payload_type_map_;
+
+ protected:
+  void RegisterRtpExtension(const RtpExtension& extension);
+  // Returns header extensions that can be parsed by the transport.
+  rtc::ArrayView<const RtpExtension> GetRegisteredExtensions() {
+    return rtp_extensions_;
+  }
+
+  // RunBaseTest overwrites the audio_state of the send and receive Call configs
+  // to simplify test code.
+  void RunBaseTest(BaseTest* test);
+
+  void CreateCalls();
+  void CreateCalls(const Call::Config& sender_config,
+                   const Call::Config& receiver_config);
+  void CreateSenderCall();
+  void CreateSenderCall(const Call::Config& config);
+  void CreateReceiverCall(const Call::Config& config);
+  void DestroyCalls();
+
+  void CreateVideoSendConfig(VideoSendStream::Config* video_config,
+                             size_t num_video_streams,
+                             size_t num_used_ssrcs,
+                             Transport* send_transport);
+  void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
+                                    size_t num_flexfec_streams,
+                                    Transport* send_transport);
+  void SetAudioConfig(const AudioSendStream::Config& config);
+
+  void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
+  void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
+  void SetReceiveUlpFecConfig(
+      VideoReceiveStreamInterface::Config* receive_config);
+
+  void CreateSendConfig(size_t num_video_streams,
+                        size_t num_audio_streams,
+                        size_t num_flexfec_streams) {
+    CreateSendConfig(num_video_streams, num_audio_streams, num_flexfec_streams,
+                     send_transport_.get());
+  }
+  void CreateSendConfig(size_t num_video_streams,
+                        size_t num_audio_streams,
+                        size_t num_flexfec_streams,
+                        Transport* send_transport);
+
+  void CreateMatchingVideoReceiveConfigs(
+      const VideoSendStream::Config& video_send_config) {
+    CreateMatchingVideoReceiveConfigs(video_send_config,
+                                      receive_transport_.get());
+  }
+  void CreateMatchingVideoReceiveConfigs(
+      const VideoSendStream::Config& video_send_config,
+      Transport* rtcp_send_transport);
+  void CreateMatchingVideoReceiveConfigs(
+      const VideoSendStream::Config& video_send_config,
+      Transport* rtcp_send_transport,
+      VideoDecoderFactory* decoder_factory,
+      absl::optional<size_t> decode_sub_stream,
+      bool receiver_reference_time_report,
+      int rtp_history_ms);
+  void AddMatchingVideoReceiveConfigs(
+      std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
+      const VideoSendStream::Config& video_send_config,
+      Transport* rtcp_send_transport,
+      VideoDecoderFactory* decoder_factory,
+      absl::optional<size_t> decode_sub_stream,
+      bool receiver_reference_time_report,
+      int rtp_history_ms);
+
+  void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
+  void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
+  static AudioReceiveStreamInterface::Config CreateMatchingAudioConfig(
+      const AudioSendStream::Config& send_config,
+      rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
+      Transport* transport,
+      std::string sync_group);
+  void CreateMatchingFecConfig(
+      Transport* transport,
+      const VideoSendStream::Config& video_send_config);
+  void CreateMatchingReceiveConfigs() {
+    CreateMatchingReceiveConfigs(receive_transport_.get());
+  }
+  void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
+
+  void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
+                                             float speed,
+                                             int framerate,
+                                             int width,
+                                             int height);
+  void CreateFrameGeneratorCapturer(int framerate, int width, int height);
+  void CreateFakeAudioDevices(
+      std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+      std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
+
+  void CreateVideoStreams();
+  void CreateVideoSendStreams();
+  void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
+  void CreateAudioStreams();
+  void CreateFlexfecStreams();
+
+  // Receiver call must be created before calling CreateSendTransport in order
+  // to set a receiver.
+  // Rtp header extensions must be registered (RegisterRtpExtension(..)) before
+  // the transport is created in order for the receiving call object receive RTP
+  // packets with extensions.
+  void CreateSendTransport(const BuiltInNetworkBehaviorConfig& config,
+                           RtpRtcpObserver* observer);
+  void CreateReceiveTransport(const BuiltInNetworkBehaviorConfig& config,
+                              RtpRtcpObserver* observer);
+
+  void ConnectVideoSourcesToStreams();
+
+  void Start();
+  void StartVideoStreams();
+  void Stop();
+  void StopVideoStreams();
+  void DestroyStreams();
+  void DestroyVideoSendStreams();
+  void SetFakeVideoCaptureRotation(VideoRotation rotation);
+
+  void SetVideoDegradation(DegradationPreference preference);
+
+  VideoSendStream::Config* GetVideoSendConfig();
+  void SetVideoSendConfig(const VideoSendStream::Config& config);
+  VideoEncoderConfig* GetVideoEncoderConfig();
+  void SetVideoEncoderConfig(const VideoEncoderConfig& config);
+  VideoSendStream* GetVideoSendStream();
+  FlexfecReceiveStream::Config* GetFlexFecConfig();
+  TaskQueueBase* task_queue() { return task_queue_.get(); }
+
+  // RtpPacketSinkInterface implementation.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+  test::RunLoop loop_;
+
+  Clock* const clock_;
+  test::ScopedKeyValueConfig field_trials_;
+
+  std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
+  std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
+  std::unique_ptr<Call> sender_call_;
+  std::unique_ptr<PacketTransport> send_transport_;
+  SimulatedNetworkInterface* send_simulated_network_ = nullptr;
+  std::vector<VideoSendStream::Config> video_send_configs_;
+  std::vector<VideoEncoderConfig> video_encoder_configs_;
+  std::vector<VideoSendStream*> video_send_streams_;
+  AudioSendStream::Config audio_send_config_;
+  AudioSendStream* audio_send_stream_;
+
+  std::unique_ptr<Call> receiver_call_;
+  std::unique_ptr<PacketTransport> receive_transport_;
+  SimulatedNetworkInterface* receive_simulated_network_ = nullptr;
+  std::vector<VideoReceiveStreamInterface::Config> video_receive_configs_;
+  std::vector<VideoReceiveStreamInterface*> video_receive_streams_;
+  std::vector<AudioReceiveStreamInterface::Config> audio_receive_configs_;
+  std::vector<AudioReceiveStreamInterface*> audio_receive_streams_;
+  std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
+  std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
+
+  test::FrameGeneratorCapturer* frame_generator_capturer_;
+  std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
+      video_sources_;
+  DegradationPreference degradation_preference_ =
+      DegradationPreference::MAINTAIN_FRAMERATE;
+
+  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
+  std::unique_ptr<NetworkStatePredictorFactoryInterface>
+      network_state_predictor_factory_;
+  std::unique_ptr<NetworkControllerFactoryInterface>
+      network_controller_factory_;
+
+  test::FunctionVideoEncoderFactory fake_encoder_factory_;
+  int fake_encoder_max_bitrate_ = -1;
+  test::FunctionVideoDecoderFactory fake_decoder_factory_;
+  std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
+  // Number of simulcast substreams.
+  size_t num_video_streams_;
+  size_t num_audio_streams_;
+  size_t num_flexfec_streams_;
+  rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
+  rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
+  test::FakeVideoRenderer fake_renderer_;
+
+
+ private:
+  absl::optional<RtpExtension> GetRtpExtensionByUri(
+      const std::string& uri) const;
+
+  void AddRtpExtensionByUri(const std::string& uri,
+                            std::vector<RtpExtension>* extensions) const;
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
+  std::vector<RtpExtension> rtp_extensions_;
+  rtc::scoped_refptr<AudioProcessing> apm_send_;
+  rtc::scoped_refptr<AudioProcessing> apm_recv_;
+  rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
+  rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
+};
+
+class BaseTest : public RtpRtcpObserver {
+ public:
+  BaseTest();
+  explicit BaseTest(TimeDelta timeout);
+  virtual ~BaseTest();
+
+  virtual void PerformTest() = 0;
+  virtual bool ShouldCreateReceivers() const = 0;
+
+  virtual size_t GetNumVideoStreams() const;
+  virtual size_t GetNumAudioStreams() const;
+  virtual size_t GetNumFlexfecStreams() const;
+
+  virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
+  virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
+  virtual void OnFakeAudioDevicesCreated(
+      TestAudioDeviceModule* send_audio_device,
+      TestAudioDeviceModule* recv_audio_device);
+
+  virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
+  virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
+
+  virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
+  virtual void OnTransportCreated(PacketTransport* to_receiver,
+                                  SimulatedNetworkInterface* sender_network,
+                                  PacketTransport* to_sender,
+                                  SimulatedNetworkInterface* receiver_network);
+
+  virtual BuiltInNetworkBehaviorConfig GetSendTransportConfig() const;
+  virtual BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const;
+
+  virtual void ModifyVideoConfigs(
+      VideoSendStream::Config* send_config,
+      std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
+      VideoEncoderConfig* encoder_config);
+  virtual void ModifyVideoCaptureStartResolution(int* width,
+                                                 int* heigt,
+                                                 int* frame_rate);
+  virtual void ModifyVideoDegradationPreference(
+      DegradationPreference* degradation_preference);
+
+  virtual void OnVideoStreamsCreated(
+      VideoSendStream* send_stream,
+      const std::vector<VideoReceiveStreamInterface*>& receive_streams);
+
+  virtual void ModifyAudioConfigs(
+      AudioSendStream::Config* send_config,
+      std::vector<AudioReceiveStreamInterface::Config>* receive_configs);
+  virtual void OnAudioStreamsCreated(
+      AudioSendStream* send_stream,
+      const std::vector<AudioReceiveStreamInterface*>& receive_streams);
+
+  virtual void ModifyFlexfecConfigs(
+      std::vector<FlexfecReceiveStream::Config>* receive_configs);
+  virtual void OnFlexfecStreamsCreated(
+      const std::vector<FlexfecReceiveStream*>& receive_streams);
+
+  virtual void OnFrameGeneratorCapturerCreated(
+      FrameGeneratorCapturer* frame_generator_capturer);
+
+  virtual void OnStreamsStopped();
+};
+
+class SendTest : public BaseTest {
+ public:
+  explicit SendTest(TimeDelta timeout);
+
+  bool ShouldCreateReceivers() const override;
+};
+
+class EndToEndTest : public BaseTest {
+ public:
+  EndToEndTest();
+  explicit EndToEndTest(TimeDelta timeout);
+
+  bool ShouldCreateReceivers() const override;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_CALL_TEST_H_
diff --git a/third_party/libwebrtc/test/configurable_frame_size_encoder.cc b/third_party/libwebrtc/test/configurable_frame_size_encoder.cc
new file mode 100644
index 0000000000..e3965ef770
--- /dev/null
+++ b/third_party/libwebrtc/test/configurable_frame_size_encoder.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/configurable_frame_size_encoder.h"
+
+#include <string.h>
+
+#include <cstdint>
+#include <type_traits>
+#include <utility>
+
+#include "api/video/encoded_image.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+ConfigurableFrameSizeEncoder::ConfigurableFrameSizeEncoder(
+    size_t max_frame_size)
+    : callback_(NULL),
+      current_frame_size_(max_frame_size),
+      codec_type_(kVideoCodecGeneric) {}
+
+ConfigurableFrameSizeEncoder::~ConfigurableFrameSizeEncoder() {}
+
+void ConfigurableFrameSizeEncoder::SetFecControllerOverride(
+    FecControllerOverride* fec_controller_override) {
+  // Ignored.
+}
+
+int32_t ConfigurableFrameSizeEncoder::InitEncode(
+    const VideoCodec* codec_settings,
+    const Settings& settings) {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::Encode(
+    const VideoFrame& inputImage,
+    const std::vector<VideoFrameType>* frame_types) {
+  EncodedImage encodedImage;
+  auto buffer = EncodedImageBuffer::Create(current_frame_size_);
+  memset(buffer->data(), 0, current_frame_size_);
+  encodedImage.SetEncodedData(buffer);
+  encodedImage._encodedHeight = inputImage.height();
+  encodedImage._encodedWidth = inputImage.width();
+  encodedImage._frameType = VideoFrameType::kVideoFrameKey;
+  encodedImage.SetTimestamp(inputImage.timestamp());
+  encodedImage.capture_time_ms_ = inputImage.render_time_ms();
+  CodecSpecificInfo specific{};
+  specific.codecType = codec_type_;
+  callback_->OnEncodedImage(encodedImage, &specific);
+  if (post_encode_callback_) {
+    (*post_encode_callback_)();
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t ConfigurableFrameSizeEncoder::Release() {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void ConfigurableFrameSizeEncoder::SetRates(
+    const RateControlParameters& parameters) {}
+
+int32_t ConfigurableFrameSizeEncoder::SetFrameSize(size_t size) {
+  current_frame_size_ = size;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void ConfigurableFrameSizeEncoder::SetCodecType(VideoCodecType codec_type) {
+  codec_type_ = codec_type;
+}
+
+void ConfigurableFrameSizeEncoder::RegisterPostEncodeCallback(
+    std::function<void(void)> post_encode_callback) {
+  post_encode_callback_ = std::move(post_encode_callback);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/configurable_frame_size_encoder.h b/third_party/libwebrtc/test/configurable_frame_size_encoder.h
new file mode 100644
index 0000000000..8dd5157b5b
--- /dev/null
+++ b/third_party/libwebrtc/test/configurable_frame_size_encoder.h
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_
+#define TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <functional>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+namespace test {
+
+class ConfigurableFrameSizeEncoder : public VideoEncoder {
+ public:
+  explicit ConfigurableFrameSizeEncoder(size_t max_frame_size);
+  ~ConfigurableFrameSizeEncoder() override;
+
+  void SetFecControllerOverride(
+      FecControllerOverride* fec_controller_override) override;
+
+  int32_t InitEncode(const VideoCodec* codec_settings,
+                     const Settings& settings) override;
+
+  int32_t Encode(const VideoFrame& input_image,
+                 const std::vector<VideoFrameType>* frame_types) override;
+
+  int32_t RegisterEncodeCompleteCallback(
+      EncodedImageCallback* callback) override;
+
+  int32_t Release() override;
+
+  void SetRates(const RateControlParameters& parameters) override;
+
+  int32_t SetFrameSize(size_t size);
+
+  void SetCodecType(VideoCodecType codec_type_);
+
+  void RegisterPostEncodeCallback(
+      std::function<void(void)> post_encode_callback);
+
+ private:
+  EncodedImageCallback* callback_;
+  absl::optional<std::function<void(void)>> post_encode_callback_;
+
+  size_t current_frame_size_;
+  VideoCodecType codec_type_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_CONFIGURABLE_FRAME_SIZE_ENCODER_H_
diff --git a/third_party/libwebrtc/test/direct_transport.cc b/third_party/libwebrtc/test/direct_transport.cc
new file mode 100644
index 0000000000..260497947c
--- /dev/null
+++ b/third_party/libwebrtc/test/direct_transport.cc
@@ -0,0 +1,145 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/direct_transport.h"
+
+#include "api/media_types.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "call/call.h"
+#include "call/fake_network_pipe.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+Demuxer::Demuxer(const std::map<uint8_t, MediaType>& payload_type_map)
+    : payload_type_map_(payload_type_map) {}
+
+MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
+                                const size_t packet_length) const {
+  if (IsRtpPacket(rtc::MakeArrayView(packet_data, packet_length))) {
+    RTC_CHECK_GE(packet_length, 2);
+    const uint8_t payload_type = packet_data[1] & 0x7f;
+    std::map<uint8_t, MediaType>::const_iterator it =
+        payload_type_map_.find(payload_type);
+    RTC_CHECK(it != payload_type_map_.end())
+        << "payload type " << static_cast<int>(payload_type) << " unknown.";
+    return it->second;
+  }
+  return MediaType::ANY;
+}
+
+DirectTransport::DirectTransport(
+    TaskQueueBase* task_queue,
+    std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+    Call* send_call,
+    const std::map<uint8_t, MediaType>& payload_type_map,
+    rtc::ArrayView<const RtpExtension> audio_extensions,
+    rtc::ArrayView<const RtpExtension> video_extensions)
+    : send_call_(send_call),
+      task_queue_(task_queue),
+      demuxer_(payload_type_map),
+      fake_network_(std::move(pipe)),
+      audio_extensions_(audio_extensions),
+      video_extensions_(video_extensions) {
+  Start();
+}
+
+DirectTransport::~DirectTransport() {
+  next_process_task_.Stop();
+}
+
+void DirectTransport::SetReceiver(PacketReceiver* receiver) {
+  fake_network_->SetReceiver(receiver);
+}
+
+bool DirectTransport::SendRtp(const uint8_t* data,
+                              size_t length,
+                              const PacketOptions& options) {
+  if (send_call_) {
+    rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
+    sent_packet.info.included_in_feedback = options.included_in_feedback;
+    sent_packet.info.included_in_allocation = options.included_in_allocation;
+    sent_packet.info.packet_size_bytes = length;
+    sent_packet.info.packet_type = rtc::PacketType::kData;
+    send_call_->OnSentPacket(sent_packet);
+  }
+
+  const RtpHeaderExtensionMap* extensions = nullptr;
+  MediaType media_type = demuxer_.GetMediaType(data, length);
+  switch (demuxer_.GetMediaType(data, length)) {
+    case webrtc::MediaType::AUDIO:
+      extensions = &audio_extensions_;
+      break;
+    case webrtc::MediaType::VIDEO:
+      extensions = &video_extensions_;
+      break;
+    default:
+      RTC_CHECK_NOTREACHED();
+  }
+  RtpPacketReceived packet(extensions, Timestamp::Micros(rtc::TimeMicros()));
+  if (media_type == MediaType::VIDEO) {
+    packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+  }
+  RTC_CHECK(packet.Parse(rtc::CopyOnWriteBuffer(data, length)));
+  fake_network_->DeliverRtpPacket(
+      media_type, std::move(packet),
+      [](const RtpPacketReceived& packet) { return false; });
+
+  MutexLock lock(&process_lock_);
+  if (!next_process_task_.Running())
+    ProcessPackets();
+  return true;
+}
+
+bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
+  fake_network_->DeliverRtcpPacket(rtc::CopyOnWriteBuffer(data, length));
+  MutexLock lock(&process_lock_);
+  if (!next_process_task_.Running())
+    ProcessPackets();
+  return true;
+}
+
+int DirectTransport::GetAverageDelayMs() {
+  return fake_network_->AverageDelay();
+}
+
+void DirectTransport::Start() {
+  RTC_DCHECK(task_queue_);
+  if (send_call_) {
+    send_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+    send_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
+  }
+}
+
+void DirectTransport::ProcessPackets() {
+  absl::optional<int64_t> initial_delay_ms =
+      fake_network_->TimeUntilNextProcess();
+  if (initial_delay_ms == absl::nullopt)
+    return;
+
+  next_process_task_ = RepeatingTaskHandle::DelayedStart(
+      task_queue_, TimeDelta::Millis(*initial_delay_ms), [this] {
+        fake_network_->Process();
+        if (auto delay_ms = fake_network_->TimeUntilNextProcess())
+          return TimeDelta::Millis(*delay_ms);
+        // Otherwise stop the task.
+        MutexLock lock(&process_lock_);
+        next_process_task_.Stop();
+        // Since this task is stopped, return value doesn't matter.
+        return TimeDelta::Zero();
+      });
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/direct_transport.h b/third_party/libwebrtc/test/direct_transport.h
new file mode 100644
index 0000000000..468e339c0a
--- /dev/null
+++ b/third_party/libwebrtc/test/direct_transport.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_DIRECT_TRANSPORT_H_
+#define TEST_DIRECT_TRANSPORT_H_
+
+#include <memory>
+
+#include "api/call/transport.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/simulated_network.h"
+#include "call/call.h"
+#include "call/simulated_packet_receiver.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+class PacketReceiver;
+
+namespace test {
+class Demuxer {
+ public:
+  explicit Demuxer(const std::map<uint8_t, MediaType>& payload_type_map);
+  ~Demuxer() = default;
+
+  Demuxer(const Demuxer&) = delete;
+  Demuxer& operator=(const Demuxer&) = delete;
+
+  MediaType GetMediaType(const uint8_t* packet_data,
+                         size_t packet_length) const;
+  const std::map<uint8_t, MediaType> payload_type_map_;
+};
+
+// Objects of this class are expected to be allocated and destroyed  on the
+// same task-queue - the one that's passed in via the constructor.
+class DirectTransport : public Transport {
+ public:
+  DirectTransport(TaskQueueBase* task_queue,
+                  std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+                  Call* send_call,
+                  const std::map<uint8_t, MediaType>& payload_type_map,
+                  rtc::ArrayView<const RtpExtension> audio_extensions,
+                  rtc::ArrayView<const RtpExtension> video_extensions);
+
+  ~DirectTransport() override;
+
+  // TODO(holmer): Look into moving this to the constructor.
+  virtual void SetReceiver(PacketReceiver* receiver);
+
+  bool SendRtp(const uint8_t* data,
+               size_t length,
+               const PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* data, size_t length) override;
+
+  int GetAverageDelayMs();
+
+ private:
+  void ProcessPackets() RTC_EXCLUSIVE_LOCKS_REQUIRED(&process_lock_);
+  void LegacySendPacket(const uint8_t* data, size_t length);
+  void Start();
+
+  Call* const send_call_;
+
+  TaskQueueBase* const task_queue_;
+
+  Mutex process_lock_;
+  RepeatingTaskHandle next_process_task_ RTC_GUARDED_BY(&process_lock_);
+
+  const Demuxer demuxer_;
+  const std::unique_ptr<SimulatedPacketReceiverInterface> fake_network_;
+  const RtpHeaderExtensionMap audio_extensions_;
+  const RtpHeaderExtensionMap video_extensions_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_DIRECT_TRANSPORT_H_
diff --git a/third_party/libwebrtc/test/direct_transport_unittest.cc b/third_party/libwebrtc/test/direct_transport_unittest.cc
new file mode 100644
index 0000000000..ab00971089
--- /dev/null
+++ b/third_party/libwebrtc/test/direct_transport_unittest.cc
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/direct_transport.h"
+
+#include <string.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+TEST(DemuxerTest, Demuxing) {
+  constexpr uint8_t kVideoPayloadType = 100;
+  constexpr uint8_t kAudioPayloadType = 101;
+  constexpr size_t kPacketSize = 12;
+  Demuxer demuxer({{kVideoPayloadType, MediaType::VIDEO},
+                   {kAudioPayloadType, MediaType::AUDIO}});
+
+  uint8_t data[kPacketSize];
+  memset(data, 0, kPacketSize);
+  data[0] = 0x80;
+  data[1] = kVideoPayloadType;
+  EXPECT_EQ(demuxer.GetMediaType(data, kPacketSize), MediaType::VIDEO);
+  data[1] = kAudioPayloadType;
+  EXPECT_EQ(demuxer.GetMediaType(data, kPacketSize), MediaType::AUDIO);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/drifting_clock.cc b/third_party/libwebrtc/test/drifting_clock.cc
new file mode 100644
index 0000000000..47c8e56916
--- /dev/null
+++ b/third_party/libwebrtc/test/drifting_clock.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/drifting_clock.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+constexpr float DriftingClock::kNoDrift;
+
+DriftingClock::DriftingClock(Clock* clock, float speed)
+    : clock_(clock), drift_(speed - 1.0f), start_time_(clock_->CurrentTime()) {
+  RTC_CHECK(clock);
+  RTC_CHECK_GT(speed, 0.0f);
+}
+
+TimeDelta DriftingClock::Drift() const {
+  auto now = clock_->CurrentTime();
+  RTC_DCHECK_GE(now, start_time_);
+  return (now - start_time_) * drift_;
+}
+
+Timestamp DriftingClock::Drift(Timestamp timestamp) const {
+  return timestamp + Drift() / 1000.;
+}
+
+NtpTime DriftingClock::Drift(NtpTime ntp_time) const {
+  // NTP precision is 1/2^32 seconds, i.e. 2^32 ntp fractions = 1 second.
+  const double kNtpFracPerMicroSecond = 4294.967296;  // = 2^32 / 10^6
+
+  uint64_t total_fractions = static_cast<uint64_t>(ntp_time);
+  total_fractions += Drift().us() * kNtpFracPerMicroSecond;
+  return NtpTime(total_fractions);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/drifting_clock.h b/third_party/libwebrtc/test/drifting_clock.h
new file mode 100644
index 0000000000..4a2500ba30
--- /dev/null
+++ b/third_party/libwebrtc/test/drifting_clock.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_DRIFTING_CLOCK_H_
+#define TEST_DRIFTING_CLOCK_H_
+
+#include <stdint.h>
+
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/ntp_time.h"
+
+namespace webrtc {
+namespace test {
+class DriftingClock : public Clock {
+ public:
+  static constexpr float kNoDrift = 1.0f;
+
+  DriftingClock(Clock* clock, float speed);
+
+  static constexpr float PercentsFaster(float percent) {
+    return 1.0f + percent / 100.0f;
+  }
+  static constexpr float PercentsSlower(float percent) {
+    return 1.0f - percent / 100.0f;
+  }
+
+  Timestamp CurrentTime() override { return Drift(clock_->CurrentTime()); }
+  NtpTime ConvertTimestampToNtpTime(Timestamp timestamp) override {
+    return Drift(clock_->ConvertTimestampToNtpTime(timestamp));
+  }
+
+ private:
+  TimeDelta Drift() const;
+  Timestamp Drift(Timestamp timestamp) const;
+  NtpTime Drift(NtpTime ntp_time) const;
+
+  Clock* const clock_;
+  const float drift_;
+  const Timestamp start_time_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_DRIFTING_CLOCK_H_
diff --git a/third_party/libwebrtc/test/encoder_settings.cc b/third_party/libwebrtc/test/encoder_settings.cc
new file mode 100644
index 0000000000..f5b298b107
--- /dev/null
+++ b/third_party/libwebrtc/test/encoder_settings.cc
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/encoder_settings.h"
+
+#include <algorithm>
+
+#include "api/scoped_refptr.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "call/rtp_config.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+const size_t DefaultVideoStreamFactory::kMaxNumberOfStreams;
+const int DefaultVideoStreamFactory::kMaxBitratePerStream[] = {150000, 450000,
+                                                               1500000};
+const int DefaultVideoStreamFactory::kDefaultMinBitratePerStream[] = {
+    30000, 200000, 700000};
+
+// static
+std::vector<VideoStream> CreateVideoStreams(
+    int width,
+    int height,
+    const webrtc::VideoEncoderConfig& encoder_config) {
+  RTC_DCHECK(encoder_config.number_of_streams <=
+             DefaultVideoStreamFactory::kMaxNumberOfStreams);
+
+  std::vector<VideoStream> stream_settings(encoder_config.number_of_streams);
+
+  int bitrate_left_bps = 0;
+  if (encoder_config.max_bitrate_bps > 0) {
+    bitrate_left_bps = encoder_config.max_bitrate_bps;
+  } else {
+    for (size_t stream_num = 0; stream_num < encoder_config.number_of_streams;
+         ++stream_num) {
+      bitrate_left_bps +=
+          DefaultVideoStreamFactory::kMaxBitratePerStream[stream_num];
+    }
+  }
+
+  for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
+    stream_settings[i].width =
+        (i + 1) * width / encoder_config.number_of_streams;
+    stream_settings[i].height =
+        (i + 1) * height / encoder_config.number_of_streams;
+    stream_settings[i].max_framerate = 30;
+    stream_settings[i].max_qp = 56;
+    stream_settings[i].min_bitrate_bps =
+        DefaultVideoStreamFactory::kDefaultMinBitratePerStream[i];
+
+    // Use configured values instead of default values if set.
+    const VideoStream stream = (i < encoder_config.simulcast_layers.size())
+                                   ? encoder_config.simulcast_layers[i]
+                                   : VideoStream();
+
+    int max_bitrate_bps =
+        stream.max_bitrate_bps > 0
+            ? stream.max_bitrate_bps
+            : DefaultVideoStreamFactory::kMaxBitratePerStream[i];
+    max_bitrate_bps = std::min(bitrate_left_bps, max_bitrate_bps);
+
+    int target_bitrate_bps = stream.target_bitrate_bps > 0
+                                 ? stream.target_bitrate_bps
+                                 : max_bitrate_bps;
+    target_bitrate_bps = std::min(max_bitrate_bps, target_bitrate_bps);
+
+    if (stream.min_bitrate_bps > 0) {
+      RTC_DCHECK_LE(stream.min_bitrate_bps, target_bitrate_bps);
+      stream_settings[i].min_bitrate_bps = stream.min_bitrate_bps;
+    }
+    if (stream.max_framerate > 0) {
+      stream_settings[i].max_framerate = stream.max_framerate;
+    }
+    if (stream.num_temporal_layers) {
+      RTC_DCHECK_GE(*stream.num_temporal_layers, 1);
+      stream_settings[i].num_temporal_layers = stream.num_temporal_layers;
+    }
+    if (stream.scale_resolution_down_by >= 1.0) {
+      stream_settings[i].width = width / stream.scale_resolution_down_by;
+      stream_settings[i].height = height / stream.scale_resolution_down_by;
+    }
+    stream_settings[i].scalability_mode = stream.scalability_mode;
+    stream_settings[i].target_bitrate_bps = target_bitrate_bps;
+    stream_settings[i].max_bitrate_bps = max_bitrate_bps;
+    stream_settings[i].active =
+        encoder_config.number_of_streams == 1 || stream.active;
+
+    bitrate_left_bps -= stream_settings[i].target_bitrate_bps;
+  }
+
+  stream_settings[encoder_config.number_of_streams - 1].max_bitrate_bps +=
+      bitrate_left_bps;
+  stream_settings[0].bitrate_priority = encoder_config.bitrate_priority;
+
+  return stream_settings;
+}
+
+DefaultVideoStreamFactory::DefaultVideoStreamFactory() {}
+
+std::vector<VideoStream> DefaultVideoStreamFactory::CreateEncoderStreams(
+    int frame_width,
+    int frame_height,
+    const webrtc::VideoEncoderConfig& encoder_config) {
+  return CreateVideoStreams(frame_width, frame_height, encoder_config);
+}
+
+void FillEncoderConfiguration(VideoCodecType codec_type,
+                              size_t num_streams,
+                              VideoEncoderConfig* configuration) {
+  RTC_DCHECK_LE(num_streams, DefaultVideoStreamFactory::kMaxNumberOfStreams);
+
+  configuration->codec_type = codec_type;
+  configuration->number_of_streams = num_streams;
+  configuration->video_stream_factory =
+      rtc::make_ref_counted<DefaultVideoStreamFactory>();
+  configuration->max_bitrate_bps = 0;
+  configuration->frame_drop_enabled = true;
+  configuration->simulcast_layers = std::vector<VideoStream>(num_streams);
+  for (size_t i = 0; i < num_streams; ++i) {
+    configuration->max_bitrate_bps +=
+        DefaultVideoStreamFactory::kMaxBitratePerStream[i];
+  }
+}
+
+VideoReceiveStreamInterface::Decoder CreateMatchingDecoder(
+    int payload_type,
+    const std::string& payload_name) {
+  VideoReceiveStreamInterface::Decoder decoder;
+  decoder.payload_type = payload_type;
+  decoder.video_format = SdpVideoFormat(payload_name);
+  return decoder;
+}
+
+VideoReceiveStreamInterface::Decoder CreateMatchingDecoder(
+    const VideoSendStream::Config& config) {
+  return CreateMatchingDecoder(config.rtp.payload_type,
+                               config.rtp.payload_name);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/encoder_settings.h b/third_party/libwebrtc/test/encoder_settings.h
new file mode 100644
index 0000000000..6dbad0fee2
--- /dev/null
+++ b/third_party/libwebrtc/test/encoder_settings.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_ENCODER_SETTINGS_H_
+#define TEST_ENCODER_SETTINGS_H_
+
+#include <stddef.h>
+
+#include <string>
+#include <vector>
+
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "video/config/video_encoder_config.h"
+
+namespace webrtc {
+namespace test {
+
+class DefaultVideoStreamFactory
+    : public VideoEncoderConfig::VideoStreamFactoryInterface {
+ public:
+  DefaultVideoStreamFactory();
+
+  static const size_t kMaxNumberOfStreams = 3;
+  // Defined as {150000, 450000, 1500000};
+  static const int kMaxBitratePerStream[];
+  // Defined as {50000, 200000, 700000};
+  static const int kDefaultMinBitratePerStream[];
+
+ private:
+  std::vector<VideoStream> CreateEncoderStreams(
+      int frame_width,
+      int frame_height,
+      const webrtc::VideoEncoderConfig& encoder_config) override;
+};
+
+// Creates `encoder_config.number_of_streams` VideoStreams where index
+// `encoder_config.number_of_streams -1` have width = `width`, height =
+// `height`. The total max bitrate of all VideoStreams is
+// `encoder_config.max_bitrate_bps`.
+std::vector<VideoStream> CreateVideoStreams(
+    int width,
+    int height,
+    const webrtc::VideoEncoderConfig& encoder_config);
+
+void FillEncoderConfiguration(VideoCodecType codec_type,
+                              size_t num_streams,
+                              VideoEncoderConfig* configuration);
+
+VideoReceiveStreamInterface::Decoder CreateMatchingDecoder(
+    int payload_type,
+    const std::string& payload_name);
+
+VideoReceiveStreamInterface::Decoder CreateMatchingDecoder(
+    const VideoSendStream::Config& config);
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_ENCODER_SETTINGS_H_
diff --git a/third_party/libwebrtc/test/explicit_key_value_config.cc b/third_party/libwebrtc/test/explicit_key_value_config.cc
new file mode 100644
index 0000000000..90690c0514
--- /dev/null
+++ b/third_party/libwebrtc/test/explicit_key_value_config.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/explicit_key_value_config.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+ExplicitKeyValueConfig::ExplicitKeyValueConfig(absl::string_view s) {
+  std::string::size_type field_start = 0;
+  while (field_start < s.size()) {
+    std::string::size_type separator_pos = s.find('/', field_start);
+    RTC_CHECK_NE(separator_pos, std::string::npos)
+        << "Missing separator '/' after field trial key.";
+    RTC_CHECK_GT(separator_pos, field_start)
+        << "Field trial key cannot be empty.";
+    std::string key(s.substr(field_start, separator_pos - field_start));
+    field_start = separator_pos + 1;
+
+    RTC_CHECK_LT(field_start, s.size())
+        << "Missing value after field trial key. String ended.";
+    separator_pos = s.find('/', field_start);
+    RTC_CHECK_NE(separator_pos, std::string::npos)
+        << "Missing terminating '/' in field trial string.";
+    RTC_CHECK_GT(separator_pos, field_start)
+        << "Field trial value cannot be empty.";
+    std::string value(s.substr(field_start, separator_pos - field_start));
+    field_start = separator_pos + 1;
+
+    key_value_map_[key] = value;
+  }
+  // This check is technically redundant due to earlier checks.
+  // We nevertheless keep the check to make it clear that the entire
+  // string has been processed, and without indexing past the end.
+  RTC_CHECK_EQ(field_start, s.size());
+}
+
+std::string ExplicitKeyValueConfig::GetValue(absl::string_view key) const {
+  auto it = key_value_map_.find(key);
+  if (it != key_value_map_.end())
+    return it->second;
+  return "";
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/explicit_key_value_config.h b/third_party/libwebrtc/test/explicit_key_value_config.h
new file mode 100644
index 0000000000..f14a10432c
--- /dev/null
+++ b/third_party/libwebrtc/test/explicit_key_value_config.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_EXPLICIT_KEY_VALUE_CONFIG_H_
+#define TEST_EXPLICIT_KEY_VALUE_CONFIG_H_
+
+#include <functional>
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/field_trials_registry.h"
+
+namespace webrtc {
+namespace test {
+
+class ExplicitKeyValueConfig : public FieldTrialsRegistry {
+ public:
+  explicit ExplicitKeyValueConfig(absl::string_view s);
+
+ private:
+  std::string GetValue(absl::string_view key) const override;
+
+  // Unlike std::less<std::string>, std::less<> is transparent and allows
+  // heterogeneous lookup directly with absl::string_view.
+  std::map<std::string, std::string, std::less<>> key_value_map_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_EXPLICIT_KEY_VALUE_CONFIG_H_
diff --git a/third_party/libwebrtc/test/fake_decoder.cc b/third_party/libwebrtc/test/fake_decoder.cc
new file mode 100644
index 0000000000..53fce37de1
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_decoder.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fake_decoder.h"
+
+#include <string.h>
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+FakeDecoder::FakeDecoder() : FakeDecoder(nullptr) {}
+
+FakeDecoder::FakeDecoder(TaskQueueFactory* task_queue_factory)
+    : callback_(nullptr),
+      width_(kDefaultWidth),
+      height_(kDefaultHeight),
+      task_queue_factory_(task_queue_factory),
+      decode_delay_ms_(0) {}
+
+bool FakeDecoder::Configure(const Settings& settings) {
+  return true;
+}
+
+int32_t FakeDecoder::Decode(const EncodedImage& input,
+                            bool missing_frames,
+                            int64_t render_time_ms) {
+  if (input._encodedWidth > 0 && input._encodedHeight > 0) {
+    width_ = input._encodedWidth;
+    height_ = input._encodedHeight;
+  }
+
+  rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(width_, height_);
+  I420Buffer::SetBlack(buffer.get());
+  VideoFrame frame = VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rotation(webrtc::kVideoRotation_0)
+                         .set_timestamp_ms(render_time_ms)
+                         .build();
+  frame.set_timestamp(input.Timestamp());
+  frame.set_ntp_time_ms(input.ntp_time_ms_);
+
+  if (decode_delay_ms_ == 0 || !task_queue_) {
+    callback_->Decoded(frame);
+  } else {
+    task_queue_->PostDelayedHighPrecisionTask(
+        [frame, this]() {
+          VideoFrame copy = frame;
+          callback_->Decoded(copy);
+        },
+        TimeDelta::Millis(decode_delay_ms_));
+  }
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+void FakeDecoder::SetDelayedDecoding(int decode_delay_ms) {
+  RTC_CHECK(task_queue_factory_);
+  if (!task_queue_) {
+    task_queue_ = task_queue_factory_->CreateTaskQueue(
+        "fake_decoder", TaskQueueFactory::Priority::NORMAL);
+  }
+  decode_delay_ms_ = decode_delay_ms;
+}
+
+int32_t FakeDecoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeDecoder::Release() {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+const char* FakeDecoder::kImplementationName = "fake_decoder";
+VideoDecoder::DecoderInfo FakeDecoder::GetDecoderInfo() const {
+  DecoderInfo info;
+  info.implementation_name = kImplementationName;
+  info.is_hardware_accelerated = true;
+  return info;
+}
+const char* FakeDecoder::ImplementationName() const {
+  return kImplementationName;
+}
+
+int32_t FakeH264Decoder::Decode(const EncodedImage& input,
+                                bool missing_frames,
+                                int64_t render_time_ms) {
+  uint8_t value = 0;
+  for (size_t i = 0; i < input.size(); ++i) {
+    uint8_t kStartCode[] = {0, 0, 0, 1};
+    if (i < input.size() - sizeof(kStartCode) &&
+        !memcmp(&input.data()[i], kStartCode, sizeof(kStartCode))) {
+      i += sizeof(kStartCode) + 1;  // Skip start code and NAL header.
+    }
+    if (input.data()[i] != value) {
+      RTC_CHECK_EQ(value, input.data()[i])
+          << "Bitstream mismatch between sender and receiver.";
+      return -1;
+    }
+    ++value;
+  }
+  return FakeDecoder::Decode(input, missing_frames, render_time_ms);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fake_decoder.h b/third_party/libwebrtc/test/fake_decoder.h
new file mode 100644
index 0000000000..cea92b49be
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_decoder.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FAKE_DECODER_H_
+#define TEST_FAKE_DECODER_H_
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeDecoder : public VideoDecoder {
+ public:
+  enum { kDefaultWidth = 320, kDefaultHeight = 180 };
+
+  FakeDecoder();
+  explicit FakeDecoder(TaskQueueFactory* task_queue_factory);
+  virtual ~FakeDecoder() {}
+
+  bool Configure(const Settings& settings) override;
+
+  int32_t Decode(const EncodedImage& input,
+                 bool missing_frames,
+                 int64_t render_time_ms) override;
+
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+
+  int32_t Release() override;
+
+  DecoderInfo GetDecoderInfo() const override;
+  const char* ImplementationName() const override;
+
+  static const char* kImplementationName;
+
+  void SetDelayedDecoding(int decode_delay_ms);
+
+ private:
+  DecodedImageCallback* callback_;
+  int width_;
+  int height_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
+  TaskQueueFactory* task_queue_factory_;
+  int decode_delay_ms_;
+};
+
+class FakeH264Decoder : public FakeDecoder {
+ public:
+  virtual ~FakeH264Decoder() {}
+
+  int32_t Decode(const EncodedImage& input,
+                 bool missing_frames,
+                 int64_t render_time_ms) override;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FAKE_DECODER_H_
diff --git a/third_party/libwebrtc/test/fake_encoded_frame.cc b/third_party/libwebrtc/test/fake_encoded_frame.cc
new file mode 100644
index 0000000000..32fa5d8ccf
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_encoded_frame.cc
@@ -0,0 +1,144 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fake_encoded_frame.h"
+
+#include <memory>
+
+#include "api/video/video_frame_type.h"
+
+namespace webrtc {
+
+void PrintTo(const EncodedFrame& frame,
+             std::ostream* os) /* no-presubmit-check TODO(webrtc:8982) */ {
+  *os << "EncodedFrame with id=" << frame.Id() << " rtp=" << frame.Timestamp()
+      << " size=" << frame.size() << " refs=[";
+  for (size_t ref = 0; ref < frame.num_references; ++ref) {
+    *os << frame.references[ref] << ",";
+  }
+  *os << "]";
+}
+
+namespace test {
+
+int64_t FakeEncodedFrame::ReceivedTime() const {
+  return received_time_;
+}
+
+int64_t FakeEncodedFrame::RenderTime() const {
+  return _renderTimeMs;
+}
+
+void FakeEncodedFrame::SetReceivedTime(int64_t received_time) {
+  received_time_ = received_time;
+}
+
+void FakeEncodedFrame::SetPayloadType(int payload_type) {
+  _payloadType = payload_type;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::Time(uint32_t rtp_timestamp) {
+  rtp_timestamp_ = rtp_timestamp;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::Id(int64_t frame_id) {
+  frame_id_ = frame_id;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::AsLast() {
+  last_spatial_layer_ = true;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::Refs(
+    const std::vector<int64_t>& references) {
+  references_ = references;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::PlayoutDelay(
+    VideoPlayoutDelay playout_delay) {
+  playout_delay_ = playout_delay;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::SpatialLayer(int spatial_layer) {
+  spatial_layer_ = spatial_layer;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::ReceivedTime(Timestamp receive_time) {
+  received_time_ = receive_time;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::Size(size_t size) {
+  size_ = size;
+  return *this;
+}
+
+std::unique_ptr<FakeEncodedFrame> FakeFrameBuilder::Build() {
+  RTC_CHECK_LE(references_.size(), EncodedFrame::kMaxFrameReferences);
+
+  auto frame = std::make_unique<FakeEncodedFrame>();
+  frame->is_last_spatial_layer = last_spatial_layer_;
+  frame->SetEncodedData(EncodedImageBuffer::Create(size_));
+
+  if (rtp_timestamp_)
+    frame->SetTimestamp(*rtp_timestamp_);
+  if (frame_id_)
+    frame->SetId(*frame_id_);
+  if (playout_delay_)
+    frame->SetPlayoutDelay(*playout_delay_);
+  frame->SetFrameType(references_.empty() ? VideoFrameType::kVideoFrameKey
+                                          : VideoFrameType::kVideoFrameDelta);
+  for (int64_t ref : references_) {
+    frame->references[frame->num_references] = ref;
+    frame->num_references++;
+  }
+  if (spatial_layer_)
+    frame->SetSpatialIndex(spatial_layer_);
+  if (received_time_)
+    frame->SetReceivedTime(received_time_->ms());
+  if (payload_type_)
+    frame->SetPayloadType(*payload_type_);
+  if (ntp_time_)
+    frame->ntp_time_ms_ = ntp_time_->ms();
+  if (rotation_)
+    frame->rotation_ = *rotation_;
+  if (packet_infos_)
+    frame->SetPacketInfos(*packet_infos_);
+  return frame;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::PayloadType(int payload_type) {
+  payload_type_ = payload_type;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::NtpTime(Timestamp ntp_time) {
+  ntp_time_ = ntp_time;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::Rotation(VideoRotation rotation) {
+  rotation_ = rotation;
+  return *this;
+}
+
+FakeFrameBuilder& FakeFrameBuilder::PacketInfos(RtpPacketInfos packet_infos) {
+  packet_infos_ = packet_infos;
+  return *this;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fake_encoded_frame.h b/third_party/libwebrtc/test/fake_encoded_frame.h
new file mode 100644
index 0000000000..a5b2aca4a1
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_encoded_frame.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FAKE_ENCODED_FRAME_H_
+#define TEST_FAKE_ENCODED_FRAME_H_
+
+#include <memory>
+#include <ostream>  // no-presubmit-check TODO(webrtc:8982)
+#include <vector>
+
+#include "api/rtp_packet_infos.h"
+#include "api/video/encoded_frame.h"
+#include "api/video/video_rotation.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+// For test printing.
+void PrintTo(const EncodedFrame& frame,
+             std::ostream* os);  // no-presubmit-check TODO(webrtc:8982)
+
+namespace test {
+
+class FakeEncodedFrame : public EncodedFrame {
+ public:
+  // Always 10ms delay and on time.
+  int64_t ReceivedTime() const override;
+  int64_t RenderTime() const override;
+
+  // Setters for protected variables.
+  void SetReceivedTime(int64_t received_time);
+  void SetPayloadType(int payload_type);
+
+ private:
+  int64_t received_time_;
+};
+
+MATCHER_P(WithId, id, "") {
+  return ::testing::Matches(::testing::Eq(id))(arg.Id());
+}
+
+MATCHER_P(FrameWithSize, id, "") {
+  return ::testing::Matches(::testing::Eq(id))(arg.size());
+}
+
+MATCHER_P(RtpTimestamp, ts, "") {
+  return ts == arg.Timestamp();
+}
+
+class FakeFrameBuilder {
+ public:
+  FakeFrameBuilder& Time(uint32_t rtp_timestamp);
+  FakeFrameBuilder& Id(int64_t frame_id);
+  FakeFrameBuilder& AsLast();
+  FakeFrameBuilder& Refs(const std::vector<int64_t>& references);
+  FakeFrameBuilder& PlayoutDelay(VideoPlayoutDelay playout_delay);
+  FakeFrameBuilder& SpatialLayer(int spatial_layer);
+  FakeFrameBuilder& ReceivedTime(Timestamp receive_time);
+  FakeFrameBuilder& Size(size_t size);
+  FakeFrameBuilder& PayloadType(int payload_type);
+  FakeFrameBuilder& NtpTime(Timestamp ntp_time);
+  FakeFrameBuilder& Rotation(VideoRotation rotation);
+  FakeFrameBuilder& PacketInfos(RtpPacketInfos packet_infos);
+  std::unique_ptr<FakeEncodedFrame> Build();
+
+ private:
+  absl::optional<uint32_t> rtp_timestamp_;
+  absl::optional<int64_t> frame_id_;
+  absl::optional<VideoPlayoutDelay> playout_delay_;
+  absl::optional<int> spatial_layer_;
+  absl::optional<Timestamp> received_time_;
+  absl::optional<int> payload_type_;
+  absl::optional<Timestamp> ntp_time_;
+  absl::optional<VideoRotation> rotation_;
+  absl::optional<RtpPacketInfos> packet_infos_;
+  std::vector<int64_t> references_;
+  bool last_spatial_layer_ = false;
+  size_t size_ = 10;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FAKE_ENCODED_FRAME_H_
diff --git a/third_party/libwebrtc/test/fake_encoder.cc b/third_party/libwebrtc/test/fake_encoder.cc
new file mode 100644
index 0000000000..bfc72c123d
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_encoder.cc
@@ -0,0 +1,441 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fake_encoder.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <string>
+
+#include "api/video/video_content_type.h"
+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/sleep.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+const int kKeyframeSizeFactor = 5;
+
+// Inverse of proportion of frames assigned to each temporal layer for all
+// possible temporal layers numbers.
+const int kTemporalLayerRateFactor[4][4] = {
+    {1, 0, 0, 0},  // 1/1
+    {2, 2, 0, 0},  // 1/2 + 1/2
+    {4, 4, 2, 0},  // 1/4 + 1/4 + 1/2
+    {8, 8, 4, 2},  // 1/8 + 1/8 + 1/4 + 1/2
+};
+
+void WriteCounter(unsigned char* payload, uint32_t counter) {
+  payload[0] = (counter & 0x00FF);
+  payload[1] = (counter & 0xFF00) >> 8;
+  payload[2] = (counter & 0xFF0000) >> 16;
+  payload[3] = (counter & 0xFF000000) >> 24;
+}
+
+}  // namespace
+
+FakeEncoder::FakeEncoder(Clock* clock)
+    : clock_(clock),
+      num_initializations_(0),
+      callback_(nullptr),
+      max_target_bitrate_kbps_(-1),
+      pending_keyframe_(true),
+      counter_(0),
+      debt_bytes_(0) {
+  for (bool& used : used_layers_) {
+    used = false;
+  }
+}
+
+void FakeEncoder::SetFecControllerOverride(
+    FecControllerOverride* fec_controller_override) {
+  // Ignored.
+}
+
+void FakeEncoder::SetMaxBitrate(int max_kbps) {
+  RTC_DCHECK_GE(max_kbps, -1);  // max_kbps == -1 disables it.
+  MutexLock lock(&mutex_);
+  max_target_bitrate_kbps_ = max_kbps;
+  SetRatesLocked(current_rate_settings_);
+}
+
+void FakeEncoder::SetQp(int qp) {
+  MutexLock lock(&mutex_);
+  qp_ = qp;
+}
+
+int32_t FakeEncoder::InitEncode(const VideoCodec* config,
+                                const Settings& settings) {
+  MutexLock lock(&mutex_);
+  config_ = *config;
+  ++num_initializations_;
+  current_rate_settings_.bitrate.SetBitrate(0, 0, config_.startBitrate * 1000);
+  current_rate_settings_.framerate_fps = config_.maxFramerate;
+  pending_keyframe_ = true;
+  last_frame_info_ = FrameInfo();
+  return 0;
+}
+
+int32_t FakeEncoder::Encode(const VideoFrame& input_image,
+                            const std::vector<VideoFrameType>* frame_types) {
+  unsigned char max_framerate;
+  unsigned char num_simulcast_streams;
+  SimulcastStream simulcast_streams[kMaxSimulcastStreams];
+  EncodedImageCallback* callback;
+  RateControlParameters rates;
+  bool keyframe;
+  uint32_t counter;
+  absl::optional<int> qp;
+  {
+    MutexLock lock(&mutex_);
+    max_framerate = config_.maxFramerate;
+    num_simulcast_streams = config_.numberOfSimulcastStreams;
+    for (int i = 0; i < num_simulcast_streams; ++i) {
+      simulcast_streams[i] = config_.simulcastStream[i];
+    }
+    callback = callback_;
+    rates = current_rate_settings_;
+    if (rates.framerate_fps <= 0.0) {
+      rates.framerate_fps = max_framerate;
+    }
+    keyframe = pending_keyframe_;
+    pending_keyframe_ = false;
+    counter = counter_++;
+    qp = qp_;
+  }
+
+  FrameInfo frame_info =
+      NextFrame(frame_types, keyframe, num_simulcast_streams, rates.bitrate,
+                simulcast_streams, static_cast<int>(rates.framerate_fps + 0.5));
+  for (uint8_t i = 0; i < frame_info.layers.size(); ++i) {
+    constexpr int kMinPayLoadLength = 14;
+    if (frame_info.layers[i].size < kMinPayLoadLength) {
+      // Drop this temporal layer.
+      continue;
+    }
+
+    auto buffer = EncodedImageBuffer::Create(frame_info.layers[i].size);
+    // Fill the buffer with arbitrary data. Write someting to make Asan happy.
+    memset(buffer->data(), 9, frame_info.layers[i].size);
+    // Write a counter to the image to make each frame unique.
+    WriteCounter(buffer->data() + frame_info.layers[i].size - 4, counter);
+
+    EncodedImage encoded;
+    encoded.SetEncodedData(buffer);
+
+    encoded.SetTimestamp(input_image.timestamp());
+    encoded._frameType = frame_info.keyframe ? VideoFrameType::kVideoFrameKey
+                                             : VideoFrameType::kVideoFrameDelta;
+    encoded._encodedWidth = simulcast_streams[i].width;
+    encoded._encodedHeight = simulcast_streams[i].height;
+    if (qp)
+      encoded.qp_ = *qp;
+    encoded.SetSpatialIndex(i);
+    CodecSpecificInfo codec_specific = EncodeHook(encoded, buffer);
+
+    if (callback->OnEncodedImage(encoded, &codec_specific).error !=
+        EncodedImageCallback::Result::OK) {
+      return -1;
+    }
+  }
+  return 0;
+}
+
+CodecSpecificInfo FakeEncoder::EncodeHook(
+    EncodedImage& encoded_image,
+    rtc::scoped_refptr<EncodedImageBuffer> buffer) {
+  CodecSpecificInfo codec_specific;
+  codec_specific.codecType = kVideoCodecGeneric;
+  return codec_specific;
+}
+
+FakeEncoder::FrameInfo FakeEncoder::NextFrame(
+    const std::vector<VideoFrameType>* frame_types,
+    bool keyframe,
+    uint8_t num_simulcast_streams,
+    const VideoBitrateAllocation& target_bitrate,
+    SimulcastStream simulcast_streams[kMaxSimulcastStreams],
+    int framerate) {
+  FrameInfo frame_info;
+  frame_info.keyframe = keyframe;
+
+  if (frame_types) {
+    for (VideoFrameType frame_type : *frame_types) {
+      if (frame_type == VideoFrameType::kVideoFrameKey) {
+        frame_info.keyframe = true;
+        break;
+      }
+    }
+  }
+
+  MutexLock lock(&mutex_);
+  for (uint8_t i = 0; i < num_simulcast_streams; ++i) {
+    if (target_bitrate.GetBitrate(i, 0) > 0) {
+      int temporal_id = last_frame_info_.layers.size() > i
+                            ? ++last_frame_info_.layers[i].temporal_id %
+                                  simulcast_streams[i].numberOfTemporalLayers
+                            : 0;
+      frame_info.layers.emplace_back(0, temporal_id);
+    }
+  }
+
+  if (last_frame_info_.layers.size() < frame_info.layers.size()) {
+    // A new keyframe is needed since a new layer will be added.
+    frame_info.keyframe = true;
+  }
+
+  for (uint8_t i = 0; i < frame_info.layers.size(); ++i) {
+    FrameInfo::SpatialLayer& layer_info = frame_info.layers[i];
+    if (frame_info.keyframe) {
+      layer_info.temporal_id = 0;
+      size_t avg_frame_size =
+          (target_bitrate.GetBitrate(i, 0) + 7) *
+          kTemporalLayerRateFactor[frame_info.layers.size() - 1][i] /
+          (8 * framerate);
+
+      // The first frame is a key frame and should be larger.
+      // Store the overshoot bytes and distribute them over the coming frames,
+      // so that we on average meet the bitrate target.
+      debt_bytes_ += (kKeyframeSizeFactor - 1) * avg_frame_size;
+      layer_info.size = kKeyframeSizeFactor * avg_frame_size;
+    } else {
+      size_t avg_frame_size =
+          (target_bitrate.GetBitrate(i, layer_info.temporal_id) + 7) *
+          kTemporalLayerRateFactor[frame_info.layers.size() - 1][i] /
+          (8 * framerate);
+      layer_info.size = avg_frame_size;
+      if (debt_bytes_ > 0) {
+        // Pay at most half of the frame size for old debts.
+        size_t payment_size = std::min(avg_frame_size / 2, debt_bytes_);
+        debt_bytes_ -= payment_size;
+        layer_info.size -= payment_size;
+      }
+    }
+  }
+  last_frame_info_ = frame_info;
+  return frame_info;
+}
+
+int32_t FakeEncoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  MutexLock lock(&mutex_);
+  callback_ = callback;
+  return 0;
+}
+
+int32_t FakeEncoder::Release() {
+  return 0;
+}
+
+void FakeEncoder::SetRates(const RateControlParameters& parameters) {
+  MutexLock lock(&mutex_);
+  SetRatesLocked(parameters);
+}
+
+void FakeEncoder::SetRatesLocked(const RateControlParameters& parameters) {
+  current_rate_settings_ = parameters;
+  int allocated_bitrate_kbps = parameters.bitrate.get_sum_kbps();
+
+  // Scale bitrate allocation to not exceed the given max target bitrate.
+  if (max_target_bitrate_kbps_ > 0 &&
+      allocated_bitrate_kbps > max_target_bitrate_kbps_) {
+    for (uint8_t spatial_idx = 0; spatial_idx < kMaxSpatialLayers;
+         ++spatial_idx) {
+      for (uint8_t temporal_idx = 0; temporal_idx < kMaxTemporalStreams;
+           ++temporal_idx) {
+        if (current_rate_settings_.bitrate.HasBitrate(spatial_idx,
+                                                      temporal_idx)) {
+          uint32_t bitrate = current_rate_settings_.bitrate.GetBitrate(
+              spatial_idx, temporal_idx);
+          bitrate = static_cast<uint32_t>(
+              (bitrate * int64_t{max_target_bitrate_kbps_}) /
+              allocated_bitrate_kbps);
+          current_rate_settings_.bitrate.SetBitrate(spatial_idx, temporal_idx,
+                                                    bitrate);
+        }
+      }
+    }
+  }
+}
+
+const char* FakeEncoder::kImplementationName = "fake_encoder";
+VideoEncoder::EncoderInfo FakeEncoder::GetEncoderInfo() const {
+  EncoderInfo info;
+  info.implementation_name = kImplementationName;
+  info.is_hardware_accelerated = true;
+  MutexLock lock(&mutex_);
+  for (int sid = 0; sid < config_.numberOfSimulcastStreams; ++sid) {
+    int number_of_temporal_layers =
+        config_.simulcastStream[sid].numberOfTemporalLayers;
+    info.fps_allocation[sid].clear();
+    for (int tid = 0; tid < number_of_temporal_layers; ++tid) {
+      // {1/4, 1/2, 1} allocation for num layers = 3.
+      info.fps_allocation[sid].push_back(255 /
+                                         (number_of_temporal_layers - tid));
+    }
+  }
+  return info;
+}
+
+int FakeEncoder::GetConfiguredInputFramerate() const {
+  MutexLock lock(&mutex_);
+  return static_cast<int>(current_rate_settings_.framerate_fps + 0.5);
+}
+
+int FakeEncoder::GetNumInitializations() const {
+  MutexLock lock(&mutex_);
+  return num_initializations_;
+}
+
+const VideoCodec& FakeEncoder::config() const {
+  MutexLock lock(&mutex_);
+  return config_;
+}
+
+FakeH264Encoder::FakeH264Encoder(Clock* clock)
+    : FakeEncoder(clock), idr_counter_(0) {}
+
+CodecSpecificInfo FakeH264Encoder::EncodeHook(
+    EncodedImage& encoded_image,
+    rtc::scoped_refptr<EncodedImageBuffer> buffer) {
+  static constexpr std::array<uint8_t, 3> kStartCode = {0, 0, 1};
+  const size_t kSpsSize = 8;
+  const size_t kPpsSize = 11;
+  const int kIdrFrequency = 10;
+  int current_idr_counter;
+  {
+    MutexLock lock(&local_mutex_);
+    current_idr_counter = idr_counter_;
+    ++idr_counter_;
+  }
+  for (size_t i = 0; i < encoded_image.size(); ++i) {
+    buffer->data()[i] = static_cast<uint8_t>(i);
+  }
+
+  if (current_idr_counter % kIdrFrequency == 0 &&
+      encoded_image.size() > kSpsSize + kPpsSize + 1 + 3 * kStartCode.size()) {
+    const size_t kSpsNalHeader = 0x67;
+    const size_t kPpsNalHeader = 0x68;
+    const size_t kIdrNalHeader = 0x65;
+    uint8_t* data = buffer->data();
+    memcpy(data, kStartCode.data(), kStartCode.size());
+    data += kStartCode.size();
+    data[0] = kSpsNalHeader;
+    data += kSpsSize;
+
+    memcpy(data, kStartCode.data(), kStartCode.size());
+    data += kStartCode.size();
+    data[0] = kPpsNalHeader;
+    data += kPpsSize;
+
+    memcpy(data, kStartCode.data(), kStartCode.size());
+    data += kStartCode.size();
+    data[0] = kIdrNalHeader;
+  } else {
+    memcpy(buffer->data(), kStartCode.data(), kStartCode.size());
+    const size_t kNalHeader = 0x41;
+    buffer->data()[kStartCode.size()] = kNalHeader;
+  }
+
+  CodecSpecificInfo codec_specific;
+  codec_specific.codecType = kVideoCodecH264;
+  codec_specific.codecSpecific.H264.packetization_mode =
+      H264PacketizationMode::NonInterleaved;
+  return codec_specific;
+}
+
+DelayedEncoder::DelayedEncoder(Clock* clock, int delay_ms)
+    : test::FakeEncoder(clock), delay_ms_(delay_ms) {
+  // The encoder could be created on a different thread than
+  // it is being used on.
+  sequence_checker_.Detach();
+}
+
+void DelayedEncoder::SetDelay(int delay_ms) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  delay_ms_ = delay_ms;
+}
+
+int32_t DelayedEncoder::Encode(const VideoFrame& input_image,
+                               const std::vector<VideoFrameType>* frame_types) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  SleepMs(delay_ms_);
+
+  return FakeEncoder::Encode(input_image, frame_types);
+}
+
+MultithreadedFakeH264Encoder::MultithreadedFakeH264Encoder(
+    Clock* clock,
+    TaskQueueFactory* task_queue_factory)
+    : test::FakeH264Encoder(clock),
+      task_queue_factory_(task_queue_factory),
+      current_queue_(0),
+      queue1_(nullptr),
+      queue2_(nullptr) {
+  // The encoder could be created on a different thread than
+  // it is being used on.
+  sequence_checker_.Detach();
+}
+
+int32_t MultithreadedFakeH264Encoder::InitEncode(const VideoCodec* config,
+                                                 const Settings& settings) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  queue1_ = task_queue_factory_->CreateTaskQueue(
+      "Queue 1", TaskQueueFactory::Priority::NORMAL);
+  queue2_ = task_queue_factory_->CreateTaskQueue(
+      "Queue 2", TaskQueueFactory::Priority::NORMAL);
+
+  return FakeH264Encoder::InitEncode(config, settings);
+}
+
+int32_t MultithreadedFakeH264Encoder::Encode(
+    const VideoFrame& input_image,
+    const std::vector<VideoFrameType>* frame_types) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  TaskQueueBase* queue =
+      (current_queue_++ % 2 == 0) ? queue1_.get() : queue2_.get();
+
+  if (!queue) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+
+  queue->PostTask([this, input_image, frame_types = *frame_types] {
+    EncodeCallback(input_image, &frame_types);
+  });
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t MultithreadedFakeH264Encoder::EncodeCallback(
+    const VideoFrame& input_image,
+    const std::vector<VideoFrameType>* frame_types) {
+  return FakeH264Encoder::Encode(input_image, frame_types);
+}
+
+int32_t MultithreadedFakeH264Encoder::Release() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  queue1_.reset();
+  queue2_.reset();
+
+  return FakeH264Encoder::Release();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fake_encoder.h b/third_party/libwebrtc/test/fake_encoder.h
new file mode 100644
index 0000000000..65e03155d7
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_encoder.h
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FAKE_ENCODER_H_
+#define TEST_FAKE_ENCODER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "api/fec_controller_override.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_bitrate_allocation.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeEncoder : public VideoEncoder {
+ public:
+  explicit FakeEncoder(Clock* clock);
+  virtual ~FakeEncoder() = default;
+
+  // Sets max bitrate. Not thread-safe, call before registering the encoder.
+  void SetMaxBitrate(int max_kbps) RTC_LOCKS_EXCLUDED(mutex_);
+  void SetQp(int qp) RTC_LOCKS_EXCLUDED(mutex_);
+
+  void SetFecControllerOverride(
+      FecControllerOverride* fec_controller_override) override;
+
+  int32_t InitEncode(const VideoCodec* config, const Settings& settings)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t Encode(const VideoFrame& input_image,
+                 const std::vector<VideoFrameType>* frame_types)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t RegisterEncodeCompleteCallback(EncodedImageCallback* callback)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  int32_t Release() override;
+  void SetRates(const RateControlParameters& parameters)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  EncoderInfo GetEncoderInfo() const override;
+
+  int GetConfiguredInputFramerate() const RTC_LOCKS_EXCLUDED(mutex_);
+  int GetNumInitializations() const RTC_LOCKS_EXCLUDED(mutex_);
+  const VideoCodec& config() const RTC_LOCKS_EXCLUDED(mutex_);
+
+  static const char* kImplementationName;
+
+ protected:
+  struct FrameInfo {
+    bool keyframe;
+    struct SpatialLayer {
+      SpatialLayer() = default;
+      SpatialLayer(int size, int temporal_id)
+          : size(size), temporal_id(temporal_id) {}
+      // Size of a current frame in the layer.
+      int size = 0;
+      // Temporal index of a current frame in the layer.
+      int temporal_id = 0;
+    };
+    std::vector<SpatialLayer> layers;
+  };
+
+  FrameInfo NextFrame(const std::vector<VideoFrameType>* frame_types,
+                      bool keyframe,
+                      uint8_t num_simulcast_streams,
+                      const VideoBitrateAllocation& target_bitrate,
+                      SimulcastStream simulcast_streams[kMaxSimulcastStreams],
+                      int framerate) RTC_LOCKS_EXCLUDED(mutex_);
+
+  // Called before the frame is passed to callback_->OnEncodedImage, to let
+  // subclasses fill out CodecSpecificInfo, possibly modify `encoded_image` or
+  // `buffer`.
+  virtual CodecSpecificInfo EncodeHook(
+      EncodedImage& encoded_image,
+      rtc::scoped_refptr<EncodedImageBuffer> buffer);
+
+  void SetRatesLocked(const RateControlParameters& parameters)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  FrameInfo last_frame_info_ RTC_GUARDED_BY(mutex_);
+  Clock* const clock_;
+
+  VideoCodec config_ RTC_GUARDED_BY(mutex_);
+  int num_initializations_ RTC_GUARDED_BY(mutex_);
+  EncodedImageCallback* callback_ RTC_GUARDED_BY(mutex_);
+  RateControlParameters current_rate_settings_ RTC_GUARDED_BY(mutex_);
+  int max_target_bitrate_kbps_ RTC_GUARDED_BY(mutex_);
+  bool pending_keyframe_ RTC_GUARDED_BY(mutex_);
+  uint32_t counter_ RTC_GUARDED_BY(mutex_);
+  mutable Mutex mutex_;
+  bool used_layers_[kMaxSimulcastStreams];
+  absl::optional<int> qp_ RTC_GUARDED_BY(mutex_);
+
+  // Current byte debt to be payed over a number of frames.
+  // The debt is acquired by keyframes overshooting the bitrate target.
+  size_t debt_bytes_;
+};
+
+class FakeH264Encoder : public FakeEncoder {
+ public:
+  explicit FakeH264Encoder(Clock* clock);
+  virtual ~FakeH264Encoder() = default;
+
+ private:
+  CodecSpecificInfo EncodeHook(
+      EncodedImage& encoded_image,
+      rtc::scoped_refptr<EncodedImageBuffer> buffer) override;
+
+  int idr_counter_ RTC_GUARDED_BY(local_mutex_);
+  Mutex local_mutex_;
+};
+
+class DelayedEncoder : public test::FakeEncoder {
+ public:
+  DelayedEncoder(Clock* clock, int delay_ms);
+  virtual ~DelayedEncoder() = default;
+
+  void SetDelay(int delay_ms);
+  int32_t Encode(const VideoFrame& input_image,
+                 const std::vector<VideoFrameType>* frame_types) override;
+
+ private:
+  int delay_ms_ RTC_GUARDED_BY(sequence_checker_);
+  SequenceChecker sequence_checker_;
+};
+
+// This class implements a multi-threaded fake encoder by posting
+// FakeH264Encoder::Encode(.) tasks to `queue1_` and `queue2_`, in an
+// alternating fashion. The class itself does not need to be thread safe,
+// as it is called from the task queue in VideoStreamEncoder.
+class MultithreadedFakeH264Encoder : public test::FakeH264Encoder {
+ public:
+  MultithreadedFakeH264Encoder(Clock* clock,
+                               TaskQueueFactory* task_queue_factory);
+  virtual ~MultithreadedFakeH264Encoder() = default;
+
+  int32_t InitEncode(const VideoCodec* config,
+                     const Settings& settings) override;
+
+  int32_t Encode(const VideoFrame& input_image,
+                 const std::vector<VideoFrameType>* frame_types) override;
+
+  int32_t EncodeCallback(const VideoFrame& input_image,
+                         const std::vector<VideoFrameType>* frame_types);
+
+  int32_t Release() override;
+
+ protected:
+  TaskQueueFactory* const task_queue_factory_;
+  int current_queue_ RTC_GUARDED_BY(sequence_checker_);
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> queue1_
+      RTC_GUARDED_BY(sequence_checker_);
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> queue2_
+      RTC_GUARDED_BY(sequence_checker_);
+  SequenceChecker sequence_checker_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FAKE_ENCODER_H_
diff --git a/third_party/libwebrtc/test/fake_texture_frame.cc b/third_party/libwebrtc/test/fake_texture_frame.cc
new file mode 100644
index 0000000000..9c17e4c1c0
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_texture_frame.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fake_texture_frame.h"
+
+#include "api/video/i420_buffer.h"
+
+namespace webrtc {
+namespace test {
+
+VideoFrame FakeNativeBuffer::CreateFrame(int width,
+                                         int height,
+                                         uint32_t timestamp,
+                                         int64_t render_time_ms,
+                                         VideoRotation rotation) {
+  return VideoFrame::Builder()
+      .set_video_frame_buffer(
+          rtc::make_ref_counted<FakeNativeBuffer>(width, height))
+      .set_timestamp_rtp(timestamp)
+      .set_timestamp_ms(render_time_ms)
+      .set_rotation(rotation)
+      .build();
+}
+
+VideoFrameBuffer::Type FakeNativeBuffer::type() const {
+  return Type::kNative;
+}
+
+int FakeNativeBuffer::width() const {
+  return width_;
+}
+
+int FakeNativeBuffer::height() const {
+  return height_;
+}
+
+rtc::scoped_refptr<I420BufferInterface> FakeNativeBuffer::ToI420() {
+  rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(width_, height_);
+  I420Buffer::SetBlack(buffer.get());
+  return buffer;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fake_texture_frame.h b/third_party/libwebrtc/test/fake_texture_frame.h
new file mode 100644
index 0000000000..1b25112e01
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_texture_frame.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_FAKE_TEXTURE_FRAME_H_
+#define TEST_FAKE_TEXTURE_FRAME_H_
+
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeNativeBuffer : public VideoFrameBuffer {
+ public:
+  static VideoFrame CreateFrame(int width,
+                                int height,
+                                uint32_t timestamp,
+                                int64_t render_time_ms,
+                                VideoRotation rotation);
+
+  FakeNativeBuffer(int width, int height) : width_(width), height_(height) {}
+
+  Type type() const override;
+  int width() const override;
+  int height() const override;
+
+ private:
+  rtc::scoped_refptr<I420BufferInterface> ToI420() override;
+
+  const int width_;
+  const int height_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  //  TEST_FAKE_TEXTURE_FRAME_H_
diff --git a/third_party/libwebrtc/test/fake_videorenderer.h b/third_party/libwebrtc/test/fake_videorenderer.h
new file mode 100644
index 0000000000..a2c953ff77
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_videorenderer.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FAKE_VIDEORENDERER_H_
+#define TEST_FAKE_VIDEORENDERER_H_
+
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeVideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  void OnFrame(const webrtc::VideoFrame& frame) override {}
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FAKE_VIDEORENDERER_H_
diff --git a/third_party/libwebrtc/test/fake_vp8_decoder.cc b/third_party/libwebrtc/test/fake_vp8_decoder.cc
new file mode 100644
index 0000000000..4c2f55a668
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_vp8_decoder.cc
@@ -0,0 +1,93 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fake_vp8_decoder.h"
+
+#include <stddef.h>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+// Read width and height from the payload of the frame if it is a key frame the
+// same way as the real VP8 decoder.
+// FakeEncoder writes width, height and frame type.
+void ParseFakeVp8(const unsigned char* data, int* width, int* height) {
+  bool key_frame = data[0] == 0;
+  if (key_frame) {
+    *width = ((data[7] << 8) + data[6]) & 0x3FFF;
+    *height = ((data[9] << 8) + data[8]) & 0x3FFF;
+  }
+}
+}  // namespace
+
+FakeVp8Decoder::FakeVp8Decoder() : callback_(nullptr), width_(0), height_(0) {}
+
+bool FakeVp8Decoder::Configure(const Settings& settings) {
+  return true;
+}
+
+int32_t FakeVp8Decoder::Decode(const EncodedImage& input,
+                               bool missing_frames,
+                               int64_t render_time_ms) {
+  constexpr size_t kMinPayLoadHeaderLength = 10;
+  if (input.size() < kMinPayLoadHeaderLength) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  ParseFakeVp8(input.data(), &width_, &height_);
+
+  VideoFrame frame =
+      VideoFrame::Builder()
+          .set_video_frame_buffer(I420Buffer::Create(width_, height_))
+          .set_rotation(webrtc::kVideoRotation_0)
+          .set_timestamp_ms(render_time_ms)
+          .build();
+  frame.set_timestamp(input.Timestamp());
+  frame.set_ntp_time_ms(input.ntp_time_ms_);
+
+  callback_->Decoded(frame, /*decode_time_ms=*/absl::nullopt,
+                     /*qp=*/absl::nullopt);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeVp8Decoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  callback_ = callback;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeVp8Decoder::Release() {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+const char* FakeVp8Decoder::kImplementationName = "fake_vp8_decoder";
+VideoDecoder::DecoderInfo FakeVp8Decoder::GetDecoderInfo() const {
+  DecoderInfo info;
+  info.implementation_name = kImplementationName;
+  info.is_hardware_accelerated = false;
+  return info;
+}
+
+const char* FakeVp8Decoder::ImplementationName() const {
+  return kImplementationName;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fake_vp8_decoder.h b/third_party/libwebrtc/test/fake_vp8_decoder.h
new file mode 100644
index 0000000000..391ee374d4
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_vp8_decoder.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FAKE_VP8_DECODER_H_
+#define TEST_FAKE_VP8_DECODER_H_
+
+#include <stdint.h>
+
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeVp8Decoder : public VideoDecoder {
+ public:
+  FakeVp8Decoder();
+  ~FakeVp8Decoder() override {}
+
+  bool Configure(const Settings& settings) override;
+
+  int32_t Decode(const EncodedImage& input,
+                 bool missing_frames,
+                 int64_t render_time_ms) override;
+
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+
+  int32_t Release() override;
+
+  DecoderInfo GetDecoderInfo() const override;
+  const char* ImplementationName() const override;
+  static const char* kImplementationName;
+
+ private:
+  DecodedImageCallback* callback_;
+  int width_;
+  int height_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FAKE_VP8_DECODER_H_
diff --git a/third_party/libwebrtc/test/fake_vp8_encoder.cc b/third_party/libwebrtc/test/fake_vp8_encoder.cc
new file mode 100644
index 0000000000..625d7a6473
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_vp8_encoder.cc
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fake_vp8_encoder.h"
+
+#include <algorithm>
+
+#include "absl/types/optional.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "api/video_codecs/vp8_temporal_layers_factory.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/utility/simulcast_utility.h"
+
+namespace {
+
+// Write width and height to the payload the same way as the real encoder does.
+// It requires that `payload` has a size of at least kMinPayLoadHeaderLength.
+void WriteFakeVp8(unsigned char* payload,
+                  int width,
+                  int height,
+                  bool key_frame) {
+  payload[0] = key_frame ? 0 : 0x01;
+
+  if (key_frame) {
+    payload[9] = (height & 0x3F00) >> 8;
+    payload[8] = (height & 0x00FF);
+
+    payload[7] = (width & 0x3F00) >> 8;
+    payload[6] = (width & 0x00FF);
+  }
+}
+}  // namespace
+
+namespace webrtc {
+
+namespace test {
+
+FakeVp8Encoder::FakeVp8Encoder(Clock* clock) : FakeEncoder(clock) {
+  sequence_checker_.Detach();
+}
+
+int32_t FakeVp8Encoder::InitEncode(const VideoCodec* config,
+                                   const Settings& settings) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  auto result = FakeEncoder::InitEncode(config, settings);
+  if (result != WEBRTC_VIDEO_CODEC_OK) {
+    return result;
+  }
+
+  Vp8TemporalLayersFactory factory;
+  frame_buffer_controller_ =
+      factory.Create(*config, settings, &fec_controller_override_);
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+int32_t FakeVp8Encoder::Release() {
+  auto result = FakeEncoder::Release();
+  sequence_checker_.Detach();
+  return result;
+}
+
+CodecSpecificInfo FakeVp8Encoder::PopulateCodecSpecific(
+    size_t size_bytes,
+    VideoFrameType frame_type,
+    int stream_idx,
+    uint32_t timestamp) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  CodecSpecificInfo codec_specific;
+  codec_specific.codecType = kVideoCodecVP8;
+  codec_specific.codecSpecific.VP8.keyIdx = kNoKeyIdx;
+  codec_specific.codecSpecific.VP8.nonReference = false;
+  if (size_bytes > 0) {
+    frame_buffer_controller_->OnEncodeDone(
+        stream_idx, timestamp, size_bytes,
+        frame_type == VideoFrameType::kVideoFrameKey, -1, &codec_specific);
+  } else {
+    frame_buffer_controller_->OnFrameDropped(stream_idx, timestamp);
+  }
+  return codec_specific;
+}
+
+CodecSpecificInfo FakeVp8Encoder::EncodeHook(
+    EncodedImage& encoded_image,
+    rtc::scoped_refptr<EncodedImageBuffer> buffer) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  uint8_t stream_idx = encoded_image.SpatialIndex().value_or(0);
+  frame_buffer_controller_->NextFrameConfig(stream_idx,
+                                            encoded_image.Timestamp());
+  CodecSpecificInfo codec_specific =
+      PopulateCodecSpecific(encoded_image.size(), encoded_image._frameType,
+                            stream_idx, encoded_image.Timestamp());
+
+  // Write width and height to the payload the same way as the real encoder
+  // does.
+  WriteFakeVp8(buffer->data(), encoded_image._encodedWidth,
+               encoded_image._encodedHeight,
+               encoded_image._frameType == VideoFrameType::kVideoFrameKey);
+  return codec_specific;
+}
+
+VideoEncoder::EncoderInfo FakeVp8Encoder::GetEncoderInfo() const {
+  EncoderInfo info;
+  info.implementation_name = "FakeVp8Encoder";
+  MutexLock lock(&mutex_);
+  for (int sid = 0; sid < config_.numberOfSimulcastStreams; ++sid) {
+    int number_of_temporal_layers =
+        config_.simulcastStream[sid].numberOfTemporalLayers;
+    info.fps_allocation[sid].clear();
+    for (int tid = 0; tid < number_of_temporal_layers; ++tid) {
+      // {1/4, 1/2, 1} allocation for num layers = 3.
+      info.fps_allocation[sid].push_back(255 /
+                                         (number_of_temporal_layers - tid));
+    }
+  }
+  return info;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fake_vp8_encoder.h b/third_party/libwebrtc/test/fake_vp8_encoder.h
new file mode 100644
index 0000000000..6aaf547379
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_vp8_encoder.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FAKE_VP8_ENCODER_H_
+#define TEST_FAKE_VP8_ENCODER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "api/fec_controller_override.h"
+#include "api/sequence_checker.h"
+#include "api/video/encoded_image.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/vp8_frame_buffer_controller.h"
+#include "api/video_codecs/vp8_temporal_layers.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+#include "test/fake_encoder.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeVp8Encoder : public FakeEncoder {
+ public:
+  explicit FakeVp8Encoder(Clock* clock);
+  virtual ~FakeVp8Encoder() = default;
+
+  int32_t InitEncode(const VideoCodec* config,
+                     const Settings& settings) override;
+
+  int32_t Release() override;
+
+  EncoderInfo GetEncoderInfo() const override;
+
+ private:
+  CodecSpecificInfo PopulateCodecSpecific(size_t size_bytes,
+                                          VideoFrameType frame_type,
+                                          int stream_idx,
+                                          uint32_t timestamp);
+
+  CodecSpecificInfo EncodeHook(
+      EncodedImage& encoded_image,
+      rtc::scoped_refptr<EncodedImageBuffer> buffer) override;
+
+  SequenceChecker sequence_checker_;
+
+  class FakeFecControllerOverride : public FecControllerOverride {
+   public:
+    ~FakeFecControllerOverride() override = default;
+
+    void SetFecAllowed(bool fec_allowed) override {}
+  };
+
+  FakeFecControllerOverride fec_controller_override_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  std::unique_ptr<Vp8FrameBufferController> frame_buffer_controller_
+      RTC_GUARDED_BY(sequence_checker_);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FAKE_VP8_ENCODER_H_
diff --git a/third_party/libwebrtc/test/fake_vp8_encoder_unittest.cc b/third_party/libwebrtc/test/fake_vp8_encoder_unittest.cc
new file mode 100644
index 0000000000..e79e8e421b
--- /dev/null
+++ b/third_party/libwebrtc/test/fake_vp8_encoder_unittest.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fake_vp8_encoder.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/test/create_simulcast_test_fixture.h"
+#include "api/test/simulcast_test_fixture.h"
+#include "api/test/video/function_video_decoder_factory.h"
+#include "api/test/video/function_video_encoder_factory.h"
+#include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
+#include "test/fake_vp8_decoder.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture() {
+  std::unique_ptr<VideoEncoderFactory> encoder_factory =
+      std::make_unique<FunctionVideoEncoderFactory>([]() {
+        return std::make_unique<FakeVp8Encoder>(Clock::GetRealTimeClock());
+      });
+  std::unique_ptr<VideoDecoderFactory> decoder_factory =
+      std::make_unique<FunctionVideoDecoderFactory>(
+          []() { return std::make_unique<FakeVp8Decoder>(); });
+  return CreateSimulcastTestFixture(std::move(encoder_factory),
+                                    std::move(decoder_factory),
+                                    SdpVideoFormat("VP8"));
+}
+}  // namespace
+
+TEST(TestFakeVp8Codec, TestKeyFrameRequestsOnAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestKeyFrameRequestsOnAllStreams();
+}
+
+TEST(TestFakeVp8Codec, TestPaddingAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingAllStreams();
+}
+
+TEST(TestFakeVp8Codec, TestPaddingTwoStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingTwoStreams();
+}
+
+TEST(TestFakeVp8Codec, TestPaddingTwoStreamsOneMaxedOut) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingTwoStreamsOneMaxedOut();
+}
+
+TEST(TestFakeVp8Codec, TestPaddingOneStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingOneStream();
+}
+
+TEST(TestFakeVp8Codec, TestPaddingOneStreamTwoMaxedOut) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestPaddingOneStreamTwoMaxedOut();
+}
+
+TEST(TestFakeVp8Codec, TestSendAllStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSendAllStreams();
+}
+
+TEST(TestFakeVp8Codec, TestDisablingStreams) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestDisablingStreams();
+}
+
+TEST(TestFakeVp8Codec, TestSwitchingToOneStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneStream();
+}
+
+TEST(TestFakeVp8Codec, TestSwitchingToOneOddStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneOddStream();
+}
+
+TEST(TestFakeVp8Codec, TestSwitchingToOneSmallStream) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSwitchingToOneSmallStream();
+}
+
+TEST(TestFakeVp8Codec, TestSpatioTemporalLayers333PatternEncoder) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestSpatioTemporalLayers333PatternEncoder();
+}
+
+TEST(TestFakeVp8Codec, TestDecodeWidthHeightSet) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestDecodeWidthHeightSet();
+}
+
+TEST(TestFakeVp8Codec,
+     TestEncoderInfoForDefaultTemporalLayerProfileHasFpsAllocation) {
+  auto fixture = CreateSpecificSimulcastTestFixture();
+  fixture->TestEncoderInfoForDefaultTemporalLayerProfileHasFpsAllocation();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/field_trial.cc b/third_party/libwebrtc/test/field_trial.cc
new file mode 100644
index 0000000000..3d6c6ac617
--- /dev/null
+++ b/third_party/libwebrtc/test/field_trial.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/field_trial.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace test {
+
+ScopedFieldTrials::ScopedFieldTrials(absl::string_view config)
+    : current_field_trials_(config),
+      previous_field_trials_(webrtc::field_trial::GetFieldTrialString()) {
+  RTC_CHECK(webrtc::field_trial::FieldTrialsStringIsValid(
+      current_field_trials_.c_str()))
+      << "Invalid field trials string: " << current_field_trials_;
+  webrtc::field_trial::InitFieldTrialsFromString(current_field_trials_.c_str());
+}
+
+ScopedFieldTrials::~ScopedFieldTrials() {
+  RTC_CHECK(
+      webrtc::field_trial::FieldTrialsStringIsValid(previous_field_trials_))
+      << "Invalid field trials string: " << previous_field_trials_;
+  webrtc::field_trial::InitFieldTrialsFromString(previous_field_trials_);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/field_trial.h b/third_party/libwebrtc/test/field_trial.h
new file mode 100644
index 0000000000..516faa0513
--- /dev/null
+++ b/third_party/libwebrtc/test/field_trial.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FIELD_TRIAL_H_
+#define TEST_FIELD_TRIAL_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+// This class is used to override field-trial configs within specific tests.
+// After this class goes out of scope previous field trials will be restored.
+class ScopedFieldTrials {
+ public:
+  explicit ScopedFieldTrials(absl::string_view config);
+  ScopedFieldTrials(const ScopedFieldTrials&) = delete;
+  ScopedFieldTrials& operator=(const ScopedFieldTrials&) = delete;
+  ~ScopedFieldTrials();
+
+ private:
+  std::string current_field_trials_;
+  const char* previous_field_trials_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FIELD_TRIAL_H_
diff --git a/third_party/libwebrtc/test/frame_forwarder.cc b/third_party/libwebrtc/test/frame_forwarder.cc
new file mode 100644
index 0000000000..e89f753bd3
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_forwarder.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/frame_forwarder.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+FrameForwarder::FrameForwarder() : sink_(nullptr) {}
+FrameForwarder::~FrameForwarder() {}
+
+void FrameForwarder::IncomingCapturedFrame(const VideoFrame& video_frame) {
+  MutexLock lock(&mutex_);
+  if (sink_)
+    sink_->OnFrame(video_frame);
+}
+
+void FrameForwarder::AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
+                                     const rtc::VideoSinkWants& wants) {
+  MutexLock lock(&mutex_);
+  AddOrUpdateSinkLocked(sink, wants);
+}
+
+void FrameForwarder::AddOrUpdateSinkLocked(
+    rtc::VideoSinkInterface<VideoFrame>* sink,
+    const rtc::VideoSinkWants& wants) {
+  RTC_DCHECK(!sink_ || sink_ == sink);
+  sink_ = sink;
+  sink_wants_ = wants;
+}
+
+void FrameForwarder::RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
+  MutexLock lock(&mutex_);
+  RTC_DCHECK_EQ(sink, sink_);
+  sink_ = nullptr;
+}
+
+rtc::VideoSinkWants FrameForwarder::sink_wants() const {
+  MutexLock lock(&mutex_);
+  return sink_wants_;
+}
+
+rtc::VideoSinkWants FrameForwarder::sink_wants_locked() const {
+  return sink_wants_;
+}
+
+bool FrameForwarder::has_sinks() const {
+  MutexLock lock(&mutex_);
+  return sink_ != nullptr;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/frame_forwarder.h b/third_party/libwebrtc/test/frame_forwarder.h
new file mode 100644
index 0000000000..6dfba9521d
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_forwarder.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_FRAME_FORWARDER_H_
+#define TEST_FRAME_FORWARDER_H_
+
+#include "api/video/video_frame.h"
+#include "api/video/video_source_interface.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace test {
+
+// FrameForwarder can be used as an implementation
+// of rtc::VideoSourceInterface<VideoFrame> where the caller controls when
+// a frame should be forwarded to its sink.
+// Currently this implementation only support one sink.
+class FrameForwarder : public rtc::VideoSourceInterface<VideoFrame> {
+ public:
+  FrameForwarder();
+  ~FrameForwarder() override;
+  // Forwards `video_frame` to the registered `sink_`.
+  virtual void IncomingCapturedFrame(const VideoFrame& video_frame)
+      RTC_LOCKS_EXCLUDED(mutex_);
+  rtc::VideoSinkWants sink_wants() const RTC_LOCKS_EXCLUDED(mutex_);
+  bool has_sinks() const RTC_LOCKS_EXCLUDED(mutex_);
+
+ protected:
+  rtc::VideoSinkWants sink_wants_locked() const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
+                       const rtc::VideoSinkWants& wants)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+  void AddOrUpdateSinkLocked(rtc::VideoSinkInterface<VideoFrame>* sink,
+                             const rtc::VideoSinkWants& wants)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink)
+      RTC_LOCKS_EXCLUDED(mutex_) override;
+
+  mutable Mutex mutex_;
+  rtc::VideoSinkInterface<VideoFrame>* sink_ RTC_GUARDED_BY(mutex_);
+  rtc::VideoSinkWants sink_wants_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FRAME_FORWARDER_H_
diff --git a/third_party/libwebrtc/test/frame_generator.cc b/third_party/libwebrtc/test/frame_generator.cc
new file mode 100644
index 0000000000..b6f16a573d
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_generator.cc
@@ -0,0 +1,438 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/frame_generator.h"
+
+#include <string.h>
+
+#include <cstdint>
+#include <cstdio>
+#include <memory>
+
+#include "api/video/i010_buffer.h"
+#include "api/video/nv12_buffer.h"
+#include "api/video/video_rotation.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/checks.h"
+#include "test/frame_utils.h"
+
+namespace webrtc {
+namespace test {
+
+SquareGenerator::SquareGenerator(int width,
+                                 int height,
+                                 OutputType type,
+                                 int num_squares)
+    : type_(type) {
+  ChangeResolution(width, height);
+  for (int i = 0; i < num_squares; ++i) {
+    squares_.emplace_back(new Square(width, height, i + 1));
+  }
+}
+
+void SquareGenerator::ChangeResolution(size_t width, size_t height) {
+  MutexLock lock(&mutex_);
+  width_ = static_cast<int>(width);
+  height_ = static_cast<int>(height);
+  RTC_CHECK(width_ > 0);
+  RTC_CHECK(height_ > 0);
+}
+
+rtc::scoped_refptr<I420Buffer> SquareGenerator::CreateI420Buffer(int width,
+                                                                 int height) {
+  rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(width, height));
+  memset(buffer->MutableDataY(), 127, height * buffer->StrideY());
+  memset(buffer->MutableDataU(), 127,
+         buffer->ChromaHeight() * buffer->StrideU());
+  memset(buffer->MutableDataV(), 127,
+         buffer->ChromaHeight() * buffer->StrideV());
+  return buffer;
+}
+
+FrameGeneratorInterface::VideoFrameData SquareGenerator::NextFrame() {
+  MutexLock lock(&mutex_);
+
+  rtc::scoped_refptr<VideoFrameBuffer> buffer = nullptr;
+  switch (type_) {
+    case OutputType::kI420:
+    case OutputType::kI010:
+    case OutputType::kNV12: {
+      buffer = CreateI420Buffer(width_, height_);
+      break;
+    }
+    case OutputType::kI420A: {
+      rtc::scoped_refptr<I420Buffer> yuv_buffer =
+          CreateI420Buffer(width_, height_);
+      rtc::scoped_refptr<I420Buffer> axx_buffer =
+          CreateI420Buffer(width_, height_);
+      buffer = WrapI420ABuffer(yuv_buffer->width(), yuv_buffer->height(),
+                               yuv_buffer->DataY(), yuv_buffer->StrideY(),
+                               yuv_buffer->DataU(), yuv_buffer->StrideU(),
+                               yuv_buffer->DataV(), yuv_buffer->StrideV(),
+                               axx_buffer->DataY(), axx_buffer->StrideY(),
+                               // To keep references alive.
+                               [yuv_buffer, axx_buffer] {});
+      break;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED() << "The given output format is not supported.";
+  }
+
+  for (const auto& square : squares_)
+    square->Draw(buffer);
+
+  if (type_ == OutputType::kI010) {
+    buffer = I010Buffer::Copy(*buffer->ToI420());
+  } else if (type_ == OutputType::kNV12) {
+    buffer = NV12Buffer::Copy(*buffer->ToI420());
+  }
+
+  return VideoFrameData(buffer, absl::nullopt);
+}
+
+SquareGenerator::Square::Square(int width, int height, int seed)
+    : random_generator_(seed),
+      x_(random_generator_.Rand(0, width)),
+      y_(random_generator_.Rand(0, height)),
+      length_(random_generator_.Rand(1, width > 4 ? width / 4 : 1)),
+      yuv_y_(random_generator_.Rand(0, 255)),
+      yuv_u_(random_generator_.Rand(0, 255)),
+      yuv_v_(random_generator_.Rand(0, 255)),
+      yuv_a_(random_generator_.Rand(0, 255)) {}
+
+void SquareGenerator::Square::Draw(
+    const rtc::scoped_refptr<VideoFrameBuffer>& frame_buffer) {
+  RTC_DCHECK(frame_buffer->type() == VideoFrameBuffer::Type::kI420 ||
+             frame_buffer->type() == VideoFrameBuffer::Type::kI420A);
+  rtc::scoped_refptr<I420BufferInterface> buffer = frame_buffer->ToI420();
+  int length_cap = std::min(buffer->height(), buffer->width()) / 4;
+  int length = std::min(length_, length_cap);
+  x_ = (x_ + random_generator_.Rand(0, 4)) % (buffer->width() - length);
+  y_ = (y_ + random_generator_.Rand(0, 4)) % (buffer->height() - length);
+  for (int y = y_; y < y_ + length; ++y) {
+    uint8_t* pos_y =
+        (const_cast<uint8_t*>(buffer->DataY()) + x_ + y * buffer->StrideY());
+    memset(pos_y, yuv_y_, length);
+  }
+
+  for (int y = y_; y < y_ + length; y = y + 2) {
+    uint8_t* pos_u = (const_cast<uint8_t*>(buffer->DataU()) + x_ / 2 +
+                      y / 2 * buffer->StrideU());
+    memset(pos_u, yuv_u_, length / 2);
+    uint8_t* pos_v = (const_cast<uint8_t*>(buffer->DataV()) + x_ / 2 +
+                      y / 2 * buffer->StrideV());
+    memset(pos_v, yuv_v_, length / 2);
+  }
+
+  if (frame_buffer->type() == VideoFrameBuffer::Type::kI420)
+    return;
+
+  // Optionally draw on alpha plane if given.
+  const webrtc::I420ABufferInterface* yuva_buffer = frame_buffer->GetI420A();
+  for (int y = y_; y < y_ + length; ++y) {
+    uint8_t* pos_y = (const_cast<uint8_t*>(yuva_buffer->DataA()) + x_ +
+                      y * yuva_buffer->StrideA());
+    memset(pos_y, yuv_a_, length);
+  }
+}
+
+YuvFileGenerator::YuvFileGenerator(std::vector<FILE*> files,
+                                   size_t width,
+                                   size_t height,
+                                   int frame_repeat_count)
+    : file_index_(0),
+      frame_index_(std::numeric_limits<size_t>::max()),
+      files_(files),
+      width_(width),
+      height_(height),
+      frame_size_(CalcBufferSize(VideoType::kI420,
+                                 static_cast<int>(width_),
+                                 static_cast<int>(height_))),
+      frame_buffer_(new uint8_t[frame_size_]),
+      frame_display_count_(frame_repeat_count),
+      current_display_count_(0) {
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
+  RTC_DCHECK_GT(frame_repeat_count, 0);
+}
+
+YuvFileGenerator::~YuvFileGenerator() {
+  for (FILE* file : files_)
+    fclose(file);
+}
+
+FrameGeneratorInterface::VideoFrameData YuvFileGenerator::NextFrame() {
+  // Empty update by default.
+  VideoFrame::UpdateRect update_rect{0, 0, 0, 0};
+  if (current_display_count_ == 0) {
+    const bool got_new_frame = ReadNextFrame();
+    // Full update on a new frame from file.
+    if (got_new_frame) {
+      update_rect = VideoFrame::UpdateRect{0, 0, static_cast<int>(width_),
+                                           static_cast<int>(height_)};
+    }
+  }
+  if (++current_display_count_ >= frame_display_count_)
+    current_display_count_ = 0;
+
+  return VideoFrameData(last_read_buffer_, update_rect);
+}
+
+bool YuvFileGenerator::ReadNextFrame() {
+  size_t prev_frame_index = frame_index_;
+  size_t prev_file_index = file_index_;
+  last_read_buffer_ = test::ReadI420Buffer(
+      static_cast<int>(width_), static_cast<int>(height_), files_[file_index_]);
+  ++frame_index_;
+  if (!last_read_buffer_) {
+    // No more frames to read in this file, rewind and move to next file.
+    rewind(files_[file_index_]);
+
+    frame_index_ = 0;
+    file_index_ = (file_index_ + 1) % files_.size();
+    last_read_buffer_ =
+        test::ReadI420Buffer(static_cast<int>(width_),
+                             static_cast<int>(height_), files_[file_index_]);
+    RTC_CHECK(last_read_buffer_);
+  }
+  return frame_index_ != prev_frame_index || file_index_ != prev_file_index;
+}
+
+NV12FileGenerator::NV12FileGenerator(std::vector<FILE*> files,
+                                     size_t width,
+                                     size_t height,
+                                     int frame_repeat_count)
+    : file_index_(0),
+      frame_index_(std::numeric_limits<size_t>::max()),
+      files_(files),
+      width_(width),
+      height_(height),
+      frame_size_(CalcBufferSize(VideoType::kNV12,
+                                 static_cast<int>(width_),
+                                 static_cast<int>(height_))),
+      frame_buffer_(new uint8_t[frame_size_]),
+      frame_display_count_(frame_repeat_count),
+      current_display_count_(0) {
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
+  RTC_DCHECK_GT(frame_repeat_count, 0);
+}
+
+NV12FileGenerator::~NV12FileGenerator() {
+  for (FILE* file : files_)
+    fclose(file);
+}
+
+FrameGeneratorInterface::VideoFrameData NV12FileGenerator::NextFrame() {
+  // Empty update by default.
+  VideoFrame::UpdateRect update_rect{0, 0, 0, 0};
+  if (current_display_count_ == 0) {
+    const bool got_new_frame = ReadNextFrame();
+    // Full update on a new frame from file.
+    if (got_new_frame) {
+      update_rect = VideoFrame::UpdateRect{0, 0, static_cast<int>(width_),
+                                           static_cast<int>(height_)};
+    }
+  }
+  if (++current_display_count_ >= frame_display_count_)
+    current_display_count_ = 0;
+
+  return VideoFrameData(last_read_buffer_, update_rect);
+}
+
+bool NV12FileGenerator::ReadNextFrame() {
+  size_t prev_frame_index = frame_index_;
+  size_t prev_file_index = file_index_;
+  last_read_buffer_ = test::ReadNV12Buffer(
+      static_cast<int>(width_), static_cast<int>(height_), files_[file_index_]);
+  ++frame_index_;
+  if (!last_read_buffer_) {
+    // No more frames to read in this file, rewind and move to next file.
+    rewind(files_[file_index_]);
+
+    frame_index_ = 0;
+    file_index_ = (file_index_ + 1) % files_.size();
+    last_read_buffer_ =
+        test::ReadNV12Buffer(static_cast<int>(width_),
+                             static_cast<int>(height_), files_[file_index_]);
+    RTC_CHECK(last_read_buffer_);
+  }
+  return frame_index_ != prev_frame_index || file_index_ != prev_file_index;
+}
+
+SlideGenerator::SlideGenerator(int width, int height, int frame_repeat_count)
+    : width_(width),
+      height_(height),
+      frame_display_count_(frame_repeat_count),
+      current_display_count_(0),
+      random_generator_(1234) {
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
+  RTC_DCHECK_GT(frame_repeat_count, 0);
+}
+
+FrameGeneratorInterface::VideoFrameData SlideGenerator::NextFrame() {
+  if (current_display_count_ == 0)
+    GenerateNewFrame();
+  if (++current_display_count_ >= frame_display_count_)
+    current_display_count_ = 0;
+
+  return VideoFrameData(buffer_, absl::nullopt);
+}
+
+void SlideGenerator::GenerateNewFrame() {
+  // The squares should have a varying order of magnitude in order
+  // to simulate variation in the slides' complexity.
+  const int kSquareNum = 1 << (4 + (random_generator_.Rand(0, 3) * 2));
+
+  buffer_ = I420Buffer::Create(width_, height_);
+  memset(buffer_->MutableDataY(), 127, height_ * buffer_->StrideY());
+  memset(buffer_->MutableDataU(), 127,
+         buffer_->ChromaHeight() * buffer_->StrideU());
+  memset(buffer_->MutableDataV(), 127,
+         buffer_->ChromaHeight() * buffer_->StrideV());
+
+  for (int i = 0; i < kSquareNum; ++i) {
+    int length = random_generator_.Rand(1, width_ > 4 ? width_ / 4 : 1);
+    // Limit the length of later squares so that they don't overwrite the
+    // previous ones too much.
+    length = (length * (kSquareNum - i)) / kSquareNum;
+
+    int x = random_generator_.Rand(0, width_ - length);
+    int y = random_generator_.Rand(0, height_ - length);
+    uint8_t yuv_y = random_generator_.Rand(0, 255);
+    uint8_t yuv_u = random_generator_.Rand(0, 255);
+    uint8_t yuv_v = random_generator_.Rand(0, 255);
+
+    for (int yy = y; yy < y + length; ++yy) {
+      uint8_t* pos_y = (buffer_->MutableDataY() + x + yy * buffer_->StrideY());
+      memset(pos_y, yuv_y, length);
+    }
+    for (int yy = y; yy < y + length; yy += 2) {
+      uint8_t* pos_u =
+          (buffer_->MutableDataU() + x / 2 + yy / 2 * buffer_->StrideU());
+      memset(pos_u, yuv_u, length / 2);
+      uint8_t* pos_v =
+          (buffer_->MutableDataV() + x / 2 + yy / 2 * buffer_->StrideV());
+      memset(pos_v, yuv_v, length / 2);
+    }
+  }
+}
+
+ScrollingImageFrameGenerator::ScrollingImageFrameGenerator(
+    Clock* clock,
+    const std::vector<FILE*>& files,
+    size_t source_width,
+    size_t source_height,
+    size_t target_width,
+    size_t target_height,
+    int64_t scroll_time_ms,
+    int64_t pause_time_ms)
+    : clock_(clock),
+      start_time_(clock->TimeInMilliseconds()),
+      scroll_time_(scroll_time_ms),
+      pause_time_(pause_time_ms),
+      num_frames_(files.size()),
+      target_width_(static_cast<int>(target_width)),
+      target_height_(static_cast<int>(target_height)),
+      current_frame_num_(num_frames_ - 1),
+      prev_frame_not_scrolled_(false),
+      current_source_frame_(nullptr, absl::nullopt),
+      current_frame_(nullptr, absl::nullopt),
+      file_generator_(files, source_width, source_height, 1) {
+  RTC_DCHECK(clock_ != nullptr);
+  RTC_DCHECK_GT(num_frames_, 0);
+  RTC_DCHECK_GE(source_height, target_height);
+  RTC_DCHECK_GE(source_width, target_width);
+  RTC_DCHECK_GE(scroll_time_ms, 0);
+  RTC_DCHECK_GE(pause_time_ms, 0);
+  RTC_DCHECK_GT(scroll_time_ms + pause_time_ms, 0);
+}
+
+FrameGeneratorInterface::VideoFrameData
+ScrollingImageFrameGenerator::NextFrame() {
+  const int64_t kFrameDisplayTime = scroll_time_ + pause_time_;
+  const int64_t now = clock_->TimeInMilliseconds();
+  int64_t ms_since_start = now - start_time_;
+
+  size_t frame_num = (ms_since_start / kFrameDisplayTime) % num_frames_;
+  UpdateSourceFrame(frame_num);
+
+  bool cur_frame_not_scrolled;
+
+  double scroll_factor;
+  int64_t time_into_frame = ms_since_start % kFrameDisplayTime;
+  if (time_into_frame < scroll_time_) {
+    scroll_factor = static_cast<double>(time_into_frame) / scroll_time_;
+    cur_frame_not_scrolled = false;
+  } else {
+    scroll_factor = 1.0;
+    cur_frame_not_scrolled = true;
+  }
+  CropSourceToScrolledImage(scroll_factor);
+
+  bool same_scroll_position =
+      prev_frame_not_scrolled_ && cur_frame_not_scrolled;
+  if (!same_scroll_position) {
+    // If scrolling is not finished yet, force full frame update.
+    current_frame_.update_rect =
+        VideoFrame::UpdateRect{0, 0, target_width_, target_height_};
+  }
+  prev_frame_not_scrolled_ = cur_frame_not_scrolled;
+
+  return current_frame_;
+}
+
+void ScrollingImageFrameGenerator::UpdateSourceFrame(size_t frame_num) {
+  VideoFrame::UpdateRect acc_update{0, 0, 0, 0};
+  while (current_frame_num_ != frame_num) {
+    current_source_frame_ = file_generator_.NextFrame();
+    if (current_source_frame_.update_rect) {
+      acc_update.Union(*current_source_frame_.update_rect);
+    }
+    current_frame_num_ = (current_frame_num_ + 1) % num_frames_;
+  }
+  current_source_frame_.update_rect = acc_update;
+}
+
+void ScrollingImageFrameGenerator::CropSourceToScrolledImage(
+    double scroll_factor) {
+  int scroll_margin_x = current_source_frame_.buffer->width() - target_width_;
+  int pixels_scrolled_x =
+      static_cast<int>(scroll_margin_x * scroll_factor + 0.5);
+  int scroll_margin_y = current_source_frame_.buffer->height() - target_height_;
+  int pixels_scrolled_y =
+      static_cast<int>(scroll_margin_y * scroll_factor + 0.5);
+
+  rtc::scoped_refptr<I420BufferInterface> i420_buffer =
+      current_source_frame_.buffer->ToI420();
+  int offset_y =
+      (i420_buffer->StrideY() * pixels_scrolled_y) + pixels_scrolled_x;
+  int offset_u = (i420_buffer->StrideU() * (pixels_scrolled_y / 2)) +
+                 (pixels_scrolled_x / 2);
+  int offset_v = (i420_buffer->StrideV() * (pixels_scrolled_y / 2)) +
+                 (pixels_scrolled_x / 2);
+
+  VideoFrame::UpdateRect update_rect =
+      current_source_frame_.update_rect->IsEmpty()
+          ? VideoFrame::UpdateRect{0, 0, 0, 0}
+          : VideoFrame::UpdateRect{0, 0, target_width_, target_height_};
+  current_frame_ = VideoFrameData(
+      WrapI420Buffer(target_width_, target_height_,
+                     &i420_buffer->DataY()[offset_y], i420_buffer->StrideY(),
+                     &i420_buffer->DataU()[offset_u], i420_buffer->StrideU(),
+                     &i420_buffer->DataV()[offset_v], i420_buffer->StrideV(),
+                     // To keep reference alive.
+                     [i420_buffer] {}),
+      update_rect);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/frame_generator.h b/third_party/libwebrtc/test/frame_generator.h
new file mode 100644
index 0000000000..9a8f08cea6
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_generator.h
@@ -0,0 +1,197 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_FRAME_GENERATOR_H_
+#define TEST_FRAME_GENERATOR_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/scoped_refptr.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/nv12_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_source_interface.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/random.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace test {
+
+// SquareGenerator is a FrameGenerator that draws a given amount of randomly
+// sized and colored squares. Between each new generated frame, the squares
+// are moved slightly towards the lower right corner.
+class SquareGenerator : public FrameGeneratorInterface {
+ public:
+  SquareGenerator(int width, int height, OutputType type, int num_squares);
+
+  void ChangeResolution(size_t width, size_t height) override;
+  VideoFrameData NextFrame() override;
+
+ private:
+  rtc::scoped_refptr<I420Buffer> CreateI420Buffer(int width, int height);
+
+  class Square {
+   public:
+    Square(int width, int height, int seed);
+
+    void Draw(const rtc::scoped_refptr<VideoFrameBuffer>& frame_buffer);
+
+   private:
+    Random random_generator_;
+    int x_;
+    int y_;
+    const int length_;
+    const uint8_t yuv_y_;
+    const uint8_t yuv_u_;
+    const uint8_t yuv_v_;
+    const uint8_t yuv_a_;
+  };
+
+  Mutex mutex_;
+  const OutputType type_;
+  int width_ RTC_GUARDED_BY(&mutex_);
+  int height_ RTC_GUARDED_BY(&mutex_);
+  std::vector<std::unique_ptr<Square>> squares_ RTC_GUARDED_BY(&mutex_);
+};
+
+class YuvFileGenerator : public FrameGeneratorInterface {
+ public:
+  YuvFileGenerator(std::vector<FILE*> files,
+                   size_t width,
+                   size_t height,
+                   int frame_repeat_count);
+
+  ~YuvFileGenerator();
+
+  VideoFrameData NextFrame() override;
+  void ChangeResolution(size_t width, size_t height) override {
+    RTC_LOG(LS_WARNING) << "YuvFileGenerator::ChangeResolution not implemented";
+  }
+
+ private:
+  // Returns true if the new frame was loaded.
+  // False only in case of a single file with a single frame in it.
+  bool ReadNextFrame();
+
+  size_t file_index_;
+  size_t frame_index_;
+  const std::vector<FILE*> files_;
+  const size_t width_;
+  const size_t height_;
+  const size_t frame_size_;
+  const std::unique_ptr<uint8_t[]> frame_buffer_;
+  const int frame_display_count_;
+  int current_display_count_;
+  rtc::scoped_refptr<I420Buffer> last_read_buffer_;
+};
+
+class NV12FileGenerator : public FrameGeneratorInterface {
+ public:
+  NV12FileGenerator(std::vector<FILE*> files,
+                    size_t width,
+                    size_t height,
+                    int frame_repeat_count);
+
+  ~NV12FileGenerator();
+
+  VideoFrameData NextFrame() override;
+  void ChangeResolution(size_t width, size_t height) override {
+    RTC_LOG(LS_WARNING)
+        << "NV12FileGenerator::ChangeResolution not implemented";
+  }
+
+ private:
+  // Returns true if the new frame was loaded.
+  // False only in case of a single file with a single frame in it.
+  bool ReadNextFrame();
+
+  size_t file_index_;
+  size_t frame_index_;
+  const std::vector<FILE*> files_;
+  const size_t width_;
+  const size_t height_;
+  const size_t frame_size_;
+  const std::unique_ptr<uint8_t[]> frame_buffer_;
+  const int frame_display_count_;
+  int current_display_count_;
+  rtc::scoped_refptr<NV12Buffer> last_read_buffer_;
+};
+
+// SlideGenerator works similarly to YuvFileGenerator but it fills the frames
+// with randomly sized and colored squares instead of reading their content
+// from files.
+class SlideGenerator : public FrameGeneratorInterface {
+ public:
+  SlideGenerator(int width, int height, int frame_repeat_count);
+
+  VideoFrameData NextFrame() override;
+  void ChangeResolution(size_t width, size_t height) override {
+    RTC_LOG(LS_WARNING) << "SlideGenerator::ChangeResolution not implemented";
+  }
+
+ private:
+  // Generates some randomly sized and colored squares scattered
+  // over the frame.
+  void GenerateNewFrame();
+
+  const int width_;
+  const int height_;
+  const int frame_display_count_;
+  int current_display_count_;
+  Random random_generator_;
+  rtc::scoped_refptr<I420Buffer> buffer_;
+};
+
+class ScrollingImageFrameGenerator : public FrameGeneratorInterface {
+ public:
+  ScrollingImageFrameGenerator(Clock* clock,
+                               const std::vector<FILE*>& files,
+                               size_t source_width,
+                               size_t source_height,
+                               size_t target_width,
+                               size_t target_height,
+                               int64_t scroll_time_ms,
+                               int64_t pause_time_ms);
+  ~ScrollingImageFrameGenerator() override = default;
+
+  VideoFrameData NextFrame() override;
+  void ChangeResolution(size_t width, size_t height) override {
+    RTC_LOG(LS_WARNING)
+        << "ScrollingImageFrameGenerator::ChangeResolution not implemented";
+  }
+
+ private:
+  void UpdateSourceFrame(size_t frame_num);
+  void CropSourceToScrolledImage(double scroll_factor);
+
+  Clock* const clock_;
+  const int64_t start_time_;
+  const int64_t scroll_time_;
+  const int64_t pause_time_;
+  const size_t num_frames_;
+  const int target_width_;
+  const int target_height_;
+
+  size_t current_frame_num_;
+  bool prev_frame_not_scrolled_;
+  VideoFrameData current_source_frame_;
+  VideoFrameData current_frame_;
+  YuvFileGenerator file_generator_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FRAME_GENERATOR_H_
diff --git a/third_party/libwebrtc/test/frame_generator_capturer.cc b/third_party/libwebrtc/test/frame_generator_capturer.cc
new file mode 100644
index 0000000000..c69fca0965
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_generator_capturer.cc
@@ -0,0 +1,323 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/frame_generator_capturer.h"
+
+#include <algorithm>
+#include <cmath>
+#include <limits>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/test/create_frame_generator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+std::string TransformFilePath(std::string path) {
+  static const std::string resource_prefix = "res://";
+  int ext_pos = path.rfind('.');
+  if (ext_pos < 0) {
+    return test::ResourcePath(path, "yuv");
+  } else if (absl::StartsWith(path, resource_prefix)) {
+    std::string name = path.substr(resource_prefix.length(), ext_pos);
+    std::string ext = path.substr(ext_pos, path.size());
+    return test::ResourcePath(name, ext);
+  }
+  return path;
+}
+}  // namespace
+
+FrameGeneratorCapturer::FrameGeneratorCapturer(
+    Clock* clock,
+    std::unique_ptr<FrameGeneratorInterface> frame_generator,
+    int target_fps,
+    TaskQueueFactory& task_queue_factory)
+    : clock_(clock),
+      sending_(true),
+      sink_wants_observer_(nullptr),
+      frame_generator_(std::move(frame_generator)),
+      source_fps_(target_fps),
+      target_capture_fps_(target_fps),
+      first_frame_capture_time_(-1),
+      task_queue_(task_queue_factory.CreateTaskQueue(
+          "FrameGenCapQ",
+          TaskQueueFactory::Priority::HIGH)) {
+  RTC_DCHECK(frame_generator_);
+  RTC_DCHECK_GT(target_fps, 0);
+}
+
+FrameGeneratorCapturer::~FrameGeneratorCapturer() {
+  Stop();
+}
+
+std::unique_ptr<FrameGeneratorCapturer> FrameGeneratorCapturer::Create(
+    Clock* clock,
+    TaskQueueFactory& task_queue_factory,
+    FrameGeneratorCapturerConfig::SquaresVideo config) {
+  return std::make_unique<FrameGeneratorCapturer>(
+      clock,
+      CreateSquareFrameGenerator(config.width, config.height,
+                                 config.pixel_format, config.num_squares),
+      config.framerate, task_queue_factory);
+}
+std::unique_ptr<FrameGeneratorCapturer> FrameGeneratorCapturer::Create(
+    Clock* clock,
+    TaskQueueFactory& task_queue_factory,
+    FrameGeneratorCapturerConfig::SquareSlides config) {
+  return std::make_unique<FrameGeneratorCapturer>(
+      clock,
+      CreateSlideFrameGenerator(
+          config.width, config.height,
+          /*frame_repeat_count*/ config.change_interval.seconds<double>() *
+              config.framerate),
+      config.framerate, task_queue_factory);
+}
+std::unique_ptr<FrameGeneratorCapturer> FrameGeneratorCapturer::Create(
+    Clock* clock,
+    TaskQueueFactory& task_queue_factory,
+    FrameGeneratorCapturerConfig::VideoFile config) {
+  RTC_CHECK(config.width && config.height);
+  return std::make_unique<FrameGeneratorCapturer>(
+      clock,
+      CreateFromYuvFileFrameGenerator({TransformFilePath(config.name)},
+                                      config.width, config.height,
+                                      /*frame_repeat_count*/ 1),
+      config.framerate, task_queue_factory);
+}
+
+std::unique_ptr<FrameGeneratorCapturer> FrameGeneratorCapturer::Create(
+    Clock* clock,
+    TaskQueueFactory& task_queue_factory,
+    FrameGeneratorCapturerConfig::ImageSlides config) {
+  std::unique_ptr<FrameGeneratorInterface> slides_generator;
+  std::vector<std::string> paths = config.paths;
+  for (std::string& path : paths)
+    path = TransformFilePath(path);
+
+  if (config.crop.width || config.crop.height) {
+    TimeDelta pause_duration =
+        config.change_interval - config.crop.scroll_duration;
+    RTC_CHECK_GE(pause_duration, TimeDelta::Zero());
+    int crop_width = config.crop.width.value_or(config.width);
+    int crop_height = config.crop.height.value_or(config.height);
+    RTC_CHECK_LE(crop_width, config.width);
+    RTC_CHECK_LE(crop_height, config.height);
+    slides_generator = CreateScrollingInputFromYuvFilesFrameGenerator(
+        clock, paths, config.width, config.height, crop_width, crop_height,
+        config.crop.scroll_duration.ms(), pause_duration.ms());
+  } else {
+    slides_generator = CreateFromYuvFileFrameGenerator(
+        paths, config.width, config.height,
+        /*frame_repeat_count*/ config.change_interval.seconds<double>() *
+            config.framerate);
+  }
+  return std::make_unique<FrameGeneratorCapturer>(
+      clock, std::move(slides_generator), config.framerate, task_queue_factory);
+}
+
+std::unique_ptr<FrameGeneratorCapturer> FrameGeneratorCapturer::Create(
+    Clock* clock,
+    TaskQueueFactory& task_queue_factory,
+    const FrameGeneratorCapturerConfig& config) {
+  if (config.video_file) {
+    return Create(clock, task_queue_factory, *config.video_file);
+  } else if (config.image_slides) {
+    return Create(clock, task_queue_factory, *config.image_slides);
+  } else if (config.squares_slides) {
+    return Create(clock, task_queue_factory, *config.squares_slides);
+  } else {
+    return Create(clock, task_queue_factory,
+                  config.squares_video.value_or(
+                      FrameGeneratorCapturerConfig::SquaresVideo()));
+  }
+}
+
+void FrameGeneratorCapturer::SetFakeRotation(VideoRotation rotation) {
+  MutexLock lock(&lock_);
+  fake_rotation_ = rotation;
+}
+
+void FrameGeneratorCapturer::SetFakeColorSpace(
+    absl::optional<ColorSpace> color_space) {
+  MutexLock lock(&lock_);
+  fake_color_space_ = color_space;
+}
+
+bool FrameGeneratorCapturer::Init() {
+  // This check is added because frame_generator_ might be file based and should
+  // not crash because a file moved.
+  if (frame_generator_.get() == nullptr)
+    return false;
+
+  frame_task_ = RepeatingTaskHandle::DelayedStart(
+      task_queue_.Get(),
+      TimeDelta::Seconds(1) / GetCurrentConfiguredFramerate(),
+      [this] {
+        InsertFrame();
+        return TimeDelta::Seconds(1) / GetCurrentConfiguredFramerate();
+      },
+      TaskQueueBase::DelayPrecision::kHigh);
+  return true;
+}
+
+void FrameGeneratorCapturer::InsertFrame() {
+  absl::optional<Resolution> resolution;
+
+  {
+    MutexLock lock(&lock_);
+    if (sending_) {
+      FrameGeneratorInterface::VideoFrameData frame_data =
+          frame_generator_->NextFrame();
+      // TODO(srte): Use more advanced frame rate control to allow arbritrary
+      // fractions.
+      int decimation =
+          std::round(static_cast<double>(source_fps_) / target_capture_fps_);
+      for (int i = 1; i < decimation; ++i)
+        frame_data = frame_generator_->NextFrame();
+
+      VideoFrame frame =
+          VideoFrame::Builder()
+              .set_video_frame_buffer(frame_data.buffer)
+              .set_rotation(fake_rotation_)
+              .set_timestamp_us(clock_->TimeInMicroseconds())
+              .set_ntp_time_ms(clock_->CurrentNtpInMilliseconds())
+              .set_update_rect(frame_data.update_rect)
+              .set_color_space(fake_color_space_)
+              .build();
+      if (first_frame_capture_time_ == -1) {
+        first_frame_capture_time_ = frame.ntp_time_ms();
+      }
+
+      resolution = Resolution{frame.width(), frame.height()};
+
+      TestVideoCapturer::OnFrame(frame);
+    }
+  }
+
+  if (resolution) {
+    MutexLock lock(&stats_lock_);
+    source_resolution_ = resolution;
+  }
+}
+
+absl::optional<FrameGeneratorCapturer::Resolution>
+FrameGeneratorCapturer::GetResolution() {
+  MutexLock lock(&stats_lock_);
+  return source_resolution_;
+}
+
+void FrameGeneratorCapturer::Start() {
+  {
+    MutexLock lock(&lock_);
+    sending_ = true;
+  }
+  if (!frame_task_.Running()) {
+    frame_task_ = RepeatingTaskHandle::Start(
+        task_queue_.Get(),
+        [this] {
+          InsertFrame();
+          return TimeDelta::Seconds(1) / GetCurrentConfiguredFramerate();
+        },
+        TaskQueueBase::DelayPrecision::kHigh);
+  }
+}
+
+void FrameGeneratorCapturer::Stop() {
+  MutexLock lock(&lock_);
+  sending_ = false;
+}
+
+void FrameGeneratorCapturer::ChangeResolution(size_t width, size_t height) {
+  MutexLock lock(&lock_);
+  frame_generator_->ChangeResolution(width, height);
+}
+
+void FrameGeneratorCapturer::ChangeFramerate(int target_framerate) {
+  MutexLock lock(&lock_);
+  RTC_CHECK(target_capture_fps_ > 0);
+  if (target_framerate > source_fps_)
+    RTC_LOG(LS_WARNING) << "Target framerate clamped from " << target_framerate
+                        << " to " << source_fps_;
+  if (source_fps_ % target_capture_fps_ != 0) {
+    int decimation =
+        std::round(static_cast<double>(source_fps_) / target_capture_fps_);
+    int effective_rate = target_capture_fps_ / decimation;
+    RTC_LOG(LS_WARNING) << "Target framerate, " << target_framerate
+                        << ", is an uneven fraction of the source rate, "
+                        << source_fps_
+                        << ". The framerate will be :" << effective_rate;
+  }
+  target_capture_fps_ = std::min(source_fps_, target_framerate);
+}
+
+void FrameGeneratorCapturer::OnOutputFormatRequest(
+    int width,
+    int height,
+    const absl::optional<int>& max_fps) {
+  TestVideoCapturer::OnOutputFormatRequest(width, height, max_fps);
+}
+
+void FrameGeneratorCapturer::SetSinkWantsObserver(SinkWantsObserver* observer) {
+  MutexLock lock(&lock_);
+  RTC_DCHECK(!sink_wants_observer_);
+  sink_wants_observer_ = observer;
+}
+
+void FrameGeneratorCapturer::AddOrUpdateSink(
+    rtc::VideoSinkInterface<VideoFrame>* sink,
+    const rtc::VideoSinkWants& wants) {
+  TestVideoCapturer::AddOrUpdateSink(sink, wants);
+  MutexLock lock(&lock_);
+  if (sink_wants_observer_) {
+    // Tests need to observe unmodified sink wants.
+    sink_wants_observer_->OnSinkWantsChanged(sink, wants);
+  }
+  UpdateFps(GetSinkWants().max_framerate_fps);
+}
+
+void FrameGeneratorCapturer::RemoveSink(
+    rtc::VideoSinkInterface<VideoFrame>* sink) {
+  TestVideoCapturer::RemoveSink(sink);
+
+  MutexLock lock(&lock_);
+  UpdateFps(GetSinkWants().max_framerate_fps);
+}
+
+void FrameGeneratorCapturer::UpdateFps(int max_fps) {
+  if (max_fps < target_capture_fps_) {
+    wanted_fps_.emplace(max_fps);
+  } else {
+    wanted_fps_.reset();
+  }
+}
+
+void FrameGeneratorCapturer::ForceFrame() {
+  // One-time non-repeating task,
+  task_queue_.PostTask([this] { InsertFrame(); });
+}
+
+int FrameGeneratorCapturer::GetCurrentConfiguredFramerate() {
+  MutexLock lock(&lock_);
+  if (wanted_fps_ && *wanted_fps_ < target_capture_fps_)
+    return *wanted_fps_;
+  return target_capture_fps_;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/frame_generator_capturer.h b/third_party/libwebrtc/test/frame_generator_capturer.h
new file mode 100644
index 0000000000..e310e40129
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_generator_capturer.h
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_FRAME_GENERATOR_CAPTURER_H_
+#define TEST_FRAME_GENERATOR_CAPTURER_H_
+
+#include <memory>
+#include <string>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "system_wrappers/include/clock.h"
+#include "test/test_video_capturer.h"
+
+namespace webrtc {
+
+namespace test {
+namespace frame_gen_cap_impl {
+template <typename T>
+class AutoOpt : public absl::optional<T> {
+ public:
+  using absl::optional<T>::optional;
+  T* operator->() {
+    if (!absl::optional<T>::has_value())
+      this->emplace(T());
+    return absl::optional<T>::operator->();
+  }
+};
+}  // namespace frame_gen_cap_impl
+struct FrameGeneratorCapturerConfig {
+  struct SquaresVideo {
+    int framerate = 30;
+    FrameGeneratorInterface::OutputType pixel_format =
+        FrameGeneratorInterface::OutputType::kI420;
+    int width = 320;
+    int height = 180;
+    int num_squares = 10;
+  };
+
+  struct SquareSlides {
+    int framerate = 30;
+    TimeDelta change_interval = TimeDelta::Seconds(10);
+    int width = 1600;
+    int height = 1200;
+  };
+
+  struct VideoFile {
+    int framerate = 30;
+    std::string name;
+    // Must be set to width and height of the source video file.
+    int width = 0;
+    int height = 0;
+  };
+
+  struct ImageSlides {
+    int framerate = 30;
+    TimeDelta change_interval = TimeDelta::Seconds(10);
+    struct Crop {
+      TimeDelta scroll_duration = TimeDelta::Seconds(0);
+      absl::optional<int> width;
+      absl::optional<int> height;
+    } crop;
+    int width = 1850;
+    int height = 1110;
+    std::vector<std::string> paths = {
+        "web_screenshot_1850_1110",
+        "presentation_1850_1110",
+        "photo_1850_1110",
+        "difficult_photo_1850_1110",
+    };
+  };
+
+  frame_gen_cap_impl::AutoOpt<SquaresVideo> squares_video;
+  frame_gen_cap_impl::AutoOpt<SquareSlides> squares_slides;
+  frame_gen_cap_impl::AutoOpt<VideoFile> video_file;
+  frame_gen_cap_impl::AutoOpt<ImageSlides> image_slides;
+};
+
+class FrameGeneratorCapturer : public TestVideoCapturer {
+ public:
+  class SinkWantsObserver {
+   public:
+    // OnSinkWantsChanged is called when FrameGeneratorCapturer::AddOrUpdateSink
+    // is called.
+    virtual void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
+                                    const rtc::VideoSinkWants& wants) = 0;
+
+   protected:
+    virtual ~SinkWantsObserver() {}
+  };
+
+  FrameGeneratorCapturer(
+      Clock* clock,
+      std::unique_ptr<FrameGeneratorInterface> frame_generator,
+      int target_fps,
+      TaskQueueFactory& task_queue_factory);
+  virtual ~FrameGeneratorCapturer();
+
+  static std::unique_ptr<FrameGeneratorCapturer> Create(
+      Clock* clock,
+      TaskQueueFactory& task_queue_factory,
+      FrameGeneratorCapturerConfig::SquaresVideo config);
+  static std::unique_ptr<FrameGeneratorCapturer> Create(
+      Clock* clock,
+      TaskQueueFactory& task_queue_factory,
+      FrameGeneratorCapturerConfig::SquareSlides config);
+  static std::unique_ptr<FrameGeneratorCapturer> Create(
+      Clock* clock,
+      TaskQueueFactory& task_queue_factory,
+      FrameGeneratorCapturerConfig::VideoFile config);
+  static std::unique_ptr<FrameGeneratorCapturer> Create(
+      Clock* clock,
+      TaskQueueFactory& task_queue_factory,
+      FrameGeneratorCapturerConfig::ImageSlides config);
+  static std::unique_ptr<FrameGeneratorCapturer> Create(
+      Clock* clock,
+      TaskQueueFactory& task_queue_factory,
+      const FrameGeneratorCapturerConfig& config);
+
+  void Start();
+  void Stop();
+  void ChangeResolution(size_t width, size_t height);
+  void ChangeFramerate(int target_framerate);
+
+  struct Resolution {
+    int width;
+    int height;
+  };
+  absl::optional<Resolution> GetResolution();
+
+  void OnOutputFormatRequest(int width,
+                             int height,
+                             const absl::optional<int>& max_fps);
+
+  void SetSinkWantsObserver(SinkWantsObserver* observer);
+
+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
+                       const rtc::VideoSinkWants& wants) override;
+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override;
+
+  void ForceFrame();
+  void SetFakeRotation(VideoRotation rotation);
+  void SetFakeColorSpace(absl::optional<ColorSpace> color_space);
+
+  int64_t first_frame_capture_time() const { return first_frame_capture_time_; }
+
+  bool Init();
+
+ private:
+  void InsertFrame();
+  static bool Run(void* obj);
+  int GetCurrentConfiguredFramerate();
+  void UpdateFps(int max_fps) RTC_EXCLUSIVE_LOCKS_REQUIRED(&lock_);
+
+  Clock* const clock_;
+  RepeatingTaskHandle frame_task_;
+  bool sending_;
+  SinkWantsObserver* sink_wants_observer_ RTC_GUARDED_BY(&lock_);
+
+  Mutex lock_;
+  std::unique_ptr<FrameGeneratorInterface> frame_generator_;
+
+  int source_fps_ RTC_GUARDED_BY(&lock_);
+  int target_capture_fps_ RTC_GUARDED_BY(&lock_);
+  absl::optional<int> wanted_fps_ RTC_GUARDED_BY(&lock_);
+  VideoRotation fake_rotation_ = kVideoRotation_0;
+  absl::optional<ColorSpace> fake_color_space_ RTC_GUARDED_BY(&lock_);
+
+  int64_t first_frame_capture_time_;
+
+  Mutex stats_lock_;
+  absl::optional<Resolution> source_resolution_ RTC_GUARDED_BY(&stats_lock_);
+
+  // Must be the last field, so it will be deconstructed first as tasks
+  // in the TaskQueue access other fields of the instance of this class.
+  rtc::TaskQueue task_queue_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FRAME_GENERATOR_CAPTURER_H_
diff --git a/third_party/libwebrtc/test/frame_generator_capturer_unittest.cc b/third_party/libwebrtc/test/frame_generator_capturer_unittest.cc
new file mode 100644
index 0000000000..d8371f4efd
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_generator_capturer_unittest.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/frame_generator_capturer.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+using ::testing::Eq;
+using ::testing::Property;
+
+constexpr int kWidth = 640;
+constexpr int kHeight = 360;
+
+class MockVideoSinkInterfaceVideoFrame
+    : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  MOCK_METHOD(void, OnFrame, (const VideoFrame& frame), (override));
+  MOCK_METHOD(void, OnDiscardedFrame, (), (override));
+};
+}  // namespace
+
+TEST(FrameGeneratorCapturerTest, CreateFromConfig) {
+  GlobalSimulatedTimeController time(Timestamp::Seconds(1000));
+  FrameGeneratorCapturerConfig config;
+  config.squares_video->width = 300;
+  config.squares_video->height = 200;
+  config.squares_video->framerate = 20;
+  auto capturer = FrameGeneratorCapturer::Create(
+      time.GetClock(), *time.GetTaskQueueFactory(), config);
+  testing::StrictMock<MockVideoSinkInterfaceVideoFrame> mock_sink;
+  capturer->AddOrUpdateSink(&mock_sink, rtc::VideoSinkWants());
+  capturer->Start();
+  EXPECT_CALL(mock_sink, OnFrame(Property(&VideoFrame::width, Eq(300))))
+      .Times(21);
+  time.AdvanceTime(TimeDelta::Seconds(1));
+}
+
+TEST(FrameGeneratorCapturerTest, OnOutputFormatRequest) {
+  GlobalSimulatedTimeController time(Timestamp::Seconds(1000));
+  FrameGeneratorCapturerConfig config;
+  config.squares_video->width = kWidth;
+  config.squares_video->height = kHeight;
+  config.squares_video->framerate = 20;
+  auto capturer = FrameGeneratorCapturer::Create(
+      time.GetClock(), *time.GetTaskQueueFactory(), config);
+  testing::StrictMock<MockVideoSinkInterfaceVideoFrame> mock_sink;
+  capturer->AddOrUpdateSink(&mock_sink, rtc::VideoSinkWants());
+  capturer->OnOutputFormatRequest(kWidth / 2, kHeight / 2, /*max_fps=*/10);
+  capturer->Start();
+  EXPECT_CALL(mock_sink, OnFrame(Property(&VideoFrame::width, Eq(kWidth / 2))))
+      .Times(11);
+  time.AdvanceTime(TimeDelta::Seconds(1));
+}
+
+TEST(FrameGeneratorCapturerTest, ChangeResolution) {
+  GlobalSimulatedTimeController time(Timestamp::Seconds(1000));
+  FrameGeneratorCapturerConfig config;
+  config.squares_video->width = kWidth;
+  config.squares_video->height = kHeight;
+  config.squares_video->framerate = 20;
+  auto capturer = FrameGeneratorCapturer::Create(
+      time.GetClock(), *time.GetTaskQueueFactory(), config);
+  EXPECT_FALSE(capturer->GetResolution());
+  capturer->Start();
+  time.AdvanceTime(TimeDelta::Seconds(1));
+  ASSERT_TRUE(capturer->GetResolution());
+  EXPECT_EQ(kWidth, capturer->GetResolution()->width);
+  EXPECT_EQ(kHeight, capturer->GetResolution()->height);
+
+  capturer->ChangeResolution(kWidth / 2, kHeight / 2);
+  time.AdvanceTime(TimeDelta::Seconds(1));
+  ASSERT_TRUE(capturer->GetResolution());
+  EXPECT_EQ(kWidth / 2, capturer->GetResolution()->width);
+  EXPECT_EQ(kHeight / 2, capturer->GetResolution()->height);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/frame_generator_unittest.cc b/third_party/libwebrtc/test/frame_generator_unittest.cc
new file mode 100644
index 0000000000..ece37a547f
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_generator_unittest.cc
@@ -0,0 +1,284 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/frame_generator.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include <cstdint>
+#include <memory>
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "api/test/create_frame_generator.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/video/video_frame_buffer.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+constexpr int kFrameWidth = 4;
+constexpr int kFrameHeight = 4;
+constexpr int y_size = kFrameWidth * kFrameHeight;
+constexpr int uv_size = ((kFrameHeight + 1) / 2) * ((kFrameWidth + 1) / 2);
+
+class FrameGeneratorTest : public ::testing::Test {
+ public:
+  void SetUp() override {
+    two_frame_yuv_filename_ =
+        test::TempFilename(test::OutputPath(), "2_frame_yuv_file");
+    one_frame_yuv_filename_ =
+        test::TempFilename(test::OutputPath(), "1_frame_yuv_file");
+    two_frame_nv12_filename_ =
+        test::TempFilename(test::OutputPath(), "2_frame_nv12_file");
+    one_frame_nv12_filename_ =
+        test::TempFilename(test::OutputPath(), "1_frame_nv12_file");
+
+    FILE* file = fopen(two_frame_yuv_filename_.c_str(), "wb");
+    WriteYuvFile(file, 0, 0, 0);
+    WriteYuvFile(file, 127, 128, 129);
+    fclose(file);
+    file = fopen(one_frame_yuv_filename_.c_str(), "wb");
+    WriteYuvFile(file, 255, 255, 255);
+    fclose(file);
+    file = fopen(two_frame_nv12_filename_.c_str(), "wb");
+    WriteNV12File(file, 0, 0, 0);
+    WriteNV12File(file, 127, 128, 129);
+    fclose(file);
+    file = fopen(one_frame_nv12_filename_.c_str(), "wb");
+    WriteNV12File(file, 255, 255, 255);
+    fclose(file);
+  }
+
+  void TearDown() override {
+    remove(one_frame_yuv_filename_.c_str());
+    remove(two_frame_yuv_filename_.c_str());
+    remove(one_frame_nv12_filename_.c_str());
+    remove(two_frame_nv12_filename_.c_str());
+  }
+
+ protected:
+  void WriteYuvFile(FILE* file, uint8_t y, uint8_t u, uint8_t v) {
+    RTC_DCHECK(file);
+    std::unique_ptr<uint8_t[]> plane_buffer(new uint8_t[y_size]);
+    memset(plane_buffer.get(), y, y_size);
+    fwrite(plane_buffer.get(), 1, y_size, file);
+    memset(plane_buffer.get(), u, uv_size);
+    fwrite(plane_buffer.get(), 1, uv_size, file);
+    memset(plane_buffer.get(), v, uv_size);
+    fwrite(plane_buffer.get(), 1, uv_size, file);
+  }
+
+  void WriteNV12File(FILE* file, uint8_t y, uint8_t u, uint8_t v) {
+    RTC_DCHECK(file);
+    uint8_t plane_buffer[y_size];
+
+    memset(&plane_buffer, y, y_size);
+    fwrite(&plane_buffer, 1, y_size, file);
+    for (size_t i = 0; i < uv_size; ++i) {
+      plane_buffer[2 * i] = u;
+      plane_buffer[2 * i + 1] = v;
+    }
+    fwrite(&plane_buffer, 1, 2 * uv_size, file);
+  }
+
+  void CheckFrameAndMutate(const FrameGeneratorInterface::VideoFrameData& frame,
+                           uint8_t y,
+                           uint8_t u,
+                           uint8_t v) {
+    // Check that frame is valid, has the correct color and timestamp are clean.
+    rtc::scoped_refptr<I420BufferInterface> i420_buffer =
+        frame.buffer->ToI420();
+    const uint8_t* buffer;
+    buffer = i420_buffer->DataY();
+    for (int i = 0; i < y_size; ++i)
+      ASSERT_EQ(y, buffer[i]);
+    buffer = i420_buffer->DataU();
+    for (int i = 0; i < uv_size; ++i)
+      ASSERT_EQ(u, buffer[i]);
+    buffer = i420_buffer->DataV();
+    for (int i = 0; i < uv_size; ++i)
+      ASSERT_EQ(v, buffer[i]);
+  }
+
+  uint64_t Hash(const FrameGeneratorInterface::VideoFrameData& frame) {
+    // Generate a 64-bit hash from the frame's buffer.
+    uint64_t hash = 19;
+    rtc::scoped_refptr<I420BufferInterface> i420_buffer =
+        frame.buffer->ToI420();
+    const uint8_t* buffer = i420_buffer->DataY();
+    for (int i = 0; i < y_size; ++i) {
+      hash = (37 * hash) + buffer[i];
+    }
+    buffer = i420_buffer->DataU();
+    for (int i = 0; i < uv_size; ++i) {
+      hash = (37 * hash) + buffer[i];
+    }
+    buffer = i420_buffer->DataV();
+    for (int i = 0; i < uv_size; ++i) {
+      hash = (37 * hash) + buffer[i];
+    }
+    return hash;
+  }
+
+  std::string two_frame_yuv_filename_;
+  std::string one_frame_yuv_filename_;
+  std::string two_frame_nv12_filename_;
+  std::string one_frame_nv12_filename_;
+};
+
+TEST_F(FrameGeneratorTest, SingleFrameYuvFile) {
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromYuvFileFrameGenerator(
+          std::vector<std::string>(1, one_frame_yuv_filename_), kFrameWidth,
+          kFrameHeight, 1));
+  CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+  CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+}
+
+TEST_F(FrameGeneratorTest, TwoFrameYuvFile) {
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromYuvFileFrameGenerator(
+          std::vector<std::string>(1, two_frame_yuv_filename_), kFrameWidth,
+          kFrameHeight, 1));
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, MultipleFrameYuvFiles) {
+  std::vector<std::string> files;
+  files.push_back(two_frame_yuv_filename_);
+  files.push_back(one_frame_yuv_filename_);
+
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromYuvFileFrameGenerator(files, kFrameWidth, kFrameHeight, 1));
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, TwoFrameYuvFileWithRepeat) {
+  const int kRepeatCount = 3;
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromYuvFileFrameGenerator(
+          std::vector<std::string>(1, two_frame_yuv_filename_), kFrameWidth,
+          kFrameHeight, kRepeatCount));
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, MultipleFrameYuvFilesWithRepeat) {
+  const int kRepeatCount = 3;
+  std::vector<std::string> files;
+  files.push_back(two_frame_yuv_filename_);
+  files.push_back(one_frame_yuv_filename_);
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromYuvFileFrameGenerator(files, kFrameWidth, kFrameHeight,
+                                      kRepeatCount));
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, SingleFrameNV12File) {
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromNV12FileFrameGenerator(
+          std::vector<std::string>(1, one_frame_nv12_filename_), kFrameWidth,
+          kFrameHeight, 1));
+  CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+  CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+}
+
+TEST_F(FrameGeneratorTest, TwoFrameNV12File) {
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromNV12FileFrameGenerator(
+          std::vector<std::string>(1, two_frame_nv12_filename_), kFrameWidth,
+          kFrameHeight, 1));
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, MultipleFrameNV12Files) {
+  std::vector<std::string> files;
+  files.push_back(two_frame_nv12_filename_);
+  files.push_back(one_frame_nv12_filename_);
+
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromNV12FileFrameGenerator(files, kFrameWidth, kFrameHeight, 1));
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, TwoFrameNV12FileWithRepeat) {
+  const int kRepeatCount = 3;
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromNV12FileFrameGenerator(
+          std::vector<std::string>(1, two_frame_nv12_filename_), kFrameWidth,
+          kFrameHeight, kRepeatCount));
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, MultipleFrameNV12FilesWithRepeat) {
+  const int kRepeatCount = 3;
+  std::vector<std::string> files;
+  files.push_back(two_frame_nv12_filename_);
+  files.push_back(one_frame_nv12_filename_);
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateFromNV12FileFrameGenerator(files, kFrameWidth, kFrameHeight,
+                                       kRepeatCount));
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 127, 128, 129);
+  for (int i = 0; i < kRepeatCount; ++i)
+    CheckFrameAndMutate(generator->NextFrame(), 255, 255, 255);
+  CheckFrameAndMutate(generator->NextFrame(), 0, 0, 0);
+}
+
+TEST_F(FrameGeneratorTest, SlideGenerator) {
+  const int kGenCount = 9;
+  const int kRepeatCount = 3;
+  std::unique_ptr<FrameGeneratorInterface> generator(
+      CreateSlideFrameGenerator(kFrameWidth, kFrameHeight, kRepeatCount));
+  uint64_t hashes[kGenCount];
+  for (int i = 0; i < kGenCount; ++i) {
+    hashes[i] = Hash(generator->NextFrame());
+  }
+  // Check that the buffer changes only every `kRepeatCount` frames.
+  for (int i = 1; i < kGenCount; ++i) {
+    if (i % kRepeatCount == 0) {
+      EXPECT_NE(hashes[i - 1], hashes[i]);
+    } else {
+      EXPECT_EQ(hashes[i - 1], hashes[i]);
+    }
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/frame_utils.cc b/third_party/libwebrtc/test/frame_utils.cc
new file mode 100644
index 0000000000..b280de1ad1
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_utils.cc
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/frame_utils.h"
+
+#include <stdio.h>
+#include <string.h>
+
+#include "api/video/i420_buffer.h"
+#include "api/video/nv12_buffer.h"
+#include "api/video/video_frame.h"
+
+namespace webrtc {
+namespace test {
+
+bool EqualPlane(const uint8_t* data1,
+                const uint8_t* data2,
+                int stride1,
+                int stride2,
+                int width,
+                int height) {
+  for (int y = 0; y < height; ++y) {
+    if (memcmp(data1, data2, width) != 0)
+      return false;
+    data1 += stride1;
+    data2 += stride2;
+  }
+  return true;
+}
+
+bool FramesEqual(const webrtc::VideoFrame& f1, const webrtc::VideoFrame& f2) {
+  if (f1.timestamp() != f2.timestamp() ||
+      f1.ntp_time_ms() != f2.ntp_time_ms() ||
+      f1.render_time_ms() != f2.render_time_ms()) {
+    return false;
+  }
+  return FrameBufsEqual(f1.video_frame_buffer(), f2.video_frame_buffer());
+}
+
+bool FrameBufsEqual(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f1,
+                    const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f2) {
+  if (f1 == f2) {
+    return true;
+  }
+  // Exlude nullptr (except if both are nullptr, as above)
+  if (!f1 || !f2) {
+    return false;
+  }
+
+  if (f1->width() != f2->width() || f1->height() != f2->height() ||
+      f1->type() != f2->type()) {
+    return false;
+  }
+
+  rtc::scoped_refptr<webrtc::I420BufferInterface> f1_i420 = f1->ToI420();
+  rtc::scoped_refptr<webrtc::I420BufferInterface> f2_i420 = f2->ToI420();
+  return EqualPlane(f1_i420->DataY(), f2_i420->DataY(), f1_i420->StrideY(),
+                    f2_i420->StrideY(), f1_i420->width(), f1_i420->height()) &&
+         EqualPlane(f1_i420->DataU(), f2_i420->DataU(), f1_i420->StrideU(),
+                    f2_i420->StrideU(), f1_i420->ChromaWidth(),
+                    f1_i420->ChromaHeight()) &&
+         EqualPlane(f1_i420->DataV(), f2_i420->DataV(), f1_i420->StrideV(),
+                    f2_i420->StrideV(), f1_i420->ChromaWidth(),
+                    f1_i420->ChromaHeight());
+}
+
+rtc::scoped_refptr<I420Buffer> ReadI420Buffer(int width, int height, FILE* f) {
+  int half_width = (width + 1) / 2;
+  rtc::scoped_refptr<I420Buffer> buffer(
+      // Explicit stride, no padding between rows.
+      I420Buffer::Create(width, height, width, half_width, half_width));
+  size_t size_y = static_cast<size_t>(width) * height;
+  size_t size_uv = static_cast<size_t>(half_width) * ((height + 1) / 2);
+
+  if (fread(buffer->MutableDataY(), 1, size_y, f) < size_y)
+    return nullptr;
+  if (fread(buffer->MutableDataU(), 1, size_uv, f) < size_uv)
+    return nullptr;
+  if (fread(buffer->MutableDataV(), 1, size_uv, f) < size_uv)
+    return nullptr;
+  return buffer;
+}
+
+rtc::scoped_refptr<NV12Buffer> ReadNV12Buffer(int width, int height, FILE* f) {
+  rtc::scoped_refptr<NV12Buffer> buffer(NV12Buffer::Create(width, height));
+  size_t size_y = static_cast<size_t>(width) * height;
+  size_t size_uv = static_cast<size_t>(width + width % 2) * ((height + 1) / 2);
+
+  if (fread(buffer->MutableDataY(), 1, size_y, f) < size_y)
+    return nullptr;
+  if (fread(buffer->MutableDataUV(), 1, size_uv, f) < size_uv)
+    return nullptr;
+  return buffer;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/frame_utils.h b/third_party/libwebrtc/test/frame_utils.h
new file mode 100644
index 0000000000..1f2b381afb
--- /dev/null
+++ b/third_party/libwebrtc/test/frame_utils.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_FRAME_UTILS_H_
+#define TEST_FRAME_UTILS_H_
+
+#include <stdint.h>
+
+#include "api/scoped_refptr.h"
+#include "api/video/nv12_buffer.h"
+
+namespace webrtc {
+class I420Buffer;
+class VideoFrame;
+class VideoFrameBuffer;
+namespace test {
+
+bool EqualPlane(const uint8_t* data1,
+                const uint8_t* data2,
+                int stride1,
+                int stride2,
+                int width,
+                int height);
+
+static inline bool EqualPlane(const uint8_t* data1,
+                              const uint8_t* data2,
+                              int stride,
+                              int width,
+                              int height) {
+  return EqualPlane(data1, data2, stride, stride, width, height);
+}
+
+bool FramesEqual(const webrtc::VideoFrame& f1, const webrtc::VideoFrame& f2);
+
+bool FrameBufsEqual(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f1,
+                    const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f2);
+
+rtc::scoped_refptr<I420Buffer> ReadI420Buffer(int width, int height, FILE*);
+
+rtc::scoped_refptr<NV12Buffer> ReadNV12Buffer(int width, int height, FILE*);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FRAME_UTILS_H_
diff --git a/third_party/libwebrtc/test/function_audio_decoder_factory.h b/third_party/libwebrtc/test/function_audio_decoder_factory.h
new file mode 100644
index 0000000000..8464f3d9aa
--- /dev/null
+++ b/third_party/libwebrtc/test/function_audio_decoder_factory.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FUNCTION_AUDIO_DECODER_FACTORY_H_
+#define TEST_FUNCTION_AUDIO_DECODER_FACTORY_H_
+
+#include <functional>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+// A decoder factory producing decoders by calling a supplied create function.
+class FunctionAudioDecoderFactory : public AudioDecoderFactory {
+ public:
+  explicit FunctionAudioDecoderFactory(
+      std::function<std::unique_ptr<AudioDecoder>()> create)
+      : create_([create](const SdpAudioFormat&,
+                         absl::optional<AudioCodecPairId> codec_pair_id) {
+          return create();
+        }) {}
+  explicit FunctionAudioDecoderFactory(
+      std::function<std::unique_ptr<AudioDecoder>(
+          const SdpAudioFormat&,
+          absl::optional<AudioCodecPairId> codec_pair_id)> create)
+      : create_(std::move(create)) {}
+
+  // Unused by tests.
+  std::vector<AudioCodecSpec> GetSupportedDecoders() override {
+    RTC_DCHECK_NOTREACHED();
+    return {};
+  }
+
+  bool IsSupportedDecoder(const SdpAudioFormat& format) override {
+    return true;
+  }
+
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) override {
+    return create_(format, codec_pair_id);
+  }
+
+ private:
+  const std::function<std::unique_ptr<AudioDecoder>(
+      const SdpAudioFormat&,
+      absl::optional<AudioCodecPairId> codec_pair_id)>
+      create_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FUNCTION_AUDIO_DECODER_FACTORY_H_
diff --git a/third_party/libwebrtc/test/fuzzers/BUILD.gn b/third_party/libwebrtc/test/fuzzers/BUILD.gn
new file mode 100644
index 0000000000..c3576713ab
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/BUILD.gn
@@ -0,0 +1,689 @@
+# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//build/config/features.gni")
+import("//testing/libfuzzer/fuzzer_test.gni")
+import("../../webrtc.gni")
+
+rtc_library("webrtc_fuzzer_main") {
+  sources = [ "webrtc_fuzzer_main.cc" ]
+  deps = [
+    "../../rtc_base:logging",
+    "//testing/libfuzzer:libfuzzer_main",
+  ]
+
+  # When WebRTC fuzzer tests are built on Chromium bots they need to link
+  # with Chromium's implementation of metrics, field trial, and system time.
+  if (build_with_chromium) {
+    deps += [
+      "../../../webrtc_overrides:field_trial",
+      "../../../webrtc_overrides:metrics",
+      "../../../webrtc_overrides:system_time",
+    ]
+  }
+}
+
+rtc_library("fuzz_data_helper") {
+  testonly = true
+  sources = [
+    "fuzz_data_helper.cc",
+    "fuzz_data_helper.h",
+  ]
+  deps = [
+    "../../api:array_view",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+  visibility = [ ":*" ]  # Only targets in this file can depend on this.
+}
+
+set_defaults("webrtc_fuzzer_test") {
+  configs = rtc_add_configs
+  absl_deps = []
+}
+
+template("webrtc_fuzzer_test") {
+  fuzzer_test(target_name) {
+    forward_variables_from(invoker, "*")
+    deps += [
+      ":fuzz_data_helper",
+      ":webrtc_fuzzer_main",
+    ]
+    additional_configs = configs
+
+    # If absl_deps is [], no action is needed. If not [], then it needs to be
+    # converted to //third_party/abseil-cpp:absl when build_with_chromium=true
+    # otherwise it just needs to be added to deps.
+    if (absl_deps != []) {
+      if (!defined(deps)) {
+        deps = []
+      }
+      if (build_with_chromium) {
+        deps += [ "//third_party/abseil-cpp:absl" ]
+      } else {
+        deps += absl_deps
+      }
+    }
+
+    if (!build_with_chromium && is_clang) {
+      suppressed_configs = [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+}
+
+webrtc_fuzzer_test("h264_depacketizer_fuzzer") {
+  sources = [ "h264_depacketizer_fuzzer.cc" ]
+  deps = [ "../../modules/rtp_rtcp" ]
+  seed_corpus = "corpora/h264-depacketizer-fuzzer-corpus"
+}
+
+webrtc_fuzzer_test("vp8_depacketizer_fuzzer") {
+  sources = [ "vp8_depacketizer_fuzzer.cc" ]
+  deps = [
+    "../../api:array_view",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_video_header",
+  ]
+}
+
+webrtc_fuzzer_test("vp9_depacketizer_fuzzer") {
+  sources = [ "vp9_depacketizer_fuzzer.cc" ]
+  deps = [
+    "../../api:array_view",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_video_header",
+  ]
+}
+
+webrtc_fuzzer_test("vp8_qp_parser_fuzzer") {
+  sources = [ "vp8_qp_parser_fuzzer.cc" ]
+  deps = [
+    "../../modules/video_coding:video_coding_utility",
+    "../../modules/video_coding/",
+  ]
+}
+
+webrtc_fuzzer_test("vp9_qp_parser_fuzzer") {
+  sources = [ "vp9_qp_parser_fuzzer.cc" ]
+  deps = [
+    "../../modules/video_coding:video_coding_utility",
+    "../../modules/video_coding/",
+  ]
+}
+
+webrtc_fuzzer_test("h264_bitstream_parser_fuzzer") {
+  sources = [ "h264_bitstream_parser_fuzzer.cc" ]
+  deps = [
+    "../../common_video",
+    "../../modules/video_coding/",
+  ]
+}
+
+webrtc_fuzzer_test("forward_error_correction_fuzzer") {
+  sources = [ "forward_error_correction_fuzzer.cc" ]
+  deps = [
+    "../../api:scoped_refptr",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:byte_buffer",
+  ]
+}
+
+webrtc_fuzzer_test("flexfec_header_reader_fuzzer") {
+  sources = [ "flexfec_header_reader_fuzzer.cc" ]
+  deps = [
+    "../../api:scoped_refptr",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+webrtc_fuzzer_test("flexfec_sender_fuzzer") {
+  sources = [ "flexfec_sender_fuzzer.cc" ]
+  deps = [
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../system_wrappers",
+  ]
+}
+
+webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") {
+  sources = [ "ulpfec_header_reader_fuzzer.cc" ]
+  deps = [
+    "../../api:scoped_refptr",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:fec_test_helper",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+webrtc_fuzzer_test("ulpfec_generator_fuzzer") {
+  sources = [ "ulpfec_generator_fuzzer.cc" ]
+  deps = [
+    "../../modules:module_api_public",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:fec_test_helper",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../system_wrappers",
+  ]
+}
+
+webrtc_fuzzer_test("ulpfec_receiver_fuzzer") {
+  sources = [ "ulpfec_receiver_fuzzer.cc" ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+webrtc_fuzzer_test("flexfec_receiver_fuzzer") {
+  sources = [ "flexfec_receiver_fuzzer.cc" ]
+  deps = [
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+webrtc_fuzzer_test("rtp_video_frame_assembler_fuzzer") {
+  sources = [ "rtp_video_frame_assembler_fuzzer.cc" ]
+  deps = [
+    "../../api/video:rtp_video_frame_assembler",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+webrtc_fuzzer_test("rtcp_receiver_fuzzer") {
+  sources = [ "rtcp_receiver_fuzzer.cc" ]
+  deps = [
+    "../../modules/rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+    "../../system_wrappers",
+  ]
+  seed_corpus = "corpora/rtcp-corpus"
+}
+
+webrtc_fuzzer_test("rtp_packet_fuzzer") {
+  sources = [ "rtp_packet_fuzzer.cc" ]
+  deps = [ "../../modules/rtp_rtcp:rtp_rtcp_format" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  seed_corpus = "corpora/rtp-corpus"
+}
+
+webrtc_fuzzer_test("rtp_packetizer_av1_fuzzer") {
+  sources = [ "rtp_packetizer_av1_fuzzer.cc" ]
+  deps = [
+    "../../api/video:video_frame_type",
+    "../../modules/rtp_rtcp:rtp_rtcp",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+  ]
+}
+
+webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
+  sources = [ "congestion_controller_feedback_fuzzer.cc" ]
+  deps = [
+    "../../modules/congestion_controller",
+    "../../modules/pacing",
+    "../../modules/remote_bitrate_estimator",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/functional:bind_front" ]
+}
+
+rtc_library("audio_decoder_fuzzer") {
+  testonly = true
+  sources = [
+    "audio_decoder_fuzzer.cc",
+    "audio_decoder_fuzzer.h",
+  ]
+  deps = [
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+webrtc_fuzzer_test("audio_decoder_g722_fuzzer") {
+  sources = [ "audio_decoder_g722_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:g722",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") {
+  sources = [ "audio_decoder_ilbc_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:ilbc",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_opus_fuzzer") {
+  sources = [ "audio_decoder_opus_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:webrtc_opus",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_opus_redundant_fuzzer") {
+  sources = [ "audio_decoder_opus_redundant_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:webrtc_opus",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_multiopus_fuzzer") {
+  sources = [ "audio_decoder_multistream_opus_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../api/audio_codecs/opus:audio_decoder_multiopus",
+    "../../api/audio_codecs/opus:audio_decoder_opus_config",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_pcm_fuzzer") {
+  sources = [ "audio_decoder_pcm_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:g711",
+  ]
+}
+
+webrtc_fuzzer_test("audio_decoder_pcm16b_fuzzer") {
+  sources = [ "audio_decoder_pcm16b_fuzzer.cc" ]
+  deps = [
+    ":audio_decoder_fuzzer",
+    "../../modules/audio_coding:pcm16b",
+  ]
+}
+
+rtc_library("audio_encoder_fuzzer") {
+  testonly = true
+  sources = [
+    "audio_encoder_fuzzer.cc",
+    "audio_encoder_fuzzer.h",
+  ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../api:array_view",
+    "../../api/audio_codecs:audio_codecs_api",
+    "../../rtc_base:buffer",
+    "../../rtc_base:checks",
+  ]
+}
+
+webrtc_fuzzer_test("audio_encoder_opus_fuzzer") {
+  sources = [ "audio_encoder_opus_fuzzer.cc" ]
+  deps = [
+    ":audio_encoder_fuzzer",
+    "../../api/audio_codecs/opus:audio_encoder_opus",
+    "../../rtc_base:checks",
+  ]
+}
+
+webrtc_fuzzer_test("turn_unwrap_fuzzer") {
+  sources = [ "turn_unwrap_fuzzer.cc" ]
+  deps = [
+    "../../media",
+    "../../media:rtc_media_base",
+    "../../media:turn_utils",
+  ]
+}
+
+webrtc_fuzzer_test("neteq_rtp_fuzzer") {
+  sources = [ "neteq_rtp_fuzzer.cc" ]
+  deps = [
+    "../../api:array_view",
+    "../../api/audio_codecs:builtin_audio_decoder_factory",
+    "../../modules/audio_coding:neteq",
+    "../../modules/audio_coding:neteq_test_tools",
+    "../../modules/audio_coding:neteq_tools_minimal",
+    "../../modules/audio_coding:pcm16b",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+webrtc_fuzzer_test("neteq_signal_fuzzer") {
+  sources = [ "neteq_signal_fuzzer.cc" ]
+  deps = [
+    "../../api:array_view",
+    "../../api/audio_codecs:builtin_audio_decoder_factory",
+    "../../modules/audio_coding:neteq",
+    "../../modules/audio_coding:neteq_test_tools",
+    "../../modules/audio_coding:neteq_tools_minimal",
+    "../../modules/audio_coding:pcm16b",
+    "../../rtc_base:random",
+    "../../rtc_base:safe_conversions",
+  ]
+}
+
+webrtc_fuzzer_test("residual_echo_detector_fuzzer") {
+  sources = [ "residual_echo_detector_fuzzer.cc" ]
+  deps = [
+    "../../api/audio:echo_detector_creator",
+    "../../rtc_base:checks",
+    "../../rtc_base:refcount",
+  ]
+}
+
+webrtc_fuzzer_test("sdp_parser_fuzzer") {
+  sources = [ "sdp_parser_fuzzer.cc" ]
+  deps = [
+    "../../api:libjingle_peerconnection_api",
+    "../../pc:libjingle_peerconnection",
+  ]
+  seed_corpus = "corpora/sdp-corpus"
+}
+
+if (!build_with_chromium) {
+  # This target depends on test infrastructure that can't be built
+  # with Chromium at the moment.
+  # TODO(bugs.chromium.org/12534): Make this fuzzer build in Chromium.
+
+  webrtc_fuzzer_test("sdp_integration_fuzzer") {
+    sources = [ "sdp_integration_fuzzer.cc" ]
+    deps = [
+      "../../api:libjingle_peerconnection_api",
+      "../../pc:integration_test_helpers",
+      "../../pc:libjingle_peerconnection",
+      "../../test:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    seed_corpus = "corpora/sdp-corpus"
+  }
+}
+
+webrtc_fuzzer_test("stun_parser_fuzzer") {
+  sources = [ "stun_parser_fuzzer.cc" ]
+  deps = [
+    "../../api/transport:stun_types",
+    "../../p2p:rtc_p2p",
+  ]
+  seed_corpus = "corpora/stun-corpus"
+  dict = "corpora/stun.tokens"
+}
+
+webrtc_fuzzer_test("stun_validator_fuzzer") {
+  sources = [ "stun_validator_fuzzer.cc" ]
+  deps = [
+    "../../api/transport:stun_types",
+    "../../p2p:rtc_p2p",
+  ]
+  seed_corpus = "corpora/stun-corpus"
+  dict = "corpora/stun.tokens"
+}
+
+webrtc_fuzzer_test("pseudotcp_parser_fuzzer") {
+  sources = [ "pseudotcp_parser_fuzzer.cc" ]
+  deps = [
+    "../../p2p:rtc_p2p",
+    "../../rtc_base:threading",
+  ]
+}
+
+rtc_library("audio_processing_fuzzer_helper") {
+  testonly = true
+  sources = [
+    "audio_processing_fuzzer_helper.cc",
+    "audio_processing_fuzzer_helper.h",
+  ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../api/audio:audio_frame_api",
+    "../../modules/audio_processing",
+    "../../modules/audio_processing:api",
+    "../../modules/audio_processing:audio_frame_proxies",
+    "../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+webrtc_fuzzer_test("audio_processing_fuzzer") {
+  sources = [ "audio_processing_configs_fuzzer.cc" ]
+  deps = [
+    ":audio_processing_fuzzer_helper",
+    "../../api:scoped_refptr",
+    "../../api/audio:aec3_factory",
+    "../../api/audio:echo_detector_creator",
+    "../../api/task_queue:default_task_queue_factory",
+    "../../modules/audio_processing",
+    "../../modules/audio_processing:api",
+    "../../modules/audio_processing:audio_buffer",
+    "../../modules/audio_processing:audioproc_test_utils",
+    "../../modules/audio_processing/aec3",
+    "../../modules/audio_processing/aec_dump",
+    "../../modules/audio_processing/aec_dump:aec_dump_impl",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:rtc_task_queue",
+    "../../rtc_base:safe_minmax",
+    "../../system_wrappers:field_trial",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+  seed_corpus = "corpora/audio_processing-corpus"
+}
+
+webrtc_fuzzer_test("audio_processing_sample_rate_fuzzer") {
+  sources = [ "audio_processing_sample_rate_fuzzer.cc" ]
+  deps = [
+    "../../api:scoped_refptr",
+    "../../api/audio:audio_frame_api",
+    "../../modules/audio_processing",
+    "../../modules/audio_processing:api",
+    "../../modules/audio_processing:audio_frame_proxies",
+    "../../modules/audio_processing:audioproc_test_utils",
+    "../../rtc_base:checks",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:safe_minmax",
+  ]
+}
+
+webrtc_fuzzer_test("agc_fuzzer") {
+  sources = [ "agc_fuzzer.cc" ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../modules/audio_processing",
+    "../../modules/audio_processing:api",
+    "../../modules/audio_processing:audio_buffer",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:safe_minmax",
+  ]
+
+  seed_corpus = "corpora/agc-corpus"
+}
+
+webrtc_fuzzer_test("aec3_config_json_fuzzer") {
+  sources = [ "aec3_config_json_fuzzer.cc" ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../api/audio:aec3_config",
+    "../../api/audio:aec3_config_json",
+  ]
+  dict = "//testing/libfuzzer/fuzzers/dicts/json.dict"
+  seed_corpus = "corpora/aec3-config-json-corpus"
+}
+
+webrtc_fuzzer_test("aec3_fuzzer") {
+  defines = []
+  if (apm_debug_dump) {
+    defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
+  } else {
+    defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
+  }
+  sources = [ "aec3_fuzzer.cc" ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../modules/audio_processing:api",
+    "../../modules/audio_processing:audio_buffer",
+    "../../modules/audio_processing/aec3",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+webrtc_fuzzer_test("comfort_noise_decoder_fuzzer") {
+  sources = [ "comfort_noise_decoder_fuzzer.cc" ]
+  deps = [
+    "../../api:array_view",
+    "../../modules/audio_coding:webrtc_cng",
+    "../../rtc_base:buffer",
+  ]
+}
+
+webrtc_fuzzer_test("rtp_depacketizer_av1_assemble_frame_fuzzer") {
+  sources = [ "rtp_depacketizer_av1_assemble_frame_fuzzer.cc" ]
+  seed_corpus = "corpora/rtp-depacketizer-av1-assemble-frame-corpus"
+  deps = [
+    ":fuzz_data_helper",
+    "../../api:array_view",
+    "../../modules/rtp_rtcp",
+  ]
+}
+
+webrtc_fuzzer_test("rtp_dependency_descriptor_fuzzer") {
+  sources = [ "rtp_dependency_descriptor_fuzzer.cc" ]
+  seed_corpus = "corpora/dependency_descriptor-corpus"
+  deps = [
+    "../../api:array_view",
+    "../../common_video/generic_frame_descriptor",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+  ]
+}
+
+webrtc_fuzzer_test("rtp_video_layers_allocation_fuzzer") {
+  sources = [ "rtp_video_layers_allocation_fuzzer.cc" ]
+  seed_corpus = "corpora/video_layers_allocation-corpus"
+  deps = [
+    "../../api:array_view",
+    "../../api/video:video_layers_allocation",
+    "../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../rtc_base:checks",
+  ]
+
+  # video_layers_allocation is an rtp header extension and thus can't be longer
+  # than 255 bytes on the wire.
+  libfuzzer_options = [ "max_len=255" ]
+}
+
+webrtc_fuzzer_test("rtp_frame_reference_finder_fuzzer") {
+  sources = [ "rtp_frame_reference_finder_fuzzer.cc" ]
+  deps = [
+    "../../api:rtp_packet_info",
+    "../../api:scoped_refptr",
+    "../../modules/video_coding/",
+    "../../system_wrappers",
+  ]
+}
+
+webrtc_fuzzer_test("frame_buffer2_fuzzer") {
+  sources = [ "frame_buffer2_fuzzer.cc" ]
+  deps = [
+    "../../api/task_queue",
+    "../../modules/video_coding:frame_buffer2",
+    "../../modules/video_coding/timing:timing_module",
+    "../../test:scoped_key_value_config",
+    "../time_controller:time_controller",
+  ]
+}
+
+webrtc_fuzzer_test("frame_buffer_fuzzer") {
+  sources = [ "frame_buffer_fuzzer.cc" ]
+  deps = [
+    ":fuzz_data_helper",
+    "../../api:array_view",
+    "../../api/video:encoded_frame",
+    "../../api/video:frame_buffer",
+    "../../rtc_base:rtc_numerics",
+    "../../test:scoped_key_value_config",
+  ]
+}
+
+webrtc_fuzzer_test("field_trial_fuzzer") {
+  sources = [ "field_trial_fuzzer.cc" ]
+  deps = [ "../../system_wrappers:field_trial" ]
+  seed_corpus = "corpora/field_trial-corpus"
+}
+
+webrtc_fuzzer_test("string_to_number_fuzzer") {
+  sources = [ "string_to_number_fuzzer.cc" ]
+  deps = [ "../../rtc_base:stringutils" ]
+  seed_corpus = "corpora/string_to_number-corpus"
+}
+
+webrtc_fuzzer_test("sctp_utils_fuzzer") {
+  sources = [ "sctp_utils_fuzzer.cc" ]
+  deps = [
+    "../../api:libjingle_peerconnection_api",
+    "../../pc:libjingle_peerconnection",
+    "../../pc:sctp_utils",
+    "../../rtc_base:copy_on_write_buffer",
+  ]
+}
+
+webrtc_fuzzer_test("dcsctp_socket_fuzzer") {
+  sources = [ "dcsctp_socket_fuzzer.cc" ]
+  deps = [
+    "../../net/dcsctp/fuzzers:dcsctp_fuzzers",
+    "../../net/dcsctp/public:socket",
+    "../../net/dcsctp/public:types",
+    "../../net/dcsctp/socket:dcsctp_socket",
+    "../../rtc_base:logging",
+  ]
+}
+
+webrtc_fuzzer_test("ssl_certificate_fuzzer") {
+  sources = [ "ssl_certificate_fuzzer.cc" ]
+  deps = [
+    "../:rtp_test_utils",
+    "../../rtc_base:ssl",
+    "../../rtc_base:stringutils",
+  ]
+}
+
+webrtc_fuzzer_test("vp8_replay_fuzzer") {
+  sources = [ "vp8_replay_fuzzer.cc" ]
+  deps = [ "utils:rtp_replayer" ]
+  seed_corpus = "corpora/rtpdump-corpus/vp8"
+}
+
+if (rtc_build_libvpx) {
+  webrtc_fuzzer_test("vp9_encoder_references_fuzzer") {
+    sources = [ "vp9_encoder_references_fuzzer.cc" ]
+    deps = [
+      "..:test_support",
+      "../../api:array_view",
+      "../../api:field_trials_view",
+      "../../api/video:video_frame",
+      "../../api/video_codecs:video_codecs_api",
+      "../../modules/video_coding:frame_dependencies_calculator",
+      "../../modules/video_coding:webrtc_libvpx_interface",
+      "../../modules/video_coding:webrtc_vp9",
+      "../../rtc_base:safe_compare",
+      rtc_libvpx_dir,
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/algorithm:container",
+      "//third_party/abseil-cpp/absl/base:core_headers",
+      "//third_party/abseil-cpp/absl/container:inlined_vector",
+    ]
+    seed_corpus = "corpora/vp9-encoder-references-corpus"
+    defines = [ "RTC_ENABLE_VP9" ]
+  }
+}
+
+webrtc_fuzzer_test("vp9_replay_fuzzer") {
+  sources = [ "vp9_replay_fuzzer.cc" ]
+  deps = [ "utils:rtp_replayer" ]
+  seed_corpus = "corpora/rtpdump-corpus/vp9"
+}
diff --git a/third_party/libwebrtc/test/fuzzers/DEPS b/third_party/libwebrtc/test/fuzzers/DEPS
new file mode 100644
index 0000000000..50b1c8adce
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+  "+audio",
+  "+pc",
+  "+net/dcsctp",
+]
diff --git a/third_party/libwebrtc/test/fuzzers/OWNERS b/third_party/libwebrtc/test/fuzzers/OWNERS
new file mode 100644
index 0000000000..3d97816b3b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/OWNERS
@@ -0,0 +1,3 @@
+danilchap@webrtc.org
+henrik.lundin@webrtc.org
+saza@webrtc.org
diff --git a/third_party/libwebrtc/test/fuzzers/aec3_config_json_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/aec3_config_json_fuzzer.cc
new file mode 100644
index 0000000000..626350c52c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/aec3_config_json_fuzzer.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string>
+
+#include "api/audio/echo_canceller3_config.h"
+#include "api/audio/echo_canceller3_config_json.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 10000) {
+    return;
+  }
+  std::string config_json(reinterpret_cast<const char*>(data), size);
+
+  EchoCanceller3Config config;
+  bool success;
+  Aec3ConfigFromJsonString(config_json, &config, &success);
+  EchoCanceller3Config::Validate(&config);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/aec3_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/aec3_fuzzer.cc
new file mode 100644
index 0000000000..a12ca30f63
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/aec3_fuzzer.cc
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "absl/types/optional.h"
+#include "modules/audio_processing/aec3/echo_canceller3.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace {
+using SampleRate = ::webrtc::AudioProcessing::NativeRate;
+
+void PrepareAudioBuffer(int sample_rate_hz,
+                        test::FuzzDataHelper* fuzz_data,
+                        AudioBuffer* buffer) {
+  float* const* channels = buffer->channels_f();
+  for (size_t i = 0; i < buffer->num_channels(); ++i) {
+    for (size_t j = 0; j < buffer->num_frames(); ++j) {
+      channels[i][j] =
+          static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0));
+    }
+  }
+  if (sample_rate_hz == 32000 || sample_rate_hz == 48000) {
+    buffer->SplitIntoFrequencyBands();
+  }
+}
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 200000) {
+    return;
+  }
+
+  test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
+
+  constexpr int kSampleRates[] = {16000, 32000, 48000};
+  const int sample_rate_hz =
+      static_cast<size_t>(fuzz_data.SelectOneOf(kSampleRates));
+
+  constexpr int kMaxNumChannels = 9;
+  const size_t num_render_channels =
+      1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
+  const size_t num_capture_channels =
+      1 + fuzz_data.ReadOrDefaultValue<uint8_t>(0) % (kMaxNumChannels - 1);
+
+  EchoCanceller3 aec3(EchoCanceller3Config(),
+                      /*multichannel_config=*/absl::nullopt, sample_rate_hz,
+                      num_render_channels, num_capture_channels);
+
+  AudioBuffer capture_audio(sample_rate_hz, num_capture_channels,
+                            sample_rate_hz, num_capture_channels,
+                            sample_rate_hz, num_capture_channels);
+  AudioBuffer render_audio(sample_rate_hz, num_render_channels, sample_rate_hz,
+                           num_render_channels, sample_rate_hz,
+                           num_render_channels);
+
+  // Fuzz frames while there is still fuzzer data.
+  while (fuzz_data.BytesLeft() > 0) {
+    bool is_capture = fuzz_data.ReadOrDefaultValue(true);
+    bool level_changed = fuzz_data.ReadOrDefaultValue(true);
+    if (is_capture) {
+      PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &capture_audio);
+      aec3.ProcessCapture(&capture_audio, level_changed);
+    } else {
+      PrepareAudioBuffer(sample_rate_hz, &fuzz_data, &render_audio);
+      aec3.AnalyzeRender(&render_audio);
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/agc_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/agc_fuzzer.cc
new file mode 100644
index 0000000000..597103993e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/agc_fuzzer.cc
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_processing/gain_control_impl.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/thread_annotations.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace {
+
+void FillAudioBuffer(size_t sample_rate_hz,
+                     test::FuzzDataHelper* fuzz_data,
+                     AudioBuffer* buffer) {
+  float* const* channels = buffer->channels_f();
+  for (size_t i = 0; i < buffer->num_channels(); ++i) {
+    for (size_t j = 0; j < buffer->num_frames(); ++j) {
+      channels[i][j] =
+          static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0));
+    }
+  }
+
+  if (sample_rate_hz != 16000) {
+    buffer->SplitIntoFrequencyBands();
+  }
+}
+
+// This function calls the GainControl functions that are overriden as private
+// in GainControlInterface.
+void FuzzGainControllerConfig(test::FuzzDataHelper* fuzz_data,
+                              GainControl* gc) {
+  GainControl::Mode modes[] = {GainControl::Mode::kAdaptiveAnalog,
+                               GainControl::Mode::kAdaptiveDigital,
+                               GainControl::Mode::kFixedDigital};
+  GainControl::Mode mode = fuzz_data->SelectOneOf(modes);
+  const bool enable_limiter = fuzz_data->ReadOrDefaultValue(true);
+  // The values are capped to comply with the API of webrtc::GainControl.
+  const int analog_level_min =
+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(0), 0, 65534);
+  const int analog_level_max =
+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(65535),
+                          analog_level_min + 1, 65535);
+  const int stream_analog_level =
+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(30000),
+                          analog_level_min, analog_level_max);
+  const int gain =
+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(30), -1, 100);
+  const int target_level_dbfs =
+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(15), -1, 35);
+
+  gc->set_mode(mode);
+  gc->enable_limiter(enable_limiter);
+  if (mode == GainControl::Mode::kAdaptiveAnalog) {
+    gc->set_analog_level_limits(analog_level_min, analog_level_max);
+    gc->set_stream_analog_level(stream_analog_level);
+  }
+  gc->set_compression_gain_db(gain);
+  gc->set_target_level_dbfs(target_level_dbfs);
+
+  static_cast<void>(gc->mode());
+  static_cast<void>(gc->analog_level_minimum());
+  static_cast<void>(gc->analog_level_maximum());
+  static_cast<void>(gc->stream_analog_level());
+  static_cast<void>(gc->compression_gain_db());
+  static_cast<void>(gc->stream_is_saturated());
+  static_cast<void>(gc->target_level_dbfs());
+  static_cast<void>(gc->is_limiter_enabled());
+}
+
+void FuzzGainController(test::FuzzDataHelper* fuzz_data, GainControlImpl* gci) {
+  using Rate = ::webrtc::AudioProcessing::NativeRate;
+  const Rate rate_kinds[] = {Rate::kSampleRate16kHz, Rate::kSampleRate32kHz,
+                             Rate::kSampleRate48kHz};
+
+  const auto sample_rate_hz =
+      static_cast<size_t>(fuzz_data->SelectOneOf(rate_kinds));
+  const size_t samples_per_frame = sample_rate_hz / 100;
+  const size_t num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
+
+  gci->Initialize(num_channels, sample_rate_hz);
+  FuzzGainControllerConfig(fuzz_data, gci);
+
+  // The audio buffer is used for both capture and render.
+  AudioBuffer audio(sample_rate_hz, num_channels, sample_rate_hz,
+                    num_channels, sample_rate_hz, num_channels);
+
+  std::vector<int16_t> packed_render_audio(samples_per_frame);
+
+  while (fuzz_data->CanReadBytes(1)) {
+    FillAudioBuffer(sample_rate_hz, fuzz_data, &audio);
+
+    const bool stream_has_echo = fuzz_data->ReadOrDefaultValue(true);
+    gci->AnalyzeCaptureAudio(audio);
+    gci->ProcessCaptureAudio(&audio, stream_has_echo);
+
+    FillAudioBuffer(sample_rate_hz, fuzz_data, &audio);
+
+    gci->PackRenderAudioBuffer(audio, &packed_render_audio);
+    gci->ProcessRenderAudio(packed_render_audio);
+  }
+}
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 200000) {
+    return;
+  }
+  test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
+  auto gci = std::make_unique<GainControlImpl>();
+  FuzzGainController(&fuzz_data, gci.get());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.cc
new file mode 100644
index 0000000000..1db332eeb5
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+#include <limits>
+
+#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+template <typename T, unsigned int B = sizeof(T)>
+bool ParseInt(const uint8_t** data, size_t* remaining_size, T* value) {
+  static_assert(std::numeric_limits<T>::is_integer, "Type must be an integer.");
+  static_assert(sizeof(T) <= sizeof(uint64_t),
+                "Cannot read wider than uint64_t.");
+  static_assert(B <= sizeof(T), "T must be at least B bytes wide.");
+  if (B > *remaining_size)
+    return false;
+  uint64_t val = ByteReader<uint64_t, B>::ReadBigEndian(*data);
+  *data += B;
+  *remaining_size -= B;
+  *value = static_cast<T>(val);
+  return true;
+}
+}  // namespace
+
+// This function reads two bytes from the beginning of `data`, interprets them
+// as the first packet length, and reads this many bytes if available. The
+// payload is inserted into the decoder, and the process continues until no more
+// data is available. Either AudioDecoder::Decode or
+// AudioDecoder::DecodeRedundant is used, depending on the value of
+// `decode_type`.
+void FuzzAudioDecoder(DecoderFunctionType decode_type,
+                      const uint8_t* data,
+                      size_t size,
+                      AudioDecoder* decoder,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded) {
+  const uint8_t* data_ptr = data;
+  size_t remaining_size = size;
+  size_t packet_len;
+  constexpr size_t kMaxNumFuzzedPackets = 200;
+  for (size_t num_packets = 0; num_packets < kMaxNumFuzzedPackets;
+       ++num_packets) {
+    if (!(ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) &&
+          packet_len <= remaining_size)) {
+      break;
+    }
+    AudioDecoder::SpeechType speech_type;
+    switch (decode_type) {
+      case DecoderFunctionType::kNormalDecode:
+        decoder->Decode(data_ptr, packet_len, sample_rate_hz, max_decoded_bytes,
+                        decoded, &speech_type);
+        break;
+      case DecoderFunctionType::kRedundantDecode:
+        decoder->DecodeRedundant(data_ptr, packet_len, sample_rate_hz,
+                                 max_decoded_bytes, decoded, &speech_type);
+        break;
+    }
+    data_ptr += packet_len;
+    remaining_size -= packet_len;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.h b/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.h
new file mode 100644
index 0000000000..8c5246bb5e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_fuzzer.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_
+#define TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+namespace webrtc {
+
+class AudioDecoder;
+
+enum class DecoderFunctionType {
+  kNormalDecode,
+  kRedundantDecode,
+};
+
+void FuzzAudioDecoder(DecoderFunctionType decode_type,
+                      const uint8_t* data,
+                      size_t size,
+                      AudioDecoder* decoder,
+                      int sample_rate_hz,
+                      size_t max_decoded_bytes,
+                      int16_t* decoded);
+
+}  // namespace webrtc
+
+#endif  // TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_g722_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_g722_fuzzer.cc
new file mode 100644
index 0000000000..08599aa333
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_g722_fuzzer.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 10000 || size < 1) {
+    return;
+  }
+
+  std::unique_ptr<AudioDecoder> dec;
+  size_t num_channels;
+  if (data[0] % 2) {
+    dec = std::make_unique<AudioDecoderG722Impl>();
+    num_channels = 1;
+  } else {
+    dec = std::make_unique<AudioDecoderG722StereoImpl>();
+    num_channels = 2;
+  }
+  // Allocate a maximum output size of 100 ms.
+  const int sample_rate_hz = dec->SampleRateHz();
+  const size_t allocated_ouput_size_samples =
+      sample_rate_hz / 10 * num_channels;
+  std::unique_ptr<int16_t[]> output =
+      std::make_unique<int16_t[]>(allocated_ouput_size_samples);
+  FuzzAudioDecoder(
+      DecoderFunctionType::kNormalDecode, data, size, dec.get(), sample_rate_hz,
+      allocated_ouput_size_samples * sizeof(int16_t), output.get());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc
new file mode 100644
index 0000000000..8548645c63
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_ilbc_fuzzer.cc
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 10000) {
+    return;
+  }
+  AudioDecoderIlbcImpl dec;
+  static const int kSampleRateHz = 8000;
+  static const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec,
+                   kSampleRateHz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_multistream_opus_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_multistream_opus_fuzzer.cc
new file mode 100644
index 0000000000..474a1cdc43
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_multistream_opus_fuzzer.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
+#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+
+AudioDecoderMultiChannelOpusConfig MakeDecoderConfig(
+    int num_channels,
+    int num_streams,
+    int coupled_streams,
+    std::vector<unsigned char> channel_mapping) {
+  AudioDecoderMultiChannelOpusConfig config;
+  config.num_channels = num_channels;
+  config.num_streams = num_streams;
+  config.coupled_streams = coupled_streams;
+  config.channel_mapping = channel_mapping;
+  return config;
+}
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  const std::vector<AudioDecoderMultiChannelOpusConfig> surround_configs = {
+      MakeDecoderConfig(1, 1, 0, {0}),  // Mono
+
+      MakeDecoderConfig(2, 2, 0, {0, 0}),  // Copy the first (of
+                                           // 2) decoded streams
+                                           // into both output
+                                           // channel 0 and output
+                                           // channel 1. Ignore
+                                           // the 2nd decoded
+                                           // stream.
+
+      MakeDecoderConfig(4, 2, 2, {0, 1, 2, 3}),             // Quad.
+      MakeDecoderConfig(6, 4, 2, {0, 4, 1, 2, 3, 5}),       // 5.1
+      MakeDecoderConfig(8, 5, 3, {0, 6, 1, 2, 3, 4, 5, 7})  // 7.1
+  };
+
+  const auto config = surround_configs[data[0] % surround_configs.size()];
+  RTC_CHECK(config.IsOk());
+  std::unique_ptr<AudioDecoder> dec =
+      AudioDecoderMultiChannelOpus::MakeAudioDecoder(config);
+  RTC_CHECK(dec);
+  const int kSampleRateHz = 48000;
+  const size_t kAllocatedOuputSizeSamples =
+      4 * kSampleRateHz / 10;  // 4x100 ms, 4 times the size of the output array
+                               // for the stereo Opus codec. It should be enough
+                               // for 8 channels.
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, dec.get(),
+                   kSampleRateHz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc
new file mode 100644
index 0000000000..a015f98b5b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_opus_fuzzer.cc
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  const size_t channels = (size % 2) + 1;  // 1 or 2 channels.
+  AudioDecoderOpusImpl dec(channels);
+  const int kSampleRateHz = 48000;
+  const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;  // 100 ms.
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, &dec,
+                   kSampleRateHz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
new file mode 100644
index 0000000000..efcba0f35d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_opus_redundant_fuzzer.cc
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  const size_t channels = (size % 2) + 1;  // 1 or 2 channels.
+  AudioDecoderOpusImpl dec(channels);
+  const int kSampleRateHz = 48000;
+  const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10;  // 100 ms.
+  int16_t output[kAllocatedOuputSizeSamples];
+  FuzzAudioDecoder(DecoderFunctionType::kRedundantDecode, data, size, &dec,
+                   kSampleRateHz, sizeof(output), output);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc
new file mode 100644
index 0000000000..6e5d6e2190
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_pcm16b_fuzzer.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 10000 || size < 2) {
+    return;
+  }
+
+  int sample_rate_hz;
+  switch (data[0] % 4) {
+    case 0:
+      sample_rate_hz = 8000;
+      break;
+    case 1:
+      sample_rate_hz = 16000;
+      break;
+    case 2:
+      sample_rate_hz = 32000;
+      break;
+    case 3:
+      sample_rate_hz = 48000;
+      break;
+    default:
+      RTC_DCHECK_NOTREACHED();
+      return;
+  }
+  const size_t num_channels = data[1] % 16 + 1;
+
+  // Two first bytes of the data are used. Move forward.
+  data += 2;
+  size -= 2;
+
+  AudioDecoderPcm16B dec(sample_rate_hz, num_channels);
+  // Allocate a maximum output size of 100 ms.
+  const size_t allocated_ouput_size_samples =
+      sample_rate_hz * num_channels / 10;
+  std::unique_ptr<int16_t[]> output =
+      std::make_unique<int16_t[]>(allocated_ouput_size_samples);
+  FuzzAudioDecoder(
+      DecoderFunctionType::kNormalDecode, data, size, &dec, sample_rate_hz,
+      allocated_ouput_size_samples * sizeof(int16_t), output.get());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_decoder_pcm_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_decoder_pcm_fuzzer.cc
new file mode 100644
index 0000000000..dbfcaf9976
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_decoder_pcm_fuzzer.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "test/fuzzers/audio_decoder_fuzzer.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 10000 || size < 2) {
+    return;
+  }
+
+  const size_t num_channels = data[0] % 16 + 1;
+
+  std::unique_ptr<AudioDecoder> dec;
+  if (data[1] % 2) {
+    dec = std::make_unique<AudioDecoderPcmU>(num_channels);
+  } else {
+    dec = std::make_unique<AudioDecoderPcmA>(num_channels);
+  }
+
+  // Two first bytes of the data are used. Move forward.
+  data += 2;
+  size -= 2;
+
+  // Allocate a maximum output size of 100 ms.
+  const size_t allocated_ouput_size_samples =
+      dec->SampleRateHz() * num_channels / 10;
+  std::unique_ptr<int16_t[]> output =
+      std::make_unique<int16_t[]>(allocated_ouput_size_samples);
+  FuzzAudioDecoder(DecoderFunctionType::kNormalDecode, data, size, dec.get(),
+                   dec->SampleRateHz(),
+                   allocated_ouput_size_samples * sizeof(int16_t),
+                   output.get());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.cc
new file mode 100644
index 0000000000..837c26df56
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.cc
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fuzzers/audio_encoder_fuzzer.h"
+
+#include <cstring>
+
+#include "rtc_base/buffer.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+
+// This function reads bytes from `data_view`, interprets them as RTP timestamp
+// and input samples, and sends them for encoding. The process continues until
+// no more data is available.
+void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
+                      std::unique_ptr<AudioEncoder> encoder) {
+  test::FuzzDataHelper data(data_view);
+  const size_t block_size_samples =
+      encoder->SampleRateHz() / 100 * encoder->NumChannels();
+  const size_t block_size_bytes = block_size_samples * sizeof(int16_t);
+  if (data_view.size() / block_size_bytes > 1000) {
+    // If the size of the fuzzer data is more than 1000 input blocks (i.e., more
+    // than 10 seconds), then don't fuzz at all for the fear of timing out.
+    return;
+  }
+
+  rtc::BufferT<int16_t> input_aligned(block_size_samples);
+  rtc::Buffer encoded;
+
+  // Each round in the loop below will need one block of samples + a 32-bit
+  // timestamp from the fuzzer input.
+  const size_t bytes_to_read = block_size_bytes + sizeof(uint32_t);
+  while (data.CanReadBytes(bytes_to_read)) {
+    const uint32_t timestamp = data.Read<uint32_t>();
+    auto byte_array = data.ReadByteArray(block_size_bytes);
+    // Align the data by copying to another array.
+    RTC_DCHECK_EQ(input_aligned.size() * sizeof(int16_t),
+                  byte_array.size() * sizeof(uint8_t));
+    memcpy(input_aligned.data(), byte_array.data(), byte_array.size());
+    auto info = encoder->Encode(timestamp, input_aligned, &encoded);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.h b/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.h
new file mode 100644
index 0000000000..0c879df4d3
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_encoder_fuzzer.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_
+#define TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_
+
+#include <memory>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+
+namespace webrtc {
+
+void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
+                      std::unique_ptr<AudioEncoder> encoder);
+
+}  // namespace webrtc
+
+#endif  // TEST_FUZZERS_AUDIO_ENCODER_FUZZER_H_
diff --git a/third_party/libwebrtc/test/fuzzers/audio_encoder_opus_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_encoder_opus_fuzzer.cc
new file mode 100644
index 0000000000..d67e6d6067
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_encoder_opus_fuzzer.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/audio_codecs/opus/audio_encoder_opus.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/audio_encoder_fuzzer.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  AudioEncoderOpus::Config config;
+  config.frame_size_ms = 20;
+  RTC_CHECK(config.IsOk());
+  constexpr int kPayloadType = 100;
+  FuzzAudioEncoder(
+      /*data_view=*/{data, size},
+      /*encoder=*/AudioEncoderOpus::MakeAudioEncoder(config, kPayloadType));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
new file mode 100644
index 0000000000..afcb4318f9
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <bitset>
+#include <string>
+
+#include "absl/memory/memory.h"
+#include "api/audio/echo_canceller3_factory.h"
+#include "api/audio/echo_detector_creator.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/task_queue.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/fuzzers/audio_processing_fuzzer_helper.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace {
+
+const std::string kFieldTrialNames[] = {
+    "WebRTC-Audio-Agc2ForceExtraSaturationMargin",
+    "WebRTC-Audio-Agc2ForceInitialSaturationMargin",
+    "WebRTC-Aec3MinErleDuringOnsetsKillSwitch",
+    "WebRTC-Aec3ShortHeadroomKillSwitch",
+};
+
+rtc::scoped_refptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
+                                              std::string* field_trial_string,
+                                              rtc::TaskQueue* worker_queue) {
+  // Parse boolean values for optionally enabling different
+  // configurable public components of APM.
+  bool use_ts = fuzz_data->ReadOrDefaultValue(true);
+  bool use_red = fuzz_data->ReadOrDefaultValue(true);
+  bool use_hpf = fuzz_data->ReadOrDefaultValue(true);
+  bool use_aec3 = fuzz_data->ReadOrDefaultValue(true);
+  bool use_aec = fuzz_data->ReadOrDefaultValue(true);
+  bool use_aecm = fuzz_data->ReadOrDefaultValue(true);
+  bool use_agc = fuzz_data->ReadOrDefaultValue(true);
+  bool use_ns = fuzz_data->ReadOrDefaultValue(true);
+  bool use_agc_limiter = fuzz_data->ReadOrDefaultValue(true);
+  bool use_agc2 = fuzz_data->ReadOrDefaultValue(true);
+  bool use_agc2_adaptive_digital = fuzz_data->ReadOrDefaultValue(true);
+
+  // Read a gain value supported by GainController2::Validate().
+  const float gain_controller2_gain_db =
+      fuzz_data->ReadOrDefaultValue<uint8_t>(0) % 50;
+
+  constexpr size_t kNumFieldTrials = arraysize(kFieldTrialNames);
+  // Verify that the read data type has enough bits to fuzz the field trials.
+  using FieldTrialBitmaskType = uint64_t;
+  static_assert(kNumFieldTrials <= sizeof(FieldTrialBitmaskType) * 8,
+                "FieldTrialBitmaskType is not large enough.");
+  std::bitset<kNumFieldTrials> field_trial_bitmask(
+      fuzz_data->ReadOrDefaultValue<FieldTrialBitmaskType>(0));
+  for (size_t i = 0; i < kNumFieldTrials; ++i) {
+    if (field_trial_bitmask[i]) {
+      *field_trial_string += kFieldTrialNames[i] + "/Enabled/";
+    }
+  }
+  field_trial::InitFieldTrialsFromString(field_trial_string->c_str());
+
+  // Ignore a few bytes. Bytes from this segment will be used for
+  // future config flag changes. We assume 40 bytes is enough for
+  // configuring the APM.
+  constexpr size_t kSizeOfConfigSegment = 40;
+  RTC_DCHECK(kSizeOfConfigSegment >= fuzz_data->BytesRead());
+  static_cast<void>(
+      fuzz_data->ReadByteArray(kSizeOfConfigSegment - fuzz_data->BytesRead()));
+
+  // Filter out incompatible settings that lead to CHECK failures.
+  if ((use_aecm && use_aec) ||          // These settings cause CHECK failure.
+      (use_aecm && use_aec3 && use_ns)  // These settings trigger webrtc:9489.
+  ) {
+    return nullptr;
+  }
+
+  std::unique_ptr<EchoControlFactory> echo_control_factory;
+  if (use_aec3) {
+    echo_control_factory.reset(new EchoCanceller3Factory());
+  }
+
+  webrtc::AudioProcessing::Config apm_config;
+  apm_config.pipeline.multi_channel_render = true;
+  apm_config.pipeline.multi_channel_capture = true;
+  apm_config.echo_canceller.enabled = use_aec || use_aecm;
+  apm_config.echo_canceller.mobile_mode = use_aecm;
+  apm_config.high_pass_filter.enabled = use_hpf;
+  apm_config.gain_controller1.enabled = use_agc;
+  apm_config.gain_controller1.enable_limiter = use_agc_limiter;
+  apm_config.gain_controller2.enabled = use_agc2;
+  apm_config.gain_controller2.fixed_digital.gain_db = gain_controller2_gain_db;
+  apm_config.gain_controller2.adaptive_digital.enabled =
+      use_agc2_adaptive_digital;
+  apm_config.noise_suppression.enabled = use_ns;
+  apm_config.transient_suppression.enabled = use_ts;
+
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetEchoControlFactory(std::move(echo_control_factory))
+          .SetEchoDetector(use_red ? CreateEchoDetector() : nullptr)
+          .SetConfig(apm_config)
+          .Create();
+
+#ifdef WEBRTC_LINUX
+  apm->AttachAecDump(AecDumpFactory::Create("/dev/null", -1, worker_queue));
+#endif
+
+  return apm;
+}
+
+TaskQueueFactory* GetTaskQueueFactory() {
+  static TaskQueueFactory* const factory =
+      CreateDefaultTaskQueueFactory().release();
+  return factory;
+}
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 400000) {
+    return;
+  }
+  test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
+  // This string must be in scope during execution, according to documentation
+  // for field_trial.h. Hence it's created here and not in CreateApm.
+  std::string field_trial_string = "";
+
+  rtc::TaskQueue worker_queue(GetTaskQueueFactory()->CreateTaskQueue(
+      "rtc-low-prio", rtc::TaskQueue::Priority::LOW));
+  auto apm = CreateApm(&fuzz_data, &field_trial_string, &worker_queue);
+
+  if (apm) {
+    FuzzAudioProcessing(&fuzz_data, std::move(apm));
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_processing_fuzzer_helper.cc b/third_party/libwebrtc/test/fuzzers/audio_processing_fuzzer_helper.cc
new file mode 100644
index 0000000000..5252918d77
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_processing_fuzzer_helper.cc
@@ -0,0 +1,143 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fuzzers/audio_processing_fuzzer_helper.h"
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <limits>
+
+#include "api/audio/audio_frame.h"
+#include "modules/audio_processing/include/audio_frame_proxies.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+bool ValidForApm(float x) {
+  return std::isfinite(x) && -1.0f <= x && x <= 1.0f;
+}
+
+void GenerateFloatFrame(test::FuzzDataHelper* fuzz_data,
+                        int input_rate,
+                        int num_channels,
+                        float* const* float_frames) {
+  const int samples_per_input_channel =
+      AudioProcessing::GetFrameSize(input_rate);
+  RTC_DCHECK_LE(samples_per_input_channel, 480);
+  for (int i = 0; i < num_channels; ++i) {
+    std::fill(float_frames[i], float_frames[i] + samples_per_input_channel, 0);
+    const size_t read_bytes = sizeof(float) * samples_per_input_channel;
+    if (fuzz_data->CanReadBytes(read_bytes)) {
+      rtc::ArrayView<const uint8_t> byte_array =
+          fuzz_data->ReadByteArray(read_bytes);
+      memmove(float_frames[i], byte_array.begin(), read_bytes);
+    }
+
+    // Sanitize input.
+    for (int j = 0; j < samples_per_input_channel; ++j) {
+      if (!ValidForApm(float_frames[i][j])) {
+        float_frames[i][j] = 0.f;
+      }
+    }
+  }
+}
+
+void GenerateFixedFrame(test::FuzzDataHelper* fuzz_data,
+                        int input_rate,
+                        int num_channels,
+                        AudioFrame* fixed_frame) {
+  const int samples_per_input_channel =
+      AudioProcessing::GetFrameSize(input_rate);
+
+  fixed_frame->samples_per_channel_ = samples_per_input_channel;
+  fixed_frame->sample_rate_hz_ = input_rate;
+  fixed_frame->num_channels_ = num_channels;
+
+  RTC_DCHECK_LE(samples_per_input_channel * num_channels,
+                AudioFrame::kMaxDataSizeSamples);
+  for (int i = 0; i < samples_per_input_channel * num_channels; ++i) {
+    fixed_frame->mutable_data()[i] = fuzz_data->ReadOrDefaultValue<int16_t>(0);
+  }
+}
+}  // namespace
+
+void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
+                         rtc::scoped_refptr<AudioProcessing> apm) {
+  AudioFrame fixed_frame;
+  // Normal usage is up to 8 channels. Allowing to fuzz one beyond this allows
+  // us to catch implicit assumptions about normal usage.
+  constexpr int kMaxNumChannels = 9;
+  std::array<std::array<float, 480>, kMaxNumChannels> float_frames;
+  std::array<float*, kMaxNumChannels> float_frame_ptrs;
+  for (int i = 0; i < kMaxNumChannels; ++i) {
+    float_frame_ptrs[i] = float_frames[i].data();
+  }
+  float* const* ptr_to_float_frames = &float_frame_ptrs[0];
+
+  constexpr int kSampleRatesHz[] = {8000,  11025, 16000, 22050,
+                                    32000, 44100, 48000};
+
+  // We may run out of fuzz data in the middle of a loop iteration. In
+  // that case, default values will be used for the rest of that
+  // iteration.
+  while (fuzz_data->CanReadBytes(1)) {
+    const bool is_float = fuzz_data->ReadOrDefaultValue(true);
+    // Decide input/output rate for this iteration.
+    const int input_rate = fuzz_data->SelectOneOf(kSampleRatesHz);
+    const int output_rate = fuzz_data->SelectOneOf(kSampleRatesHz);
+
+    const uint8_t stream_delay = fuzz_data->ReadOrDefaultValue<uint8_t>(0);
+    // API call needed for AECM to run.
+    apm->set_stream_delay_ms(stream_delay);
+
+    const bool key_pressed = fuzz_data->ReadOrDefaultValue(true);
+    apm->set_stream_key_pressed(key_pressed);
+
+    // Make the APM call depending on capture/render mode and float /
+    // fix interface.
+    const bool is_capture = fuzz_data->ReadOrDefaultValue(true);
+
+    // Fill the arrays with audio samples from the data.
+    int apm_return_code = AudioProcessing::Error::kNoError;
+    if (is_float) {
+      const int num_channels =
+          fuzz_data->ReadOrDefaultValue<uint8_t>(1) % kMaxNumChannels;
+
+      GenerateFloatFrame(fuzz_data, input_rate, num_channels,
+                         ptr_to_float_frames);
+      if (is_capture) {
+        apm_return_code = apm->ProcessStream(
+            ptr_to_float_frames, StreamConfig(input_rate, num_channels),
+            StreamConfig(output_rate, num_channels), ptr_to_float_frames);
+      } else {
+        apm_return_code = apm->ProcessReverseStream(
+            ptr_to_float_frames, StreamConfig(input_rate, num_channels),
+            StreamConfig(output_rate, num_channels), ptr_to_float_frames);
+      }
+    } else {
+      const int num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
+      GenerateFixedFrame(fuzz_data, input_rate, num_channels, &fixed_frame);
+
+      if (is_capture) {
+        apm_return_code = ProcessAudioFrame(apm.get(), &fixed_frame);
+      } else {
+        apm_return_code = ProcessReverseAudioFrame(apm.get(), &fixed_frame);
+      }
+    }
+
+    // Cover stats gathering code paths.
+    static_cast<void>(apm->GetStatistics(true /*has_remote_tracks*/));
+
+    RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/audio_processing_fuzzer_helper.h b/third_party/libwebrtc/test/fuzzers/audio_processing_fuzzer_helper.h
new file mode 100644
index 0000000000..a604db8cef
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_processing_fuzzer_helper.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
+#define TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
+
+#include <memory>
+
+#include "modules/audio_processing/include/audio_processing.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+namespace webrtc {
+
+void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
+                         rtc::scoped_refptr<AudioProcessing> apm);
+
+}  // namespace webrtc
+
+#endif  // TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
diff --git a/third_party/libwebrtc/test/fuzzers/audio_processing_sample_rate_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
new file mode 100644
index 0000000000..ca3946988c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/audio_processing_sample_rate_fuzzer.cc
@@ -0,0 +1,162 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <array>
+#include <cmath>
+#include <limits>
+
+#include "modules/audio_processing/include/audio_processing.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace {
+constexpr int kMaxNumChannels = 2;
+// APM supported max rate is 384000 Hz, using a limit slightly above lets the
+// fuzzer exercise the handling of too high rates.
+constexpr int kMaxSampleRateHz = 400000;
+constexpr int kMaxSamplesPerChannel = kMaxSampleRateHz / 100;
+
+void GenerateFloatFrame(test::FuzzDataHelper& fuzz_data,
+                        int input_rate,
+                        int num_channels,
+                        float* const* float_frames) {
+  const int samples_per_input_channel =
+      AudioProcessing::GetFrameSize(input_rate);
+  RTC_DCHECK_LE(samples_per_input_channel, kMaxSamplesPerChannel);
+  for (int i = 0; i < num_channels; ++i) {
+    float channel_value;
+    fuzz_data.CopyTo<float>(&channel_value);
+    std::fill(float_frames[i], float_frames[i] + samples_per_input_channel,
+              channel_value);
+  }
+}
+
+void GenerateFixedFrame(test::FuzzDataHelper& fuzz_data,
+                        int input_rate,
+                        int num_channels,
+                        int16_t* fixed_frames) {
+  const int samples_per_input_channel =
+      AudioProcessing::GetFrameSize(input_rate);
+  RTC_DCHECK_LE(samples_per_input_channel, kMaxSamplesPerChannel);
+  // Write interleaved samples.
+  for (int ch = 0; ch < num_channels; ++ch) {
+    const int16_t channel_value = fuzz_data.ReadOrDefaultValue<int16_t>(0);
+    for (int i = ch; i < samples_per_input_channel * num_channels;
+         i += num_channels) {
+      fixed_frames[i] = channel_value;
+    }
+  }
+}
+
+// No-op processor used to influence APM input/output pipeline decisions based
+// on what submodules are present.
+class NoopCustomProcessing : public CustomProcessing {
+ public:
+  NoopCustomProcessing() {}
+  ~NoopCustomProcessing() override {}
+  void Initialize(int sample_rate_hz, int num_channels) override {}
+  void Process(AudioBuffer* audio) override {}
+  std::string ToString() const override { return ""; }
+  void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {}
+};
+}  // namespace
+
+// This fuzzer is directed at fuzzing unexpected input and output sample rates
+// of APM. For example, the sample rate 22050 Hz is processed by APM in frames
+// of floor(22050/100) = 220 samples. This is not exactly 10 ms of audio
+// content, and may break assumptions commonly made on the APM frame size.
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 100) {
+    return;
+  }
+  test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
+
+  std::unique_ptr<CustomProcessing> capture_processor =
+      fuzz_data.ReadOrDefaultValue(true)
+          ? std::make_unique<NoopCustomProcessing>()
+          : nullptr;
+  std::unique_ptr<CustomProcessing> render_processor =
+      fuzz_data.ReadOrDefaultValue(true)
+          ? std::make_unique<NoopCustomProcessing>()
+          : nullptr;
+  rtc::scoped_refptr<AudioProcessing> apm =
+      AudioProcessingBuilderForTesting()
+          .SetConfig({.pipeline = {.multi_channel_render = true,
+                                   .multi_channel_capture = true}})
+          .SetCapturePostProcessing(std::move(capture_processor))
+          .SetRenderPreProcessing(std::move(render_processor))
+          .Create();
+  RTC_DCHECK(apm);
+
+  std::array<int16_t, kMaxSamplesPerChannel * kMaxNumChannels> fixed_frame;
+  std::array<std::array<float, kMaxSamplesPerChannel>, kMaxNumChannels>
+      float_frames;
+  std::array<float*, kMaxNumChannels> float_frame_ptrs;
+  for (int i = 0; i < kMaxNumChannels; ++i) {
+    float_frame_ptrs[i] = float_frames[i].data();
+  }
+  float* const* ptr_to_float_frames = &float_frame_ptrs[0];
+
+  // Choose whether to fuzz the float or int16_t interfaces of APM.
+  const bool is_float = fuzz_data.ReadOrDefaultValue(true);
+
+  // We may run out of fuzz data in the middle of a loop iteration. In
+  // that case, default values will be used for the rest of that
+  // iteration.
+  while (fuzz_data.CanReadBytes(1)) {
+    // Decide input/output rate for this iteration.
+    const int input_rate = static_cast<int>(
+        fuzz_data.ReadOrDefaultValue<size_t>(8000) % kMaxSampleRateHz);
+    const int output_rate = static_cast<int>(
+        fuzz_data.ReadOrDefaultValue<size_t>(8000) % kMaxSampleRateHz);
+    const int num_channels = fuzz_data.ReadOrDefaultValue(true) ? 2 : 1;
+
+    // Since render and capture calls have slightly different reinitialization
+    // procedures, we let the fuzzer choose the order.
+    const bool is_capture = fuzz_data.ReadOrDefaultValue(true);
+
+    int apm_return_code = AudioProcessing::Error::kNoError;
+    if (is_float) {
+      GenerateFloatFrame(fuzz_data, input_rate, num_channels,
+                         ptr_to_float_frames);
+
+      if (is_capture) {
+        apm_return_code = apm->ProcessStream(
+            ptr_to_float_frames, StreamConfig(input_rate, num_channels),
+            StreamConfig(output_rate, num_channels), ptr_to_float_frames);
+      } else {
+        apm_return_code = apm->ProcessReverseStream(
+            ptr_to_float_frames, StreamConfig(input_rate, num_channels),
+            StreamConfig(output_rate, num_channels), ptr_to_float_frames);
+      }
+    } else {
+      GenerateFixedFrame(fuzz_data, input_rate, num_channels,
+                         fixed_frame.data());
+
+      if (is_capture) {
+        apm_return_code = apm->ProcessStream(
+            fixed_frame.data(), StreamConfig(input_rate, num_channels),
+            StreamConfig(output_rate, num_channels), fixed_frame.data());
+      } else {
+        apm_return_code = apm->ProcessReverseStream(
+            fixed_frame.data(), StreamConfig(input_rate, num_channels),
+            StreamConfig(output_rate, num_channels), fixed_frame.data());
+      }
+    }
+    // APM may flag an error on unsupported audio formats, but should not crash.
+    RTC_DCHECK(apm_return_code == AudioProcessing::kNoError ||
+               apm_return_code == AudioProcessing::kBadSampleRateError);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc
new file mode 100644
index 0000000000..7f44af99fb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+
+#include "api/array_view.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "rtc_base/buffer.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+void FuzzOneInputTest(rtc::ArrayView<const uint8_t> data) {
+  FuzzDataHelper fuzz_data(data);
+  ComfortNoiseDecoder cng_decoder;
+
+  while (1) {
+    if (!fuzz_data.CanReadBytes(1))
+      break;
+    const uint8_t sid_frame_len = fuzz_data.Read<uint8_t>();
+    auto sid_frame = fuzz_data.ReadByteArray(sid_frame_len);
+    if (sid_frame.empty())
+      break;
+    cng_decoder.UpdateSid(sid_frame);
+    if (!fuzz_data.CanReadBytes(3))
+      break;
+    constexpr bool kTrueOrFalse[] = {true, false};
+    const bool new_period = fuzz_data.SelectOneOf(kTrueOrFalse);
+    constexpr size_t kOutputSizes[] = {80, 160, 320, 480};
+    const size_t output_size = fuzz_data.SelectOneOf(kOutputSizes);
+    const size_t num_generate_calls =
+        std::min(fuzz_data.Read<uint8_t>(), static_cast<uint8_t>(17));
+    rtc::BufferT<int16_t> output(output_size);
+    for (size_t i = 0; i < num_generate_calls; ++i) {
+      cng_decoder.Generate(output, new_period);
+    }
+  }
+}
+
+}  // namespace
+}  // namespace test
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 5000) {
+    return;
+  }
+  test::FuzzOneInputTest(rtc::ArrayView<const uint8_t>(data, size));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_fec_config.json b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_fec_config.json
new file mode 100644
index 0000000000..59b5db9446
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_fec_config.json
@@ -0,0 +1,151 @@
+[
+   {
+      "decoders" : [
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "1"
+               },
+               {
+                  "packetization-mode" : "1"
+               },
+               {
+                  "profile-level-id" : "42001f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 100
+         },
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "1"
+               },
+               {
+                  "packetization-mode" : "0"
+               },
+               {
+                  "profile-level-id" : "42001f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 102
+         },
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "1"
+               },
+               {
+                  "packetization-mode" : "1"
+               },
+               {
+                  "profile-level-id" : "42e01f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 127
+         },
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "1"
+               },
+               {
+                  "packetization-mode" : "0"
+               },
+               {
+                  "profile-level-id" : "42e01f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 125
+         },
+         {
+            "codec_params" : [],
+            "payload_name" : "VP8",
+            "payload_type" : 96
+         },
+         {
+            "codec_params" : [],
+            "payload_name" : "VP9",
+            "payload_type" : 98
+         }
+      ],
+      "render_delay_ms" : 10,
+      "rtp" : {
+         "extensions" : [
+            {
+               "encrypt" : false,
+               "id" : 5,
+               "uri" : "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
+            },
+            {
+               "encrypt" : false,
+               "id" : 3,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
+            },
+            {
+               "encrypt" : false,
+               "id" : 6,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
+            },
+            {
+               "encrypt" : false,
+               "id" : 7,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"
+            },
+            {
+               "encrypt" : false,
+               "id" : 8,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/video-timing"
+            },
+            {
+               "encrypt" : false,
+               "id" : 4,
+               "uri" : "urn:3gpp:video-orientation"
+            },
+            {
+               "encrypt" : false,
+               "id" : 2,
+               "uri" : "urn:ietf:params:rtp-hdrext:toffset"
+            }
+         ],
+         "local_ssrc" : 1,
+         "lntf" : {
+            "enabled": false,
+         },
+         "nack" : {
+            "rtp_history_ms" : 1000
+         },
+         "red_payload_type" : -1,
+         "remb" : true,
+         "remote_ssrc" : 2736493666,
+         "rtcp_mode" : "RtcpMode::kReducedSize",
+         "rtx_payload_types" : [
+            {
+               "97" : 96
+            },
+            {
+               "99" : 98
+            },
+            {
+               "101" : 100
+            },
+            {
+               "107" : 125
+            },
+            {
+               "122" : 127
+            },
+            {
+               "123" : 102
+            }
+         ],
+         "rtx_ssrc" : 885796452,
+         "transport_cc" : true,
+         "ulpfec_payload_type" : -1
+      },
+      "target_delay_ms" : 0
+   }
+]
diff --git a/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_non_interleaved_config.json b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_non_interleaved_config.json
new file mode 100644
index 0000000000..9cb5bd767b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_non_interleaved_config.json
@@ -0,0 +1,66 @@
+[
+   {
+      "decoders" : [
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "0"
+               },
+               {
+                  "packetization-mode" : "0"
+               },
+               {
+                  "profile-level-id" : "42001f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 100
+         },
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "1"
+               },
+               {
+                  "packetization-mode" : "1"
+               },
+               {
+                  "profile-level-id" : "42e01f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 102
+         }
+      ],
+      "render_delay_ms" : 10,
+      "rtp" : {
+         "extensions" : [],
+         "local_ssrc" : 1,
+         "lntf" : {
+            "enabled": false,
+         },
+         "nack" : {
+            "rtp_history_ms" : 1000
+         },
+         "red_payload_type" : 125,
+         "remb" : true,
+         "remote_ssrc" : 1989790381,
+         "rtcp_mode" : "RtcpMode::kReducedSize",
+         "rtx_payload_types" : [
+            {
+               "101" : 100
+            },
+            {
+               "122" : 125
+            },
+            {
+               "123" : 127
+            }
+         ],
+         "rtx_ssrc" : 1406083315,
+         "transport_cc" : true,
+         "ulpfec_payload_type" : -1
+      },
+      "target_delay_ms" : 0
+   }
+]
diff --git a/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_single_nal_config.json b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_single_nal_config.json
new file mode 100644
index 0000000000..f3ae6bbafc
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/h264_single_nal_config.json
@@ -0,0 +1,66 @@
+[
+   {
+      "decoders" : [
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "1"
+               },
+               {
+                  "packetization-mode" : "1"
+               },
+               {
+                  "profile-level-id" : "42001f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 100
+         },
+         {
+            "codec_params" : [
+               {
+                  "level-asymmetry-allowed" : "1"
+               },
+               {
+                  "packetization-mode" : "1"
+               },
+               {
+                  "profile-level-id" : "42e01f"
+               }
+            ],
+            "payload_name" : "H264",
+            "payload_type" : 102
+         }
+      ],
+      "render_delay_ms" : 10,
+      "rtp" : {
+         "extensions" : [],
+         "local_ssrc" : 1,
+         "lntf" : {
+            "enabled": false,
+         },
+         "nack" : {
+            "rtp_history_ms" : 1000
+         },
+         "red_payload_type" : 125,
+         "remb" : true,
+         "remote_ssrc" : 1989790381,
+         "rtcp_mode" : "RtcpMode::kReducedSize",
+         "rtx_payload_types" : [
+            {
+               "101" : 100
+            },
+            {
+               "122" : 125
+            },
+            {
+               "123" : 127
+            }
+         ],
+         "rtx_ssrc" : 1406083315,
+         "transport_cc" : true,
+         "ulpfec_payload_type" : -1
+      },
+      "target_delay_ms" : 0
+   }
+]
diff --git a/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json
new file mode 100644
index 0000000000..0a5eef8b09
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp8_config.json
@@ -0,0 +1,29 @@
+[{
+      "decoders" : [
+         {
+            "codec_params" : [],
+            "payload_name" : "VP8",
+            "payload_type" : 125
+         }
+      ],
+      "render_delay_ms" : 10,
+      "rtp" : {
+         "extensions" : [],
+         "local_ssrc" : 7331,
+         "lntf" : {
+            "enabled": false,
+         },
+         "nack" : {
+            "rtp_history_ms" : 1000
+         },
+         "red_payload_type" : -1,
+         "remb" : true,
+         "remote_ssrc" : 1337,
+         "rtcp_mode" : "RtcpMode::kCompound",
+         "rtx_payload_types" : [],
+         "rtx_ssrc" : 100,
+         "transport_cc" : true,
+         "ulpfec_payload_type" : -1
+      },
+      "target_delay_ms" : 0
+}]
diff --git a/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp8_fec_config.json b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp8_fec_config.json
new file mode 100644
index 0000000000..3d2b66301d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp8_fec_config.json
@@ -0,0 +1,73 @@
+[
+   {
+      "decoders" : [
+         {
+            "codec_params" : [],
+            "payload_name" : "VP8",
+            "payload_type" : 96
+         }
+      ],
+      "render_delay_ms" : 10,
+      "rtp" : {
+         "extensions" : [
+            {
+               "encrypt" : false,
+               "id" : 5,
+               "uri" : "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
+            },
+            {
+               "encrypt" : false,
+               "id" : 3,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
+            },
+            {
+               "encrypt" : false,
+               "id" : 6,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
+            },
+            {
+               "encrypt" : false,
+               "id" : 7,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"
+            },
+            {
+               "encrypt" : false,
+               "id" : 8,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/video-timing"
+            },
+            {
+               "encrypt" : false,
+               "id" : 4,
+               "uri" : "urn:3gpp:video-orientation"
+            },
+            {
+               "encrypt" : false,
+               "id" : 2,
+               "uri" : "urn:ietf:params:rtp-hdrext:toffset"
+            }
+         ],
+         "local_ssrc" : 1,
+         "lntf" : {
+            "enabled": false,
+         },
+         "nack" : {
+            "rtp_history_ms" : 1000
+         },
+         "red_payload_type" : -1,
+         "remb" : true,
+         "remote_ssrc" : 2672243158,
+         "rtcp_mode" : "RtcpMode::kReducedSize",
+         "rtx_payload_types" : [
+            {
+               "97" : 96
+            },
+            {
+               "99" : 98
+            }
+         ],
+         "rtx_ssrc" : 1807563126,
+         "transport_cc" : true
+      },
+      "target_delay_ms" : 0
+   }
+]
diff --git a/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp9_config.json b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp9_config.json
new file mode 100644
index 0000000000..a9c88a2402
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp9_config.json
@@ -0,0 +1,29 @@
+[{
+      "decoders" : [
+         {
+            "codec_params" : [],
+            "payload_name" : "VP9",
+            "payload_type" : 124
+         }
+      ],
+      "render_delay_ms" : 10,
+      "rtp" : {
+         "extensions" : [],
+         "local_ssrc" : 7331,
+         "lntf" : {
+            "enabled": false,
+         },
+         "nack" : {
+            "rtp_history_ms" : 1000
+         },
+         "red_payload_type" : -1,
+         "remb" : true,
+         "remote_ssrc" : 1337,
+         "rtcp_mode" : "RtcpMode::kCompound",
+         "rtx_payload_types" : [],
+         "rtx_ssrc" : 100,
+         "transport_cc" : true,
+         "ulpfec_payload_type" : -1
+      },
+      "target_delay_ms" : 0
+}]
diff --git a/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp9_fec_config.json b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp9_fec_config.json
new file mode 100644
index 0000000000..d089aa8fec
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/configs/replay_packet_fuzzer/vp9_fec_config.json
@@ -0,0 +1,79 @@
+[
+   {
+      "decoders" : [
+         {
+            "codec_params" : [],
+            "payload_name" : "VP9",
+            "payload_type" : 98
+         },
+         {
+            "codec_params" : [],
+            "payload_name" : "VP8",
+            "payload_type" : 96
+         }
+      ],
+      "render_delay_ms" : 10,
+      "rtp" : {
+         "extensions" : [
+            {
+               "encrypt" : false,
+               "id" : 5,
+               "uri" : "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"
+            },
+            {
+               "encrypt" : false,
+               "id" : 3,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
+            },
+            {
+               "encrypt" : false,
+               "id" : 6,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay"
+            },
+            {
+               "encrypt" : false,
+               "id" : 7,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"
+            },
+            {
+               "encrypt" : false,
+               "id" : 8,
+               "uri" : "http://www.webrtc.org/experiments/rtp-hdrext/video-timing"
+            },
+            {
+               "encrypt" : false,
+               "id" : 4,
+               "uri" : "urn:3gpp:video-orientation"
+            },
+            {
+               "encrypt" : false,
+               "id" : 2,
+               "uri" : "urn:ietf:params:rtp-hdrext:toffset"
+            }
+         ],
+         "local_ssrc" : 1,
+         "lntf" : {
+            "enabled": false,
+         },
+         "nack" : {
+            "rtp_history_ms" : 1000
+         },
+         "red_payload_type" : -1,
+         "remb" : true,
+         "remote_ssrc" : 2678204013,
+         "rtcp_mode" : "RtcpMode::kReducedSize",
+         "rtx_payload_types" : [
+            {
+               "97" : 96
+            },
+            {
+               "99" : 98
+            }
+         ],
+         "rtx_ssrc" : 1110725867,
+         "transport_cc" : true,
+         "ulpfec_payload_type" : -1
+      },
+      "target_delay_ms" : 0
+   }
+]
diff --git a/third_party/libwebrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc
new file mode 100644
index 0000000000..e266b85fa4
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/congestion_controller_feedback_fuzzer.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "absl/functional/bind_front.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  size_t i = 0;
+  if (size < sizeof(int64_t) + sizeof(uint8_t) + sizeof(uint32_t))
+    return;
+  SimulatedClock clock(data[i++]);
+  PacketRouter packet_router;
+  ReceiveSideCongestionController cc(
+      &clock,
+      absl::bind_front(&PacketRouter::SendCombinedRtcpPacket, &packet_router),
+      absl::bind_front(&PacketRouter::SendRemb, &packet_router), nullptr);
+  RTPHeader header;
+  header.ssrc = ByteReader<uint32_t>::ReadBigEndian(&data[i]);
+  i += sizeof(uint32_t);
+  header.extension.hasTransportSequenceNumber = true;
+  int64_t arrival_time_ms = std::min<int64_t>(
+      std::max<int64_t>(ByteReader<int64_t>::ReadBigEndian(&data[i]), 0),
+      std::numeric_limits<int64_t>::max() / 2);
+  i += sizeof(int64_t);
+  const size_t kMinPacketSize =
+      sizeof(size_t) + sizeof(uint16_t) + sizeof(uint8_t);
+  while (i + kMinPacketSize < size) {
+    size_t payload_size = ByteReader<size_t>::ReadBigEndian(&data[i]) % 1500;
+    i += sizeof(size_t);
+    header.extension.transportSequenceNumber =
+        ByteReader<uint16_t>::ReadBigEndian(&data[i]);
+    i += sizeof(uint16_t);
+    cc.OnReceivedPacket(arrival_time_ms, payload_size, header);
+    clock.AdvanceTimeMilliseconds(5);
+    arrival_time_ms += ByteReader<uint8_t>::ReadBigEndian(&data[i]);
+    i += sizeof(uint8_t);
+    cc.MaybeProcess();
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/README b/third_party/libwebrtc/test/fuzzers/corpora/README
new file mode 100644
index 0000000000..cc87025ff6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/README
@@ -0,0 +1,37 @@
+This is a collection of corpora for various WebRTC fuzzers. To use
+them, the gn targets define seed_corpus=$corpus_dir, which causes the
+ClusterFuzz upload bot to bundle $corpus_dir and upload it.
+
+The format is simple: one file per test case. Specific notes are
+included below.
+
+### SDP ###
+This corpus was initially assembled manually from the following
+sources:
+
+  - curl --silent https://www.ietf.org/rfc/rfc4317.txt | grep '^[ a-z]*=[^=]*$' | sed 's/^[[:space:]]*//' | awk -v RS='(^|\n)v=' '/./ {print "v="$0 > NR".sdp"}'
+  - all the SDPs used in the parser unit tests
+  - some manually gathered SDPs from Firefox and Opera
+
+The SDP tokens come from:
+
+ -  grep "^static const " webrtc/api/webrtcsdp.cc | cut -d'=' -f2 | cut -d ';' -f1 | tr -d '"' | tr -d "'" | tr -d ' ' | sort -u | grep -v '^(\n|\r|\r\n)$|^$' | sed -e 's/^/"/' -e 's/$/"/' | tail -n +2
+
+### STUN ###
+This corpus was initially assembled from the STUN unit tests, together
+with a crash that it found relatively quickly.
+
+### RT(C)P ###
+This corpus was initially assembled from the unittests. RTCP was
+minimised first.
+
+There is also rt(c?)p-corpus-with-extra-byte, in which each sample is
+prefixed by the byte 0xff. Some of the rtp fuzzers need to decide
+which header extensions to enable, and the first byte of the fuzz data
+is used for this.
+
+### PseudoTCP ###
+Very small corpus minimised from the unit tests.
+
+### SCTP ###
+This corpus was extracted from a few manually recorder wireshark dumps.
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/aec3-config-json-corpus/default.json b/third_party/libwebrtc/test/fuzzers/corpora/aec3-config-json-corpus/default.json
new file mode 100644
index 0000000000..4dcbcee53a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/aec3-config-json-corpus/default.json
@@ -0,0 +1 @@
+{"aec3": {"buffering": {"excess_render_detection_interval_blocks": 250,"max_allowed_excess_render_blocks": 8},"delay": {"default_delay": 5,"down_sampling_factor": 4,"num_filters": 5,"delay_headroom_samples": 32,"hysteresis_limit_blocks": 1,"fixed_capture_delay_samples": 0,"delay_estimate_smoothing": 0.7,"delay_candidate_detection_threshold": 0.2,"delay_selection_thresholds": {"initial": 5,"converged": 20},"use_external_delay_estimator": false,"downmix_before_delay_estimation": true,"log_warning_on_delay_changes": false},"filter": {"main": [13,5e-05,0.05,0.001,2,2.00753e+07],"shadow": [13,0.7,2.00753e+07],"main_initial": [12,0.005,0.5,0.001,2,2.00753e+07],"shadow_initial": [12,0.9,2.00753e+07],"config_change_duration_blocks": 250,"initial_state_seconds": 2.5,"conservative_initial_phase": false,"enable_shadow_filter_output_usage": true},"erle": {"min": 1,"max_l": 4,"max_h": 1.5,"onset_detection": true,"num_sections": 1,"clamp_quality_estimate_to_zero": true,"clamp_quality_estimate_to_one": true},"ep_strength": {"default_gain": 1,"default_len": 0.83,"echo_can_saturate": true,"bounded_erl": false},"echo_audibility": {"low_render_limit": 256,"normal_render_limit": 64,"floor_power": 128,"audibility_threshold_lf": 10,"audibility_threshold_mf": 10,"audibility_threshold_hf": 10,"use_stationarity_properties": false,"use_stationarity_properties_at_init": false},"render_levels": {"active_render_limit": 100,"poor_excitation_render_limit": 150,"poor_excitation_render_limit_ds8": 20},"echo_removal_control": {"has_clock_drift": false,"linear_and_stable_echo_path": false},"echo_model": {"noise_floor_hold": 50,"min_noise_floor_power": 1.6384e+06,"stationary_gate_slope": 10,"noise_gate_power": 27509.4,"noise_gate_slope": 0.3,"render_pre_window_size": 1,"render_post_window_size": 1},"suppressor": {"nearend_average_blocks": 4,"normal_tuning": {"mask_lf": [0.3,0.4,0.3],"mask_hf": [0.07,0.1,0.3],"max_inc_factor": 2,"max_dec_factor_lf": 0.25},"nearend_tuning": {"mask_lf": [1.09,1.1,0.3],"mask_hf": [0.1,0.3,0.3],"max_inc_factor": 2,"max_dec_factor_lf": 0.25},"dominant_nearend_detection": {"enr_threshold": 0.25,"enr_exit_threshold": 10,"snr_threshold": 30,"hold_duration": 50,"trigger_threshold": 12,"use_during_initial_phase": 1},"high_bands_suppression": {"enr_threshold": 1,"max_gain_during_echo": 1},"floor_first_increase": 1e-05,"enforce_transparent": false,"enforce_empty_higher_bands": false}}}
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-1 b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-1
new file mode 100644
index 0000000000..cda107c534
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-2 b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-2
new file mode 100644
index 0000000000..bf1a98dccd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-2
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-3 b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-3
new file mode 100644
index 0000000000..f28283502a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-3
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-4 b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-4
new file mode 100644
index 0000000000..7b5d1cd790
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/agc-corpus/agc-4
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-0 b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-0
new file mode 100644
index 0000000000..da32f0c29c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-0
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-1 b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-1
new file mode 100644
index 0000000000..097af39c3f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-2 b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-2
new file mode 100644
index 0000000000..a9f4011d3f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-2
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-3 b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-3
new file mode 100644
index 0000000000..b4ba1578cd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-3
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-0 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-0
new file mode 100644
index 0000000000..fc56e0e591
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-0
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-1 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-1
new file mode 100644
index 0000000000..b5c147f9f3
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-10 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-10
new file mode 100644
index 0000000000..ced48741aa
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-10
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-11 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-11
new file mode 100644
index 0000000000..53afe97468
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-11
@@ -0,0 +1 @@
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°Ë±±o
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diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-12 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-12
new file mode 100644
index 0000000000..4d126eacc3
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-12
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-13 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-13
new file mode 100644
index 0000000000..847e67b79a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-13
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-14 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-14
new file mode 100644
index 0000000000..f5efdd1317
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-14
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-15 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-15
new file mode 100644
index 0000000000..830dc8a679
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-15
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-16 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-16
new file mode 100644
index 0000000000..06017917dc
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-16
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-2 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-2
new file mode 100644
index 0000000000..43cf72619c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-2
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-3 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-3
new file mode 100644
index 0000000000..a2e8e0c9ae
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-3
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-4 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-4
new file mode 100644
index 0000000000..ff5785f0c3
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-4
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-5 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-5
new file mode 100644
index 0000000000..f1196c98aa
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-5
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-6 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-6
new file mode 100644
index 0000000000..88ce0a1600
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-6
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-7 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-7
new file mode 100644
index 0000000000..9adb5c2125
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-7
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-8 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-8
new file mode 100644
index 0000000000..5a8f929a61
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-8
@@ -0,0 +1,2 @@
+ÿÿÿÞ
+‚
ÿ
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-9 b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-9
new file mode 100644
index 0000000000..bde1b127b3
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/dependency_descriptor-corpus/dependency-descriptor-9
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-0 b/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-0
new file mode 100644
index 0000000000..73e1d22f10
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-0
@@ -0,0 +1 @@
+WebRTC-DecoderDataDumpDirectory/Enabled/
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-1 b/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-1
new file mode 100644
index 0000000000..997888a6fd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-1
@@ -0,0 +1 @@
+WebRTC-DecoderDataDumpDirectory/Disabled/
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-2 b/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-2
new file mode 100644
index 0000000000..7e897a418d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/field_trial-corpus/field-trial-2
@@ -0,0 +1 @@
+WebRTC-DecoderDataDumpDirectory/Disabled/WebRTC-IPv6Default/Enabled/
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/h264-depacketizer-fuzzer-corpus/h264-0 b/third_party/libwebrtc/test/fuzzers/corpora/h264-depacketizer-fuzzer-corpus/h264-0
new file mode 100644
index 0000000000..dbe089f278
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/h264-depacketizer-fuzzer-corpus/h264-0
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/h264-depacketizer-fuzzer-corpus/h264-1 b/third_party/libwebrtc/test/fuzzers/corpora/h264-depacketizer-fuzzer-corpus/h264-1
new file mode 100644
index 0000000000..dcb9c476f0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/h264-depacketizer-fuzzer-corpus/h264-1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/1.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/1.mdns
new file mode 100644
index 0000000000..6e1789a83b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/1.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/10.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/10.mdns
new file mode 100644
index 0000000000..f5fa6f8d17
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/10.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/11.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/11.mdns
new file mode 100644
index 0000000000..f72f28c13c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/11.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/12.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/12.mdns
new file mode 100644
index 0000000000..9efa76d64a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/12.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/13.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/13.mdns
new file mode 100644
index 0000000000..538a5a62dc
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/13.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/14.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/14.mdns
new file mode 100644
index 0000000000..d4ce4efea8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/14.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/15.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/15.mdns
new file mode 100644
index 0000000000..ff2810565b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/15.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/16.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/16.mdns
new file mode 100644
index 0000000000..a1a02d7f8e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/16.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/17.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/17.mdns
new file mode 100644
index 0000000000..ba30f7bc2f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/17.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/18.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/18.mdns
new file mode 100644
index 0000000000..7cbdd3e7a0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/18.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/19.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/19.mdns
new file mode 100644
index 0000000000..f70eaa3ab2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/19.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/2.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/2.mdns
new file mode 100644
index 0000000000..7d259c2ea9
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/2.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/20.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/20.mdns
new file mode 100644
index 0000000000..6681f943e4
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/20.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/3.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/3.mdns
new file mode 100644
index 0000000000..3ac4fd6c98
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/3.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/4.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/4.mdns
new file mode 100644
index 0000000000..3207842f48
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/4.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/5.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/5.mdns
new file mode 100644
index 0000000000..871ea41405
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/5.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/6.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/6.mdns
new file mode 100644
index 0000000000..ee2f8eca1c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/6.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/7.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/7.mdns
new file mode 100644
index 0000000000..d37b935654
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/7.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/8.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/8.mdns
new file mode 100644
index 0000000000..dd2f976afd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/8.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/9.mdns b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/9.mdns
new file mode 100644
index 0000000000..a01e729342
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/mdns-corpus/9.mdns
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/pseudotcp-corpus/785b96587d0eb44dd5d75b7a886f37e2ac504511 b/third_party/libwebrtc/test/fuzzers/corpora/pseudotcp-corpus/785b96587d0eb44dd5d75b7a886f37e2ac504511
new file mode 100644
index 0000000000..21f5cffa12
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/pseudotcp-corpus/785b96587d0eb44dd5d75b7a886f37e2ac504511
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/0.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/0.rtcp
new file mode 100644
index 0000000000..802eecf052
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/0.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/1.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/1.rtcp
new file mode 100644
index 0000000000..c062793b3c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/1.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/10.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/10.rtcp
new file mode 100644
index 0000000000..fdc7780218
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/10.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/11.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/11.rtcp
new file mode 100644
index 0000000000..994f721dca
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/11.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/12.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/12.rtcp
new file mode 100644
index 0000000000..557fe9c23a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/12.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/13.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/13.rtcp
new file mode 100644
index 0000000000..0f1b1809a7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/13.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/14.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/14.rtcp
new file mode 100644
index 0000000000..df6bfed86c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/14.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/15.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/15.rtcp
new file mode 100644
index 0000000000..3b31b3079d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/15.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/16.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/16.rtcp
new file mode 100644
index 0000000000..0496ddf910
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/16.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/17.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/17.rtcp
new file mode 100644
index 0000000000..755b7198ba
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/17.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/18.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/18.rtcp
new file mode 100644
index 0000000000..04fd1e3417
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/18.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/19.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/19.rtcp
new file mode 100644
index 0000000000..8ae9812f10
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/19.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/2.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/2.rtcp
new file mode 100644
index 0000000000..ba38a0a599
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/2.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/20.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/20.rtcp
new file mode 100644
index 0000000000..debb7de940
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/20.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/21.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/21.rtcp
new file mode 100644
index 0000000000..3fcbd405e2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/21.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/22.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/22.rtcp
new file mode 100644
index 0000000000..3defd8c253
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/22.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/23.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/23.rtcp
new file mode 100644
index 0000000000..211ccbd5be
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/23.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/24.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/24.rtcp
new file mode 100644
index 0000000000..8ded9be7fd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/24.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/25.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/25.rtcp
new file mode 100644
index 0000000000..b5c8146cef
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/25.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/26.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/26.rtcp
new file mode 100644
index 0000000000..0fd4f25511
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/26.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/27.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/27.rtcp
new file mode 100644
index 0000000000..2c8bb63c9c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/27.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/28.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/28.rtcp
new file mode 100644
index 0000000000..6a20bc27d8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/28.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/29.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/29.rtcp
new file mode 100644
index 0000000000..76fd214197
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/29.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/3.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/3.rtcp
new file mode 100644
index 0000000000..72307e08bd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/3.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/30.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/30.rtcp
new file mode 100644
index 0000000000..cfa38faa67
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/30.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/31.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/31.rtcp
new file mode 100644
index 0000000000..8abf725057
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/31.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/32.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/32.rtcp
new file mode 100644
index 0000000000..76518c88dd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/32.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/33.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/33.rtcp
new file mode 100644
index 0000000000..92964965a6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/33.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/34.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/34.rtcp
new file mode 100644
index 0000000000..9045c158b5
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/34.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/35.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/35.rtcp
new file mode 100644
index 0000000000..f86df02448
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/35.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/36.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/36.rtcp
new file mode 100644
index 0000000000..d274d416a2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/36.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/37.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/37.rtcp
new file mode 100644
index 0000000000..b4f04f4069
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/37.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/38.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/38.rtcp
new file mode 100644
index 0000000000..8d65e0ff35
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/38.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/39.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/39.rtcp
new file mode 100644
index 0000000000..fb969f85f0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/39.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/4.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/4.rtcp
new file mode 100644
index 0000000000..5170185122
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/4.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/40.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/40.rtcp
new file mode 100644
index 0000000000..279047c6b2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/40.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/41.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/41.rtcp
new file mode 100644
index 0000000000..09a8a589bd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/41.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/42.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/42.rtcp
new file mode 100644
index 0000000000..f727b190a7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/42.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/43.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/43.rtcp
new file mode 100644
index 0000000000..9aab33a48a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/43.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/44.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/44.rtcp
new file mode 100644
index 0000000000..2eba529c3a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/44.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/45.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/45.rtcp
new file mode 100644
index 0000000000..50c0beabed
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/45.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/46.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/46.rtcp
new file mode 100644
index 0000000000..2de424a5d1
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/46.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/47.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/47.rtcp
new file mode 100644
index 0000000000..71d530b41f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/47.rtcp
@@ -0,0 +1 @@
+€
\ No newline at end of file
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/48.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/48.rtcp
new file mode 100644
index 0000000000..665695029a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/48.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/49.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/49.rtcp
new file mode 100644
index 0000000000..33b0f3d873
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/49.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/5.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/5.rtcp
new file mode 100644
index 0000000000..46bef3242a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/5.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/50.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/50.rtcp
new file mode 100644
index 0000000000..5bec125397
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/50.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/51.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/51.rtcp
new file mode 100644
index 0000000000..ff94fb84b7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/51.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/52.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/52.rtcp
new file mode 100644
index 0000000000..55cc76cd81
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/52.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/53.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/53.rtcp
new file mode 100644
index 0000000000..ac11523c94
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/53.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/54.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/54.rtcp
new file mode 100644
index 0000000000..f3f0bb8fa1
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/54.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/55.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/55.rtcp
new file mode 100644
index 0000000000..e69de29bb2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/55.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/56.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/56.rtcp
new file mode 100644
index 0000000000..f0012f1efa
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/56.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/57.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/57.rtcp
new file mode 100644
index 0000000000..61517f01d8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/57.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/58.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/58.rtcp
new file mode 100644
index 0000000000..3688310a05
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/58.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/59.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/59.rtcp
new file mode 100644
index 0000000000..e914f83b16
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/59.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/6.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/6.rtcp
new file mode 100644
index 0000000000..dcad8f6b40
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/6.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/60.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/60.rtcp
new file mode 100644
index 0000000000..8e999e7832
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/60.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/61.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/61.rtcp
new file mode 100644
index 0000000000..eb38d237aa
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/61.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/62.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/62.rtcp
new file mode 100644
index 0000000000..6df94b715f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/62.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/63.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/63.rtcp
new file mode 100644
index 0000000000..a3b2acb3a8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/63.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/64.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/64.rtcp
new file mode 100644
index 0000000000..4d50c0f4ae
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/64.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/65.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/65.rtcp
new file mode 100644
index 0000000000..537d10c6ce
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/65.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/66.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/66.rtcp
new file mode 100644
index 0000000000..f280f3ca09
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/66.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/7.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/7.rtcp
new file mode 100644
index 0000000000..f8f74e12c6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/7.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/8.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/8.rtcp
new file mode 100644
index 0000000000..f0ceba83f4
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/8.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/9.rtcp b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/9.rtcp
new file mode 100644
index 0000000000..7cc07893c3
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtcp-corpus/9.rtcp
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-0 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-0
new file mode 100644
index 0000000000..c93ce82ec0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-0
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-1 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-1
new file mode 100644
index 0000000000..84834aefea
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-2 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-2
new file mode 100644
index 0000000000..3a5aedba9b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-2
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-3 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-3
new file mode 100644
index 0000000000..300309ff37
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-3
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-4 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-4
new file mode 100644
index 0000000000..84834aefea
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-4
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-5 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-5
new file mode 100644
index 0000000000..7dd6632007
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-5
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-6 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-6
new file mode 100644
index 0000000000..3a1145cd0f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-6
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-7 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-7
new file mode 100644
index 0000000000..6c4a9feb52
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-7
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-8 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-8
new file mode 100644
index 0000000000..4b069389b1
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-corpus/rtp-8
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtp-depacketizer-av1-assemble-frame-corpus/av1-assemble-frame-0 b/third_party/libwebrtc/test/fuzzers/corpora/rtp-depacketizer-av1-assemble-frame-corpus/av1-assemble-frame-0
new file mode 100644
index 0000000000..540a770e29
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtp-depacketizer-av1-assemble-frame-corpus/av1-assemble-frame-0
@@ -0,0 +1 @@
+ô0ÿÿÿ¸¸¸
\ No newline at end of file
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtpdump-corpus/vp8/vp8.rtpdump b/third_party/libwebrtc/test/fuzzers/corpora/rtpdump-corpus/vp8/vp8.rtpdump
new file mode 100644
index 0000000000..328559820d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtpdump-corpus/vp8/vp8.rtpdump
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/rtpdump-corpus/vp9/vp9.rtpdump b/third_party/libwebrtc/test/fuzzers/corpora/rtpdump-corpus/vp9/vp9.rtpdump
new file mode 100644
index 0000000000..4dc780cd16
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/rtpdump-corpus/vp9/vp9.rtpdump
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-ack-sack.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-ack-sack.bin
new file mode 100644
index 0000000000..4374f5aad5
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-ack-sack.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data-data-data.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data-data-data.bin
new file mode 100644
index 0000000000..1f1d0be301
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data-data-data.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data-data.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data-data.bin
new file mode 100644
index 0000000000..21a0c22837
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data-data.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data.bin
new file mode 100644
index 0000000000..fc8600106e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/cookie-echo-data.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/data-fragment1.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/data-fragment1.bin
new file mode 100644
index 0000000000..bec7b289e7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/data-fragment1.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/forward-tsn.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/forward-tsn.bin
new file mode 100644
index 0000000000..ab98a0a4a7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/forward-tsn.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/heartbeat-ack.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/heartbeat-ack.bin
new file mode 100644
index 0000000000..59200abe5e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/heartbeat-ack.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/heartbeat.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/heartbeat.bin
new file mode 100644
index 0000000000..cef8cfe929
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/heartbeat.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/init-ack.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/init-ack.bin
new file mode 100644
index 0000000000..80438434d0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/init-ack.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/init.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/init.bin
new file mode 100644
index 0000000000..3fb4977d58
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/init.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/re-config.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/re-config.bin
new file mode 100644
index 0000000000..74c74f3377
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/re-config.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/sack-data.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/sack-data.bin
new file mode 100644
index 0000000000..fe4de63863
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/sack-data.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/sack-gap-ack-1.bin b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/sack-gap-ack-1.bin
new file mode 100644
index 0000000000..08494c1515
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sctp-packet-corpus/sack-gap-ack-1.bin
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/10.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/10.sdp
new file mode 100644
index 0000000000..36319c4476
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/10.sdp
@@ -0,0 +1,11 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:97 iLBC/8000
+m=audio 49172 RTP/AVP 98
+a=rtpmap:98 telephone-event/8000
+a=sendonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/11.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/11.sdp
new file mode 100644
index 0000000000..c3dbce401f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/11.sdp
@@ -0,0 +1,10 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=audio 49174 RTP/AVP 98
+a=rtpmap:98 telephone-event/8000
+a=recvonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/12.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/12.sdp
new file mode 100644
index 0000000000..26d4ff6266
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/12.sdp
@@ -0,0 +1,9 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 51372 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/13.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/13.sdp
new file mode 100644
index 0000000000..f625e3890c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/13.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49174 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 49170 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/14.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/14.sdp
new file mode 100644
index 0000000000..c36b9fe6cb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/14.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 newhost.biloxi.example.com
+t=0 0
+m=audio 49178 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 49188 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/15.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/15.sdp
new file mode 100644
index 0000000000..26d4ff6266
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/15.sdp
@@ -0,0 +1,9 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 51372 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/16.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/16.sdp
new file mode 100644
index 0000000000..1d187a0dc5
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/16.sdp
@@ -0,0 +1,10 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=audio 51372 RTP/AVP 97 101
+a=rtpmap:97 iLBC/8000
+a=rtpmap:101 telephone-event/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/17.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/17.sdp
new file mode 100644
index 0000000000..a328d16a16
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/17.sdp
@@ -0,0 +1,10 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 0 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=audio 49170 RTP/AVP 97 101
+a=rtpmap:97 iLBC/8000
+a=rtpmap:101 telephone-event/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/18.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/18.sdp
new file mode 100644
index 0000000000..1e0d3f5705
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/18.sdp
@@ -0,0 +1,9 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 99
+a=rtpmap:99 iLBC/8000
+m=video 51372 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/19.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/19.sdp
new file mode 100644
index 0000000000..a9a5e12138
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/19.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 99
+a=rtpmap:99 iLBC/8000
+m=video 51374 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/2.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/2.sdp
new file mode 100644
index 0000000000..96c4975ec0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/2.sdp
@@ -0,0 +1,12 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 8 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:97 iLBC/8000
+m=video 51372 RTP/AVP 31 32
+a=rtpmap:31 H261/90000
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/20.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/20.sdp
new file mode 100644
index 0000000000..b2c072af1d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/20.sdp
@@ -0,0 +1,10 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 99
+a=rtpmap:99 iLBC/8000
+m=video 51372 RTP/AVP 31 32
+a=rtpmap:31 H261/90000
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/21.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/21.sdp
new file mode 100644
index 0000000000..28e949133e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/21.sdp
@@ -0,0 +1,10 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 99
+a=rtpmap:99 iLBC/8000
+m=video 51374 RTP/AVP 31 32
+a=rtpmap:31 H261/90000
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/22.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/22.sdp
new file mode 100644
index 0000000000..96c4975ec0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/22.sdp
@@ -0,0 +1,12 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 8 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:97 iLBC/8000
+m=video 51372 RTP/AVP 31 32
+a=rtpmap:31 H261/90000
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/23.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/23.sdp
new file mode 100644
index 0000000000..36c7d70148
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/23.sdp
@@ -0,0 +1,10 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49174 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=video 49172 RTP/AVP 32
+c=IN IP4 otherhost.biloxi.example.com
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/24.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/24.sdp
new file mode 100644
index 0000000000..6cf5dc5894
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/24.sdp
@@ -0,0 +1,8 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/25.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/25.sdp
new file mode 100644
index 0000000000..ac1ec85b29
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/25.sdp
@@ -0,0 +1,8 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 placeholder.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+a=sendonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/26.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/26.sdp
new file mode 100644
index 0000000000..f0eb0d62a7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/26.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/27.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/27.sdp
new file mode 100644
index 0000000000..1e4f3b3e19
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/27.sdp
@@ -0,0 +1,7 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49178 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/28.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/28.sdp
new file mode 100644
index 0000000000..36319c4476
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/28.sdp
@@ -0,0 +1,11 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:97 iLBC/8000
+m=audio 49172 RTP/AVP 98
+a=rtpmap:98 telephone-event/8000
+a=sendonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/29.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/29.sdp
new file mode 100644
index 0000000000..c3dbce401f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/29.sdp
@@ -0,0 +1,10 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=audio 49174 RTP/AVP 98
+a=rtpmap:98 telephone-event/8000
+a=recvonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/3.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/3.sdp
new file mode 100644
index 0000000000..b0142f5dcf
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/3.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49174 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=video 49170 RTP/AVP 32
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/30.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/30.sdp
new file mode 100644
index 0000000000..0798631d2e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/30.sdp
@@ -0,0 +1,11 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+a=sendonly
+m=audio 49174 RTP/AVP 98
+a=rtpmap:98 telephone-event/8000
+a=recvonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/31.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/31.sdp
new file mode 100644
index 0000000000..78efdd8fcb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/31.sdp
@@ -0,0 +1,11 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:97 iLBC/8000
+m=audio 49172 RTP/AVP 98
+a=rtpmap:98 telephone-event/8000
+a=sendonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/32.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/32.sdp
new file mode 100644
index 0000000000..6cf5dc5894
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/32.sdp
@@ -0,0 +1,8 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/33.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/33.sdp
new file mode 100644
index 0000000000..23f462f5a6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/33.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/34.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/34.sdp
new file mode 100644
index 0000000000..b52b947486
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/34.sdp
@@ -0,0 +1,11 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=audio 48282 RTP/AVP 98
+c=IN IP4 mediaserver.biloxi.example.com
+a=rtpmap:98 telephone-event/8000
+a=recvonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/35.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/35.sdp
new file mode 100644
index 0000000000..07ad4f8758
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/35.sdp
@@ -0,0 +1,11 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=audio 49172 RTP/AVP 98
+c=IN IP4 host.atlanta.example.com
+a=rtpmap:98 telephone-event/8000
+a=sendonly
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/36.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/36.sdp
new file mode 100644
index 0000000000..c75c6977bb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/36.sdp
@@ -0,0 +1,7 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/37.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/37.sdp
new file mode 100644
index 0000000000..d05dbd61fd
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/37.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/38.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/38.sdp
new file mode 100644
index 0000000000..253ac0705a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/38.sdp
@@ -0,0 +1,9 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=video 49172 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/39.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/39.sdp
new file mode 100644
index 0000000000..57d3203c74
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/39.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=video 49168 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/4.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/4.sdp
new file mode 100644
index 0000000000..96c4975ec0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/4.sdp
@@ -0,0 +1,12 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 8 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:97 iLBC/8000
+m=video 51372 RTP/AVP 31 32
+a=rtpmap:31 H261/90000
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/40.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/40.sdp
new file mode 100644
index 0000000000..26d4ff6266
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/40.sdp
@@ -0,0 +1,9 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 51372 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/41.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/41.sdp
new file mode 100644
index 0000000000..f625e3890c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/41.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49174 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 49170 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/42.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/42.sdp
new file mode 100644
index 0000000000..2dda9bb252
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/42.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49174 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 0 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/43.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/43.sdp
new file mode 100644
index 0000000000..83a2bf9e66
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/43.sdp
@@ -0,0 +1,9 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+m=video 0 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/44.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/44.sdp
new file mode 100644
index 0000000000..e187d7df7d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/44.sdp
@@ -0,0 +1,5 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/45.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/45.sdp
new file mode 100644
index 0000000000..9e344f035a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/45.sdp
@@ -0,0 +1,5 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/46.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/46.sdp
new file mode 100644
index 0000000000..8a63ca0992
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/46.sdp
@@ -0,0 +1,7 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/47.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/47.sdp
new file mode 100644
index 0000000000..66dadbb93f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/47.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/48.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/48.sdp
new file mode 100644
index 0000000000..6df76ab942
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/48.sdp
@@ -0,0 +1,7 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 0.0.0.0
+t=0 0
+m=audio 23442 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/49.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/49.sdp
new file mode 100644
index 0000000000..23f462f5a6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/49.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/5.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/5.sdp
new file mode 100644
index 0000000000..264ab8024f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/5.sdp
@@ -0,0 +1,10 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 0 8
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+m=video 0 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/50.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/50.sdp
new file mode 100644
index 0000000000..8a63ca0992
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/50.sdp
@@ -0,0 +1,7 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/51.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/51.sdp
new file mode 100644
index 0000000000..23f462f5a6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/51.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/52.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/52.sdp
new file mode 100644
index 0000000000..7c1730eb1e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/52.sdp
@@ -0,0 +1,7 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/53.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/53.sdp
new file mode 100644
index 0000000000..599157691a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/53.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 0.0.0.0
+t=0 0
+m=audio 9322 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/54.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/54.sdp
new file mode 100644
index 0000000000..66dadbb93f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/54.sdp
@@ -0,0 +1,7 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/55.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/55.sdp
new file mode 100644
index 0000000000..bd724c471b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/55.sdp
@@ -0,0 +1,8 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 97
+a=rtpmap:97 iLBC/8000
+
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/6.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/6.sdp
new file mode 100644
index 0000000000..3f80345e30
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/6.sdp
@@ -0,0 +1,9 @@
+v=0
+o=alice 2890844526 2890844527 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 51372 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=video 0 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/7.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/7.sdp
new file mode 100644
index 0000000000..920ad8229e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/7.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844565 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 0
+a=rtpmap:0 PCMU/8000
+m=video 0 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/8.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/8.sdp
new file mode 100644
index 0000000000..96c4975ec0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/8.sdp
@@ -0,0 +1,12 @@
+v=0
+o=alice 2890844526 2890844526 IN IP4 host.atlanta.example.com
+s=
+c=IN IP4 host.atlanta.example.com
+t=0 0
+m=audio 49170 RTP/AVP 0 8 97
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:97 iLBC/8000
+m=video 51372 RTP/AVP 31 32
+a=rtpmap:31 H261/90000
+a=rtpmap:32 MPV/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/9.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/9.sdp
new file mode 100644
index 0000000000..a9a5e12138
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/9.sdp
@@ -0,0 +1,9 @@
+v=0
+o=bob 2808844564 2808844564 IN IP4 host.biloxi.example.com
+s=
+c=IN IP4 host.biloxi.example.com
+t=0 0
+m=audio 49172 RTP/AVP 99
+a=rtpmap:99 iLBC/8000
+m=video 51374 RTP/AVP 31
+a=rtpmap:31 H261/90000
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/firefox-1.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/firefox-1.sdp
new file mode 100644
index 0000000000..6d664bc051
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/firefox-1.sdp
@@ -0,0 +1,58 @@
+v=0
+o=mozilla...THIS_IS_SDPARTA-46.0.1 5115930144083302970 0 IN IP4 0.0.0.0
+s=-
+t=0 0
+a=fingerprint:sha-256 24:67:5E:1B:9A:B9:CF:36:C5:30:8F:35:F7:B1:50:66:88:81:92:CB:29:BA:53:A5:02:C8:0A:A5:4E:9C:AE:D9
+a=group:BUNDLE sdparta_0 sdparta_1 sdparta_2
+a=ice-options:trickle
+a=msid-semantic:WMS *
+m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8
+c=IN IP4 0.0.0.0
+a=sendrecv
+a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
+a=fmtp:109 maxplaybackrate=48000;stereo=1
+a=ice-pwd:b9f3911b591ae61e2d7f6af0531fd2a3
+a=ice-ufrag:3edc9012
+a=mid:sdparta_0
+a=msid:{258e92fb-547c-40ca-92e9-efe0cedb4cba} {bd1fafff-bfd0-40d4-b0a3-2a87cff307ee}
+a=rtcp-mux
+a=rtpmap:109 opus/48000/2
+a=rtpmap:9 G722/8000/1
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=setup:actpass
+a=ssrc:2121669360 cname:{387b0735-bde2-43a4-8484-7f5663b60f24}
+m=video 9 UDP/TLS/RTP/SAVPF 120 126 97
+c=IN IP4 0.0.0.0
+a=sendrecv
+a=fmtp:126 profile-level-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1
+a=fmtp:97 profile-level-id=42e01f;level-asymmetry-allowed=1
+a=fmtp:120 max-fs=12288;max-fr=60
+a=ice-pwd:b9f3911b591ae61e2d7f6af0531fd2a3
+a=ice-ufrag:3edc9012
+a=mid:sdparta_1
+a=msid:{258e92fb-547c-40ca-92e9-efe0cedb4cba} {9e8f5867-a9aa-4489-8bd4-3a8a57a5e592}
+a=rtcp-fb:120 nack
+a=rtcp-fb:120 nack pli
+a=rtcp-fb:120 ccm fir
+a=rtcp-fb:126 nack
+a=rtcp-fb:126 nack pli
+a=rtcp-fb:126 ccm fir
+a=rtcp-fb:97 nack
+a=rtcp-fb:97 nack pli
+a=rtcp-fb:97 ccm fir
+a=rtcp-mux
+a=rtpmap:120 VP8/90000
+a=rtpmap:126 H264/90000
+a=rtpmap:97 H264/90000
+a=setup:actpass
+a=ssrc:2158832026 cname:{387b0735-bde2-43a4-8484-7f5663b60f24}
+m=application 9 DTLS/SCTP 5000
+c=IN IP4 0.0.0.0
+a=sendrecv
+a=ice-pwd:b9f3911b591ae61e2d7f6af0531fd2a3
+a=ice-ufrag:3edc9012
+a=mid:sdparta_2
+a=sctpmap:5000 webrtc-datachannel 256
+a=setup:actpass
+a=ssrc:2670959794 cname:{387b0735-bde2-43a4-8484-7f5663b60f24}
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/firefox-2.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/firefox-2.sdp
new file mode 100644
index 0000000000..07806eb225
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/firefox-2.sdp
@@ -0,0 +1,43 @@
+v=0
+o=mozilla...THIS_IS_SDPARTA-46.0.1 3068771576687940834 0 IN IP4 0.0.0.0
+s=-
+t=0 0
+a=fingerprint:sha-256 AD:87:B3:11:E4:E2:BA:EF:D2:3F:2E:AC:24:57:8E:DC:1F:67:41:29:44:C4:96:E3:62:90:CC:90:59:CA:2C:84
+a=group:BUNDLE sdparta_0 sdparta_1 sdparta_2
+a=ice-options:trickle
+a=msid-semantic:WMS *
+m=audio 9 UDP/TLS/RTP/SAVPF 109
+c=IN IP4 0.0.0.0
+a=recvonly
+a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
+a=fmtp:109 maxplaybackrate=48000;stereo=1
+a=ice-pwd:ff4c4dc6fe92e1f22d2c10352d8967d5
+a=ice-ufrag:a539544b
+a=mid:sdparta_0
+a=rtcp-mux
+a=rtpmap:109 opus/48000/2
+a=setup:active
+a=ssrc:600995474 cname:{5b598a29-a81b-4ffe-a2c5-507778057e7a}
+m=video 9 UDP/TLS/RTP/SAVPF 120
+c=IN IP4 0.0.0.0
+a=recvonly
+a=fmtp:120 max-fs=12288;max-fr=60
+a=ice-pwd:ff4c4dc6fe92e1f22d2c10352d8967d5
+a=ice-ufrag:a539544b
+a=mid:sdparta_1
+a=rtcp-fb:120 nack
+a=rtcp-fb:120 nack pli
+a=rtcp-fb:120 ccm fir
+a=rtcp-mux
+a=rtpmap:120 VP8/90000
+a=setup:active
+a=ssrc:3480150809 cname:{5b598a29-a81b-4ffe-a2c5-507778057e7a}
+m=application 9 DTLS/SCTP 5000
+c=IN IP4 0.0.0.0
+a=sendrecv
+a=ice-pwd:ff4c4dc6fe92e1f22d2c10352d8967d5
+a=ice-ufrag:a539544b
+a=mid:sdparta_2
+a=sctpmap:5000 webrtc-datachannel 256
+a=setup:active
+a=ssrc:3021788991 cname:{5b598a29-a81b-4ffe-a2c5-507778057e7a}
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/opera-1.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/opera-1.sdp
new file mode 100644
index 0000000000..42d15e81f4
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/opera-1.sdp
@@ -0,0 +1,85 @@
+v=0
+o=- 1656229333038673902 2 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=group:BUNDLE audio video data
+a=msid-semantic: WMS Ppsa09YmDLBombOh5e8HqfqxEIPF69a46Hd4
+m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:1Jyk4q3nLIL5NiMx
+a=ice-pwd:GL8/iarMqPIhImfnsG2dyXlH
+a=fingerprint:sha-256 5A:16:96:94:B2:AC:60:27:64:C5:FE:46:6C:02:C0:CD:49:E3:E2:0B:5B:C9:D4:86:C4:B3:A4:F2:23:80:7A:DA
+a=setup:actpass
+a=mid:audio
+a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
+a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+a=sendrecv
+a=rtcp-mux
+a=rtpmap:111 opus/48000/2
+a=rtcp-fb:111 transport-cc
+a=fmtp:111 minptime=10; useinbandfec=1
+a=rtpmap:103 ISAC/16000
+a=rtpmap:104 ISAC/32000
+a=rtpmap:9 G722/8000
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:106 CN/32000
+a=rtpmap:105 CN/16000
+a=rtpmap:13 CN/8000
+a=rtpmap:126 telephone-event/8000
+a=maxptime:60
+a=ssrc:2233075910 cname:VhHMGYCjn4alR9zP
+a=ssrc:2233075910 msid:Ppsa09YmDLBombOh5e8HqfqxEIPF69a46Hd4 689d3496-0896-4d52-bce6-8e90512a368b
+a=ssrc:2233075910 mslabel:Ppsa09YmDLBombOh5e8HqfqxEIPF69a46Hd4
+a=ssrc:2233075910 label:689d3496-0896-4d52-bce6-8e90512a368b
+m=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:1Jyk4q3nLIL5NiMx
+a=ice-pwd:GL8/iarMqPIhImfnsG2dyXlH
+a=fingerprint:sha-256 5A:16:96:94:B2:AC:60:27:64:C5:FE:46:6C:02:C0:CD:49:E3:E2:0B:5B:C9:D4:86:C4:B3:A4:F2:23:80:7A:DA
+a=setup:actpass
+a=mid:video
+a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
+a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+a=extmap:4 urn:3gpp:video-orientation
+a=sendrecv
+a=rtcp-mux
+a=rtpmap:100 VP8/90000
+a=rtcp-fb:100 ccm fir
+a=rtcp-fb:100 nack
+a=rtcp-fb:100 nack pli
+a=rtcp-fb:100 goog-remb
+a=rtcp-fb:100 transport-cc
+a=rtpmap:101 VP9/90000
+a=rtcp-fb:101 ccm fir
+a=rtcp-fb:101 nack
+a=rtcp-fb:101 nack pli
+a=rtcp-fb:101 goog-remb
+a=rtcp-fb:101 transport-cc
+a=rtpmap:116 red/90000
+a=rtpmap:117 ulpfec/90000
+a=rtpmap:96 rtx/90000
+a=fmtp:96 apt=100
+a=rtpmap:97 rtx/90000
+a=fmtp:97 apt=101
+a=rtpmap:98 rtx/90000
+a=fmtp:98 apt=116
+a=ssrc-group:FID 50498894 2399294607
+a=ssrc:50498894 cname:VhHMGYCjn4alR9zP
+a=ssrc:50498894 msid:Ppsa09YmDLBombOh5e8HqfqxEIPF69a46Hd4 1aef96f4-fc4c-4f86-98f3-0fbf4f625f70
+a=ssrc:50498894 mslabel:Ppsa09YmDLBombOh5e8HqfqxEIPF69a46Hd4
+a=ssrc:50498894 label:1aef96f4-fc4c-4f86-98f3-0fbf4f625f70
+a=ssrc:2399294607 cname:VhHMGYCjn4alR9zP
+a=ssrc:2399294607 msid:Ppsa09YmDLBombOh5e8HqfqxEIPF69a46Hd4 1aef96f4-fc4c-4f86-98f3-0fbf4f625f70
+a=ssrc:2399294607 mslabel:Ppsa09YmDLBombOh5e8HqfqxEIPF69a46Hd4
+a=ssrc:2399294607 label:1aef96f4-fc4c-4f86-98f3-0fbf4f625f70
+m=application 9 DTLS/SCTP 5000
+c=IN IP4 0.0.0.0
+a=ice-ufrag:1Jyk4q3nLIL5NiMx
+a=ice-pwd:GL8/iarMqPIhImfnsG2dyXlH
+a=fingerprint:sha-256 5A:16:96:94:B2:AC:60:27:64:C5:FE:46:6C:02:C0:CD:49:E3:E2:0B:5B:C9:D4:86:C4:B3:A4:F2:23:80:7A:DA
+a=setup:actpass
+a=mid:data
+a=sctpmap:5000 webrtc-datachannel 1024
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/opera-2.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/opera-2.sdp
new file mode 100644
index 0000000000..6ed4e3b4a8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/opera-2.sdp
@@ -0,0 +1,73 @@
+v=0
+o=- 2013283641453412290 2 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=group:BUNDLE audio video data
+a=msid-semantic: WMS
+m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:YVa3KTlFDCwsfPOQ
+a=ice-pwd:ByUn1Od88VokVM0rtQ/bbeZa
+a=fingerprint:sha-256 5A:16:96:94:B2:AC:60:27:64:C5:FE:46:6C:02:C0:CD:49:E3:E2:0B:5B:C9:D4:86:C4:B3:A4:F2:23:80:7A:DA
+a=setup:active
+a=mid:audio
+a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
+a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+a=recvonly
+a=rtcp-mux
+a=rtpmap:111 opus/48000/2
+a=rtcp-fb:111 transport-cc
+a=fmtp:111 minptime=10; useinbandfec=1
+a=rtpmap:103 ISAC/16000
+a=rtpmap:104 ISAC/32000
+a=rtpmap:9 G722/8000
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:106 CN/32000
+a=rtpmap:105 CN/16000
+a=rtpmap:13 CN/8000
+a=rtpmap:126 telephone-event/8000
+a=maxptime:60
+m=video 9 UDP/TLS/RTP/SAVPF 100 101 116 117 96 97 98
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:YVa3KTlFDCwsfPOQ
+a=ice-pwd:ByUn1Od88VokVM0rtQ/bbeZa
+a=fingerprint:sha-256 5A:16:96:94:B2:AC:60:27:64:C5:FE:46:6C:02:C0:CD:49:E3:E2:0B:5B:C9:D4:86:C4:B3:A4:F2:23:80:7A:DA
+a=setup:active
+a=mid:video
+a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
+a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+a=extmap:4 urn:3gpp:video-orientation
+a=recvonly
+a=rtcp-mux
+a=rtpmap:100 VP8/90000
+a=rtcp-fb:100 ccm fir
+a=rtcp-fb:100 nack
+a=rtcp-fb:100 nack pli
+a=rtcp-fb:100 goog-remb
+a=rtcp-fb:100 transport-cc
+a=rtpmap:101 VP9/90000
+a=rtcp-fb:101 ccm fir
+a=rtcp-fb:101 nack
+a=rtcp-fb:101 nack pli
+a=rtcp-fb:101 goog-remb
+a=rtcp-fb:101 transport-cc
+a=rtpmap:116 red/90000
+a=rtpmap:117 ulpfec/90000
+a=rtpmap:96 rtx/90000
+a=fmtp:96 apt=100
+a=rtpmap:97 rtx/90000
+a=fmtp:97 apt=101
+a=rtpmap:98 rtx/90000
+a=fmtp:98 apt=116
+m=application 9 DTLS/SCTP 5000
+c=IN IP4 0.0.0.0
+b=AS:30
+a=ice-ufrag:YVa3KTlFDCwsfPOQ
+a=ice-pwd:ByUn1Od88VokVM0rtQ/bbeZa
+a=fingerprint:sha-256 5A:16:96:94:B2:AC:60:27:64:C5:FE:46:6C:02:C0:CD:49:E3:E2:0B:5B:C9:D4:86:C4:B3:A4:F2:23:80:7A:DA
+a=setup:active
+a=mid:data
+a=sctpmap:5000 webrtc-datachannel 1024
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/simulcast.1.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/simulcast.1.sdp
new file mode 100644
index 0000000000..e7a219812b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/simulcast.1.sdp
@@ -0,0 +1,41 @@
+v=0
+o=- 367669362084170381 2 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=group:BUNDLE 0
+a=msid-semantic: WMS VfhSdt9LWGwoduWpoASvxGyAGEQFAkQe1hT1
+m=video 9 UDP/TLS/RTP/SAVPF 96
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:OLwt
+a=ice-pwd:kjGBqIFYs8UqCyfmJ7nEJw/Q
+a=ice-options:trickle
+a=fingerprint:sha-256 B4:AB:3E:01:90:D1:FC:22:8F:05:6A:01:76:B6:C5:0D:45:88:1A:E1:99:AF:18:43:E6:18:BB:D0:92:51:9B:0E
+a=setup:actpass
+a=mid:0
+a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
+a=extmap:13 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+a=extmap:12 urn:3gpp:video-orientation
+a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
+a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
+a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
+a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
+a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
+a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
+a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
+a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
+a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
+a=sendrecv
+a=msid:VfhSdt9LWGwoduWpoASvxGyAGEQFAkQe1hT1 a8c06601-e9ed-4312-a7d4-283e078c5966
+a=rtcp-mux
+a=rtcp-rsize
+a=rtpmap:96 VP8/90000
+a=rtcp-fb:96 goog-remb
+a=rtcp-fb:96 transport-cc
+a=rtcp-fb:96 ccm fir
+a=rtcp-fb:96 nack
+a=rtcp-fb:96 nack pli
+a=rid:f send
+a=rid:h send
+a=rid:q send
+a=simulcast:send f;h;q
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/simulcast.2.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/simulcast.2.sdp
new file mode 100644
index 0000000000..52879f9e66
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/simulcast.2.sdp
@@ -0,0 +1,41 @@
+v=0
+o=- 367669362084170381 2 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=group:BUNDLE 0
+a=msid-semantic: WMS VfhSdt9LWGwoduWpoASvxGyAGEQFAkQe1hT1
+m=video 9 UDP/TLS/RTP/SAVPF 96
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:OLwt
+a=ice-pwd:kjGBqIFYs8UqCyfmJ7nEJw/Q
+a=ice-options:trickle
+a=fingerprint:sha-256 B4:AB:3E:01:90:D1:FC:22:8F:05:6A:01:76:B6:C5:0D:45:88:1A:E1:99:AF:18:43:E6:18:BB:D0:92:51:9B:0E
+a=setup:actpass
+a=mid:0
+a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
+a=extmap:13 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+a=extmap:12 urn:3gpp:video-orientation
+a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
+a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
+a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
+a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
+a=extmap:8 http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07
+a=extmap:9 http://www.webrtc.org/experiments/rtp-hdrext/color-space
+a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
+a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
+a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
+a=sendrecv
+a=msid:VfhSdt9LWGwoduWpoASvxGyAGEQFAkQe1hT1 a8c06601-e9ed-4312-a7d4-283e078c5966
+a=rtcp-mux
+a=rtcp-rsize
+a=rtpmap:96 VP8/90000
+a=rtcp-fb:96 goog-remb
+a=rtcp-fb:96 transport-cc
+a=rtcp-fb:96 ccm fir
+a=rtcp-fb:96 nack
+a=rtcp-fb:96 nack pli
+a=rid:f recv
+a=rid:h recv
+a=rid:q recv
+a=simulcast:recv f;h;q
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-1.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-1.sdp
new file mode 100644
index 0000000000..ce349ac30f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-1.sdp
@@ -0,0 +1,52 @@
+v=0
+o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=msid-semantic: WMS local_stream_1
+m=audio 2345 RTP/SAVPF 111 103 104
+c=IN IP4 74.125.127.126
+a=rtcp:2347 IN IP4 74.125.127.126
+a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1234 typ host generation 2
+a=candidate:a0+B/1 2 udp 2130706432 192.168.1.5 1235 typ host generation 2
+a=candidate:a0+B/2 1 udp 2130706432 ::1 1238 typ host generation 2
+a=candidate:a0+B/2 2 udp 2130706432 ::1 1239 typ host generation 2
+a=candidate:a0+B/3 1 udp 2130706432 74.125.127.126 2345 typ srflx raddr 192.168.1.5 rport 2346 generation 2
+a=candidate:a0+B/3 2 udp 2130706432 74.125.127.126 2347 typ srflx raddr 192.168.1.5 rport 2348 generation 2
+a=ice-ufrag:ufrag_voice
+a=ice-pwd:pwd_voice
+a=mid:audio_content_name
+a=sendrecv
+a=rtcp-mux
+a=rtcp-rsize
+a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 dummy_session_params
+a=rtpmap:111 opus/48000/2
+a=rtpmap:103 ISAC/16000
+a=rtpmap:104 ISAC/32000
+a=ssrc:1 cname:stream_1_cname
+a=ssrc:1 msid:local_stream_1 audio_track_id_1
+a=ssrc:1 mslabel:local_stream_1
+a=ssrc:1 label:audio_track_id_1
+m=video 3457 RTP/SAVPF 120
+c=IN IP4 74.125.224.39
+a=rtcp:3456 IN IP4 74.125.224.39
+a=candidate:a0+B/1 2 udp 2130706432 192.168.1.5 1236 typ host generation 2
+a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1237 typ host generation 2
+a=candidate:a0+B/2 2 udp 2130706432 ::1 1240 typ host generation 2
+a=candidate:a0+B/2 1 udp 2130706432 ::1 1241 typ host generation 2
+a=candidate:a0+B/4 2 udp 2130706432 74.125.224.39 3456 typ relay generation 2
+a=candidate:a0+B/4 1 udp 2130706432 74.125.224.39 3457 typ relay generation 2
+a=ice-ufrag:ufrag_video
+a=ice-pwd:pwd_video
+a=mid:video_content_name
+a=sendrecv
+a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32
+a=rtpmap:120 VP8/90000
+a=ssrc-group:FEC 2 3
+a=ssrc:2 cname:stream_1_cname
+a=ssrc:2 msid:local_stream_1 video_track_id_1
+a=ssrc:2 mslabel:local_stream_1
+a=ssrc:2 label:video_track_id_1
+a=ssrc:3 cname:stream_1_cname
+a=ssrc:3 msid:local_stream_1 video_track_id_1
+a=ssrc:3 mslabel:local_stream_1
+a=ssrc:3 label:video_track_id_1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-2.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-2.sdp
new file mode 100644
index 0000000000..a35392513f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-2.sdp
@@ -0,0 +1,40 @@
+v=0
+o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=msid-semantic: WMS local_stream_1
+m=audio 9 RTP/SAVPF 111 103 104
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:ufrag_voice
+a=ice-pwd:pwd_voice
+a=mid:audio_content_name
+a=sendrecv
+a=rtcp-mux
+a=rtcp-rsize
+a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 dummy_session_params
+a=rtpmap:111 opus/48000/2
+a=rtpmap:103 ISAC/16000
+a=rtpmap:104 ISAC/32000
+a=ssrc:1 cname:stream_1_cname
+a=ssrc:1 msid:local_stream_1 audio_track_id_1
+a=ssrc:1 mslabel:local_stream_1
+a=ssrc:1 label:audio_track_id_1
+m=video 9 RTP/SAVPF 120
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:ufrag_video
+a=ice-pwd:pwd_video
+a=mid:video_content_name
+a=sendrecv
+a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32
+a=rtpmap:120 VP8/90000
+a=ssrc-group:FEC 2 3
+a=ssrc:2 cname:stream_1_cname
+a=ssrc:2 msid:local_stream_1 video_track_id_1
+a=ssrc:2 mslabel:local_stream_1
+a=ssrc:2 label:video_track_id_1
+a=ssrc:3 cname:stream_1_cname
+a=ssrc:3 msid:local_stream_1 video_track_id_1
+a=ssrc:3 mslabel:local_stream_1
+a=ssrc:3 label:video_track_id_1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-3.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-3.sdp
new file mode 100644
index 0000000000..50f8e55ef7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-3.sdp
@@ -0,0 +1,13 @@
+m=application 9 RTP/SAVPF 101
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:ufrag_data
+a=ice-pwd:pwd_data
+a=mid:data_content_name
+a=sendrecv
+a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5
+a=rtpmap:101 google-data/90000
+a=ssrc:10 cname:data_channel_cname
+a=ssrc:10 msid:data_channel data_channeld0
+a=ssrc:10 mslabel:data_channel
+a=ssrc:10 label:data_channeld0
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-4.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-4.sdp
new file mode 100644
index 0000000000..709db42c00
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-4.sdp
@@ -0,0 +1,11 @@
+v=0
+o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=msid-semantic: WMS
+m=audio 9 RTP/SAVPF 111 103 104
+c=IN IP4 0.0.0.0
+a=x-google-flag:conference
+m=video 9 RTP/SAVPF 120
+c=IN IP4 0.0.0.0
+a=x-google-flag:conference
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-5.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-5.sdp
new file mode 100644
index 0000000000..c440386062
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-5.sdp
@@ -0,0 +1,5 @@
+v=0
+o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=msid-semantic: WMS local_stream
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-6.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-6.sdp
new file mode 100644
index 0000000000..e07b5647bf
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-6.sdp
@@ -0,0 +1,12 @@
+m=audio 9 RTP/SAVPF 111
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:ufrag_voice
+a=ice-pwd:pwd_voice
+a=mid:audio_content_name
+a=sendrecv
+a=rtpmap:111 opus/48000/2
+a=ssrc:1 cname:stream_1_cname
+a=ssrc:1 msid:local_stream audio_track_id_1
+a=ssrc:1 mslabel:local_stream
+a=ssrc:1 label:audio_track_id_1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-7.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-7.sdp
new file mode 100644
index 0000000000..8bdacc2baa
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-7.sdp
@@ -0,0 +1,12 @@
+m=video 9 RTP/SAVPF 120
+c=IN IP4 0.0.0.0
+a=rtcp:9 IN IP4 0.0.0.0
+a=ice-ufrag:ufrag_video
+a=ice-pwd:pwd_video
+a=mid:video_content_name
+a=sendrecv
+a=rtpmap:120 VP8/90000
+a=ssrc:2 cname:stream_1_cname
+a=ssrc:2 msid:local_stream video_track_id_1
+a=ssrc:2 mslabel:local_stream
+a=ssrc:2 label:video_track_id_1
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-8.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-8.sdp
new file mode 100644
index 0000000000..9688fc46a2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-8.sdp
@@ -0,0 +1,64 @@
+v=0
+o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=msid-semantic: WMS local_stream_1 local_stream_2
+m=audio 2345 RTP/SAVPF 111 103 104
+c=IN IP4 74.125.127.126
+a=rtcp:2347 IN IP4 74.125.127.126
+a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1234 typ host generation 2
+a=candidate:a0+B/1 2 udp 2130706432 192.168.1.5 1235 typ host generation 2
+a=candidate:a0+B/2 1 udp 2130706432 ::1 1238 typ host generation 2
+a=candidate:a0+B/2 2 udp 2130706432 ::1 1239 typ host generation 2
+a=candidate:a0+B/3 1 udp 2130706432 74.125.127.126 2345 typ srflx raddr 192.168.1.5 rport 2346 generation 2
+a=candidate:a0+B/3 2 udp 2130706432 74.125.127.126 2347 typ srflx raddr 192.168.1.5 rport 2348 generation 2
+a=ice-ufrag:ufrag_voice
+a=ice-pwd:pwd_voice
+a=mid:audio_content_name
+a=sendrecv
+a=rtcp-mux
+a=rtcp-rsize
+a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 dummy_session_params
+a=rtpmap:111 opus/48000/2
+a=rtpmap:103 ISAC/16000
+a=rtpmap:104 ISAC/32000
+a=ssrc:1 cname:stream_1_cname
+a=ssrc:1 msid:local_stream_1 audio_track_id_1
+a=ssrc:1 mslabel:local_stream_1
+a=ssrc:1 label:audio_track_id_1
+a=ssrc:4 cname:stream_2_cname
+a=ssrc:4 msid:local_stream_2 audio_track_id_2
+a=ssrc:4 mslabel:local_stream_2
+a=ssrc:4 label:audio_track_id_2
+m=video 3457 RTP/SAVPF 120
+c=IN IP4 74.125.224.39
+a=rtcp:3456 IN IP4 74.125.224.39
+a=candidate:a0+B/1 2 udp 2130706432 192.168.1.5 1236 typ host generation 2
+a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1237 typ host generation 2
+a=candidate:a0+B/2 2 udp 2130706432 ::1 1240 typ host generation 2
+a=candidate:a0+B/2 1 udp 2130706432 ::1 1241 typ host generation 2
+a=candidate:a0+B/4 2 udp 2130706432 74.125.224.39 3456 typ relay generation 2
+a=candidate:a0+B/4 1 udp 2130706432 74.125.224.39 3457 typ relay generation 2
+a=ice-ufrag:ufrag_video
+a=ice-pwd:pwd_video
+a=mid:video_content_name
+a=sendrecv
+a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32
+a=rtpmap:120 VP8/90000
+a=ssrc-group:FEC 2 3
+a=ssrc:2 cname:stream_1_cname
+a=ssrc:2 msid:local_stream_1 video_track_id_1
+a=ssrc:2 mslabel:local_stream_1
+a=ssrc:2 label:video_track_id_1
+a=ssrc:3 cname:stream_1_cname
+a=ssrc:3 msid:local_stream_1 video_track_id_1
+a=ssrc:3 mslabel:local_stream_1
+a=ssrc:3 label:video_track_id_1
+a=ssrc:5 cname:stream_2_cname
+a=ssrc:5 msid:local_stream_2 video_track_id_2
+a=ssrc:5 mslabel:local_stream_2
+a=ssrc:5 label:video_track_id_2
+a=ssrc:6 cname:stream_2_cname
+a=ssrc:6 msid:local_stream_2 video_track_id_3
+a=ssrc:6 mslabel:local_stream_2
+a=ssrc:6 label:video_track_id_3
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-9.sdp b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-9.sdp
new file mode 100644
index 0000000000..8c7c4c7c3d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp-corpus/unittest-9.sdp
@@ -0,0 +1,66 @@
+v=0
+o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1
+s=-
+t=0 0
+a=msid-semantic: WMS local_stream_1 local_stream_2
+m=audio 2345 RTP/SAVPF 111 103 104
+c=IN IP4 74.125.127.126
+a=rtcp:2347 IN IP4 74.125.127.126
+a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1234 typ host generation 2
+a=candidate:a0+B/1 2 udp 2130706432 192.168.1.5 1235 typ host generation 2
+a=candidate:a0+B/2 1 udp 2130706432 ::1 1238 typ host generation 2
+a=candidate:a0+B/2 2 udp 2130706432 ::1 1239 typ host generation 2
+a=candidate:a0+B/3 1 udp 2130706432 74.125.127.126 2345 typ srflx raddr 192.168.1.5 rport 2346 generation 2
+a=candidate:a0+B/3 2 udp 2130706432 74.125.127.126 2347 typ srflx raddr 192.168.1.5 rport 2348 generation 2
+a=ice-ufrag:ufrag_voice
+a=ice-pwd:pwd_voice
+a=mid:audio_content_name
+a=msid:local_stream_1 audio_track_id_1
+a=sendrecv
+a=rtcp-mux
+a=rtcp-rsize
+a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 dummy_session_params
+a=rtpmap:111 opus/48000/2
+a=rtpmap:103 ISAC/16000
+a=rtpmap:104 ISAC/32000
+a=ssrc:1 cname:stream_1_cname
+a=ssrc:1 msid:local_stream_1 audio_track_id_1
+a=ssrc:1 mslabel:local_stream_1
+a=ssrc:1 label:audio_track_id_1
+a=ssrc:4 cname:stream_2_cname
+a=ssrc:4 msid:local_stream_2 audio_track_id_2
+a=ssrc:4 mslabel:local_stream_2
+a=ssrc:4 label:audio_track_id_2
+m=video 3457 RTP/SAVPF 120
+c=IN IP4 74.125.224.39
+a=rtcp:3456 IN IP4 74.125.224.39
+a=candidate:a0+B/1 2 udp 2130706432 192.168.1.5 1236 typ host generation 2
+a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1237 typ host generation 2
+a=candidate:a0+B/2 2 udp 2130706432 ::1 1240 typ host generation 2
+a=candidate:a0+B/2 1 udp 2130706432 ::1 1241 typ host generation 2
+a=candidate:a0+B/4 2 udp 2130706432 74.125.224.39 3456 typ relay generation 2
+a=candidate:a0+B/4 1 udp 2130706432 74.125.224.39 3457 typ relay generation 2
+a=ice-ufrag:ufrag_video
+a=ice-pwd:pwd_video
+a=mid:video_content_name
+a=msid:local_stream_1 video_track_id_1
+a=sendrecv
+a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32
+a=rtpmap:120 VP8/90000
+a=ssrc-group:FEC 2 3
+a=ssrc:2 cname:stream_1_cname
+a=ssrc:2 msid:local_stream_1 video_track_id_1
+a=ssrc:2 mslabel:local_stream_1
+a=ssrc:2 label:video_track_id_1
+a=ssrc:3 cname:stream_1_cname
+a=ssrc:3 msid:local_stream_1 video_track_id_1
+a=ssrc:3 mslabel:local_stream_1
+a=ssrc:3 label:video_track_id_1
+a=ssrc:5 cname:stream_2_cname
+a=ssrc:5 msid:local_stream_2 video_track_id_2
+a=ssrc:5 mslabel:local_stream_2
+a=ssrc:5 label:video_track_id_2
+a=ssrc:6 cname:stream_2_cname
+a=ssrc:6 msid:local_stream_2 video_track_id_3
+a=ssrc:6 mslabel:local_stream_2
+a=ssrc:6 label:video_track_id_3
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/sdp.tokens b/third_party/libwebrtc/test/fuzzers/corpora/sdp.tokens
new file mode 100644
index 0000000000..ddfdad5811
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/sdp.tokens
@@ -0,0 +1,56 @@
+"0.0.0.0"
+"127.0.0.1"
+"application"
+"audio"
+"candidate"
+"cname"
+"conference"
+"crypto"
+"default"
+"extmap"
+"fingerprint"
+"fmtp"
+"generation"
+"group"
+"host"
+"ice-lite"
+"ice-options"
+"ice-pwd"
+"ice-ufrag"
+"inactive"
+"IP4"
+"IP6"
+"ISAC"
+"label"
+"mid"
+"msid"
+"msid-semantic"
+"mslabel"
+"network-cost"
+"network-id"
+"prflx"
+"pwd"
+"raddr"
+"recvonly"
+"relay"
+"rport"
+"rtcp"
+"rtcp-fb"
+"rtcp-mux"
+"rtcp-rsize"
+"rtpmap"
+"sctpmap"
+"sctp-port"
+"sendonly"
+"sendrecv"
+"setup"
+"srflx"
+"ssrc"
+"ssrc-group"
+"tcptype"
+"typ"
+"ufrag"
+"video"
+"webrtc-datachannel"
+"WMS"
+"x-google-flag"
\ No newline at end of file
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/0 b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/0
new file mode 100644
index 0000000000..fbfcbf9105
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/0
@@ -0,0 +1 @@
+90742757
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/1 b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/1
new file mode 100644
index 0000000000..587611a1d6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/1
@@ -0,0 +1 @@
+27.70727677
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/2 b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/2
new file mode 100644
index 0000000000..45a4fb75db
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/2
@@ -0,0 +1 @@
+8
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/3 b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/3
new file mode 100644
index 0000000000..a0a162047c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/3
@@ -0,0 +1 @@
+0.0001
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/4 b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/4
new file mode 100644
index 0000000000..da24ab1137
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/4
@@ -0,0 +1 @@
+27277272727272727272
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/5 b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/5
new file mode 100644
index 0000000000..575c0a830c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/string_to_number-corpus/5
@@ -0,0 +1 @@
+999999999999999999999999999999999999999999999999999999999999999999999999999999
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/0.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/0.stun
new file mode 100644
index 0000000000..205997706f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/0.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/1.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/1.stun
new file mode 100644
index 0000000000..1f82c03ffa
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/1.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/10.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/10.stun
new file mode 100644
index 0000000000..a6b06b2be8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/10.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/11.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/11.stun
new file mode 100644
index 0000000000..1f82c03ffa
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/11.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/12.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/12.stun
new file mode 100644
index 0000000000..cb91baa250
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/12.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/13.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/13.stun
new file mode 100644
index 0000000000..63298fce7d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/13.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/14.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/14.stun
new file mode 100644
index 0000000000..31f9f732d4
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/14.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/15.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/15.stun
new file mode 100644
index 0000000000..3d15a67193
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/15.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/16.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/16.stun
new file mode 100644
index 0000000000..8ccf0af26d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/16.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/17.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/17.stun
new file mode 100644
index 0000000000..9c3efa0b80
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/17.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/2.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/2.stun
new file mode 100644
index 0000000000..50fe614adb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/2.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/3.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/3.stun
new file mode 100644
index 0000000000..50fe614adb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/3.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/4.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/4.stun
new file mode 100644
index 0000000000..a6b06b2be8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/4.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/5.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/5.stun
new file mode 100644
index 0000000000..c0a79fa7ab
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/5.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/6.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/6.stun
new file mode 100644
index 0000000000..1f43a4787d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/6.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/7.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/7.stun
new file mode 100644
index 0000000000..f5e824a68d
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/7.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/8.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/8.stun
new file mode 100644
index 0000000000..99f4e3bcf8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/8.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/9.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/9.stun
new file mode 100644
index 0000000000..cbbcab4e74
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/9.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/validator-crash-1.stun b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/validator-crash-1.stun
new file mode 100644
index 0000000000..5428c38d75
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun-corpus/validator-crash-1.stun
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/stun.tokens b/third_party/libwebrtc/test/fuzzers/corpora/stun.tokens
new file mode 100644
index 0000000000..e01130f12e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/stun.tokens
@@ -0,0 +1,2 @@
+"2112A442"
+"12214A24"
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/video_layers_allocation-corpus/vla-0 b/third_party/libwebrtc/test/fuzzers/corpora/video_layers_allocation-corpus/vla-0
new file mode 100644
index 0000000000..1b6fdf78f6
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/video_layers_allocation-corpus/vla-0
@@ -0,0 +1 @@
+
@2
?ÿÿ
\ No newline at end of file
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/0cee4d5fd2905dc1fb2979f10a9724265b7075e2 b/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/0cee4d5fd2905dc1fb2979f10a9724265b7075e2
new file mode 100644
index 0000000000..febe4ad130
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/0cee4d5fd2905dc1fb2979f10a9724265b7075e2
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/a1c75436e1872a23391d58316d88c45da0fb7682 b/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/a1c75436e1872a23391d58316d88c45da0fb7682
new file mode 100644
index 0000000000..fc538ac83f
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/a1c75436e1872a23391d58316d88c45da0fb7682
diff --git a/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/a8b3fb7be82395c9462684c766841d668dc0029f b/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/a8b3fb7be82395c9462684c766841d668dc0029f
new file mode 100644
index 0000000000..1bd09373c8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/corpora/vp9-encoder-references-corpus/a8b3fb7be82395c9462684c766841d668dc0029f
diff --git a/third_party/libwebrtc/test/fuzzers/dcsctp_packet_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/dcsctp_packet_fuzzer.cc
new file mode 100644
index 0000000000..2fc3fe10f1
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/dcsctp_packet_fuzzer.cc
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/packet/chunk/chunk.h"
+#include "net/dcsctp/packet/sctp_packet.h"
+
+namespace webrtc {
+using dcsctp::SctpPacket;
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  absl::optional<SctpPacket> c =
+      SctpPacket::Parse(rtc::ArrayView<const uint8_t>(data, size),
+                        /*disable_checksum_verification=*/true);
+
+  if (!c.has_value()) {
+    return;
+  }
+
+  for (const SctpPacket::ChunkDescriptor& desc : c->descriptors()) {
+    dcsctp::DebugConvertChunkToString(desc.data);
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/dcsctp_socket_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/dcsctp_socket_fuzzer.cc
new file mode 100644
index 0000000000..390cbb7f6c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/dcsctp_socket_fuzzer.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "net/dcsctp/fuzzers/dcsctp_fuzzers.h"
+#include "net/dcsctp/public/dcsctp_message.h"
+#include "net/dcsctp/public/dcsctp_options.h"
+#include "net/dcsctp/public/dcsctp_socket.h"
+#include "net/dcsctp/socket/dcsctp_socket.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  dcsctp::dcsctp_fuzzers::FuzzerCallbacks cb;
+  dcsctp::DcSctpOptions options;
+  options.disable_checksum_verification = true;
+  dcsctp::DcSctpSocket socket("A", cb, nullptr, options);
+
+  dcsctp::dcsctp_fuzzers::FuzzSocket(socket, cb,
+                                     rtc::ArrayView<const uint8_t>(data, size));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/field_trial_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/field_trial_fuzzer.cc
new file mode 100644
index 0000000000..74fe65ce56
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/field_trial_fuzzer.cc
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  std::string field_trial(reinterpret_cast<const char*>(data), size);
+  field_trial::InitFieldTrialsFromString(field_trial.c_str());
+  field_trial::FindFullName(field_trial);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/flexfec_header_reader_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/flexfec_header_reader_fuzzer.cc
new file mode 100644
index 0000000000..854cc8b811
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/flexfec_header_reader_fuzzer.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+
+namespace webrtc {
+
+using Packet = ForwardErrorCorrection::Packet;
+using ReceivedFecPacket = ForwardErrorCorrection::ReceivedFecPacket;
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  ReceivedFecPacket packet;
+  packet.pkt = rtc::scoped_refptr<Packet>(new Packet());
+  const size_t packet_size =
+      std::min(size, static_cast<size_t>(IP_PACKET_SIZE));
+  packet.pkt->data.SetSize(packet_size);
+  packet.pkt->data.EnsureCapacity(IP_PACKET_SIZE);
+  memcpy(packet.pkt->data.MutableData(), data, packet_size);
+
+  FlexfecHeaderReader flexfec_reader;
+  flexfec_reader.ReadFecHeader(&packet);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/flexfec_receiver_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
new file mode 100644
index 0000000000..67d603d3fc
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+namespace {
+class DummyCallback : public RecoveredPacketReceiver {
+  void OnRecoveredPacket(const RtpPacketReceived& packet) override {}
+};
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  constexpr size_t kMinDataNeeded = 12;
+  if (size < kMinDataNeeded || size > 2000) {
+    return;
+  }
+
+  uint32_t flexfec_ssrc;
+  memcpy(&flexfec_ssrc, data + 0, 4);
+  uint16_t flexfec_seq_num;
+  memcpy(&flexfec_seq_num, data + 4, 2);
+  uint32_t media_ssrc;
+  memcpy(&media_ssrc, data + 6, 4);
+  uint16_t media_seq_num;
+  memcpy(&media_seq_num, data + 10, 2);
+
+  DummyCallback callback;
+  FlexfecReceiver receiver(flexfec_ssrc, media_ssrc, &callback);
+
+  std::unique_ptr<uint8_t[]> packet;
+  size_t packet_length;
+  size_t i = kMinDataNeeded;
+  while (i < size) {
+    packet_length = kRtpHeaderSize + data[i++];
+    packet = std::unique_ptr<uint8_t[]>(new uint8_t[packet_length]);
+    if (i + packet_length >= size) {
+      break;
+    }
+    memcpy(packet.get(), data + i, packet_length);
+    i += packet_length;
+    if (i < size && data[i++] % 2 == 0) {
+      // Simulate FlexFEC packet.
+      ByteWriter<uint16_t>::WriteBigEndian(packet.get() + 2, flexfec_seq_num++);
+      ByteWriter<uint32_t>::WriteBigEndian(packet.get() + 8, flexfec_ssrc);
+    } else {
+      // Simulate media packet.
+      ByteWriter<uint16_t>::WriteBigEndian(packet.get() + 2, media_seq_num++);
+      ByteWriter<uint32_t>::WriteBigEndian(packet.get() + 8, media_ssrc);
+    }
+    RtpPacketReceived parsed_packet;
+    if (parsed_packet.Parse(packet.get(), packet_length)) {
+      receiver.OnRtpPacket(parsed_packet);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/flexfec_sender_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/flexfec_sender_fuzzer.cc
new file mode 100644
index 0000000000..8ddd1c0fe0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/flexfec_sender_fuzzer.cc
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/rtp_rtcp/include/flexfec_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr int kFlexfecPayloadType = 123;
+constexpr uint32_t kMediaSsrc = 1234;
+constexpr uint32_t kFlexfecSsrc = 5678;
+const char kNoMid[] = "";
+const std::vector<RtpExtension> kNoRtpHeaderExtensions;
+const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  size_t i = 0;
+  if (size < 5 || size > 200) {
+    return;
+  }
+  SimulatedClock clock(1 + data[i++]);
+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
+                       kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
+                       nullptr /* rtp_state */, &clock);
+  FecProtectionParams params = {
+      data[i++], static_cast<int>(data[i++] % 100),
+      data[i++] <= 127 ? kFecMaskRandom : kFecMaskBursty};
+  sender.SetProtectionParameters(params, params);
+  uint16_t seq_num = data[i++];
+
+  while (i + 1 < size) {
+    // Everything past the base RTP header (12 bytes) is payload,
+    // from the perspective of FlexFEC.
+    size_t payload_size = data[i++];
+    if (i + kRtpHeaderSize + payload_size >= size)
+      break;
+    std::unique_ptr<uint8_t[]> packet(
+        new uint8_t[kRtpHeaderSize + payload_size]);
+    memcpy(packet.get(), &data[i], kRtpHeaderSize + payload_size);
+    i += kRtpHeaderSize + payload_size;
+    ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++);
+    ByteWriter<uint32_t>::WriteBigEndian(&packet[8], kMediaSsrc);
+    RtpPacketToSend rtp_packet(nullptr);
+    if (!rtp_packet.Parse(packet.get(), kRtpHeaderSize + payload_size))
+      break;
+    sender.AddPacketAndGenerateFec(rtp_packet);
+    sender.GetFecPackets();
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc
new file mode 100644
index 0000000000..04a459bc71
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/forward_error_correction_fuzzer.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "rtc_base/byte_buffer.h"
+
+namespace webrtc {
+
+namespace {
+constexpr uint32_t kMediaSsrc = 100200300;
+constexpr uint32_t kFecSsrc = 111222333;
+
+constexpr size_t kPacketSize = 50;
+constexpr size_t kMaxPacketsInBuffer = 48;
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 5000) {
+    return;
+  }
+  // Object under test.
+  std::unique_ptr<ForwardErrorCorrection> fec =
+      ForwardErrorCorrection::CreateFlexfec(kFecSsrc, kMediaSsrc);
+
+  // Entropy from fuzzer.
+  rtc::ByteBufferReader fuzz_buffer(reinterpret_cast<const char*>(data), size);
+
+  // Initial stream state.
+  uint16_t media_seqnum;
+  if (!fuzz_buffer.ReadUInt16(&media_seqnum))
+    return;
+  const uint16_t original_media_seqnum = media_seqnum;
+  uint16_t fec_seqnum;
+  if (!fuzz_buffer.ReadUInt16(&fec_seqnum))
+    return;
+
+  // Existing packets in the packet buffer.
+  ForwardErrorCorrection::RecoveredPacketList recovered_packets;
+  uint8_t num_existing_recovered_packets;
+  if (!fuzz_buffer.ReadUInt8(&num_existing_recovered_packets))
+    return;
+  for (size_t i = 0; i < num_existing_recovered_packets % kMaxPacketsInBuffer;
+       ++i) {
+    ForwardErrorCorrection::RecoveredPacket* recovered_packet =
+        new ForwardErrorCorrection::RecoveredPacket();
+    recovered_packet->pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
+        new ForwardErrorCorrection::Packet());
+    recovered_packet->pkt->data.SetSize(kPacketSize);
+    memset(recovered_packet->pkt->data.MutableData(), 0, kPacketSize);
+    recovered_packet->ssrc = kMediaSsrc;
+    recovered_packet->seq_num = media_seqnum++;
+    recovered_packets.emplace_back(recovered_packet);
+  }
+
+  // New packets received from the network.
+  ForwardErrorCorrection::ReceivedPacket received_packet;
+  received_packet.pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
+      new ForwardErrorCorrection::Packet());
+  received_packet.pkt->data.SetSize(kPacketSize);
+  received_packet.pkt->data.EnsureCapacity(IP_PACKET_SIZE);
+  uint8_t* packet_buffer = received_packet.pkt->data.MutableData();
+  uint8_t reordering;
+  uint16_t seq_num_diff;
+  uint8_t packet_type;
+  uint8_t packet_loss;
+  while (true) {
+    if (!fuzz_buffer.ReadBytes(reinterpret_cast<char*>(packet_buffer),
+                               kPacketSize)) {
+      return;
+    }
+    if (!fuzz_buffer.ReadUInt8(&reordering))
+      return;
+    if (!fuzz_buffer.ReadUInt16(&seq_num_diff))
+      return;
+    if (!fuzz_buffer.ReadUInt8(&packet_type))
+      return;
+    if (!fuzz_buffer.ReadUInt8(&packet_loss))
+      return;
+
+    if (reordering % 10 != 0)
+      seq_num_diff = 0;
+
+    if (packet_type % 2 == 0) {
+      received_packet.is_fec = true;
+      received_packet.ssrc = kFecSsrc;
+      received_packet.seq_num = seq_num_diff + fec_seqnum++;
+
+      // Overwrite parts of the FlexFEC header for fuzzing efficiency.
+      packet_buffer[0] = 0;                                       // R, F bits.
+      ByteWriter<uint8_t>::WriteBigEndian(&packet_buffer[8], 1);  // SSRCCount.
+      ByteWriter<uint32_t>::WriteBigEndian(&packet_buffer[12],
+                                           kMediaSsrc);  // SSRC_i.
+      ByteWriter<uint16_t>::WriteBigEndian(
+          &packet_buffer[16], original_media_seqnum);  // SN base_i.
+    } else {
+      received_packet.is_fec = false;
+      received_packet.ssrc = kMediaSsrc;
+      received_packet.seq_num = seq_num_diff + media_seqnum++;
+    }
+
+    if (packet_loss % 10 == 0)
+      continue;
+
+    fec->DecodeFec(received_packet, &recovered_packets);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/frame_buffer2_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/frame_buffer2_fuzzer.cc
new file mode 100644
index 0000000000..ec1bbbb4c1
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/frame_buffer2_fuzzer.cc
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "api/task_queue/task_queue_base.h"
+#include "modules/video_coding/frame_buffer2.h"
+#include "modules/video_coding/timing/timing.h"
+#include "test/scoped_key_value_config.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+namespace {
+
+// When DataReader runs out of data provided in the constructor it will
+// just set/return 0 instead.
+struct DataReader {
+  DataReader(const uint8_t* data, size_t size) : data_(data), size_(size) {}
+
+  void CopyTo(void* destination, size_t dest_size) {
+    memset(destination, 0, dest_size);
+
+    size_t bytes_to_copy = std::min(size_ - offset_, dest_size);
+    memcpy(destination, data_ + offset_, bytes_to_copy);
+    offset_ += bytes_to_copy;
+  }
+
+  template <typename T>
+  T GetNum() {
+    T res;
+    if (offset_ + sizeof(res) < size_) {
+      memcpy(&res, data_ + offset_, sizeof(res));
+      offset_ += sizeof(res);
+      return res;
+    }
+
+    offset_ = size_;
+    return T(0);
+  }
+
+  bool MoreToRead() { return offset_ < size_; }
+
+  const uint8_t* const data_;
+  size_t size_;
+  size_t offset_ = 0;
+};
+
+class FuzzyFrameObject : public EncodedFrame {
+ public:
+  FuzzyFrameObject() {}
+  ~FuzzyFrameObject() {}
+
+  int64_t ReceivedTime() const override { return 0; }
+  int64_t RenderTime() const override { return _renderTimeMs; }
+};
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 10000) {
+    return;
+  }
+  DataReader reader(data, size);
+  GlobalSimulatedTimeController time_controller(Timestamp::Seconds(0));
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue =
+      time_controller.GetTaskQueueFactory()->CreateTaskQueue(
+          "time_tq", TaskQueueFactory::Priority::NORMAL);
+  test::ScopedKeyValueConfig field_trials;
+  VCMTiming timing(time_controller.GetClock(), field_trials);
+  video_coding::FrameBuffer frame_buffer(time_controller.GetClock(), &timing,
+                                         field_trials);
+
+  bool next_frame_task_running = false;
+
+  while (reader.MoreToRead()) {
+    if (reader.GetNum<uint8_t>() % 2) {
+      std::unique_ptr<FuzzyFrameObject> frame(new FuzzyFrameObject());
+      frame->SetId(reader.GetNum<int64_t>());
+      frame->SetSpatialIndex(reader.GetNum<uint8_t>() % 5);
+      frame->SetTimestamp(reader.GetNum<uint32_t>());
+      frame->num_references =
+          reader.GetNum<uint8_t>() % EncodedFrame::kMaxFrameReferences;
+
+      for (size_t r = 0; r < frame->num_references; ++r)
+        frame->references[r] = reader.GetNum<int64_t>();
+
+      frame_buffer.InsertFrame(std::move(frame));
+    } else {
+      if (!next_frame_task_running) {
+        next_frame_task_running = true;
+        bool keyframe_required = reader.GetNum<uint8_t>() % 2;
+        int max_wait_time_ms = reader.GetNum<uint8_t>();
+        task_queue->PostTask([&task_queue, &frame_buffer,
+                              &next_frame_task_running, keyframe_required,
+                              max_wait_time_ms] {
+          frame_buffer.NextFrame(
+              max_wait_time_ms, keyframe_required, task_queue.get(),
+              [&next_frame_task_running](std::unique_ptr<EncodedFrame> frame) {
+                next_frame_task_running = false;
+              });
+        });
+      }
+    }
+
+    time_controller.AdvanceTime(TimeDelta::Millis(reader.GetNum<uint8_t>()));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/frame_buffer_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/frame_buffer_fuzzer.cc
new file mode 100644
index 0000000000..e58d5e9f98
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/frame_buffer_fuzzer.cc
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/array_view.h"
+#include "api/video/encoded_frame.h"
+#include "api/video/frame_buffer.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+namespace {
+class FuzzyFrameObject : public EncodedFrame {
+ public:
+  int64_t ReceivedTime() const override { return 0; }
+  int64_t RenderTime() const override { return 0; }
+};
+
+constexpr int kFrameIdLength = 1 << 15;
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 10000) {
+    return;
+  }
+
+  test::ScopedKeyValueConfig field_trials;
+  FrameBuffer buffer(/*max_frame_slots=*/100, /*max_decode_history=*/1000,
+                     field_trials);
+  test::FuzzDataHelper helper(rtc::MakeArrayView(data, size));
+  SeqNumUnwrapper<uint16_t, kFrameIdLength> unwrapper;
+
+  while (helper.BytesLeft() > 0) {
+    int action = helper.ReadOrDefaultValue<uint8_t>(0) % 6;
+
+    switch (action) {
+      case 0: {
+        buffer.LastContinuousFrameId();
+        break;
+      }
+      case 1: {
+        buffer.LastContinuousTemporalUnitFrameId();
+        break;
+      }
+      case 2: {
+        buffer.DecodableTemporalUnitsInfo();
+        break;
+      }
+      case 3: {
+        buffer.ExtractNextDecodableTemporalUnit();
+        break;
+      }
+      case 4: {
+        buffer.DropNextDecodableTemporalUnit();
+        break;
+      }
+      case 5: {
+        auto frame = std::make_unique<FuzzyFrameObject>();
+        frame->SetTimestamp(helper.ReadOrDefaultValue<uint32_t>(0));
+        int64_t wire_id =
+            helper.ReadOrDefaultValue<uint16_t>(0) & (kFrameIdLength - 1);
+        frame->SetId(unwrapper.Unwrap(wire_id));
+        frame->is_last_spatial_layer = helper.ReadOrDefaultValue<bool>(false);
+
+        frame->num_references = helper.ReadOrDefaultValue<uint8_t>(0) %
+                                EncodedFrame::kMaxFrameReferences;
+
+        for (uint8_t i = 0; i < frame->num_references; ++i) {
+          frame->references[i] = helper.ReadOrDefaultValue<int64_t>(0);
+        }
+
+        buffer.InsertFrame(std::move(frame));
+        break;
+      }
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/fuzz_data_helper.cc b/third_party/libwebrtc/test/fuzzers/fuzz_data_helper.cc
new file mode 100644
index 0000000000..866f7bc4b8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/fuzz_data_helper.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace test {
+
+FuzzDataHelper::FuzzDataHelper(rtc::ArrayView<const uint8_t> data)
+    : data_(data) {}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/fuzz_data_helper.h b/third_party/libwebrtc/test/fuzzers/fuzz_data_helper.h
new file mode 100644
index 0000000000..cedc31d0a4
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/fuzz_data_helper.h
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FUZZERS_FUZZ_DATA_HELPER_H_
+#define TEST_FUZZERS_FUZZ_DATA_HELPER_H_
+
+#include <limits>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace test {
+
+// Helper class to take care of the fuzzer input, read from it, and keep track
+// of when the end of the data has been reached.
+class FuzzDataHelper {
+ public:
+  explicit FuzzDataHelper(rtc::ArrayView<const uint8_t> data);
+
+  // Returns true if n bytes can be read.
+  bool CanReadBytes(size_t n) const { return data_ix_ + n <= data_.size(); }
+
+  // Reads and returns data of type T.
+  template <typename T>
+  T Read() {
+    RTC_CHECK(CanReadBytes(sizeof(T)));
+    T x = ByteReader<T>::ReadLittleEndian(&data_[data_ix_]);
+    data_ix_ += sizeof(T);
+    return x;
+  }
+
+  // Reads and returns data of type T. Returns default_value if not enough
+  // fuzzer input remains to read a T.
+  template <typename T>
+  T ReadOrDefaultValue(T default_value) {
+    if (!CanReadBytes(sizeof(T))) {
+      return default_value;
+    }
+    return Read<T>();
+  }
+
+  // Like ReadOrDefaultValue, but replaces the value 0 with default_value.
+  template <typename T>
+  T ReadOrDefaultValueNotZero(T default_value) {
+    static_assert(std::is_integral<T>::value, "");
+    T x = ReadOrDefaultValue(default_value);
+    return x == 0 ? default_value : x;
+  }
+
+  // Returns one of the elements from the provided input array. The selection
+  // is based on the fuzzer input data. If not enough fuzzer data is available,
+  // the method will return the first element in the input array. The reason for
+  // not flagging this as an error is to allow the method to be called from
+  // class constructors, and in constructors we typically do not handle
+  // errors. The code will work anyway, and the fuzzer will likely see that
+  // providing more data will actually make this method return something else.
+  template <typename T, size_t N>
+  T SelectOneOf(const T (&select_from)[N]) {
+    static_assert(N <= std::numeric_limits<uint8_t>::max(), "");
+    // Read an index between 0 and select_from.size() - 1 from the fuzzer data.
+    uint8_t index = ReadOrDefaultValue<uint8_t>(0) % N;
+    return select_from[index];
+  }
+
+  rtc::ArrayView<const uint8_t> ReadByteArray(size_t bytes) {
+    if (!CanReadBytes(bytes)) {
+      return rtc::ArrayView<const uint8_t>(nullptr, 0);
+    }
+    const size_t index_to_return = data_ix_;
+    data_ix_ += bytes;
+    return data_.subview(index_to_return, bytes);
+  }
+
+  // If sizeof(T) > BytesLeft then the remaining bytes will be used and the rest
+  // of the object will be zero initialized.
+  template <typename T>
+  void CopyTo(T* object) {
+    memset(object, 0, sizeof(T));
+
+    size_t bytes_to_copy = std::min(BytesLeft(), sizeof(T));
+    memcpy(object, data_.data() + data_ix_, bytes_to_copy);
+    data_ix_ += bytes_to_copy;
+  }
+
+  size_t BytesRead() const { return data_ix_; }
+
+  size_t BytesLeft() const { return data_.size() - data_ix_; }
+
+ private:
+  rtc::ArrayView<const uint8_t> data_;
+  size_t data_ix_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FUZZERS_FUZZ_DATA_HELPER_H_
diff --git a/third_party/libwebrtc/test/fuzzers/h264_bitstream_parser_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/h264_bitstream_parser_fuzzer.cc
new file mode 100644
index 0000000000..cd1128c0b4
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/h264_bitstream_parser_fuzzer.cc
@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <stdint.h>
+
+#include "common_video/h264/h264_bitstream_parser.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  H264BitstreamParser h264_bitstream_parser;
+  h264_bitstream_parser.ParseBitstream(
+      rtc::ArrayView<const uint8_t>(data, size));
+  h264_bitstream_parser.GetLastSliceQp();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/h264_depacketizer_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/h264_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..97127228ed
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/h264_depacketizer_fuzzer.cc
@@ -0,0 +1,19 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_h264.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 200000)
+    return;
+  VideoRtpDepacketizerH264 depacketizer;
+  depacketizer.Parse(rtc::CopyOnWriteBuffer(data, size));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/neteq_rtp_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/neteq_rtp_fuzzer.cc
new file mode 100644
index 0000000000..348c84f040
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/neteq_rtp_fuzzer.cc
@@ -0,0 +1,184 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <cmath>
+#include <cstring>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/audio_checksum.h"
+#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+constexpr int kPayloadType = 95;
+
+class SineGenerator : public EncodeNetEqInput::Generator {
+ public:
+  explicit SineGenerator(int sample_rate_hz)
+      : sample_rate_hz_(sample_rate_hz) {}
+
+  rtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
+    if (samples_.size() < num_samples) {
+      samples_.resize(num_samples);
+    }
+
+    rtc::ArrayView<int16_t> output(samples_.data(), num_samples);
+    for (auto& x : output) {
+      x = static_cast<int16_t>(2000.0 * std::sin(phase_));
+      phase_ += 2 * kPi * kFreqHz / sample_rate_hz_;
+    }
+    return output;
+  }
+
+ private:
+  static constexpr int kFreqHz = 300;  // The sinewave frequency.
+  const int sample_rate_hz_;
+  const double kPi = std::acos(-1);
+  std::vector<int16_t> samples_;
+  double phase_ = 0.0;
+};
+
+class FuzzRtpInput : public NetEqInput {
+ public:
+  explicit FuzzRtpInput(rtc::ArrayView<const uint8_t> data) : data_(data) {
+    AudioEncoderPcm16B::Config config;
+    config.payload_type = kPayloadType;
+    config.sample_rate_hz = 32000;
+    std::unique_ptr<AudioEncoder> encoder(new AudioEncoderPcm16B(config));
+    std::unique_ptr<EncodeNetEqInput::Generator> generator(
+        new SineGenerator(config.sample_rate_hz));
+    input_.reset(new EncodeNetEqInput(std::move(generator), std::move(encoder),
+                                      std::numeric_limits<int64_t>::max()));
+    packet_ = input_->PopPacket();
+    FuzzHeader();
+    MaybeFuzzPayload();
+  }
+
+  absl::optional<int64_t> NextPacketTime() const override {
+    return packet_->time_ms;
+  }
+
+  absl::optional<int64_t> NextOutputEventTime() const override {
+    return input_->NextOutputEventTime();
+  }
+
+  std::unique_ptr<PacketData> PopPacket() override {
+    RTC_DCHECK(packet_);
+    std::unique_ptr<PacketData> packet_to_return = std::move(packet_);
+    packet_ = input_->PopPacket();
+    FuzzHeader();
+    MaybeFuzzPayload();
+    return packet_to_return;
+  }
+
+  void AdvanceOutputEvent() override { return input_->AdvanceOutputEvent(); }
+
+  bool ended() const override { return ended_; }
+
+  absl::optional<RTPHeader> NextHeader() const override {
+    RTC_DCHECK(packet_);
+    return packet_->header;
+  }
+
+ private:
+  void FuzzHeader() {
+    constexpr size_t kNumBytesToFuzz = 11;
+    if (data_ix_ + kNumBytesToFuzz > data_.size()) {
+      ended_ = true;
+      return;
+    }
+    RTC_DCHECK(packet_);
+    const size_t start_ix = data_ix_;
+    packet_->header.payloadType =
+        ByteReader<uint8_t>::ReadLittleEndian(&data_[data_ix_]);
+    packet_->header.payloadType &= 0x7F;
+    data_ix_ += sizeof(uint8_t);
+    packet_->header.sequenceNumber =
+        ByteReader<uint16_t>::ReadLittleEndian(&data_[data_ix_]);
+    data_ix_ += sizeof(uint16_t);
+    packet_->header.timestamp =
+        ByteReader<uint32_t>::ReadLittleEndian(&data_[data_ix_]);
+    data_ix_ += sizeof(uint32_t);
+    packet_->header.ssrc =
+        ByteReader<uint32_t>::ReadLittleEndian(&data_[data_ix_]);
+    data_ix_ += sizeof(uint32_t);
+    RTC_CHECK_EQ(data_ix_ - start_ix, kNumBytesToFuzz);
+  }
+
+  void MaybeFuzzPayload() {
+    // Read one byte of fuzz data to determine how many payload bytes to fuzz.
+    if (data_ix_ + 1 > data_.size()) {
+      ended_ = true;
+      return;
+    }
+    size_t bytes_to_fuzz = data_[data_ix_++];
+
+    // Restrict number of bytes to fuzz to 16; a reasonably low number enough to
+    // cover a few RED headers. Also don't write outside the payload length.
+    bytes_to_fuzz = std::min(bytes_to_fuzz % 16, packet_->payload.size());
+
+    if (bytes_to_fuzz == 0)
+      return;
+
+    if (data_ix_ + bytes_to_fuzz > data_.size()) {
+      ended_ = true;
+      return;
+    }
+
+    std::memcpy(packet_->payload.data(), &data_[data_ix_], bytes_to_fuzz);
+    data_ix_ += bytes_to_fuzz;
+  }
+
+  bool ended_ = false;
+  rtc::ArrayView<const uint8_t> data_;
+  size_t data_ix_ = 0;
+  std::unique_ptr<EncodeNetEqInput> input_;
+  std::unique_ptr<PacketData> packet_;
+};
+}  // namespace
+
+void FuzzOneInputTest(const uint8_t* data, size_t size) {
+  std::unique_ptr<FuzzRtpInput> input(
+      new FuzzRtpInput(rtc::ArrayView<const uint8_t>(data, size)));
+  std::unique_ptr<AudioChecksum> output(new AudioChecksum);
+  NetEqTest::Callbacks callbacks;
+  NetEq::Config config;
+  auto codecs = NetEqTest::StandardDecoderMap();
+  // kPayloadType is the payload type that will be used for encoding. Verify
+  // that it is included in the standard decoder map, and that it points to the
+  // expected decoder type.
+  const auto it = codecs.find(kPayloadType);
+  RTC_CHECK(it != codecs.end());
+  RTC_CHECK(it->second == SdpAudioFormat("L16", 32000, 1));
+
+  NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs,
+                 /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
+                 std::move(input), std::move(output), callbacks);
+  test.Run();
+}
+
+}  // namespace test
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 70000) {
+    return;
+  }
+  test::FuzzOneInputTest(data, size);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc
new file mode 100644
index 0000000000..8653f137a2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/neteq_signal_fuzzer.cc
@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cmath>
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/audio_checksum.h"
+#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/random.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+// Generate a mixture of sine wave and gaussian noise.
+class SineAndNoiseGenerator : public EncodeNetEqInput::Generator {
+ public:
+  // The noise generator is seeded with a value from the fuzzer data, but 0 is
+  // avoided (since it is not allowed by the Random class).
+  SineAndNoiseGenerator(int sample_rate_hz, FuzzDataHelper* fuzz_data)
+      : sample_rate_hz_(sample_rate_hz),
+        fuzz_data_(*fuzz_data),
+        noise_generator_(fuzz_data_.ReadOrDefaultValueNotZero<uint64_t>(1)) {}
+
+  // Generates num_samples of the sine-gaussian mixture.
+  rtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
+    if (samples_.size() < num_samples) {
+      samples_.resize(num_samples);
+    }
+
+    rtc::ArrayView<int16_t> output(samples_.data(), num_samples);
+    // Randomize an amplitude between 0 and 32768; use 65000/2 if we are out of
+    // fuzzer data.
+    const float amplitude = fuzz_data_.ReadOrDefaultValue<uint16_t>(65000) / 2;
+    // Randomize a noise standard deviation between 0 and 1999.
+    const float noise_std = fuzz_data_.ReadOrDefaultValue<uint16_t>(0) % 2000;
+    for (auto& x : output) {
+      x = rtc::saturated_cast<int16_t>(amplitude * std::sin(phase_) +
+                                       noise_generator_.Gaussian(0, noise_std));
+      phase_ += 2 * kPi * kFreqHz / sample_rate_hz_;
+    }
+    return output;
+  }
+
+ private:
+  static constexpr int kFreqHz = 300;  // The sinewave frequency.
+  const int sample_rate_hz_;
+  const double kPi = std::acos(-1);
+  std::vector<int16_t> samples_;
+  double phase_ = 0.0;
+  FuzzDataHelper& fuzz_data_;
+  Random noise_generator_;
+};
+
+class FuzzSignalInput : public NetEqInput {
+ public:
+  explicit FuzzSignalInput(FuzzDataHelper* fuzz_data,
+                           int sample_rate,
+                           uint8_t payload_type)
+      : fuzz_data_(*fuzz_data) {
+    AudioEncoderPcm16B::Config config;
+    config.payload_type = payload_type;
+    config.sample_rate_hz = sample_rate;
+    std::unique_ptr<AudioEncoder> encoder(new AudioEncoderPcm16B(config));
+    std::unique_ptr<EncodeNetEqInput::Generator> generator(
+        new SineAndNoiseGenerator(config.sample_rate_hz, fuzz_data));
+    input_.reset(new EncodeNetEqInput(std::move(generator), std::move(encoder),
+                                      std::numeric_limits<int64_t>::max()));
+    packet_ = input_->PopPacket();
+
+    // Select an output event period. This is how long time we wait between each
+    // call to NetEq::GetAudio. 10 ms is nominal, 9 and 11 ms will both lead to
+    // clock drift (in different directions).
+    constexpr int output_event_periods[] = {9, 10, 11};
+    output_event_period_ms_ = fuzz_data_.SelectOneOf(output_event_periods);
+  }
+
+  absl::optional<int64_t> NextPacketTime() const override {
+    return packet_->time_ms;
+  }
+
+  absl::optional<int64_t> NextOutputEventTime() const override {
+    return next_output_event_ms_;
+  }
+
+  std::unique_ptr<PacketData> PopPacket() override {
+    RTC_DCHECK(packet_);
+    std::unique_ptr<PacketData> packet_to_return = std::move(packet_);
+    do {
+      packet_ = input_->PopPacket();
+      // If the next value from the fuzzer input is 0, the packet is discarded
+      // and the next one is pulled from the source.
+    } while (fuzz_data_.CanReadBytes(1) && fuzz_data_.Read<uint8_t>() == 0);
+    if (fuzz_data_.CanReadBytes(1)) {
+      // Generate jitter by setting an offset for the arrival time.
+      const int8_t arrival_time_offset_ms = fuzz_data_.Read<int8_t>();
+      // The arrival time can not be before the previous packets.
+      packet_->time_ms = std::max(packet_to_return->time_ms,
+                                  packet_->time_ms + arrival_time_offset_ms);
+    } else {
+      // Mark that we are at the end of the test. However, the current packet is
+      // still valid (but it may not have been fuzzed as expected).
+      ended_ = true;
+    }
+    return packet_to_return;
+  }
+
+  void AdvanceOutputEvent() override {
+    next_output_event_ms_ += output_event_period_ms_;
+  }
+
+  bool ended() const override { return ended_; }
+
+  absl::optional<RTPHeader> NextHeader() const override {
+    RTC_DCHECK(packet_);
+    return packet_->header;
+  }
+
+ private:
+  bool ended_ = false;
+  FuzzDataHelper& fuzz_data_;
+  std::unique_ptr<EncodeNetEqInput> input_;
+  std::unique_ptr<PacketData> packet_;
+  int64_t next_output_event_ms_ = 0;
+  int64_t output_event_period_ms_ = 10;
+};
+
+template <class T>
+bool MapHas(const std::map<int, T>& m, int key, const T& value) {
+  const auto it = m.find(key);
+  return (it != m.end() && it->second == value);
+}
+
+}  // namespace
+
+void FuzzOneInputTest(const uint8_t* data, size_t size) {
+  if (size < 1 || size > 65000) {
+    return;
+  }
+
+  FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
+
+  // Allowed sample rates and payload types used in the test.
+  std::pair<int, uint8_t> rate_types[] = {
+      {8000, 93}, {16000, 94}, {32000, 95}, {48000, 96}};
+  const auto rate_type = fuzz_data.SelectOneOf(rate_types);
+  const int sample_rate = rate_type.first;
+  const uint8_t payload_type = rate_type.second;
+
+  // Set up the input signal generator.
+  std::unique_ptr<FuzzSignalInput> input(
+      new FuzzSignalInput(&fuzz_data, sample_rate, payload_type));
+
+  // Output sink for the test.
+  std::unique_ptr<AudioChecksum> output(new AudioChecksum);
+
+  // Configure NetEq and the NetEqTest object.
+  NetEqTest::Callbacks callbacks;
+  NetEq::Config config;
+  config.enable_post_decode_vad = true;
+  config.enable_fast_accelerate = true;
+  auto codecs = NetEqTest::StandardDecoderMap();
+  // rate_types contains the payload types that will be used for encoding.
+  // Verify that they all are included in the standard decoder map, and that
+  // they point to the expected decoder types.
+  RTC_CHECK(
+      MapHas(codecs, rate_types[0].second, SdpAudioFormat("l16", 8000, 1)));
+  RTC_CHECK(
+      MapHas(codecs, rate_types[1].second, SdpAudioFormat("l16", 16000, 1)));
+  RTC_CHECK(
+      MapHas(codecs, rate_types[2].second, SdpAudioFormat("l16", 32000, 1)));
+  RTC_CHECK(
+      MapHas(codecs, rate_types[3].second, SdpAudioFormat("l16", 48000, 1)));
+
+  NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs,
+                 /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
+                 std::move(input), std::move(output), callbacks);
+  test.Run();
+}
+
+}  // namespace test
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  test::FuzzOneInputTest(data, size);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/pseudotcp_parser_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/pseudotcp_parser_fuzzer.cc
new file mode 100644
index 0000000000..78ddf0e455
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/pseudotcp_parser_fuzzer.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "p2p/base/pseudo_tcp.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+class FakeIPseudoTcpNotify : public cricket::IPseudoTcpNotify {
+ public:
+  void OnTcpOpen(cricket::PseudoTcp* tcp) {}
+  void OnTcpReadable(cricket::PseudoTcp* tcp) {}
+  void OnTcpWriteable(cricket::PseudoTcp* tcp) {}
+  void OnTcpClosed(cricket::PseudoTcp* tcp, uint32_t error) {}
+
+  cricket::IPseudoTcpNotify::WriteResult TcpWritePacket(cricket::PseudoTcp* tcp,
+                                                        const char* buffer,
+                                                        size_t len) {
+    return cricket::IPseudoTcpNotify::WriteResult::WR_SUCCESS;
+  }
+};
+
+struct Environment {
+  explicit Environment(cricket::IPseudoTcpNotify* notifier)
+      : ptcp(notifier, 0) {}
+
+  // We need the thread to avoid some uninteresting crashes, since the
+  // production code expects there to be a thread object available.
+  rtc::AutoThread thread;
+  cricket::PseudoTcp ptcp;
+};
+
+Environment* env = new Environment(new FakeIPseudoTcpNotify());
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  env->ptcp.NotifyPacket(reinterpret_cast<const char*>(data), size);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/residual_echo_detector_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/residual_echo_detector_fuzzer.cc
new file mode 100644
index 0000000000..0efe81f220
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/residual_echo_detector_fuzzer.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <math.h>
+#include <string.h>
+
+#include <algorithm>
+#include <bitset>
+#include <vector>
+
+#include "api/audio/echo_detector_creator.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  // Number of times to update the echo detector.
+  constexpr size_t kNrOfUpdates = 7;
+  // Each round of updates requires a call to both AnalyzeRender and
+  // AnalyzeCapture, so the amount of needed input bytes doubles. Also, two
+  // bytes are used to set the call order.
+  constexpr size_t kNrOfNeededInputBytes = 2 * kNrOfUpdates * sizeof(float) + 2;
+  // The maximum audio energy that an audio frame can have is equal to the
+  // number of samples in the frame multiplied by 2^30. We use a single sample
+  // to represent an audio frame in this test, so it should have a maximum value
+  // equal to the square root of that value.
+  const float maxFuzzedValue = sqrtf(20 * 48) * 32768;
+  if (size < kNrOfNeededInputBytes) {
+    return;
+  }
+  size_t read_idx = 0;
+  // Use the first two bytes to choose the call order.
+  uint16_t call_order_int;
+  memcpy(&call_order_int, &data[read_idx], 2);
+  read_idx += 2;
+  std::bitset<16> call_order(call_order_int);
+
+  rtc::scoped_refptr<EchoDetector> echo_detector = CreateEchoDetector();
+  std::vector<float> input(1);
+  // Call AnalyzeCaptureAudio once to prevent the flushing of the buffer.
+  echo_detector->AnalyzeCaptureAudio(input);
+  for (size_t i = 0; i < 2 * kNrOfUpdates; ++i) {
+    // Convert 4 input bytes to a float.
+    RTC_DCHECK_LE(read_idx + sizeof(float), size);
+    memcpy(input.data(), &data[read_idx], sizeof(float));
+    read_idx += sizeof(float);
+    if (!isfinite(input[0]) || fabs(input[0]) > maxFuzzedValue) {
+      // Ignore infinity, nan values and values that are unrealistically large.
+      continue;
+    }
+    if (call_order[i]) {
+      echo_detector->AnalyzeRenderAudio(input);
+    } else {
+      echo_detector->AnalyzeCaptureAudio(input);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtcp_receiver_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
new file mode 100644
index 0000000000..8bad9e456a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+#include "modules/rtp_rtcp/source/rtcp_receiver.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kRtcpIntervalMs = 1000;
+
+// RTCP is typically sent over UDP, which has a maximum payload length
+// of 65535 bytes. We err on the side of caution and check a bit above that.
+constexpr size_t kMaxInputLenBytes = 66000;
+
+class NullModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
+ public:
+  void SetTmmbn(std::vector<rtcp::TmmbItem>) override {}
+  void OnRequestSendReport() override {}
+  void OnReceivedNack(const std::vector<uint16_t>&) override {}
+  void OnReceivedRtcpReportBlocks(const ReportBlockList&) override {}
+};
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > kMaxInputLenBytes) {
+    return;
+  }
+
+  NullModuleRtpRtcp rtp_rtcp_module;
+  SimulatedClock clock(1234);
+
+  RtpRtcpInterface::Configuration config;
+  config.clock = &clock;
+  config.rtcp_report_interval_ms = kRtcpIntervalMs;
+  config.local_media_ssrc = 1;
+
+  RTCPReceiver receiver(config, &rtp_rtcp_module);
+
+  receiver.IncomingPacket(data, size);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
new file mode 100644
index 0000000000..168e7b606b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_depacketizer_av1_assemble_frame_fuzzer.cc
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  std::vector<rtc::ArrayView<const uint8_t>> rtp_payloads;
+
+  // Convert plain array of bytes into array of array bytes.
+  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
+  while (fuzz_input.CanReadBytes(sizeof(uint16_t))) {
+    // In practice one rtp payload can be up to ~1200 - 1500 bytes. Majority
+    // of the payload is just copied. To make fuzzing more efficient limit the
+    // size of rtp payload to realistic value.
+    uint16_t next_size = fuzz_input.Read<uint16_t>() % 1200;
+    if (next_size > fuzz_input.BytesLeft()) {
+      next_size = fuzz_input.BytesLeft();
+    }
+    rtp_payloads.push_back(fuzz_input.ReadByteArray(next_size));
+  }
+  // Run code under test.
+  VideoRtpDepacketizerAv1().AssembleFrame(rtp_payloads);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
new file mode 100644
index 0000000000..82404f7264
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_dependency_descriptor_fuzzer.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+#include <memory>
+#include <utility>
+
+#include "api/array_view.h"
+#include "common_video/generic_frame_descriptor/generic_frame_info.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  FrameDependencyStructure structure1;
+  // nullptr during 1st while loop, after that should point to structure1.
+  const FrameDependencyStructure* structure1_ptr = nullptr;
+  std::unique_ptr<const FrameDependencyStructure> structure2;
+
+  test::FuzzDataHelper fuzz_data(rtc::MakeArrayView(data, size));
+  while (fuzz_data.CanReadBytes(1)) {
+    // Treat next byte as size of the next extension. That aligns how
+    // two-byte rtp header extension sizes are written.
+    size_t next_size = fuzz_data.Read<uint8_t>();
+    auto raw =
+        fuzz_data.ReadByteArray(std::min(next_size, fuzz_data.BytesLeft()));
+
+    // Read the random input.
+    DependencyDescriptor descriptor1;
+    if (!RtpDependencyDescriptorExtension::Parse(raw, structure1_ptr,
+                                                 &descriptor1)) {
+      // Ignore invalid buffer and move on.
+      continue;
+    }
+    if (descriptor1.attached_structure) {
+      structure1 = *descriptor1.attached_structure;
+      structure1_ptr = &structure1;
+    }
+    RTC_CHECK(structure1_ptr);
+
+    // Write parsed descriptor back into raw buffer.
+    size_t value_size =
+        RtpDependencyDescriptorExtension::ValueSize(structure1, descriptor1);
+    // Check `writer` use minimal number of bytes to pack the descriptor by
+    // checking it doesn't use more than reader consumed.
+    RTC_CHECK_LE(value_size, raw.size());
+    uint8_t some_memory[256];
+    // That should be true because value_size <= next_size < 256
+    RTC_CHECK_LT(value_size, 256);
+    rtc::ArrayView<uint8_t> write_buffer(some_memory, value_size);
+    RTC_CHECK(RtpDependencyDescriptorExtension::Write(write_buffer, structure1,
+                                                      descriptor1));
+
+    // Parse what Write assembled.
+    // Unlike random input that should always succeed.
+    DependencyDescriptor descriptor2;
+    RTC_CHECK(RtpDependencyDescriptorExtension::Parse(
+        write_buffer, structure2.get(), &descriptor2));
+    // Check descriptor1 and descriptor2 have same values.
+    RTC_CHECK_EQ(descriptor1.first_packet_in_frame,
+                 descriptor2.first_packet_in_frame);
+    RTC_CHECK_EQ(descriptor1.last_packet_in_frame,
+                 descriptor2.last_packet_in_frame);
+    RTC_CHECK_EQ(descriptor1.attached_structure != nullptr,
+                 descriptor2.attached_structure != nullptr);
+    // Using value_or would miss invalid corner case when one value is nullopt
+    // while another one is 0, but for other errors would produce much nicer
+    // error message than using RTC_CHECK(optional1 == optional2);
+    // If logger would support pretty printing optional values, value_or can be
+    // removed.
+    RTC_CHECK_EQ(descriptor1.active_decode_targets_bitmask.value_or(0),
+                 descriptor2.active_decode_targets_bitmask.value_or(0));
+    RTC_CHECK_EQ(descriptor1.frame_number, descriptor2.frame_number);
+    RTC_CHECK(descriptor1.resolution == descriptor2.resolution);
+    RTC_CHECK(descriptor1.frame_dependencies == descriptor2.frame_dependencies);
+
+    if (descriptor2.attached_structure) {
+      structure2 = std::move(descriptor2.attached_structure);
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
new file mode 100644
index 0000000000..fdb4aa5f3c
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
@@ -0,0 +1,154 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "api/rtp_packet_infos.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+
+namespace webrtc {
+
+namespace {
+class DataReader {
+ public:
+  DataReader(const uint8_t* data, size_t size) : data_(data), size_(size) {}
+
+  template <typename T>
+  void CopyTo(T* object) {
+    static_assert(std::is_pod<T>(), "");
+    uint8_t* destination = reinterpret_cast<uint8_t*>(object);
+    size_t object_size = sizeof(T);
+    size_t num_bytes = std::min(size_ - offset_, object_size);
+    memcpy(destination, data_ + offset_, num_bytes);
+    offset_ += num_bytes;
+
+    // If we did not have enough data, fill the rest with 0.
+    object_size -= num_bytes;
+    memset(destination + num_bytes, 0, object_size);
+  }
+
+  template <typename T>
+  T GetNum() {
+    T res;
+    if (offset_ + sizeof(res) < size_) {
+      memcpy(&res, data_ + offset_, sizeof(res));
+      offset_ += sizeof(res);
+      return res;
+    }
+
+    offset_ = size_;
+    return T(0);
+  }
+
+  bool MoreToRead() { return offset_ < size_; }
+
+ private:
+  const uint8_t* data_;
+  size_t size_;
+  size_t offset_ = 0;
+};
+
+absl::optional<RTPVideoHeader::GenericDescriptorInfo>
+GenerateGenericFrameDependencies(DataReader* reader) {
+  absl::optional<RTPVideoHeader::GenericDescriptorInfo> result;
+  uint8_t flags = reader->GetNum<uint8_t>();
+  if (flags & 0b1000'0000) {
+    // i.e. with 50% chance there are no generic dependencies.
+    // in such case codec-specfic code path of the RtpFrameReferenceFinder will
+    // be validated.
+    return result;
+  }
+
+  result.emplace();
+  result->frame_id = reader->GetNum<int32_t>();
+  result->spatial_index = (flags & 0b0111'0000) >> 4;
+  result->temporal_index = (flags & 0b0000'1110) >> 1;
+
+  // Larger than supported by the RtpFrameReferenceFinder.
+  int num_diffs = (reader->GetNum<uint8_t>() % 16);
+  for (int i = 0; i < num_diffs; ++i) {
+    result->dependencies.push_back(result->frame_id -
+                                   (reader->GetNum<uint16_t>() % (1 << 14)));
+  }
+
+  return result;
+}
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  DataReader reader(data, size);
+  RtpFrameReferenceFinder reference_finder;
+
+  auto codec = static_cast<VideoCodecType>(reader.GetNum<uint8_t>() % 5);
+
+  while (reader.MoreToRead()) {
+    uint16_t first_seq_num = reader.GetNum<uint16_t>();
+    uint16_t last_seq_num = reader.GetNum<uint16_t>();
+    bool marker_bit = reader.GetNum<uint8_t>();
+
+    RTPVideoHeader video_header;
+    switch (reader.GetNum<uint8_t>() % 3) {
+      case 0:
+        video_header.frame_type = VideoFrameType::kEmptyFrame;
+        break;
+      case 1:
+        video_header.frame_type = VideoFrameType::kVideoFrameKey;
+        break;
+      case 2:
+        video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+        break;
+    }
+
+    switch (codec) {
+      case kVideoCodecVP8:
+        reader.CopyTo(
+            &video_header.video_type_header.emplace<RTPVideoHeaderVP8>());
+        break;
+      case kVideoCodecVP9:
+        reader.CopyTo(
+            &video_header.video_type_header.emplace<RTPVideoHeaderVP9>());
+        break;
+      case kVideoCodecH264:
+        reader.CopyTo(
+            &video_header.video_type_header.emplace<RTPVideoHeaderH264>());
+        break;
+      default:
+        break;
+    }
+
+    video_header.generic = GenerateGenericFrameDependencies(&reader);
+
+    // clang-format off
+    auto frame = std::make_unique<RtpFrameObject>(
+        first_seq_num,
+        last_seq_num,
+        marker_bit,
+        /*times_nacked=*/0,
+        /*first_packet_received_time=*/0,
+        /*last_packet_received_time=*/0,
+        /*rtp_timestamp=*/0,
+        /*ntp_time_ms=*/0,
+        VideoSendTiming(),
+        /*payload_type=*/0,
+        codec,
+        kVideoRotation_0,
+        VideoContentType::UNSPECIFIED,
+        video_header,
+        /*color_space=*/absl::nullopt,
+        RtpPacketInfos(),
+        EncodedImageBuffer::Create(/*size=*/0));
+    // clang-format on
+
+    reference_finder.ManageFrame(std::move(frame));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc
new file mode 100644
index 0000000000..5d117529bb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc
@@ -0,0 +1,182 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <bitset>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+
+namespace webrtc {
+// We decide which header extensions to register by reading four bytes
+// from the beginning of `data` and interpreting it as a bitmask over
+// the RTPExtensionType enum. This assert ensures four bytes are enough.
+static_assert(kRtpExtensionNumberOfExtensions <= 32,
+              "Insufficient bits read to configure all header extensions. Add "
+              "an extra byte and update the switches.");
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size <= 4)
+    return;
+
+  // Don't use the configuration byte as part of the packet.
+  std::bitset<32> extensionMask(*reinterpret_cast<const uint32_t*>(data));
+  data += 4;
+  size -= 4;
+
+  RtpPacketReceived::ExtensionManager extensions(/*extmap_allow_mixed=*/true);
+  // Start at local_id = 1 since 0 is an invalid extension id.
+  int local_id = 1;
+  // Skip i = 0 since it maps to kRtpExtensionNone.
+  for (int i = 1; i < kRtpExtensionNumberOfExtensions; i++) {
+    RTPExtensionType extension_type = static_cast<RTPExtensionType>(i);
+    if (extensionMask[i]) {
+      // Extensions are registered with an ID, which you signal to the
+      // peer so they know what to expect. This code only cares about
+      // parsing so the value of the ID isn't relevant.
+      extensions.RegisterByType(local_id++, extension_type);
+    }
+  }
+
+  RtpPacketReceived packet(&extensions);
+  packet.Parse(data, size);
+
+  // Call packet accessors because they have extra checks.
+  packet.Marker();
+  packet.PayloadType();
+  packet.SequenceNumber();
+  packet.Timestamp();
+  packet.Ssrc();
+  packet.Csrcs();
+
+  // Each extension has its own getter. It is supported behaviour to
+  // call GetExtension on an extension which was not registered, so we
+  // don't check the bitmask here.
+  for (int i = 0; i < kRtpExtensionNumberOfExtensions; i++) {
+    switch (static_cast<RTPExtensionType>(i)) {
+      case kRtpExtensionNone:
+      case kRtpExtensionNumberOfExtensions:
+        break;
+      case kRtpExtensionTransmissionTimeOffset:
+        int32_t offset;
+        packet.GetExtension<TransmissionOffset>(&offset);
+        break;
+      case kRtpExtensionAudioLevel:
+        bool voice_activity;
+        uint8_t audio_level;
+        packet.GetExtension<AudioLevel>(&voice_activity, &audio_level);
+        break;
+#if !defined(WEBRTC_MOZILLA_BUILD)
+      case kRtpExtensionCsrcAudioLevel: {
+        std::vector<uint8_t> audio_levels;
+        packet.GetExtension<CsrcAudioLevel>(&audio_levels);
+        break;
+      }
+#endif
+      case kRtpExtensionAbsoluteSendTime:
+        uint32_t sendtime;
+        packet.GetExtension<AbsoluteSendTime>(&sendtime);
+        break;
+      case kRtpExtensionAbsoluteCaptureTime: {
+        AbsoluteCaptureTime extension;
+        packet.GetExtension<AbsoluteCaptureTimeExtension>(&extension);
+        break;
+      }
+      case kRtpExtensionVideoRotation:
+        uint8_t rotation;
+        packet.GetExtension<VideoOrientation>(&rotation);
+        break;
+      case kRtpExtensionTransportSequenceNumber:
+        uint16_t seqnum;
+        packet.GetExtension<TransportSequenceNumber>(&seqnum);
+        break;
+      case kRtpExtensionTransportSequenceNumber02: {
+        uint16_t seqnum;
+        absl::optional<FeedbackRequest> feedback_request;
+        packet.GetExtension<TransportSequenceNumberV2>(&seqnum,
+                                                       &feedback_request);
+        break;
+      }
+      case kRtpExtensionPlayoutDelay: {
+        VideoPlayoutDelay playout;
+        packet.GetExtension<PlayoutDelayLimits>(&playout);
+        break;
+      }
+      case kRtpExtensionVideoContentType:
+        VideoContentType content_type;
+        packet.GetExtension<VideoContentTypeExtension>(&content_type);
+        break;
+      case kRtpExtensionVideoTiming: {
+        VideoSendTiming timing;
+        packet.GetExtension<VideoTimingExtension>(&timing);
+        break;
+      }
+      case kRtpExtensionRtpStreamId: {
+        std::string rsid;
+        packet.GetExtension<RtpStreamId>(&rsid);
+        break;
+      }
+      case kRtpExtensionRepairedRtpStreamId: {
+        std::string rsid;
+        packet.GetExtension<RepairedRtpStreamId>(&rsid);
+        break;
+      }
+      case kRtpExtensionMid: {
+        std::string mid;
+        packet.GetExtension<RtpMid>(&mid);
+        break;
+      }
+      case kRtpExtensionGenericFrameDescriptor: {
+        RtpGenericFrameDescriptor descriptor;
+        packet.GetExtension<RtpGenericFrameDescriptorExtension00>(&descriptor);
+        break;
+      }
+      case kRtpExtensionColorSpace: {
+        ColorSpace color_space;
+        packet.GetExtension<ColorSpaceExtension>(&color_space);
+        break;
+      }
+      case kRtpExtensionInbandComfortNoise: {
+        absl::optional<uint8_t> noise_level;
+        packet.GetExtension<InbandComfortNoiseExtension>(&noise_level);
+        break;
+      }
+      case kRtpExtensionVideoLayersAllocation: {
+        VideoLayersAllocation allocation;
+        packet.GetExtension<RtpVideoLayersAllocationExtension>(&allocation);
+        break;
+      }
+      case kRtpExtensionVideoFrameTrackingId: {
+        uint16_t tracking_id;
+        packet.GetExtension<VideoFrameTrackingIdExtension>(&tracking_id);
+        break;
+      }
+      case kRtpExtensionDependencyDescriptor:
+        // This extension requires state to read and so complicated that
+        // deserves own fuzzer.
+        break;
+#if defined(WEBRTC_MOZILLA_BUILD)
+      case kRtpExtensionCsrcAudioLevel: {
+        CsrcAudioLevelList levels;
+        packet.GetExtension<CsrcAudioLevel>(&levels);
+        break;
+      }
+#endif
+    }
+  }
+
+  // Check that zero-ing mutable extensions wouldn't cause any problems.
+  packet.ZeroMutableExtensions();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc
new file mode 100644
index 0000000000..e5550c1279
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_packetizer_av1_fuzzer.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/rtp_rtcp/source/rtp_packetizer_av1.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  // Read uint8_t to be sure reduction_lens are much smaller than
+  // max_payload_len and thus limits structure is valid.
+  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.single_packet_reduction_len =
+      fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  const VideoFrameType kFrameTypes[] = {VideoFrameType::kVideoFrameKey,
+                                        VideoFrameType::kVideoFrameDelta};
+  VideoFrameType frame_type = fuzz_input.SelectOneOf(kFrameTypes);
+
+  // Main function under test: RtpPacketizerAv1's constructor.
+  RtpPacketizerAv1 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+                              limits, frame_type,
+                              /*is_last_frame_in_picture=*/true);
+
+  size_t num_packets = packetizer.NumPackets();
+  if (num_packets == 0) {
+    return;
+  }
+  // When packetization was successful, validate NextPacket function too.
+  // While at it, check that packets respect the payload size limits.
+  RtpPacketToSend rtp_packet(nullptr);
+  // Single packet.
+  if (num_packets == 1) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+    RTC_CHECK_LE(rtp_packet.payload_size(),
+                 limits.max_payload_len - limits.single_packet_reduction_len);
+    return;
+  }
+  // First packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.first_packet_reduction_len);
+  // Middle packets.
+  for (size_t i = 1; i < num_packets - 1; ++i) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet))
+        << "Failed to get packet#" << i;
+    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
+        << "Packet #" << i << " exceeds it's limit";
+  }
+  // Last packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.last_packet_reduction_len);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc
new file mode 100644
index 0000000000..6ab6b9a905
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_video_frame_assembler_fuzzer.cc
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2023 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+#include <cstddef>
+#include <cstdint>
+
+#include "api/video/rtp_video_frame_assembler.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size == 0) {
+    return;
+  }
+  RtpHeaderExtensionMap extensions;
+  extensions.Register<RtpDependencyDescriptorExtension>(1);
+  extensions.Register<RtpGenericFrameDescriptorExtension00>(2);
+  RtpPacketReceived rtp_packet(&extensions);
+
+  RtpVideoFrameAssembler assembler(
+      static_cast<RtpVideoFrameAssembler::PayloadFormat>(data[0] % 6));
+
+  for (size_t i = 1; i < size;) {
+    size_t packet_size = std::min<size_t>(size - i, 300);
+    if (rtp_packet.Parse(data + i, packet_size)) {
+      assembler.InsertPacket(rtp_packet);
+    }
+    i += packet_size;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
new file mode 100644
index 0000000000..ae8b8728fb
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstddef>
+#include <cstdint>
+#include <limits>
+
+#include "api/array_view.h"
+#include "api/video/video_layers_allocation.h"
+#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  // Video layers allocation is an rtp header extension.
+  // Per https://datatracker.ietf.org/doc/html/rfc8285#section-4.3
+  // rtp header extension uses up to one byte to store the size, i.e.
+  // maximum size of any rtp header extension is 255 bytes.
+  constexpr int kMaxSize = std::numeric_limits<uint8_t>::max();
+  if (size > kMaxSize) {
+    return;
+  }
+  auto raw = rtc::MakeArrayView(data, size);
+
+  VideoLayersAllocation allocation1;
+  if (!RtpVideoLayersAllocationExtension::Parse(raw, &allocation1)) {
+    // Ignore invalid buffer and move on.
+    return;
+  }
+
+  // Write parsed allocation back into raw buffer.
+  size_t value_size = RtpVideoLayersAllocationExtension::ValueSize(allocation1);
+  // Check `writer` use minimal number of bytes to pack the extension by
+  // checking it doesn't use more than reader consumed.
+  RTC_CHECK_LE(value_size, raw.size());
+  uint8_t some_memory[kMaxSize];
+  RTC_CHECK_LE(value_size, kMaxSize);
+  rtc::ArrayView<uint8_t> write_buffer(some_memory, value_size);
+  RTC_CHECK(
+      RtpVideoLayersAllocationExtension::Write(write_buffer, allocation1));
+
+  // Parse what Write assembled.
+  // Unlike random input that should always succeed.
+  VideoLayersAllocation allocation2;
+  RTC_CHECK(
+      RtpVideoLayersAllocationExtension::Parse(write_buffer, &allocation2));
+
+  RTC_CHECK_EQ(allocation1.rtp_stream_index, allocation2.rtp_stream_index);
+  RTC_CHECK_EQ(allocation1.resolution_and_frame_rate_is_valid,
+               allocation2.resolution_and_frame_rate_is_valid);
+  RTC_CHECK_EQ(allocation1.active_spatial_layers.size(),
+               allocation2.active_spatial_layers.size());
+  RTC_CHECK(allocation1 == allocation2);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/sctp_utils_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/sctp_utils_fuzzer.cc
new file mode 100644
index 0000000000..249707514e
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/sctp_utils_fuzzer.cc
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+#include "api/data_channel_interface.h"
+#include "pc/sctp_utils.h"
+#include "rtc_base/copy_on_write_buffer.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  rtc::CopyOnWriteBuffer payload(data, size);
+  std::string label;
+  DataChannelInit config;
+  IsOpenMessage(payload);
+  ParseDataChannelOpenMessage(payload, &label, &config);
+  ParseDataChannelOpenAckMessage(payload);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/sdp_integration_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/sdp_integration_fuzzer.cc
new file mode 100644
index 0000000000..ece4b50505
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/sdp_integration_fuzzer.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/strings/string_view.h"
+#include "pc/test/integration_test_helpers.h"
+
+namespace webrtc {
+
+class FuzzerTest : public PeerConnectionIntegrationBaseTest {
+ public:
+  FuzzerTest()
+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
+
+  void RunNegotiateCycle(absl::string_view message) {
+    CreatePeerConnectionWrappers();
+    // Note - we do not do test.ConnectFakeSignaling(); all signals
+    // generated are discarded.
+
+    auto srd_observer =
+        rtc::make_ref_counted<FakeSetRemoteDescriptionObserver>();
+
+    SdpParseError error;
+    std::unique_ptr<SessionDescriptionInterface> sdp(
+        CreateSessionDescription("offer", std::string(message), &error));
+    caller()->pc()->SetRemoteDescription(std::move(sdp), srd_observer);
+    // Wait a short time for observer to be called. Timeout is short
+    // because the fuzzer should be trying many branches.
+    EXPECT_TRUE_WAIT(srd_observer->called(), 100);
+
+    // If set-remote-description was successful, try to answer.
+    auto sld_observer =
+        rtc::make_ref_counted<FakeSetLocalDescriptionObserver>();
+    if (srd_observer->error().ok()) {
+      caller()->pc()->SetLocalDescription(sld_observer);
+      EXPECT_TRUE_WAIT(sld_observer->called(), 100);
+    }
+    // If there is an EXPECT failure, die here.
+    RTC_CHECK(!HasFailure());
+  }
+
+  // This test isn't using the test definition macros, so we have to
+  // define the TestBody() function, even though we don't need it.
+  void TestBody() override {}
+};
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 16384) {
+    return;
+  }
+
+  FuzzerTest test;
+  test.RunNegotiateCycle(
+      absl::string_view(reinterpret_cast<const char*>(data), size));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/sdp_parser_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/sdp_parser_fuzzer.cc
new file mode 100644
index 0000000000..c85eab4047
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/sdp_parser_fuzzer.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/jsep_session_description.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > 16384) {
+    return;
+  }
+  std::string message(reinterpret_cast<const char*>(data), size);
+  webrtc::SdpParseError error;
+
+  std::unique_ptr<webrtc::SessionDescriptionInterface> sdp(
+      CreateSessionDescription("offer", message, &error));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/ssl_certificate_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/ssl_certificate_fuzzer.cc
new file mode 100644
index 0000000000..4bab5c8f02
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/ssl_certificate_fuzzer.cc
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+#include "rtc_base/message_digest.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/string_encode.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  std::string pem_certificate(reinterpret_cast<const char*>(data), size);
+
+  std::unique_ptr<rtc::SSLCertificate> cert =
+      rtc::SSLCertificate::FromPEMString(pem_certificate);
+
+  if (cert == nullptr) {
+    return;
+  }
+
+  cert->Clone();
+  cert->GetStats();
+  cert->ToPEMString();
+  cert->CertificateExpirationTime();
+
+  std::string algorithm;
+  cert->GetSignatureDigestAlgorithm(&algorithm);
+
+  unsigned char digest[rtc::MessageDigest::kMaxSize];
+  size_t digest_len;
+  cert->ComputeDigest(algorithm, digest, rtc::MessageDigest::kMaxSize,
+                      &digest_len);
+
+  rtc::Buffer der_buffer;
+  cert->ToDER(&der_buffer);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/string_to_number_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/string_to_number_fuzzer.cc
new file mode 100644
index 0000000000..28b36a73ce
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/string_to_number_fuzzer.cc
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <string>
+
+#include "rtc_base/string_to_number.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  std::string number_to_parse(reinterpret_cast<const char*>(data), size);
+  rtc::StringToNumber<int8_t>(number_to_parse);
+  rtc::StringToNumber<int16_t>(number_to_parse);
+  rtc::StringToNumber<int32_t>(number_to_parse);
+  rtc::StringToNumber<int64_t>(number_to_parse);
+  rtc::StringToNumber<uint8_t>(number_to_parse);
+  rtc::StringToNumber<uint16_t>(number_to_parse);
+  rtc::StringToNumber<uint32_t>(number_to_parse);
+  rtc::StringToNumber<uint64_t>(number_to_parse);
+  rtc::StringToNumber<float>(number_to_parse);
+  rtc::StringToNumber<double>(number_to_parse);
+  rtc::StringToNumber<long double>(number_to_parse);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc
new file mode 100644
index 0000000000..6ca9eac8b2
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/stun_parser_fuzzer.cc
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/transport/stun.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  const char* message = reinterpret_cast<const char*>(data);
+
+  // Normally we'd check the integrity first, but those checks are
+  // fuzzed separately in stun_validator_fuzzer.cc. We still want to
+  // fuzz this target since the integrity checks could be forged by a
+  // malicious adversary who receives a call.
+  std::unique_ptr<cricket::IceMessage> stun_msg(new cricket::IceMessage());
+  rtc::ByteBufferReader buf(message, size);
+  stun_msg->Read(&buf);
+  stun_msg->ValidateMessageIntegrity("");
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/stun_validator_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/stun_validator_fuzzer.cc
new file mode 100644
index 0000000000..421638db1b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/stun_validator_fuzzer.cc
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/transport/stun.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  const char* message = reinterpret_cast<const char*>(data);
+
+  cricket::StunMessage::ValidateFingerprint(message, size);
+  cricket::StunMessage::ValidateMessageIntegrityForTesting(message, size, "");
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/turn_unwrap_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/turn_unwrap_fuzzer.cc
new file mode 100644
index 0000000000..47ee7fd205
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/turn_unwrap_fuzzer.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "media/base/turn_utils.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  size_t content_position;
+  size_t content_size;
+  cricket::UnwrapTurnPacket(data, size, &content_position, &content_size);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/ulpfec_generator_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/ulpfec_generator_fuzzer.cc
new file mode 100644
index 0000000000..43d9450918
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/ulpfec_generator_fuzzer.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/fec_test_helper.h"
+#include "modules/rtp_rtcp/source/ulpfec_generator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+namespace {
+constexpr uint8_t kFecPayloadType = 96;
+constexpr uint8_t kRedPayloadType = 97;
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  SimulatedClock clock(1);
+  UlpfecGenerator generator(kRedPayloadType, kFecPayloadType, &clock);
+  size_t i = 0;
+  if (size < 4)
+    return;
+  FecProtectionParams params = {
+      data[i++] % 128, static_cast<int>(data[i++] % 10), kFecMaskBursty};
+  generator.SetProtectionParameters(params, params);
+  uint16_t seq_num = data[i++];
+  uint16_t prev_seq_num = 0;
+  while (i + 3 < size) {
+    size_t rtp_header_length = data[i++] % 10 + 12;
+    size_t payload_size = data[i++] % 10;
+    if (i + payload_size + rtp_header_length + 2 > size)
+      break;
+    rtc::CopyOnWriteBuffer packet(&data[i], payload_size + rtp_header_length);
+    packet.EnsureCapacity(IP_PACKET_SIZE);
+    // Write a valid parsable header (version = 2, no padding, no extensions,
+    // no CSRCs).
+    ByteWriter<uint8_t>::WriteBigEndian(packet.MutableData(), 2 << 6);
+    // Make sure sequence numbers are increasing.
+    ByteWriter<uint16_t>::WriteBigEndian(packet.MutableData() + 2, seq_num++);
+    i += payload_size + rtp_header_length;
+    const bool protect = data[i++] % 2 == 1;
+
+    // Check the sequence numbers are monotonic. In rare case the packets number
+    // may loop around and in the same FEC-protected group the packet sequence
+    // number became out of order.
+    if (protect && IsNewerSequenceNumber(seq_num, prev_seq_num) &&
+        seq_num < prev_seq_num + kUlpfecMaxMediaPackets) {
+      RtpPacketToSend rtp_packet(nullptr);
+      // Check that we actually have a parsable packet, we want to fuzz FEC
+      // logic, not RTP header parsing.
+      RTC_CHECK(rtp_packet.Parse(packet));
+      generator.AddPacketAndGenerateFec(rtp_packet);
+      prev_seq_num = seq_num;
+    }
+
+    generator.GetFecPackets();
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/ulpfec_header_reader_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/ulpfec_header_reader_fuzzer.cc
new file mode 100644
index 0000000000..243cb4ed70
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/ulpfec_header_reader_fuzzer.cc
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+
+#include "api/scoped_refptr.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/forward_error_correction.h"
+#include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
+
+namespace webrtc {
+
+using Packet = ForwardErrorCorrection::Packet;
+using ReceivedFecPacket = ForwardErrorCorrection::ReceivedFecPacket;
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  ReceivedFecPacket packet;
+  packet.pkt = rtc::scoped_refptr<Packet>(new Packet());
+  const size_t packet_size =
+      std::min(size, static_cast<size_t>(IP_PACKET_SIZE));
+  packet.pkt->data.SetSize(packet_size);
+  packet.pkt->data.EnsureCapacity(IP_PACKET_SIZE);
+  memcpy(packet.pkt->data.MutableData(), data, packet_size);
+
+  UlpfecHeaderReader ulpfec_reader;
+  ulpfec_reader.ReadFecHeader(&packet);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
new file mode 100644
index 0000000000..0a29ba3259
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <algorithm>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/ulpfec_receiver.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+
+namespace {
+class DummyCallback : public RecoveredPacketReceiver {
+  void OnRecoveredPacket(const RtpPacketReceived& packet) override {}
+};
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  constexpr size_t kMinDataNeeded = 12;
+  if (size < kMinDataNeeded || size > 2000) {
+    return;
+  }
+
+  uint32_t ulpfec_ssrc = ByteReader<uint32_t>::ReadLittleEndian(data + 0);
+  uint16_t ulpfec_seq_num = ByteReader<uint16_t>::ReadLittleEndian(data + 4);
+  uint32_t media_ssrc = ByteReader<uint32_t>::ReadLittleEndian(data + 6);
+  uint16_t media_seq_num = ByteReader<uint16_t>::ReadLittleEndian(data + 10);
+
+  DummyCallback callback;
+  UlpfecReceiver receiver(ulpfec_ssrc, 0, &callback, Clock::GetRealTimeClock());
+
+  test::FuzzDataHelper fuzz_data(rtc::MakeArrayView(data, size));
+  while (fuzz_data.CanReadBytes(kMinDataNeeded)) {
+    size_t packet_length = kRtpHeaderSize + fuzz_data.Read<uint8_t>();
+    auto raw_packet = fuzz_data.ReadByteArray(packet_length);
+
+    RtpPacketReceived parsed_packet;
+    if (!parsed_packet.Parse(raw_packet))
+      continue;
+
+    // Overwrite the fields for the sequence number and SSRC with
+    // consistent values for either a received UlpFEC packet or received media
+    // packet. (We're still relying on libfuzzer to manage to generate packet
+    // headers that interact together; this just ensures that we have two
+    // consistent streams).
+    if (fuzz_data.ReadOrDefaultValue<uint8_t>(0) % 2 == 0) {
+      // Simulate UlpFEC packet.
+      parsed_packet.SetSequenceNumber(ulpfec_seq_num++);
+      parsed_packet.SetSsrc(ulpfec_ssrc);
+    } else {
+      // Simulate media packet.
+      parsed_packet.SetSequenceNumber(media_seq_num++);
+      parsed_packet.SetSsrc(media_ssrc);
+    }
+
+    receiver.AddReceivedRedPacket(parsed_packet);
+  }
+
+  receiver.ProcessReceivedFec();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/utils/BUILD.gn b/third_party/libwebrtc/test/fuzzers/utils/BUILD.gn
new file mode 100644
index 0000000000..c5744fc33b
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/utils/BUILD.gn
@@ -0,0 +1,47 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("rtp_replayer") {
+  testonly = true
+  sources = [
+    "rtp_replayer.cc",
+    "rtp_replayer.h",
+  ]
+  deps = [
+    "../../../api/rtc_event_log",
+    "../../../api/task_queue:default_task_queue_factory",
+    "../../../api/test/video:function_video_factory",
+    "../../../api/transport:field_trial_based_config",
+    "../../../api/units:timestamp",
+    "../../../api/video_codecs:video_codecs_api",
+    "../../../call",
+    "../../../call:call_interfaces",
+    "../../../common_video",
+    "../../../media:rtc_internal_video_codecs",
+    "../../../modules/rtp_rtcp:rtp_rtcp_format",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:rtc_base_tests_utils",
+    "../../../rtc_base:rtc_json",
+    "../../../rtc_base:timeutils",
+    "../../../system_wrappers",
+    "../../../test:call_config_utils",
+    "../../../test:encoder_settings",
+    "../../../test:fake_video_codecs",
+    "../../../test:null_transport",
+    "../../../test:rtp_test_utils",
+    "../../../test:run_loop",
+    "../../../test:run_test",
+    "../../../test:run_test_interface",
+    "../../../test:test_renderer",
+    "../../../test:test_support",
+    "../../../test:video_test_common",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory:memory" ]
+}
diff --git a/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc
new file mode 100644
index 0000000000..12743d89d9
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.cc
@@ -0,0 +1,200 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/fuzzers/utils/rtp_replayer.h"
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/strings/json.h"
+#include "system_wrappers/include/clock.h"
+#include "test/call_config_utils.h"
+#include "test/encoder_settings.h"
+#include "test/fake_decoder.h"
+#include "test/rtp_file_reader.h"
+#include "test/run_loop.h"
+
+namespace webrtc {
+namespace test {
+
+void RtpReplayer::Replay(const std::string& replay_config_filepath,
+                         const uint8_t* rtp_dump_data,
+                         size_t rtp_dump_size) {
+  auto stream_state = std::make_unique<StreamState>();
+  std::vector<VideoReceiveStreamInterface::Config> receive_stream_configs =
+      ReadConfigFromFile(replay_config_filepath, &(stream_state->transport));
+  return Replay(std::move(stream_state), std::move(receive_stream_configs),
+                rtp_dump_data, rtp_dump_size);
+}
+
+void RtpReplayer::Replay(
+    std::unique_ptr<StreamState> stream_state,
+    std::vector<VideoReceiveStreamInterface::Config> receive_stream_configs,
+    const uint8_t* rtp_dump_data,
+    size_t rtp_dump_size) {
+  RunLoop loop;
+  rtc::ScopedBaseFakeClock fake_clock;
+
+  // Work around: webrtc calls webrtc::Random(clock.TimeInMicroseconds())
+  // everywhere and Random expects non-zero seed. Let's set the clock non-zero
+  // to make them happy.
+  fake_clock.SetTime(webrtc::Timestamp::Millis(1));
+
+  // Attempt to create an RtpReader from the input file.
+  auto rtp_reader = CreateRtpReader(rtp_dump_data, rtp_dump_size);
+  if (rtp_reader == nullptr) {
+    RTC_LOG(LS_ERROR) << "Failed to create the rtp_reader";
+    return;
+  }
+
+  RtpHeaderExtensionMap extensions(/*extmap_allow_mixed=*/true);
+  // Skip i = 0 since it maps to kRtpExtensionNone.
+  for (int i = 1; i < kRtpExtensionNumberOfExtensions; i++) {
+    RTPExtensionType extension_type = static_cast<RTPExtensionType>(i);
+    // Extensions are registered with an ID, which you signal to the
+    // peer so they know what to expect. This code only cares about
+    // parsing so the value of the ID isn't relevant.
+    extensions.RegisterByType(i, extension_type);
+  }
+
+  // Setup the video streams based on the configuration.
+  webrtc::RtcEventLogNull event_log;
+  std::unique_ptr<TaskQueueFactory> task_queue_factory =
+      CreateDefaultTaskQueueFactory();
+  Call::Config call_config(&event_log);
+  call_config.task_queue_factory = task_queue_factory.get();
+  FieldTrialBasedConfig field_trials;
+  call_config.trials = &field_trials;
+  std::unique_ptr<Call> call(Call::Create(call_config));
+  SetupVideoStreams(&receive_stream_configs, stream_state.get(), call.get());
+
+  // Start replaying the provided stream now that it has been configured.
+  for (const auto& receive_stream : stream_state->receive_streams) {
+    receive_stream->Start();
+  }
+
+  ReplayPackets(&fake_clock, call.get(), rtp_reader.get(), extensions);
+
+  for (const auto& receive_stream : stream_state->receive_streams) {
+    call->DestroyVideoReceiveStream(receive_stream);
+  }
+}
+
+std::vector<VideoReceiveStreamInterface::Config>
+RtpReplayer::ReadConfigFromFile(const std::string& replay_config,
+                                Transport* transport) {
+  Json::CharReaderBuilder factory;
+  std::unique_ptr<Json::CharReader> json_reader =
+      absl::WrapUnique(factory.newCharReader());
+  Json::Value json_configs;
+  Json::String errors;
+  if (!json_reader->parse(replay_config.data(),
+                          replay_config.data() + replay_config.length(),
+                          &json_configs, &errors)) {
+    RTC_LOG(LS_ERROR)
+        << "Error parsing JSON replay configuration for the fuzzer: " << errors;
+    return {};
+  }
+
+  std::vector<VideoReceiveStreamInterface::Config> receive_stream_configs;
+  receive_stream_configs.reserve(json_configs.size());
+  for (const auto& json : json_configs) {
+    receive_stream_configs.push_back(
+        ParseVideoReceiveStreamJsonConfig(transport, json));
+  }
+  return receive_stream_configs;
+}
+
+void RtpReplayer::SetupVideoStreams(
+    std::vector<VideoReceiveStreamInterface::Config>* receive_stream_configs,
+    StreamState* stream_state,
+    Call* call) {
+  stream_state->decoder_factory = std::make_unique<InternalDecoderFactory>();
+  for (auto& receive_config : *receive_stream_configs) {
+    // Attach the decoder for the corresponding payload type in the config.
+    for (auto& decoder : receive_config.decoders) {
+      decoder = test::CreateMatchingDecoder(decoder.payload_type,
+                                            decoder.video_format.name);
+    }
+
+    // Create the window to display the rendered video.
+    stream_state->sinks.emplace_back(
+        test::VideoRenderer::Create("Fuzzing WebRTC Video Config", 640, 480));
+    // Create a receive stream for this config.
+    receive_config.renderer = stream_state->sinks.back().get();
+    receive_config.decoder_factory = stream_state->decoder_factory.get();
+    stream_state->receive_streams.emplace_back(
+        call->CreateVideoReceiveStream(std::move(receive_config)));
+  }
+}
+
+std::unique_ptr<test::RtpFileReader> RtpReplayer::CreateRtpReader(
+    const uint8_t* rtp_dump_data,
+    size_t rtp_dump_size) {
+  std::unique_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create(
+      test::RtpFileReader::kRtpDump, rtp_dump_data, rtp_dump_size, {}));
+  if (!rtp_reader) {
+    RTC_LOG(LS_ERROR) << "Unable to open input file with any supported format";
+    return nullptr;
+  }
+  return rtp_reader;
+}
+
+void RtpReplayer::ReplayPackets(
+    rtc::FakeClock* clock,
+    Call* call,
+    test::RtpFileReader* rtp_reader,
+    const RtpPacketReceived::ExtensionManager& extensions) {
+  int64_t replay_start_ms = -1;
+
+  while (true) {
+    int64_t now_ms = rtc::TimeMillis();
+    if (replay_start_ms == -1) {
+      replay_start_ms = now_ms;
+    }
+
+    test::RtpPacket packet;
+    if (!rtp_reader->NextPacket(&packet)) {
+      break;
+    }
+
+    int64_t deliver_in_ms = replay_start_ms + packet.time_ms - now_ms;
+    if (deliver_in_ms > 0) {
+      // StatsCounter::ReportMetricToAggregatedCounter is O(elapsed time).
+      // Set an upper limit to prevent waste time.
+      clock->AdvanceTime(webrtc::TimeDelta::Millis(
+          std::min(deliver_in_ms, static_cast<int64_t>(100))));
+    }
+
+    RtpPacketReceived received_packet(
+        &extensions, Timestamp::Micros(clock->TimeNanos() / 1000));
+    if (!received_packet.Parse(packet.data, packet.length)) {
+      RTC_LOG(LS_ERROR) << "Packet error, corrupt packets or incorrect setup?";
+      break;
+    }
+
+    call->Receiver()->DeliverRtpPacket(
+        MediaType::VIDEO, std::move(received_packet),
+        [&](const RtpPacketReceived& parsed_packet) {
+          RTC_LOG(LS_ERROR) << "Unknown SSRC: " << parsed_packet.Ssrc();
+          return false;
+        });
+  }
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h
new file mode 100644
index 0000000000..ae94a640a5
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/utils/rtp_replayer.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
+#define TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
+
+#include <stdio.h>
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/test/video/function_video_decoder_factory.h"
+#include "api/video_codecs/video_decoder.h"
+#include "call/call.h"
+#include "media/engine/internal_decoder_factory.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/time_utils.h"
+#include "test/null_transport.h"
+#include "test/rtp_file_reader.h"
+#include "test/test_video_capturer.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace test {
+
+// The RtpReplayer is a utility for fuzzing the RTP/RTCP receiver stack in
+// WebRTC. It achieves this by accepting a set of Receiver configurations and
+// an RtpDump (consisting of both RTP and RTCP packets). The `rtp_dump` is
+// passed in as a buffer to allow simple mutation fuzzing directly on the dump.
+class RtpReplayer final {
+ public:
+  // Holds all the important stream information required to emulate the WebRTC
+  // rtp receival code path.
+  struct StreamState {
+    test::NullTransport transport;
+    std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
+    std::vector<VideoReceiveStreamInterface*> receive_streams;
+    std::unique_ptr<VideoDecoderFactory> decoder_factory;
+  };
+
+  // Construct an RtpReplayer from a JSON replay configuration file.
+  static void Replay(const std::string& replay_config_filepath,
+                     const uint8_t* rtp_dump_data,
+                     size_t rtp_dump_size);
+
+  // Construct an RtpReplayer from  a set of
+  // VideoReceiveStreamInterface::Configs. Note the stream_state.transport must
+  // be set for each receiver stream.
+  static void Replay(
+      std::unique_ptr<StreamState> stream_state,
+      std::vector<VideoReceiveStreamInterface::Config> receive_stream_config,
+      const uint8_t* rtp_dump_data,
+      size_t rtp_dump_size);
+
+ private:
+  // Reads the replay configuration from Json.
+  static std::vector<VideoReceiveStreamInterface::Config> ReadConfigFromFile(
+      const std::string& replay_config,
+      Transport* transport);
+
+  // Configures the stream state based on the receiver configurations.
+  static void SetupVideoStreams(
+      std::vector<VideoReceiveStreamInterface::Config>* receive_stream_configs,
+      StreamState* stream_state,
+      Call* call);
+
+  // Creates a new RtpReader which can read the RtpDump
+  static std::unique_ptr<test::RtpFileReader> CreateRtpReader(
+      const uint8_t* rtp_dump_data,
+      size_t rtp_dump_size);
+
+  // Replays each packet to from the RtpDump.
+  static void ReplayPackets(rtc::FakeClock* clock,
+                            Call* call,
+                            test::RtpFileReader* rtp_reader,
+                            const RtpHeaderExtensionMap& extensions);
+};  // class RtpReplayer
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_FUZZERS_UTILS_RTP_REPLAYER_H_
diff --git a/third_party/libwebrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..1691b55cc0
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/vp8_depacketizer_fuzzer.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp8.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  RTPVideoHeader video_header;
+  VideoRtpDepacketizerVp8::ParseRtpPayload(rtc::MakeArrayView(data, size),
+                                           &video_header);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc
new file mode 100644
index 0000000000..2ecfd820c8
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/vp8_qp_parser_fuzzer.cc
@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/video_coding/utility/vp8_header_parser.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  int qp;
+  vp8::GetQp(data, size, &qp);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/vp8_replay_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/vp8_replay_fuzzer.cc
new file mode 100644
index 0000000000..819b9626f9
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/vp8_replay_fuzzer.cc
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "test/fuzzers/utils/rtp_replayer.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  auto stream_state = std::make_unique<test::RtpReplayer::StreamState>();
+  VideoReceiveStreamInterface::Config vp8_config(&(stream_state->transport));
+
+  VideoReceiveStreamInterface::Decoder vp8_decoder;
+  vp8_decoder.video_format = SdpVideoFormat("VP8");
+  vp8_decoder.payload_type = 125;
+  vp8_config.decoders.push_back(std::move(vp8_decoder));
+
+  vp8_config.rtp.local_ssrc = 7731;
+  vp8_config.rtp.remote_ssrc = 1337;
+  vp8_config.rtp.rtx_ssrc = 100;
+  vp8_config.rtp.nack.rtp_history_ms = 1000;
+  vp8_config.rtp.lntf.enabled = true;
+
+  std::vector<VideoReceiveStreamInterface::Config> replay_configs;
+  replay_configs.push_back(std::move(vp8_config));
+
+  test::RtpReplayer::Replay(std::move(stream_state), std::move(replay_configs),
+                            data, size);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc
new file mode 100644
index 0000000000..ae36a94931
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/vp9_depacketizer_fuzzer.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  RTPVideoHeader video_header;
+  VideoRtpDepacketizerVp9::ParseRtpPayload(rtc::MakeArrayView(data, size),
+                                           &video_header);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/vp9_encoder_references_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/vp9_encoder_references_fuzzer.cc
new file mode 100644
index 0000000000..09848ae4de
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/vp9_encoder_references_fuzzer.cc
@@ -0,0 +1,624 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdint.h>
+
+#include "absl/algorithm/container.h"
+#include "absl/base/macros.h"
+#include "absl/container/inlined_vector.h"
+#include "api/array_view.h"
+#include "api/field_trials_view.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/codecs/interface/libvpx_interface.h"
+#include "modules/video_coding/codecs/vp9/libvpx_vp9_encoder.h"
+#include "modules/video_coding/frame_dependencies_calculator.h"
+#include "rtc_base/numerics/safe_compare.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+// Fuzzer simulates various svc configurations and libvpx encoder dropping
+// layer frames.
+// Validates vp9 encoder wrapper produces consistent frame references.
+namespace webrtc {
+namespace {
+
+using test::FuzzDataHelper;
+
+constexpr int kBitrateEnabledBps = 100'000;
+
+class FrameValidator : public EncodedImageCallback {
+ public:
+  ~FrameValidator() override = default;
+
+  Result OnEncodedImage(const EncodedImage& encoded_image,
+                        const CodecSpecificInfo* codec_specific_info) override {
+    RTC_CHECK(codec_specific_info);
+    RTC_CHECK_EQ(codec_specific_info->codecType, kVideoCodecVP9);
+    if (codec_specific_info->codecSpecific.VP9.first_frame_in_picture) {
+      ++picture_id_;
+    }
+    int64_t frame_id = frame_id_++;
+    LayerFrame& layer_frame = frames_[frame_id % kMaxFrameHistorySize];
+    layer_frame.picture_id = picture_id_;
+    layer_frame.spatial_id = encoded_image.SpatialIndex().value_or(0);
+    layer_frame.frame_id = frame_id;
+    layer_frame.temporal_id =
+        codec_specific_info->codecSpecific.VP9.temporal_idx;
+    if (layer_frame.temporal_id == kNoTemporalIdx) {
+      layer_frame.temporal_id = 0;
+    }
+    layer_frame.vp9_non_ref_for_inter_layer_pred =
+        codec_specific_info->codecSpecific.VP9.non_ref_for_inter_layer_pred;
+    CheckVp9References(layer_frame, codec_specific_info->codecSpecific.VP9);
+
+    if (codec_specific_info->generic_frame_info.has_value()) {
+      absl::InlinedVector<int64_t, 5> frame_dependencies =
+          dependencies_calculator_.FromBuffersUsage(
+              frame_id,
+              codec_specific_info->generic_frame_info->encoder_buffers);
+
+      CheckGenericReferences(frame_dependencies,
+                             *codec_specific_info->generic_frame_info);
+      CheckGenericAndCodecSpecificReferencesAreConsistent(
+          frame_dependencies, *codec_specific_info, layer_frame);
+    }
+
+    return Result(Result::OK);
+  }
+
+ private:
+  // With 4 spatial layers and patterns up to 8 pictures, it should be enough to
+  // keep the last 32 frames to validate dependencies.
+  static constexpr size_t kMaxFrameHistorySize = 32;
+  struct LayerFrame {
+    int64_t frame_id;
+    int64_t picture_id;
+    int spatial_id;
+    int temporal_id;
+    bool vp9_non_ref_for_inter_layer_pred;
+  };
+
+  void CheckVp9References(const LayerFrame& layer_frame,
+                          const CodecSpecificInfoVP9& vp9_info) {
+    if (layer_frame.frame_id == 0) {
+      RTC_CHECK(!vp9_info.inter_layer_predicted);
+    } else {
+      const LayerFrame& previous_frame = Frame(layer_frame.frame_id - 1);
+      if (vp9_info.inter_layer_predicted) {
+        RTC_CHECK(!previous_frame.vp9_non_ref_for_inter_layer_pred);
+        RTC_CHECK_EQ(layer_frame.picture_id, previous_frame.picture_id);
+      }
+      if (previous_frame.picture_id == layer_frame.picture_id) {
+        RTC_CHECK_GT(layer_frame.spatial_id, previous_frame.spatial_id);
+        // The check below would fail for temporal shift structures. Remove it
+        // or move it to !flexible_mode section when vp9 encoder starts
+        // supporting such structures.
+        RTC_CHECK_EQ(layer_frame.temporal_id, previous_frame.temporal_id);
+      }
+    }
+    if (!vp9_info.flexible_mode) {
+      if (vp9_info.gof.num_frames_in_gof > 0) {
+        gof_.CopyGofInfoVP9(vp9_info.gof);
+      }
+      RTC_CHECK_EQ(gof_.temporal_idx[vp9_info.gof_idx],
+                   layer_frame.temporal_id);
+    }
+  }
+
+  void CheckGenericReferences(rtc::ArrayView<const int64_t> frame_dependencies,
+                              const GenericFrameInfo& generic_info) const {
+    for (int64_t dependency_frame_id : frame_dependencies) {
+      RTC_CHECK_GE(dependency_frame_id, 0);
+      const LayerFrame& dependency = Frame(dependency_frame_id);
+      RTC_CHECK_GE(generic_info.spatial_id, dependency.spatial_id);
+      RTC_CHECK_GE(generic_info.temporal_id, dependency.temporal_id);
+    }
+  }
+
+  void CheckGenericAndCodecSpecificReferencesAreConsistent(
+      rtc::ArrayView<const int64_t> frame_dependencies,
+      const CodecSpecificInfo& info,
+      const LayerFrame& layer_frame) const {
+    const CodecSpecificInfoVP9& vp9_info = info.codecSpecific.VP9;
+    const GenericFrameInfo& generic_info = *info.generic_frame_info;
+
+    RTC_CHECK_EQ(generic_info.spatial_id, layer_frame.spatial_id);
+    RTC_CHECK_EQ(generic_info.temporal_id, layer_frame.temporal_id);
+    auto picture_id_diffs =
+        rtc::MakeArrayView(vp9_info.p_diff, vp9_info.num_ref_pics);
+    RTC_CHECK_EQ(
+        frame_dependencies.size(),
+        picture_id_diffs.size() + (vp9_info.inter_layer_predicted ? 1 : 0));
+    for (int64_t dependency_frame_id : frame_dependencies) {
+      RTC_CHECK_GE(dependency_frame_id, 0);
+      const LayerFrame& dependency = Frame(dependency_frame_id);
+      if (dependency.spatial_id != layer_frame.spatial_id) {
+        RTC_CHECK(vp9_info.inter_layer_predicted);
+        RTC_CHECK_EQ(layer_frame.picture_id, dependency.picture_id);
+        RTC_CHECK_GT(layer_frame.spatial_id, dependency.spatial_id);
+      } else {
+        RTC_CHECK(vp9_info.inter_pic_predicted);
+        RTC_CHECK_EQ(layer_frame.spatial_id, dependency.spatial_id);
+        RTC_CHECK(absl::c_linear_search(
+            picture_id_diffs, layer_frame.picture_id - dependency.picture_id));
+      }
+    }
+  }
+
+  const LayerFrame& Frame(int64_t frame_id) const {
+    auto& frame = frames_[frame_id % kMaxFrameHistorySize];
+    RTC_CHECK_EQ(frame.frame_id, frame_id);
+    return frame;
+  }
+
+  GofInfoVP9 gof_;
+  int64_t frame_id_ = 0;
+  int64_t picture_id_ = 1;
+  FrameDependenciesCalculator dependencies_calculator_;
+  LayerFrame frames_[kMaxFrameHistorySize];
+};
+
+class FieldTrials : public FieldTrialsView {
+ public:
+  explicit FieldTrials(FuzzDataHelper& config)
+      : flags_(config.ReadOrDefaultValue<uint8_t>(0)) {}
+
+  ~FieldTrials() override = default;
+  std::string Lookup(absl::string_view key) const override {
+    static constexpr absl::string_view kBinaryFieldTrials[] = {
+        "WebRTC-Vp9ExternalRefCtrl",
+        "WebRTC-Vp9IssueKeyFrameOnLayerDeactivation",
+    };
+    for (size_t i = 0; i < ABSL_ARRAYSIZE(kBinaryFieldTrials); ++i) {
+      if (key == kBinaryFieldTrials[i]) {
+        return (flags_ & (1u << i)) ? "Enabled" : "Disabled";
+      }
+    }
+
+    // Ignore following field trials.
+    if (key == "WebRTC-CongestionWindow" ||
+        key == "WebRTC-UseBaseHeavyVP8TL3RateAllocation" ||
+        key == "WebRTC-SimulcastUpswitchHysteresisPercent" ||
+        key == "WebRTC-SimulcastScreenshareUpswitchHysteresisPercent" ||
+        key == "WebRTC-VideoRateControl" ||
+        key == "WebRTC-VP9-PerformanceFlags" ||
+        key == "WebRTC-VP9VariableFramerateScreenshare" ||
+        key == "WebRTC-VP9QualityScaler") {
+      return "";
+    }
+    // Crash when using unexpected field trial to decide if it should be fuzzed
+    // or have a constant value.
+    RTC_CHECK(false) << "Unfuzzed field trial " << key << "\n";
+  }
+
+ private:
+  const uint8_t flags_;
+};
+
+VideoCodec CodecSettings(FuzzDataHelper& rng) {
+  uint16_t config = rng.ReadOrDefaultValue<uint16_t>(0);
+  // Test up to to 4 spatial and 4 temporal layers.
+  int num_spatial_layers = 1 + (config & 0b11);
+  int num_temporal_layers = 1 + ((config >> 2) & 0b11);
+
+  VideoCodec codec_settings = {};
+  codec_settings.codecType = kVideoCodecVP9;
+  codec_settings.maxFramerate = 30;
+  codec_settings.width = 320 << (num_spatial_layers - 1);
+  codec_settings.height = 180 << (num_spatial_layers - 1);
+  if (num_spatial_layers > 1) {
+    for (int sid = 0; sid < num_spatial_layers; ++sid) {
+      SpatialLayer& spatial_layer = codec_settings.spatialLayers[sid];
+      codec_settings.width = 320 << sid;
+      codec_settings.height = 180 << sid;
+      spatial_layer.width = codec_settings.width;
+      spatial_layer.height = codec_settings.height;
+      spatial_layer.targetBitrate = kBitrateEnabledBps * num_temporal_layers;
+      spatial_layer.maxFramerate = codec_settings.maxFramerate;
+      spatial_layer.numberOfTemporalLayers = num_temporal_layers;
+    }
+  }
+  codec_settings.VP9()->numberOfSpatialLayers = num_spatial_layers;
+  codec_settings.VP9()->numberOfTemporalLayers = num_temporal_layers;
+  int inter_layer_pred = (config >> 4) & 0b11;
+  // There are only 3 valid values.
+  codec_settings.VP9()->interLayerPred = static_cast<InterLayerPredMode>(
+      inter_layer_pred < 3 ? inter_layer_pred : 0);
+  codec_settings.VP9()->flexibleMode = (config & (1u << 6)) != 0;
+  codec_settings.SetFrameDropEnabled((config & (1u << 7)) != 0);
+  codec_settings.mode = VideoCodecMode::kRealtimeVideo;
+  return codec_settings;
+}
+
+VideoEncoder::Settings EncoderSettings() {
+  return VideoEncoder::Settings(VideoEncoder::Capabilities(false),
+                                /*number_of_cores=*/1,
+                                /*max_payload_size=*/0);
+}
+
+bool IsSupported(int num_spatial_layers,
+                 int num_temporal_layers,
+                 const VideoBitrateAllocation& allocation) {
+  // VP9 encoder doesn't support certain configurations.
+  // BitrateAllocator shouldn't produce them.
+  if (allocation.get_sum_bps() == 0) {
+    // Ignore allocation that turns off all the layers.
+    // In such a case it is up to upper layer code not to call Encode.
+    return false;
+  }
+
+  for (int tid = 0; tid < num_temporal_layers; ++tid) {
+    int min_enabled_spatial_id = -1;
+    int max_enabled_spatial_id = -1;
+    int num_enabled_spatial_layers = 0;
+    for (int sid = 0; sid < num_spatial_layers; ++sid) {
+      if (allocation.GetBitrate(sid, tid) > 0) {
+        if (min_enabled_spatial_id == -1) {
+          min_enabled_spatial_id = sid;
+        }
+        max_enabled_spatial_id = sid;
+        ++num_enabled_spatial_layers;
+      }
+    }
+    if (num_enabled_spatial_layers == 0) {
+      // Each temporal layer should be enabled because skipping a full frame is
+      // not supported in non-flexible mode.
+      return false;
+    }
+    if (max_enabled_spatial_id - min_enabled_spatial_id + 1 !=
+        num_enabled_spatial_layers) {
+      // To avoid odd spatial dependencies, there should be no gaps in active
+      // spatial layers.
+      return false;
+    }
+  }
+
+  return true;
+}
+
+struct LibvpxState {
+  LibvpxState() {
+    pkt.kind = VPX_CODEC_CX_FRAME_PKT;
+    pkt.data.frame.buf = pkt_buffer;
+    pkt.data.frame.sz = ABSL_ARRAYSIZE(pkt_buffer);
+    layer_id.spatial_layer_id = -1;
+  }
+
+  uint8_t pkt_buffer[1000] = {};
+  vpx_codec_enc_cfg_t config = {};
+  vpx_codec_priv_output_cx_pkt_cb_pair_t callback = {};
+  vpx_image_t img = {};
+  vpx_svc_ref_frame_config_t ref_config = {};
+  vpx_svc_layer_id_t layer_id = {};
+  vpx_svc_frame_drop_t frame_drop = {};
+  vpx_codec_cx_pkt pkt = {};
+};
+
+class StubLibvpx : public LibvpxInterface {
+ public:
+  explicit StubLibvpx(LibvpxState* state) : state_(state) { RTC_CHECK(state_); }
+
+  vpx_codec_err_t codec_enc_config_default(vpx_codec_iface_t* iface,
+                                           vpx_codec_enc_cfg_t* cfg,
+                                           unsigned int usage) const override {
+    state_->config = *cfg;
+    return VPX_CODEC_OK;
+  }
+
+  vpx_codec_err_t codec_enc_init(vpx_codec_ctx_t* ctx,
+                                 vpx_codec_iface_t* iface,
+                                 const vpx_codec_enc_cfg_t* cfg,
+                                 vpx_codec_flags_t flags) const override {
+    RTC_CHECK(ctx);
+    ctx->err = VPX_CODEC_OK;
+    return VPX_CODEC_OK;
+  }
+
+  vpx_image_t* img_wrap(vpx_image_t* img,
+                        vpx_img_fmt_t fmt,
+                        unsigned int d_w,
+                        unsigned int d_h,
+                        unsigned int stride_align,
+                        unsigned char* img_data) const override {
+    state_->img.fmt = fmt;
+    state_->img.d_w = d_w;
+    state_->img.d_h = d_h;
+    return &state_->img;
+  }
+
+  vpx_codec_err_t codec_encode(vpx_codec_ctx_t* ctx,
+                               const vpx_image_t* img,
+                               vpx_codec_pts_t pts,
+                               uint64_t duration,
+                               vpx_enc_frame_flags_t flags,
+                               uint64_t deadline) const override {
+    if (flags & VPX_EFLAG_FORCE_KF) {
+      state_->pkt.data.frame.flags = VPX_FRAME_IS_KEY;
+    } else {
+      state_->pkt.data.frame.flags = 0;
+    }
+    state_->pkt.data.frame.duration = duration;
+    return VPX_CODEC_OK;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                void* param) const override {
+    if (ctrl_id == VP9E_REGISTER_CX_CALLBACK) {
+      state_->callback =
+          *reinterpret_cast<vpx_codec_priv_output_cx_pkt_cb_pair_t*>(param);
+    }
+    return VPX_CODEC_OK;
+  }
+
+  vpx_codec_err_t codec_control(
+      vpx_codec_ctx_t* ctx,
+      vp8e_enc_control_id ctrl_id,
+      vpx_svc_ref_frame_config_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_REF_FRAME_CONFIG:
+        state_->ref_config = *param;
+        break;
+      case VP9E_GET_SVC_REF_FRAME_CONFIG:
+        *param = state_->ref_config;
+        break;
+      default:
+        break;
+    }
+    return VPX_CODEC_OK;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_svc_layer_id_t* param) const override {
+    switch (ctrl_id) {
+      case VP9E_SET_SVC_LAYER_ID:
+        state_->layer_id = *param;
+        break;
+      case VP9E_GET_SVC_LAYER_ID:
+        *param = state_->layer_id;
+        break;
+      default:
+        break;
+    }
+    return VPX_CODEC_OK;
+  }
+
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_svc_frame_drop_t* param) const override {
+    if (ctrl_id == VP9E_SET_SVC_FRAME_DROP_LAYER) {
+      state_->frame_drop = *param;
+    }
+    return VPX_CODEC_OK;
+  }
+
+  vpx_codec_err_t codec_enc_config_set(
+      vpx_codec_ctx_t* ctx,
+      const vpx_codec_enc_cfg_t* cfg) const override {
+    state_->config = *cfg;
+    return VPX_CODEC_OK;
+  }
+
+  vpx_image_t* img_alloc(vpx_image_t* img,
+                         vpx_img_fmt_t fmt,
+                         unsigned int d_w,
+                         unsigned int d_h,
+                         unsigned int align) const override {
+    return nullptr;
+  }
+  void img_free(vpx_image_t* img) const override {}
+  vpx_codec_err_t codec_enc_init_multi(vpx_codec_ctx_t* ctx,
+                                       vpx_codec_iface_t* iface,
+                                       vpx_codec_enc_cfg_t* cfg,
+                                       int num_enc,
+                                       vpx_codec_flags_t flags,
+                                       vpx_rational_t* dsf) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_destroy(vpx_codec_ctx_t* ctx) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                uint32_t param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                int param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                int* param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_roi_map* param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_active_map* param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_scaling_mode* param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_svc_extra_cfg_t* param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(
+      vpx_codec_ctx_t* ctx,
+      vp8e_enc_control_id ctrl_id,
+      vpx_svc_spatial_layer_sync_t* param) const override {
+    return VPX_CODEC_OK;
+  }
+  vpx_codec_err_t codec_control(vpx_codec_ctx_t* ctx,
+                                vp8e_enc_control_id ctrl_id,
+                                vpx_rc_funcs_t* param) const override {
+    return VPX_CODEC_OK;
+  }
+  const vpx_codec_cx_pkt_t* codec_get_cx_data(
+      vpx_codec_ctx_t* ctx,
+      vpx_codec_iter_t* iter) const override {
+    return nullptr;
+  }
+  const char* codec_error_detail(vpx_codec_ctx_t* ctx) const override {
+    return nullptr;
+  }
+  const char* codec_error(vpx_codec_ctx_t* ctx) const override {
+    return nullptr;
+  }
+  const char* codec_err_to_string(vpx_codec_err_t err) const override {
+    return nullptr;
+  }
+
+ private:
+  LibvpxState* const state_;
+};
+
+enum Actions {
+  kEncode,
+  kSetRates,
+};
+
+// When a layer frame is marked for drop, drops all layer frames from that
+// pictures with larger spatial ids.
+constexpr bool DropAbove(uint8_t layers_mask, int sid) {
+  uint8_t full_mask = (uint8_t{1} << (sid + 1)) - 1;
+  return (layers_mask & full_mask) != full_mask;
+}
+// inline unittests
+static_assert(DropAbove(0b1011, /*sid=*/0) == false, "");
+static_assert(DropAbove(0b1011, /*sid=*/1) == false, "");
+static_assert(DropAbove(0b1011, /*sid=*/2) == true, "");
+static_assert(DropAbove(0b1011, /*sid=*/3) == true, "");
+
+// When a layer frame is marked for drop, drops all layer frames from that
+// pictures with smaller spatial ids.
+constexpr bool DropBelow(uint8_t layers_mask, int sid, int num_layers) {
+  return (layers_mask >> sid) != (1 << (num_layers - sid)) - 1;
+}
+// inline unittests
+static_assert(DropBelow(0b1101, /*sid=*/0, 4) == true, "");
+static_assert(DropBelow(0b1101, /*sid=*/1, 4) == true, "");
+static_assert(DropBelow(0b1101, /*sid=*/2, 4) == false, "");
+static_assert(DropBelow(0b1101, /*sid=*/3, 4) == false, "");
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  FuzzDataHelper helper(rtc::MakeArrayView(data, size));
+
+  FrameValidator validator;
+  FieldTrials field_trials(helper);
+  // Setup call callbacks for the fake
+  LibvpxState state;
+
+  // Initialize encoder
+  LibvpxVp9Encoder encoder(cricket::VideoCodec(),
+                           std::make_unique<StubLibvpx>(&state), field_trials);
+  VideoCodec codec = CodecSettings(helper);
+  if (encoder.InitEncode(&codec, EncoderSettings()) != WEBRTC_VIDEO_CODEC_OK) {
+    return;
+  }
+  RTC_CHECK_EQ(encoder.RegisterEncodeCompleteCallback(&validator),
+               WEBRTC_VIDEO_CODEC_OK);
+  {
+    // Enable all the layers initially. Encoder doesn't support producing
+    // frames when no layers are enabled.
+    LibvpxVp9Encoder::RateControlParameters parameters;
+    parameters.framerate_fps = 30.0;
+    for (int sid = 0; sid < codec.VP9()->numberOfSpatialLayers; ++sid) {
+      for (int tid = 0; tid < codec.VP9()->numberOfTemporalLayers; ++tid) {
+        parameters.bitrate.SetBitrate(sid, tid, kBitrateEnabledBps);
+      }
+    }
+    encoder.SetRates(parameters);
+  }
+
+  std::vector<VideoFrameType> frame_types(1);
+  VideoFrame fake_image = VideoFrame::Builder()
+                              .set_video_frame_buffer(I420Buffer::Create(
+                                  int{codec.width}, int{codec.height}))
+                              .build();
+
+  // Start producing frames at random.
+  while (helper.CanReadBytes(1)) {
+    uint8_t action = helper.Read<uint8_t>();
+    switch (action & 0b11) {
+      case kEncode: {
+        // bitmask of the action: SSSS-K00, where
+        // four S bit indicate which spatial layers should be produced,
+        // K bit indicates if frame should be a key frame.
+        frame_types[0] = (action & 0b100) ? VideoFrameType::kVideoFrameKey
+                                          : VideoFrameType::kVideoFrameDelta;
+        encoder.Encode(fake_image, &frame_types);
+        uint8_t encode_spatial_layers = (action >> 4);
+        for (size_t sid = 0; sid < state.config.ss_number_layers; ++sid) {
+          if (state.config.ss_target_bitrate[sid] == 0) {
+            // Don't encode disabled spatial layers.
+            continue;
+          }
+          bool drop = true;
+          switch (state.frame_drop.framedrop_mode) {
+            case FULL_SUPERFRAME_DROP:
+              drop = encode_spatial_layers == 0;
+              break;
+            case LAYER_DROP:
+              drop = (encode_spatial_layers & (1 << sid)) == 0;
+              break;
+            case CONSTRAINED_LAYER_DROP:
+              drop = DropBelow(encode_spatial_layers, sid,
+                               state.config.ss_number_layers);
+              break;
+            case CONSTRAINED_FROM_ABOVE_DROP:
+              drop = DropAbove(encode_spatial_layers, sid);
+              break;
+          }
+          if (!drop) {
+            state.layer_id.spatial_layer_id = sid;
+            state.callback.output_cx_pkt(&state.pkt, state.callback.user_priv);
+          }
+        }
+      } break;
+      case kSetRates: {
+        // bitmask of the action: (S2)(S1)(S0)01,
+        // where Sx is number of temporal layers to enable for spatial layer x
+        // In pariculat Sx = 0 indicates spatial layer x should be disabled.
+        LibvpxVp9Encoder::RateControlParameters parameters;
+        parameters.framerate_fps = 30.0;
+        for (int sid = 0; sid < codec.VP9()->numberOfSpatialLayers; ++sid) {
+          int temporal_layers = (action >> ((1 + sid) * 2)) & 0b11;
+          for (int tid = 0; tid < temporal_layers; ++tid) {
+            parameters.bitrate.SetBitrate(sid, tid, kBitrateEnabledBps);
+          }
+        }
+        if (IsSupported(codec.VP9()->numberOfSpatialLayers,
+                        codec.VP9()->numberOfTemporalLayers,
+                        parameters.bitrate)) {
+          encoder.SetRates(parameters);
+        }
+      } break;
+      default:
+        // Unspecificed values are noop.
+        break;
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/vp9_qp_parser_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/vp9_qp_parser_fuzzer.cc
new file mode 100644
index 0000000000..80dfe15b16
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/vp9_qp_parser_fuzzer.cc
@@ -0,0 +1,17 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  ParseUncompressedVp9Header(rtc::MakeArrayView(data, size));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/vp9_replay_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/vp9_replay_fuzzer.cc
new file mode 100644
index 0000000000..fc10d9ffc7
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/vp9_replay_fuzzer.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "test/fuzzers/utils/rtp_replayer.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  auto stream_state = std::make_unique<test::RtpReplayer::StreamState>();
+  VideoReceiveStreamInterface::Config vp9_config(&(stream_state->transport));
+
+  VideoReceiveStreamInterface::Decoder vp9_decoder;
+  vp9_decoder.video_format = SdpVideoFormat("VP9");
+  vp9_decoder.payload_type = 124;
+  vp9_config.decoders.push_back(std::move(vp9_decoder));
+
+  vp9_config.rtp.local_ssrc = 7731;
+  vp9_config.rtp.remote_ssrc = 1337;
+  vp9_config.rtp.rtx_ssrc = 100;
+  vp9_config.rtp.nack.rtp_history_ms = 1000;
+
+  std::vector<VideoReceiveStreamInterface::Config> replay_configs;
+  replay_configs.push_back(std::move(vp9_config));
+
+  test::RtpReplayer::Replay(std::move(stream_state), std::move(replay_configs),
+                            data, size);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/fuzzers/webrtc_fuzzer_main.cc b/third_party/libwebrtc/test/fuzzers/webrtc_fuzzer_main.cc
new file mode 100644
index 0000000000..a52dd231be
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/webrtc_fuzzer_main.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file is intended to provide a common interface for fuzzing functions.
+// It's intended to set sane defaults, such as removing logging for further
+// fuzzing efficiency.
+
+#include "rtc_base/logging.h"
+
+namespace {
+bool g_initialized = false;
+void InitializeWebRtcFuzzDefaults() {
+  if (g_initialized)
+    return;
+
+// Remove default logging to prevent huge slowdowns.
+// TODO(pbos): Disable in Chromium: http://crbug.com/561667
+#if !defined(WEBRTC_CHROMIUM_BUILD)
+  rtc::LogMessage::LogToDebug(rtc::LS_NONE);
+#endif  // !defined(WEBRTC_CHROMIUM_BUILD)
+
+  g_initialized = true;
+}
+}  // namespace
+
+namespace webrtc {
+extern void FuzzOneInput(const uint8_t* data, size_t size);
+}  // namespace webrtc
+
+extern "C" int LLVMFuzzerTestOneInput(const unsigned char* data, size_t size) {
+  InitializeWebRtcFuzzDefaults();
+  webrtc::FuzzOneInput(data, size);
+  return 0;
+}
diff --git a/third_party/libwebrtc/test/gl/gl_renderer.cc b/third_party/libwebrtc/test/gl/gl_renderer.cc
new file mode 100644
index 0000000000..10162ee056
--- /dev/null
+++ b/third_party/libwebrtc/test/gl/gl_renderer.cc
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/gl/gl_renderer.h"
+
+#include <string.h>
+
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+GlRenderer::GlRenderer()
+    : is_init_(false), buffer_(NULL), width_(0), height_(0) {}
+
+void GlRenderer::Init() {
+  RTC_DCHECK(!is_init_);
+  is_init_ = true;
+
+  glGenTextures(1, &texture_);
+}
+
+void GlRenderer::Destroy() {
+  if (!is_init_) {
+    return;
+  }
+
+  is_init_ = false;
+
+  delete[] buffer_;
+  buffer_ = NULL;
+
+  glDeleteTextures(1, &texture_);
+}
+
+void GlRenderer::ResizeViewport(size_t width, size_t height) {
+  // TODO(pbos): Aspect ratio, letterbox the video.
+  glViewport(0, 0, width, height);
+
+  glMatrixMode(GL_PROJECTION);
+  glLoadIdentity();
+  glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
+  glOrtho(0.0f, 1.0f, 1.0f, 0.0f, -1.0f, 1.0f);
+  glMatrixMode(GL_MODELVIEW);
+}
+
+void GlRenderer::ResizeVideo(size_t width, size_t height) {
+  RTC_DCHECK(is_init_);
+  width_ = width;
+  height_ = height;
+
+  buffer_size_ = width * height * 4;  // BGRA
+
+  delete[] buffer_;
+  buffer_ = new uint8_t[buffer_size_];
+  RTC_DCHECK(buffer_);
+  memset(buffer_, 0, buffer_size_);
+  glBindTexture(GL_TEXTURE_2D, texture_);
+  glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_BASE_LEVEL, 0);
+  glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAX_LEVEL, 0);
+  glTexImage2D(GL_TEXTURE_2D, 0, GL_RGB, width, height, 0, GL_BGRA,
+               GL_UNSIGNED_INT_8_8_8_8, static_cast<GLvoid*>(buffer_));
+}
+
+void GlRenderer::OnFrame(const webrtc::VideoFrame& frame) {
+  RTC_DCHECK(is_init_);
+
+  if (static_cast<size_t>(frame.width()) != width_ ||
+      static_cast<size_t>(frame.height()) != height_) {
+    ResizeVideo(frame.width(), frame.height());
+  }
+
+  webrtc::ConvertFromI420(frame, VideoType::kBGRA, 0, buffer_);
+
+  glEnable(GL_TEXTURE_2D);
+  glBindTexture(GL_TEXTURE_2D, texture_);
+  glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width_, height_, GL_BGRA,
+                  GL_UNSIGNED_INT_8_8_8_8, buffer_);
+
+  glClear(GL_COLOR_BUFFER_BIT | GL_DEPTH_BUFFER_BIT);
+
+  glLoadIdentity();
+
+  glBegin(GL_QUADS);
+  {
+    glTexCoord2f(0.0f, 0.0f);
+    glVertex3f(0.0f, 0.0f, 0.0f);
+
+    glTexCoord2f(0.0f, 1.0f);
+    glVertex3f(0.0f, 1.0f, 0.0f);
+
+    glTexCoord2f(1.0f, 1.0f);
+    glVertex3f(1.0f, 1.0f, 0.0f);
+
+    glTexCoord2f(1.0f, 0.0f);
+    glVertex3f(1.0f, 0.0f, 0.0f);
+  }
+  glEnd();
+
+  glBindTexture(GL_TEXTURE_2D, 0);
+  glFlush();
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/gl/gl_renderer.h b/third_party/libwebrtc/test/gl/gl_renderer.h
new file mode 100644
index 0000000000..8338591244
--- /dev/null
+++ b/third_party/libwebrtc/test/gl/gl_renderer.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_GL_GL_RENDERER_H_
+#define TEST_GL_GL_RENDERER_H_
+
+#ifdef WEBRTC_MAC
+#include <OpenGL/gl.h>
+#else
+#include <GL/gl.h>
+#endif
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/video/video_frame.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace test {
+
+class GlRenderer : public VideoRenderer {
+ public:
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+ protected:
+  GlRenderer();
+
+  void Init();
+  void Destroy();
+
+  void ResizeViewport(size_t width, size_t height);
+
+ private:
+  bool is_init_;
+  uint8_t* buffer_;
+  GLuint texture_;
+  size_t width_, height_, buffer_size_;
+
+  void ResizeVideo(size_t width, size_t height);
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_GL_GL_RENDERER_H_
diff --git a/third_party/libwebrtc/test/gmock.h b/third_party/libwebrtc/test/gmock.h
new file mode 100644
index 0000000000..f137d080a4
--- /dev/null
+++ b/third_party/libwebrtc/test/gmock.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_GMOCK_H_
+#define TEST_GMOCK_H_
+
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "testing/gmock/include/gmock/gmock.h"
+RTC_POP_IGNORING_WUNDEF()
+
+#endif  // TEST_GMOCK_H_
diff --git a/third_party/libwebrtc/test/gtest.h b/third_party/libwebrtc/test/gtest.h
new file mode 100644
index 0000000000..fa4396420e
--- /dev/null
+++ b/third_party/libwebrtc/test/gtest.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_GTEST_H_
+#define TEST_GTEST_H_
+
+#include "rtc_base/ignore_wundef.h"
+
+RTC_PUSH_IGNORING_WUNDEF()
+#include "testing/gtest/include/gtest/gtest-spi.h"
+#include "testing/gtest/include/gtest/gtest.h"
+RTC_POP_IGNORING_WUNDEF()
+
+// GTEST_HAS_DEATH_TEST is set to 1 when death tests are supported, but appears
+// to be left unset if they're not supported. Rather than depend on this, we
+// set it to 0 ourselves here.
+#ifndef GTEST_HAS_DEATH_TEST
+#define GTEST_HAS_DEATH_TEST 0
+#endif
+
+#endif  // TEST_GTEST_H_
diff --git a/third_party/libwebrtc/test/ios/Info.plist b/third_party/libwebrtc/test/ios/Info.plist
new file mode 100644
index 0000000000..fe06a5e005
--- /dev/null
+++ b/third_party/libwebrtc/test/ios/Info.plist
@@ -0,0 +1,47 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>CFBundleDevelopmentRegion</key>
+	<string>English</string>
+	<key>CFBundleDisplayName</key>
+	<string>${PRODUCT_NAME}</string>
+	<key>CFBundleExecutable</key>
+	<string>${EXECUTABLE_NAME}</string>
+	<key>CFBundleIdentifier</key>
+	<string>${BUNDLE_IDENTIFIER}</string>
+	<key>CFBundleInfoDictionaryVersion</key>
+	<string>6.0</string>
+	<key>CFBundleName</key>
+	<string>${PRODUCT_NAME}</string>
+	<key>CFBundlePackageType</key>
+	<string>APPL</string>
+	<key>CFBundleShortVersionString</key>
+	<string>1.0</string>
+	<key>CFBundleSignature</key>
+	<string>????</string>
+	<key>CFBundleVersion</key>
+	<string>1.0</string>
+	<key>LSRequiresIPhoneOS</key>
+	<true/>
+	<key>NSAppTransportSecurity</key>
+	<dict>
+		<key>NSAllowsArbitraryLoads</key>
+		<true/>
+	</dict>
+	<key>UIFileSharingEnabled</key>
+	<true/>
+	<key>UISupportedInterfaceOrientation</key>
+	<array>
+		<string>UIInterfaceOrientationPortrait</string>
+		<string>UIInterfaceOrientationLandscapeLeft</string>
+		<string>UIInterfaceOrientationLandscapeRight</string>
+	</array>
+	<key>UIBackgroundModes</key>
+	<array>
+		<string>fetch</string>
+	</array>
+	<key>NSMicrophoneUsageDescription</key>
+	<string>For testing purposes</string>
+</dict>
+</plist>
diff --git a/third_party/libwebrtc/test/ios/coverage_util_ios.h b/third_party/libwebrtc/test/ios/coverage_util_ios.h
new file mode 100644
index 0000000000..a17b69dca8
--- /dev/null
+++ b/third_party/libwebrtc/test/ios/coverage_util_ios.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_IOS_COVERAGE_UTIL_IOS_H_
+#define TEST_IOS_COVERAGE_UTIL_IOS_H_
+
+namespace rtc {
+namespace test {
+
+// In debug builds, if IOS_ENABLE_COVERAGE is defined, sets the filename of the
+// coverage file. Otherwise, it does nothing.
+void ConfigureCoverageReportPath();
+
+}  // namespace test
+}  // namespace rtc
+
+#endif  // TEST_IOS_COVERAGE_UTIL_IOS_H_
diff --git a/third_party/libwebrtc/test/ios/coverage_util_ios.mm b/third_party/libwebrtc/test/ios/coverage_util_ios.mm
new file mode 100644
index 0000000000..c21a16def2
--- /dev/null
+++ b/third_party/libwebrtc/test/ios/coverage_util_ios.mm
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#ifdef WEBRTC_IOS_ENABLE_COVERAGE
+extern "C" void __llvm_profile_set_filename(const char* name);
+#endif
+
+namespace rtc {
+namespace test {
+
+void ConfigureCoverageReportPath() {
+#ifdef WEBRTC_IOS_ENABLE_COVERAGE
+  static dispatch_once_t once_token;
+  dispatch_once(&once_token, ^{
+    // Writes the profraw file to the Documents directory, where the app has
+    // write rights.
+    NSArray* paths =
+        NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
+    NSString* documents_directory = [paths firstObject];
+    NSString* file_name = [documents_directory stringByAppendingPathComponent:@"coverage.profraw"];
+
+    // For documentation, see:
+    // http://clang.llvm.org/docs/SourceBasedCodeCoverage.html
+    __llvm_profile_set_filename([file_name cStringUsingEncoding:NSUTF8StringEncoding]);
+
+    // Print the path for easier retrieval.
+    NSLog(@"Coverage data at %@.", file_name);
+  });
+#endif  // ifdef WEBRTC_IOS_ENABLE_COVERAGE
+}
+
+}  // namespace test
+}  // namespace rtc
diff --git a/third_party/libwebrtc/test/ios/google_test_runner.mm b/third_party/libwebrtc/test/ios/google_test_runner.mm
new file mode 100644
index 0000000000..87b7f7dfd7
--- /dev/null
+++ b/third_party/libwebrtc/test/ios/google_test_runner.mm
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Copied from Chromium base/test/ios/google_test_runner.mm to avoid
+// the //base dependency. The protocol is required to run iOS Unittest.
+
+#import <UIKit/UIKit.h>
+#import <XCTest/XCTest.h>
+
+#import "test/ios/google_test_runner_delegate.h"
+
+#if !defined(__has_feature) || !__has_feature(objc_arc)
+#error "This file requires ARC support."
+#endif
+
+@interface GoogleTestRunner : XCTestCase
+@end
+
+@implementation GoogleTestRunner
+
+- (void)testRunGoogleTests {
+  self.continueAfterFailure = false;
+
+  id appDelegate = UIApplication.sharedApplication.delegate;
+  XCTAssertTrue([appDelegate conformsToProtocol:@protocol(GoogleTestRunnerDelegate)]);
+
+  id<GoogleTestRunnerDelegate> runnerDelegate =
+      static_cast<id<GoogleTestRunnerDelegate>>(appDelegate);
+  XCTAssertTrue(runnerDelegate.supportsRunningGoogleTests);
+  XCTAssertTrue([runnerDelegate runGoogleTests] == 0);
+}
+
+@end
diff --git a/third_party/libwebrtc/test/ios/google_test_runner_delegate.h b/third_party/libwebrtc/test/ios/google_test_runner_delegate.h
new file mode 100644
index 0000000000..78ee59e028
--- /dev/null
+++ b/third_party/libwebrtc/test/ios/google_test_runner_delegate.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_IOS_GOOGLE_TEST_RUNNER_DELEGATE_H_
+#define TEST_IOS_GOOGLE_TEST_RUNNER_DELEGATE_H_
+
+// Copied from Chromium base/test/ios/google_test_runner_delegate.h
+// to avoid the //base dependency. This protocol is required
+// to run iOS Unittest.
+@protocol GoogleTestRunnerDelegate
+
+// Returns YES if this delegate supports running GoogleTests via a call to
+// `runGoogleTests`.
+@property(nonatomic, readonly, assign) BOOL supportsRunningGoogleTests;
+
+// Runs GoogleTests and returns the final exit code.
+- (int)runGoogleTests;
+
+@end
+
+#endif  // TEST_IOS_GOOGLE_TEST_RUNNER_DELEGATE_H_
diff --git a/third_party/libwebrtc/test/ios/test_support.h b/third_party/libwebrtc/test/ios/test_support.h
new file mode 100644
index 0000000000..5ac731393f
--- /dev/null
+++ b/third_party/libwebrtc/test/ios/test_support.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_IOS_TEST_SUPPORT_H_
+#define TEST_IOS_TEST_SUPPORT_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+
+namespace rtc {
+namespace test {
+// Launches an iOS app that serves as a host for a test suite.
+// This is necessary as iOS doesn't like processes without a gui
+// running for longer than a few seconds.
+void RunTestsFromIOSApp();
+void InitTestSuite(int (*test_suite)(void),
+                   int argc,
+                   char* argv[],
+                   bool save_chartjson_result,
+                   bool export_perf_results_new_api,
+                   std::string webrtc_test_metrics_output_path,
+                   absl::optional<std::vector<std::string>> metrics_to_plot);
+
+// Returns true if unittests should be run by the XCTest runnner.
+bool ShouldRunIOSUnittestsWithXCTest();
+
+}  // namespace test
+}  // namespace rtc
+
+#endif  // TEST_IOS_TEST_SUPPORT_H_
diff --git a/third_party/libwebrtc/test/ios/test_support.mm b/third_party/libwebrtc/test/ios/test_support.mm
new file mode 100644
index 0000000000..d3c9ee0c74
--- /dev/null
+++ b/third_party/libwebrtc/test/ios/test_support.mm
@@ -0,0 +1,217 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+#include "api/test/metrics/chrome_perf_dashboard_metrics_exporter.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metrics_exporter.h"
+#include "api/test/metrics/metrics_set_proto_file_exporter.h"
+#include "api/test/metrics/print_result_proxy_metrics_exporter.h"
+#include "api/test/metrics/stdout_metrics_exporter.h"
+#include "test/ios/coverage_util_ios.h"
+#include "test/ios/google_test_runner_delegate.h"
+#include "test/ios/test_support.h"
+#include "test/testsupport/perf_test.h"
+
+#import "sdk/objc/helpers/NSString+StdString.h"
+
+// Springboard will kill any iOS app that fails to check in after launch within
+// a given time. Starting a UIApplication before invoking TestSuite::Run
+// prevents this from happening.
+
+// InitIOSRunHook saves the TestSuite and argc/argv, then invoking
+// RunTestsFromIOSApp calls UIApplicationMain(), providing an application
+// delegate class: WebRtcUnitTestDelegate. The delegate implements
+// application:didFinishLaunchingWithOptions: to invoke the TestSuite's Run
+// method.
+
+// Since the executable isn't likely to be a real iOS UI, the delegate puts up a
+// window displaying the app name. If a bunch of apps using MainHook are being
+// run in a row, this provides an indication of which one is currently running.
+
+// If enabled, runs unittests using the XCTest test runner.
+const char kEnableRunIOSUnittestsWithXCTest[] = "enable-run-ios-unittests-with-xctest";
+
+static int (*g_test_suite)(void) = NULL;
+static int g_argc;
+static char **g_argv;
+static bool g_write_perf_output;
+static bool g_export_perf_results_new_api;
+static std::string g_webrtc_test_metrics_output_path;
+static absl::optional<bool> g_is_xctest;
+static absl::optional<std::vector<std::string>> g_metrics_to_plot;
+
+@interface UIApplication (Testing)
+- (void)_terminateWithStatus:(int)status;
+@end
+
+@interface WebRtcUnitTestDelegate : NSObject <GoogleTestRunnerDelegate> {
+  UIWindow *_window;
+}
+- (void)runTests;
+@end
+
+@implementation WebRtcUnitTestDelegate
+
+- (BOOL)application:(UIApplication *)application
+    didFinishLaunchingWithOptions:(NSDictionary *)launchOptions {
+  CGRect bounds = [[UIScreen mainScreen] bounds];
+
+  _window = [[UIWindow alloc] initWithFrame:bounds];
+  [_window setBackgroundColor:[UIColor whiteColor]];
+  [_window makeKeyAndVisible];
+
+  // Add a label with the app name.
+  UILabel *label = [[UILabel alloc] initWithFrame:bounds];
+  label.text = [[NSProcessInfo processInfo] processName];
+  label.textAlignment = NSTextAlignmentCenter;
+  [_window addSubview:label];
+
+  // An NSInternalInconsistencyException is thrown if the app doesn't have a
+  // root view controller. Set an empty one here.
+  [_window setRootViewController:[[UIViewController alloc] init]];
+
+  if (!rtc::test::ShouldRunIOSUnittestsWithXCTest()) {
+    // When running in XCTest mode, XCTest will invoke `runGoogleTest` directly.
+    // Otherwise, schedule a call to `runTests`.
+    [self performSelector:@selector(runTests) withObject:nil afterDelay:0.1];
+  }
+
+  return YES;
+}
+
+- (BOOL)supportsRunningGoogleTests {
+  return rtc::test::ShouldRunIOSUnittestsWithXCTest();
+}
+
+- (int)runGoogleTests {
+  rtc::test::ConfigureCoverageReportPath();
+
+  int exitStatus = g_test_suite();
+
+  NSArray<NSString *> *outputDirectories =
+      NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
+  std::vector<std::unique_ptr<webrtc::test::MetricsExporter>> exporters;
+  if (g_export_perf_results_new_api) {
+    exporters.push_back(std::make_unique<webrtc::test::StdoutMetricsExporter>());
+    if (g_write_perf_output) {
+      // Stores data into a proto file under the app's document directory.
+      NSString *fileName = @"perftest-output.pb";
+      if ([outputDirectories count] != 0) {
+        NSString *outputPath = [outputDirectories[0] stringByAppendingPathComponent:fileName];
+
+        exporters.push_back(std::make_unique<webrtc::test::ChromePerfDashboardMetricsExporter>(
+            [NSString stdStringForString:outputPath]));
+      }
+    }
+    if (!g_webrtc_test_metrics_output_path.empty()) {
+      RTC_CHECK_EQ(g_webrtc_test_metrics_output_path.find('/'), std::string::npos)
+          << "On iOS, --webrtc_test_metrics_output_path must only be a file name.";
+      if ([outputDirectories count] != 0) {
+        NSString *fileName = [NSString stringWithCString:g_webrtc_test_metrics_output_path.c_str()
+                                                encoding:[NSString defaultCStringEncoding]];
+        NSString *outputPath = [outputDirectories[0] stringByAppendingPathComponent:fileName];
+        exporters.push_back(std::make_unique<webrtc::test::MetricsSetProtoFileExporter>(
+            webrtc::test::MetricsSetProtoFileExporter::Options(
+                [NSString stdStringForString:outputPath])));
+      }
+    }
+  } else {
+    exporters.push_back(std::make_unique<webrtc::test::PrintResultProxyMetricsExporter>());
+  }
+  webrtc::test::ExportPerfMetric(*webrtc::test::GetGlobalMetricsLogger(), std::move(exporters));
+  if (!g_export_perf_results_new_api) {
+    if (g_write_perf_output) {
+      // Stores data into a proto file under the app's document directory.
+      NSString *fileName = @"perftest-output.pb";
+      if ([outputDirectories count] != 0) {
+        NSString *outputPath = [outputDirectories[0] stringByAppendingPathComponent:fileName];
+
+        if (!webrtc::test::WritePerfResults([NSString stdStringForString:outputPath])) {
+          return 1;
+        }
+      }
+    }
+    if (g_metrics_to_plot) {
+      webrtc::test::PrintPlottableResults(*g_metrics_to_plot);
+    }
+  }
+
+  return exitStatus;
+}
+
+- (void)runTests {
+  RTC_DCHECK(!rtc::test::ShouldRunIOSUnittestsWithXCTest());
+  rtc::test::ConfigureCoverageReportPath();
+
+  int exitStatus = [self runGoogleTests];
+
+  // If a test app is too fast, it will exit before Instruments has has a
+  // a chance to initialize and no test results will be seen.
+  // TODO(crbug.com/137010): Figure out how much time is actually needed, and
+  // sleep only to make sure that much time has elapsed since launch.
+  [NSThread sleepUntilDate:[NSDate dateWithTimeIntervalSinceNow:2.0]];
+
+  // Use the hidden selector to try and cleanly take down the app (otherwise
+  // things can think the app crashed even on a zero exit status).
+  UIApplication *application = [UIApplication sharedApplication];
+  [application _terminateWithStatus:exitStatus];
+
+  exit(exitStatus);
+}
+
+@end
+namespace rtc {
+namespace test {
+
+// Note: This is not thread safe, and must be called from the same thread as
+// runTests above.
+void InitTestSuite(int (*test_suite)(void),
+                   int argc,
+                   char *argv[],
+                   bool write_perf_output,
+                   bool export_perf_results_new_api,
+                   std::string webrtc_test_metrics_output_path,
+                   absl::optional<std::vector<std::string>> metrics_to_plot) {
+  g_test_suite = test_suite;
+  g_argc = argc;
+  g_argv = argv;
+  g_write_perf_output = write_perf_output;
+  g_export_perf_results_new_api = export_perf_results_new_api;
+  g_webrtc_test_metrics_output_path = webrtc_test_metrics_output_path;
+  g_metrics_to_plot = std::move(metrics_to_plot);
+}
+
+void RunTestsFromIOSApp() {
+  @autoreleasepool {
+    exit(UIApplicationMain(g_argc, g_argv, nil, @"WebRtcUnitTestDelegate"));
+  }
+}
+
+bool ShouldRunIOSUnittestsWithXCTest() {
+  if (g_is_xctest.has_value()) {
+    return g_is_xctest.value();
+  }
+
+  char **argv = g_argv;
+  while (*argv != nullptr) {
+    if (strstr(*argv, kEnableRunIOSUnittestsWithXCTest) != nullptr) {
+      g_is_xctest = absl::optional<bool>(true);
+      return true;
+    }
+    argv++;
+  }
+  g_is_xctest = absl::optional<bool>(false);
+  return false;
+}
+
+}  // namespace test
+}  // namespace rtc
diff --git a/third_party/libwebrtc/test/layer_filtering_transport.cc b/third_party/libwebrtc/test/layer_filtering_transport.cc
new file mode 100644
index 0000000000..f5e5ee3002
--- /dev/null
+++ b/third_party/libwebrtc/test/layer_filtering_transport.cc
@@ -0,0 +1,186 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/layer_filtering_transport.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "api/rtp_headers.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "modules/video_coding/codecs/interface/common_constants.h"
+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+LayerFilteringTransport::LayerFilteringTransport(
+    TaskQueueBase* task_queue,
+    std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+    Call* send_call,
+    uint8_t vp8_video_payload_type,
+    uint8_t vp9_video_payload_type,
+    int selected_tl,
+    int selected_sl,
+    const std::map<uint8_t, MediaType>& payload_type_map,
+    uint32_t ssrc_to_filter_min,
+    uint32_t ssrc_to_filter_max,
+    rtc::ArrayView<const RtpExtension> audio_extensions,
+    rtc::ArrayView<const RtpExtension> video_extensions)
+    : DirectTransport(task_queue,
+                      std::move(pipe),
+                      send_call,
+                      payload_type_map,
+                      audio_extensions,
+                      video_extensions),
+      vp8_video_payload_type_(vp8_video_payload_type),
+      vp9_video_payload_type_(vp9_video_payload_type),
+      vp8_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP8)),
+      vp9_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP9)),
+      selected_tl_(selected_tl),
+      selected_sl_(selected_sl),
+      discarded_last_packet_(false),
+      ssrc_to_filter_min_(ssrc_to_filter_min),
+      ssrc_to_filter_max_(ssrc_to_filter_max) {}
+
+LayerFilteringTransport::LayerFilteringTransport(
+    TaskQueueBase* task_queue,
+    std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+    Call* send_call,
+    uint8_t vp8_video_payload_type,
+    uint8_t vp9_video_payload_type,
+    int selected_tl,
+    int selected_sl,
+    const std::map<uint8_t, MediaType>& payload_type_map,
+    rtc::ArrayView<const RtpExtension> audio_extensions,
+    rtc::ArrayView<const RtpExtension> video_extensions)
+    : LayerFilteringTransport(task_queue,
+                              std::move(pipe),
+                              send_call,
+                              vp8_video_payload_type,
+                              vp9_video_payload_type,
+                              selected_tl,
+                              selected_sl,
+                              payload_type_map,
+                              /*ssrc_to_filter_min=*/0,
+                              /*ssrc_to_filter_max=*/0xFFFFFFFF,
+                              audio_extensions,
+                              video_extensions) {}
+
+bool LayerFilteringTransport::DiscardedLastPacket() const {
+  return discarded_last_packet_;
+}
+
+bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
+                                      size_t length,
+                                      const PacketOptions& options) {
+  if (selected_tl_ == -1 && selected_sl_ == -1) {
+    // Nothing to change, forward the packet immediately.
+    return test::DirectTransport::SendRtp(packet, length, options);
+  }
+
+  RtpPacket rtp_packet;
+  rtp_packet.Parse(packet, length);
+
+  if (rtp_packet.Ssrc() < ssrc_to_filter_min_ ||
+      rtp_packet.Ssrc() > ssrc_to_filter_max_) {
+    // Nothing to change, forward the packet immediately.
+    return test::DirectTransport::SendRtp(packet, length, options);
+  }
+
+  if (rtp_packet.PayloadType() == vp8_video_payload_type_ ||
+      rtp_packet.PayloadType() == vp9_video_payload_type_) {
+    const bool is_vp8 = rtp_packet.PayloadType() == vp8_video_payload_type_;
+    VideoRtpDepacketizer& depacketizer =
+        is_vp8 ? *vp8_depacketizer_ : *vp9_depacketizer_;
+    if (auto parsed_payload = depacketizer.Parse(rtp_packet.PayloadBuffer())) {
+      int temporal_idx;
+      int spatial_idx;
+      bool non_ref_for_inter_layer_pred;
+      bool end_of_frame;
+
+      if (is_vp8) {
+        temporal_idx = absl::get<RTPVideoHeaderVP8>(
+                           parsed_payload->video_header.video_type_header)
+                           .temporalIdx;
+        spatial_idx = kNoSpatialIdx;
+        num_active_spatial_layers_ = 1;
+        non_ref_for_inter_layer_pred = false;
+        end_of_frame = true;
+      } else {
+        const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
+            parsed_payload->video_header.video_type_header);
+        temporal_idx = vp9_header.temporal_idx;
+        spatial_idx = vp9_header.spatial_idx;
+        non_ref_for_inter_layer_pred = vp9_header.non_ref_for_inter_layer_pred;
+        end_of_frame = vp9_header.end_of_frame;
+        if (vp9_header.ss_data_available) {
+          RTC_DCHECK(vp9_header.temporal_idx == kNoTemporalIdx ||
+                     vp9_header.temporal_idx == 0);
+          num_active_spatial_layers_ = vp9_header.num_spatial_layers;
+        }
+      }
+
+      if (spatial_idx == kNoSpatialIdx)
+        num_active_spatial_layers_ = 1;
+
+      RTC_CHECK_GT(num_active_spatial_layers_, 0);
+
+      if (selected_sl_ >= 0 &&
+          spatial_idx ==
+              std::min(num_active_spatial_layers_ - 1, selected_sl_) &&
+          end_of_frame) {
+        // This layer is now the last in the superframe.
+        rtp_packet.SetMarker(true);
+      } else {
+        const bool higher_temporal_layer =
+            (selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
+             temporal_idx > selected_tl_);
+
+        const bool higher_spatial_layer =
+            (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
+             spatial_idx > selected_sl_);
+
+        // Filter out non-reference lower spatial layers since they are not
+        // needed for decoding of target spatial layer.
+        const bool lower_non_ref_spatial_layer =
+            (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
+             spatial_idx <
+                 std::min(num_active_spatial_layers_ - 1, selected_sl_) &&
+             non_ref_for_inter_layer_pred);
+
+        if (higher_temporal_layer || higher_spatial_layer ||
+            lower_non_ref_spatial_layer) {
+          // Truncate packet to a padding packet.
+          rtp_packet.SetPayloadSize(0);
+          rtp_packet.SetPadding(1);
+          rtp_packet.SetMarker(false);
+          discarded_last_packet_ = true;
+        }
+      }
+    } else {
+      RTC_DCHECK_NOTREACHED() << "Parse error";
+    }
+  }
+
+  return test::DirectTransport::SendRtp(rtp_packet.data(), rtp_packet.size(),
+                                        options);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/layer_filtering_transport.h b/third_party/libwebrtc/test/layer_filtering_transport.h
new file mode 100644
index 0000000000..3aefd7159b
--- /dev/null
+++ b/third_party/libwebrtc/test/layer_filtering_transport.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_LAYER_FILTERING_TRANSPORT_H_
+#define TEST_LAYER_FILTERING_TRANSPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <map>
+#include <memory>
+
+#include "api/call/transport.h"
+#include "api/media_types.h"
+#include "call/call.h"
+#include "call/simulated_packet_receiver.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "test/direct_transport.h"
+
+namespace webrtc {
+
+namespace test {
+
+class LayerFilteringTransport : public test::DirectTransport {
+ public:
+  LayerFilteringTransport(
+      TaskQueueBase* task_queue,
+      std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+      Call* send_call,
+      uint8_t vp8_video_payload_type,
+      uint8_t vp9_video_payload_type,
+      int selected_tl,
+      int selected_sl,
+      const std::map<uint8_t, MediaType>& payload_type_map,
+      uint32_t ssrc_to_filter_min,
+      uint32_t ssrc_to_filter_max,
+      rtc::ArrayView<const RtpExtension> audio_extensions,
+      rtc::ArrayView<const RtpExtension> video_extensions);
+  LayerFilteringTransport(
+      TaskQueueBase* task_queue,
+      std::unique_ptr<SimulatedPacketReceiverInterface> pipe,
+      Call* send_call,
+      uint8_t vp8_video_payload_type,
+      uint8_t vp9_video_payload_type,
+      int selected_tl,
+      int selected_sl,
+      const std::map<uint8_t, MediaType>& payload_type_map,
+      rtc::ArrayView<const RtpExtension> audio_extensions,
+      rtc::ArrayView<const RtpExtension> video_extensions);
+  bool DiscardedLastPacket() const;
+  bool SendRtp(const uint8_t* data,
+               size_t length,
+               const PacketOptions& options) override;
+
+ private:
+  // Used to distinguish between VP8 and VP9.
+  const uint8_t vp8_video_payload_type_;
+  const uint8_t vp9_video_payload_type_;
+  const std::unique_ptr<VideoRtpDepacketizer> vp8_depacketizer_;
+  const std::unique_ptr<VideoRtpDepacketizer> vp9_depacketizer_;
+  // Discard or invalidate all temporal/spatial layers with id greater than the
+  // selected one. -1 to disable filtering.
+  const int selected_tl_;
+  const int selected_sl_;
+  bool discarded_last_packet_;
+  int num_active_spatial_layers_;
+  const uint32_t ssrc_to_filter_min_;
+  const uint32_t ssrc_to_filter_max_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_LAYER_FILTERING_TRANSPORT_H_
diff --git a/third_party/libwebrtc/test/linux/glx_renderer.cc b/third_party/libwebrtc/test/linux/glx_renderer.cc
new file mode 100644
index 0000000000..509a6c286e
--- /dev/null
+++ b/third_party/libwebrtc/test/linux/glx_renderer.cc
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/linux/glx_renderer.h"
+
+#include <X11/Xlib.h>
+#include <X11/Xutil.h>
+#include <stdlib.h>
+
+namespace webrtc {
+namespace test {
+
+GlxRenderer::GlxRenderer(size_t width, size_t height)
+    : width_(width), height_(height), display_(NULL), context_(NULL) {
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
+}
+
+GlxRenderer::~GlxRenderer() {
+  Destroy();
+}
+
+bool GlxRenderer::Init(const char* window_title) {
+  if ((display_ = XOpenDisplay(NULL)) == NULL) {
+    Destroy();
+    return false;
+  }
+
+  int screen = DefaultScreen(display_);
+
+  XVisualInfo* vi;
+  int attr_list[] = {
+      GLX_DOUBLEBUFFER, GLX_RGBA, GLX_RED_SIZE,   4,  GLX_GREEN_SIZE, 4,
+      GLX_BLUE_SIZE,    4,        GLX_DEPTH_SIZE, 16, None,
+  };
+
+  if ((vi = glXChooseVisual(display_, screen, attr_list)) == NULL) {
+    Destroy();
+    return false;
+  }
+
+  context_ = glXCreateContext(display_, vi, 0, true);
+  if (context_ == NULL) {
+    Destroy();
+    return false;
+  }
+
+  XSetWindowAttributes window_attributes;
+  window_attributes.colormap = XCreateColormap(
+      display_, RootWindow(display_, vi->screen), vi->visual, AllocNone);
+  window_attributes.border_pixel = 0;
+  window_attributes.event_mask = StructureNotifyMask | ExposureMask;
+  window_ = XCreateWindow(display_, RootWindow(display_, vi->screen), 0, 0,
+                          width_, height_, 0, vi->depth, InputOutput,
+                          vi->visual, CWBorderPixel | CWColormap | CWEventMask,
+                          &window_attributes);
+  XFree(vi);
+
+  XSetStandardProperties(display_, window_, window_title, window_title, None,
+                         NULL, 0, NULL);
+
+  Atom wm_delete = XInternAtom(display_, "WM_DELETE_WINDOW", True);
+  if (wm_delete != None) {
+    XSetWMProtocols(display_, window_, &wm_delete, 1);
+  }
+
+  XMapRaised(display_, window_);
+
+  if (!glXMakeCurrent(display_, window_, context_)) {
+    Destroy();
+    return false;
+  }
+  GlRenderer::Init();
+  if (!glXMakeCurrent(display_, None, NULL)) {
+    Destroy();
+    return false;
+  }
+
+  Resize(width_, height_);
+  return true;
+}
+
+void GlxRenderer::Destroy() {
+  if (context_ != NULL) {
+    glXMakeCurrent(display_, window_, context_);
+    GlRenderer::Destroy();
+    glXMakeCurrent(display_, None, NULL);
+    glXDestroyContext(display_, context_);
+    context_ = NULL;
+  }
+
+  if (display_ != NULL) {
+    XCloseDisplay(display_);
+    display_ = NULL;
+  }
+}
+
+GlxRenderer* GlxRenderer::Create(const char* window_title,
+                                 size_t width,
+                                 size_t height) {
+  GlxRenderer* glx_renderer = new GlxRenderer(width, height);
+  if (!glx_renderer->Init(window_title)) {
+    // TODO(pbos): Add GLX-failed warning here?
+    delete glx_renderer;
+    return NULL;
+  }
+  return glx_renderer;
+}
+
+void GlxRenderer::Resize(size_t width, size_t height) {
+  width_ = width;
+  height_ = height;
+  if (!glXMakeCurrent(display_, window_, context_)) {
+    abort();
+  }
+  GlRenderer::ResizeViewport(width_, height_);
+  if (!glXMakeCurrent(display_, None, NULL)) {
+    abort();
+  }
+
+  XSizeHints* size_hints = XAllocSizeHints();
+  if (size_hints == NULL) {
+    abort();
+  }
+  size_hints->flags = PAspect;
+  size_hints->min_aspect.x = size_hints->max_aspect.x = width_;
+  size_hints->min_aspect.y = size_hints->max_aspect.y = height_;
+  XSetWMNormalHints(display_, window_, size_hints);
+  XFree(size_hints);
+
+  XWindowChanges wc;
+  wc.width = static_cast<int>(width);
+  wc.height = static_cast<int>(height);
+  XConfigureWindow(display_, window_, CWWidth | CWHeight, &wc);
+}
+
+void GlxRenderer::OnFrame(const webrtc::VideoFrame& frame) {
+  if (static_cast<size_t>(frame.width()) != width_ ||
+      static_cast<size_t>(frame.height()) != height_) {
+    Resize(static_cast<size_t>(frame.width()),
+           static_cast<size_t>(frame.height()));
+  }
+
+  XEvent event;
+  if (!glXMakeCurrent(display_, window_, context_)) {
+    abort();
+  }
+  while (XPending(display_)) {
+    XNextEvent(display_, &event);
+    switch (event.type) {
+      case ConfigureNotify:
+        GlRenderer::ResizeViewport(event.xconfigure.width,
+                                   event.xconfigure.height);
+        break;
+      default:
+        break;
+    }
+  }
+
+  GlRenderer::OnFrame(frame);
+  glXSwapBuffers(display_, window_);
+
+  if (!glXMakeCurrent(display_, None, NULL)) {
+    abort();
+  }
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/linux/glx_renderer.h b/third_party/libwebrtc/test/linux/glx_renderer.h
new file mode 100644
index 0000000000..8add60d964
--- /dev/null
+++ b/third_party/libwebrtc/test/linux/glx_renderer.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_LINUX_GLX_RENDERER_H_
+#define TEST_LINUX_GLX_RENDERER_H_
+
+#include <GL/glx.h>
+#include <X11/X.h>
+#include <X11/Xlib.h>
+#include <stddef.h>
+
+#include "api/video/video_frame.h"
+#include "test/gl/gl_renderer.h"
+
+namespace webrtc {
+namespace test {
+
+class GlxRenderer : public GlRenderer {
+ public:
+  static GlxRenderer* Create(const char* window_title,
+                             size_t width,
+                             size_t height);
+  virtual ~GlxRenderer();
+
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+ private:
+  GlxRenderer(size_t width, size_t height);
+
+  bool Init(const char* window_title);
+  void Resize(size_t width, size_t height);
+  void Destroy();
+
+  size_t width_, height_;
+
+  Display* display_;
+  Window window_;
+  GLXContext context_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_LINUX_GLX_RENDERER_H_
diff --git a/third_party/libwebrtc/test/linux/video_renderer_linux.cc b/third_party/libwebrtc/test/linux/video_renderer_linux.cc
new file mode 100644
index 0000000000..74c95235df
--- /dev/null
+++ b/third_party/libwebrtc/test/linux/video_renderer_linux.cc
@@ -0,0 +1,28 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <stddef.h>
+
+#include "test/linux/glx_renderer.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace test {
+
+VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title,
+                                                     size_t width,
+                                                     size_t height) {
+  GlxRenderer* glx_renderer = GlxRenderer::Create(window_title, width, height);
+  if (glx_renderer != NULL) {
+    return glx_renderer;
+  }
+  return NULL;
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/logging/BUILD.gn b/third_party/libwebrtc/test/logging/BUILD.gn
new file mode 100644
index 0000000000..301c0e59c0
--- /dev/null
+++ b/third_party/libwebrtc/test/logging/BUILD.gn
@@ -0,0 +1,35 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("log_writer") {
+  testonly = true
+  visibility = [ "*" ]
+  sources = [
+    "file_log_writer.cc",
+    "file_log_writer.h",
+    "log_writer.cc",
+    "log_writer.h",
+    "memory_log_writer.cc",
+    "memory_log_writer.h",
+  ]
+
+  deps = [
+    "../../api:libjingle_logging_api",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:rtc_base_tests_utils",
+    "../../rtc_base:stringutils",
+    "../../test:fileutils",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
diff --git a/third_party/libwebrtc/test/logging/file_log_writer.cc b/third_party/libwebrtc/test/logging/file_log_writer.cc
new file mode 100644
index 0000000000..9189e1630d
--- /dev/null
+++ b/third_party/libwebrtc/test/logging/file_log_writer.cc
@@ -0,0 +1,65 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/logging/file_log_writer.h"
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace webrtc_impl {
+
+FileLogWriter::FileLogWriter(absl::string_view file_path)
+    : out_(std::fopen(std::string(file_path).c_str(), "wb")) {
+  RTC_CHECK(out_ != nullptr)
+      << "Failed to open file: '" << file_path << "' for writing.";
+}
+
+FileLogWriter::~FileLogWriter() {
+  std::fclose(out_);
+}
+
+bool FileLogWriter::IsActive() const {
+  return true;
+}
+
+bool FileLogWriter::Write(absl::string_view value) {
+  // We don't expect the write to fail. If it does, we don't want to risk
+  // silently ignoring it.
+  RTC_CHECK_EQ(std::fwrite(value.data(), 1, value.size(), out_), value.size())
+      << "fwrite failed unexpectedly: " << errno;
+  return true;
+}
+
+void FileLogWriter::Flush() {
+  RTC_CHECK_EQ(fflush(out_), 0) << "fflush failed unexpectedly: " << errno;
+}
+
+}  // namespace webrtc_impl
+
+FileLogWriterFactory::FileLogWriterFactory(absl::string_view base_path)
+    : base_path_(base_path) {
+  for (size_t i = 0; i < base_path.size(); ++i) {
+    if (base_path[i] == '/')
+      test::CreateDir(base_path.substr(0, i));
+  }
+}
+
+FileLogWriterFactory::~FileLogWriterFactory() {}
+
+std::unique_ptr<RtcEventLogOutput> FileLogWriterFactory::Create(
+    absl::string_view filename) {
+  return std::make_unique<webrtc_impl::FileLogWriter>(base_path_ +
+                                                      std::string(filename));
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/logging/file_log_writer.h b/third_party/libwebrtc/test/logging/file_log_writer.h
new file mode 100644
index 0000000000..c49b96ceff
--- /dev/null
+++ b/third_party/libwebrtc/test/logging/file_log_writer.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_LOGGING_FILE_LOG_WRITER_H_
+#define TEST_LOGGING_FILE_LOG_WRITER_H_
+
+#include <cstdio>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "test/logging/log_writer.h"
+
+namespace webrtc {
+namespace webrtc_impl {
+class FileLogWriter final : public RtcEventLogOutput {
+ public:
+  explicit FileLogWriter(absl::string_view file_path);
+  ~FileLogWriter() final;
+  bool IsActive() const override;
+  bool Write(absl::string_view value) override;
+  void Flush() override;
+
+ private:
+  std::FILE* const out_;
+};
+}  // namespace webrtc_impl
+class FileLogWriterFactory final : public LogWriterFactoryInterface {
+ public:
+  explicit FileLogWriterFactory(absl::string_view base_path);
+  ~FileLogWriterFactory() final;
+
+  std::unique_ptr<RtcEventLogOutput> Create(
+      absl::string_view filename) override;
+
+ private:
+  const std::string base_path_;
+  std::vector<std::unique_ptr<webrtc_impl::FileLogWriter>> writers_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_LOGGING_FILE_LOG_WRITER_H_
diff --git a/third_party/libwebrtc/test/logging/log_writer.cc b/third_party/libwebrtc/test/logging/log_writer.cc
new file mode 100644
index 0000000000..d9b8c1e68f
--- /dev/null
+++ b/third_party/libwebrtc/test/logging/log_writer.cc
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/logging/log_writer.h"
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+
+LogWriterFactoryAddPrefix::LogWriterFactoryAddPrefix(
+    LogWriterFactoryInterface* base,
+    absl::string_view prefix)
+    : base_factory_(base), prefix_(prefix) {}
+
+std::unique_ptr<RtcEventLogOutput> LogWriterFactoryAddPrefix::Create(
+    absl::string_view filename) {
+  return base_factory_->Create(prefix_ + std::string(filename));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/logging/log_writer.h b/third_party/libwebrtc/test/logging/log_writer.h
new file mode 100644
index 0000000000..335dab353f
--- /dev/null
+++ b/third_party/libwebrtc/test/logging/log_writer.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_LOGGING_LOG_WRITER_H_
+#define TEST_LOGGING_LOG_WRITER_H_
+
+#include <stdarg.h>
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/rtc_event_log_output.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+template <class... Args>
+inline void LogWriteFormat(RtcEventLogOutput* out_, const char* fmt, ...) {
+  va_list args, copy;
+  va_start(args, fmt);
+  va_copy(copy, args);
+  const int predicted_length = std::vsnprintf(nullptr, 0, fmt, copy);
+  va_end(copy);
+
+  RTC_DCHECK_GE(predicted_length, 0);
+  std::string out_str(predicted_length, '\0');
+  if (predicted_length > 0) {
+    // Pass "+ 1" to vsnprintf to include space for the '\0'.
+    const int actual_length =
+        std::vsnprintf(&out_str.front(), predicted_length + 1, fmt, args);
+    RTC_DCHECK_GE(actual_length, 0);
+  }
+  va_end(args);
+  out_->Write(out_str);
+}
+
+class LogWriterFactoryInterface {
+ public:
+  virtual std::unique_ptr<RtcEventLogOutput> Create(
+      absl::string_view filename) = 0;
+  virtual ~LogWriterFactoryInterface() = default;
+};
+
+class LogWriterFactoryAddPrefix : public LogWriterFactoryInterface {
+ public:
+  LogWriterFactoryAddPrefix(LogWriterFactoryInterface* base,
+                            absl::string_view prefix);
+  std::unique_ptr<RtcEventLogOutput> Create(
+      absl::string_view filename) override;
+
+ private:
+  LogWriterFactoryInterface* const base_factory_;
+  const std::string prefix_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_LOGGING_LOG_WRITER_H_
diff --git a/third_party/libwebrtc/test/logging/memory_log_writer.cc b/third_party/libwebrtc/test/logging/memory_log_writer.cc
new file mode 100644
index 0000000000..eae9223c77
--- /dev/null
+++ b/third_party/libwebrtc/test/logging/memory_log_writer.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/logging/memory_log_writer.h"
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+class MemoryLogWriter final : public RtcEventLogOutput {
+ public:
+  explicit MemoryLogWriter(std::map<std::string, std::string>* target,
+                           absl::string_view filename)
+      : target_(target), filename_(filename) {}
+  ~MemoryLogWriter() final { target_->insert({filename_, std::move(buffer_)}); }
+  bool IsActive() const override { return true; }
+  bool Write(absl::string_view value) override {
+    buffer_.append(value.data(), value.size());
+    return true;
+  }
+  void Flush() override {}
+
+ private:
+  std::map<std::string, std::string>* const target_;
+  const std::string filename_;
+  std::string buffer_;
+};
+
+class MemoryLogWriterFactory final : public LogWriterFactoryInterface {
+ public:
+  explicit MemoryLogWriterFactory(std::map<std::string, std::string>* target)
+      : target_(target) {}
+  ~MemoryLogWriterFactory() override {}
+  std::unique_ptr<RtcEventLogOutput> Create(
+      absl::string_view filename) override {
+    return std::make_unique<MemoryLogWriter>(target_, filename);
+  }
+
+ private:
+  std::map<std::string, std::string>* const target_;
+};
+
+}  // namespace
+
+MemoryLogStorage::MemoryLogStorage() {}
+
+MemoryLogStorage::~MemoryLogStorage() {}
+
+std::unique_ptr<LogWriterFactoryInterface> MemoryLogStorage::CreateFactory() {
+  return std::make_unique<MemoryLogWriterFactory>(&logs_);
+}
+
+// namespace webrtc_impl
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/logging/memory_log_writer.h b/third_party/libwebrtc/test/logging/memory_log_writer.h
new file mode 100644
index 0000000000..e795b2fd10
--- /dev/null
+++ b/third_party/libwebrtc/test/logging/memory_log_writer.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_LOGGING_MEMORY_LOG_WRITER_H_
+#define TEST_LOGGING_MEMORY_LOG_WRITER_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "test/logging/log_writer.h"
+
+namespace webrtc {
+
+// Allows creating log writer factories that creates log writers that saves
+// their content to memory. When the log writers are destroyed, their content is
+// saved to the logs_ member of this class. The intended usage is to keep this
+// class alive after the created factories and writers have been destroyed and
+// then use logs() to access all the saved logs.
+class MemoryLogStorage {
+ public:
+  MemoryLogStorage();
+  ~MemoryLogStorage();
+  std::unique_ptr<LogWriterFactoryInterface> CreateFactory();
+  const std::map<std::string, std::string>& logs() { return logs_; }
+
+ private:
+  std::map<std::string, std::string> logs_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_LOGGING_MEMORY_LOG_WRITER_H_
diff --git a/third_party/libwebrtc/test/mac/Info.plist b/third_party/libwebrtc/test/mac/Info.plist
new file mode 100644
index 0000000000..8a2b5cf0a0
--- /dev/null
+++ b/third_party/libwebrtc/test/mac/Info.plist
@@ -0,0 +1,16 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>CFBundleIdentifier</key>
+	<string>org.webrtc.video_loopback</string>
+	<key>CFBundleName</key>
+	<string>video_loopback</string>
+	<key>CFBundlePackageType</key>
+	<string>APPL</string>
+	<key>NSCameraUsageDescription</key>
+	<string>Camera access needed for video calling</string>
+	<key>NSMicrophoneUsageDescription</key>
+	<string>Microphone access needed for video calling</string>
+</dict>
+</plist>
diff --git a/third_party/libwebrtc/test/mac/run_test.mm b/third_party/libwebrtc/test/mac/run_test.mm
new file mode 100644
index 0000000000..38c6c8f8c1
--- /dev/null
+++ b/third_party/libwebrtc/test/mac/run_test.mm
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Cocoa/Cocoa.h>
+
+#include "test/run_test.h"
+
+// Converting a C++ function pointer to an Objective-C block.
+typedef void(^TestBlock)();
+TestBlock functionToBlock(void(*function)()) {
+  return [^(void) { function(); } copy];
+}
+
+// Class calling the test function on the platform specific thread.
+@interface TestRunner : NSObject {
+  BOOL running_;
+}
+- (void)runAllTests:(TestBlock)ignored;
+- (BOOL)running;
+@end
+
+@implementation TestRunner
+- (id)init {
+  self = [super init];
+  if (self) {
+    running_ = YES;
+  }
+  return self;
+}
+
+- (void)runAllTests:(TestBlock)testBlock {
+  @autoreleasepool {
+    testBlock();
+    running_ = NO;
+  }
+}
+
+- (BOOL)running {
+  return running_;
+}
+@end
+
+namespace webrtc {
+namespace test {
+
+void RunTest(void(*test)()) {
+  @autoreleasepool {
+    [NSApplication sharedApplication];
+
+    // Convert the function pointer to an Objective-C block and call on a
+    // separate thread, to avoid blocking the main thread.
+    TestRunner *testRunner = [[TestRunner alloc] init];
+    TestBlock testBlock = functionToBlock(test);
+    [NSThread detachNewThreadSelector:@selector(runAllTests:)
+                             toTarget:testRunner
+                           withObject:testBlock];
+
+    NSRunLoop *runLoop = [NSRunLoop currentRunLoop];
+    while ([testRunner running] && [runLoop runMode:NSDefaultRunLoopMode
+                                         beforeDate:[NSDate dateWithTimeIntervalSinceNow:0.1]])
+      ;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/mac/video_renderer_mac.h b/third_party/libwebrtc/test/mac/video_renderer_mac.h
new file mode 100644
index 0000000000..8e629b0a49
--- /dev/null
+++ b/third_party/libwebrtc/test/mac/video_renderer_mac.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MAC_VIDEO_RENDERER_MAC_H_
+#define TEST_MAC_VIDEO_RENDERER_MAC_H_
+
+#include "test/gl/gl_renderer.h"
+
+@class CocoaWindow;
+
+namespace webrtc {
+namespace test {
+
+class MacRenderer : public GlRenderer {
+ public:
+  MacRenderer();
+  virtual ~MacRenderer();
+
+  MacRenderer(const MacRenderer&) = delete;
+  MacRenderer& operator=(const MacRenderer&) = delete;
+
+  bool Init(const char* window_title, int width, int height);
+
+  // Implements GlRenderer.
+  void OnFrame(const VideoFrame& frame) override;
+
+ private:
+  CocoaWindow* window_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_MAC_VIDEO_RENDERER_MAC_H_
diff --git a/third_party/libwebrtc/test/mac/video_renderer_mac.mm b/third_party/libwebrtc/test/mac/video_renderer_mac.mm
new file mode 100644
index 0000000000..7103375383
--- /dev/null
+++ b/third_party/libwebrtc/test/mac/video_renderer_mac.mm
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/mac/video_renderer_mac.h"
+
+#import <Cocoa/Cocoa.h>
+
+// Creates a Cocoa Window with an OpenGL context, used together with an OpenGL
+// renderer.
+@interface CocoaWindow : NSObject {
+ @private
+  NSWindow *window_;
+  NSOpenGLContext *context_;
+  NSString *title_;
+  int width_;
+  int height_;
+}
+
+- (id)initWithTitle:(NSString *)title width:(int)width height:(int)height;
+// 'createWindow' must be called on the main thread.
+- (void)createWindow:(NSObject *)ignored;
+- (void)makeCurrentContext;
+
+@end
+
+@implementation CocoaWindow
+  static NSInteger nextXOrigin_;
+  static NSInteger nextYOrigin_;
+
+- (id)initWithTitle:(NSString *)title width:(int)width height:(int)height {
+  if (self = [super init]) {
+    title_ = title;
+    width_ = width;
+    height_ = height;
+  }
+  return self;
+}
+
+- (void)createWindow:(NSObject *)ignored {
+  NSInteger xOrigin = nextXOrigin_;
+  NSRect screenFrame = [[NSScreen mainScreen] frame];
+  if (nextXOrigin_ + width_ < screenFrame.size.width) {
+    nextXOrigin_ += width_;
+  } else {
+    xOrigin = 0;
+    nextXOrigin_ = 0;
+    nextYOrigin_ += height_;
+  }
+  if (nextYOrigin_ + height_ > screenFrame.size.height) {
+    xOrigin = 0;
+    nextXOrigin_ = 0;
+    nextYOrigin_ = 0;
+  }
+  NSInteger yOrigin = nextYOrigin_;
+  NSRect windowFrame = NSMakeRect(xOrigin, yOrigin, width_, height_);
+  window_ = [[NSWindow alloc] initWithContentRect:windowFrame
+                                        styleMask:NSWindowStyleMaskTitled
+                                          backing:NSBackingStoreBuffered
+                                            defer:NO];
+
+  NSRect viewFrame = NSMakeRect(0, 0, width_, height_);
+  NSOpenGLView *view = [[NSOpenGLView alloc] initWithFrame:viewFrame pixelFormat:nil];
+  context_ = [view openGLContext];
+
+  [[window_ contentView] addSubview:view];
+  [window_ setTitle:title_];
+  [window_ makeKeyAndOrderFront:NSApp];
+}
+
+- (void)makeCurrentContext {
+  [context_ makeCurrentContext];
+}
+
+@end
+
+namespace webrtc {
+namespace test {
+
+VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title,
+                                                     size_t width,
+                                                     size_t height) {
+  MacRenderer* renderer = new MacRenderer();
+  if (!renderer->Init(window_title, width, height)) {
+    delete renderer;
+    return NULL;
+  }
+  return renderer;
+}
+
+MacRenderer::MacRenderer()
+    : window_(NULL) {}
+
+MacRenderer::~MacRenderer() {
+  GlRenderer::Destroy();
+}
+
+bool MacRenderer::Init(const char* window_title, int width, int height) {
+  window_ = [[CocoaWindow alloc]
+      initWithTitle:[NSString stringWithUTF8String:window_title]
+                                             width:width
+                                            height:height];
+  if (!window_)
+    return false;
+  [window_ performSelectorOnMainThread:@selector(createWindow:)
+                            withObject:nil
+                         waitUntilDone:YES];
+
+  [window_ makeCurrentContext];
+  GlRenderer::Init();
+  GlRenderer::ResizeViewport(width, height);
+  return true;
+}
+
+void MacRenderer::OnFrame(const VideoFrame& frame) {
+  [window_ makeCurrentContext];
+  GlRenderer::OnFrame(frame);
+}
+
+}  // test
+}  // webrtc
diff --git a/third_party/libwebrtc/test/mac_capturer.h b/third_party/libwebrtc/test/mac_capturer.h
new file mode 100644
index 0000000000..3d7ee77b45
--- /dev/null
+++ b/third_party/libwebrtc/test/mac_capturer.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_MAC_CAPTURER_H_
+#define TEST_MAC_CAPTURER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/media_stream_interface.h"
+#include "api/scoped_refptr.h"
+#include "modules/video_capture/video_capture.h"
+#include "rtc_base/thread.h"
+#include "test/test_video_capturer.h"
+
+namespace webrtc {
+namespace test {
+
+class MacCapturer : public TestVideoCapturer,
+                    public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  static MacCapturer* Create(size_t width,
+                             size_t height,
+                             size_t target_fps,
+                             size_t capture_device_index);
+  ~MacCapturer() override;
+
+  void OnFrame(const VideoFrame& frame) override;
+
+ private:
+  MacCapturer(size_t width,
+              size_t height,
+              size_t target_fps,
+              size_t capture_device_index);
+  void Destroy();
+
+  void* capturer_;
+  void* adapter_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_MAC_CAPTURER_H_
diff --git a/third_party/libwebrtc/test/mac_capturer.mm b/third_party/libwebrtc/test/mac_capturer.mm
new file mode 100644
index 0000000000..da8e9b76b6
--- /dev/null
+++ b/third_party/libwebrtc/test/mac_capturer.mm
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/mac_capturer.h"
+
+#import "sdk/objc/base/RTCVideoCapturer.h"
+#import "sdk/objc/components/capturer/RTCCameraVideoCapturer.h"
+#import "sdk/objc/native/api/video_capturer.h"
+#import "sdk/objc/native/src/objc_frame_buffer.h"
+
+@interface RTCTestVideoSourceAdapter : NSObject <RTC_OBJC_TYPE (RTCVideoCapturerDelegate)>
+@property(nonatomic) webrtc::test::MacCapturer *capturer;
+@end
+
+@implementation RTCTestVideoSourceAdapter
+@synthesize capturer = _capturer;
+
+- (void)capturer:(RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer
+    didCaptureVideoFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  const int64_t timestamp_us = frame.timeStampNs / rtc::kNumNanosecsPerMicrosec;
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer =
+      rtc::make_ref_counted<webrtc::ObjCFrameBuffer>(frame.buffer);
+  _capturer->OnFrame(webrtc::VideoFrame::Builder()
+                         .set_video_frame_buffer(buffer)
+                         .set_rotation(webrtc::kVideoRotation_0)
+                         .set_timestamp_us(timestamp_us)
+                         .build());
+}
+
+@end
+
+namespace {
+
+AVCaptureDeviceFormat *SelectClosestFormat(AVCaptureDevice *device, size_t width, size_t height) {
+  NSArray<AVCaptureDeviceFormat *> *formats =
+      [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:device];
+  AVCaptureDeviceFormat *selectedFormat = nil;
+  int currentDiff = INT_MAX;
+  for (AVCaptureDeviceFormat *format in formats) {
+    CMVideoDimensions dimension = CMVideoFormatDescriptionGetDimensions(format.formatDescription);
+    int diff =
+        std::abs((int64_t)width - dimension.width) + std::abs((int64_t)height - dimension.height);
+    if (diff < currentDiff) {
+      selectedFormat = format;
+      currentDiff = diff;
+    }
+  }
+  return selectedFormat;
+}
+
+}  // namespace
+
+namespace webrtc {
+namespace test {
+
+MacCapturer::MacCapturer(size_t width,
+                         size_t height,
+                         size_t target_fps,
+                         size_t capture_device_index) {
+  RTCTestVideoSourceAdapter *adapter = [[RTCTestVideoSourceAdapter alloc] init];
+  adapter_ = (__bridge_retained void *)adapter;
+  adapter.capturer = this;
+
+  RTC_OBJC_TYPE(RTCCameraVideoCapturer) *capturer =
+      [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) alloc] initWithDelegate:adapter];
+  capturer_ = (__bridge_retained void *)capturer;
+
+  AVCaptureDevice *device =
+      [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) captureDevices] objectAtIndex:capture_device_index];
+  AVCaptureDeviceFormat *format = SelectClosestFormat(device, width, height);
+  [capturer startCaptureWithDevice:device format:format fps:target_fps];
+}
+
+MacCapturer *MacCapturer::Create(size_t width,
+                                 size_t height,
+                                 size_t target_fps,
+                                 size_t capture_device_index) {
+  return new MacCapturer(width, height, target_fps, capture_device_index);
+}
+
+void MacCapturer::Destroy() {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunused-variable"
+  RTCTestVideoSourceAdapter *adapter = (__bridge_transfer RTCTestVideoSourceAdapter *)adapter_;
+  RTC_OBJC_TYPE(RTCCameraVideoCapturer) *capturer =
+      (__bridge_transfer RTC_OBJC_TYPE(RTCCameraVideoCapturer) *)capturer_;
+  [capturer stopCapture];
+#pragma clang diagnostic pop
+}
+
+MacCapturer::~MacCapturer() {
+  Destroy();
+}
+
+void MacCapturer::OnFrame(const VideoFrame &frame) {
+  TestVideoCapturer::OnFrame(frame);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/mappable_native_buffer.cc b/third_party/libwebrtc/test/mappable_native_buffer.cc
new file mode 100644
index 0000000000..1b171e604b
--- /dev/null
+++ b/third_party/libwebrtc/test/mappable_native_buffer.cc
@@ -0,0 +1,185 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/mappable_native_buffer.h"
+
+#include "absl/algorithm/container.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/nv12_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_rotation.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+class NV12BufferWithDidConvertToI420 : public NV12Buffer {
+ public:
+  NV12BufferWithDidConvertToI420(int width, int height)
+      : NV12Buffer(width, height), did_convert_to_i420_(false) {}
+
+  bool did_convert_to_i420() const { return did_convert_to_i420_; }
+
+  rtc::scoped_refptr<I420BufferInterface> ToI420() override {
+    did_convert_to_i420_ = true;
+    return NV12Buffer::ToI420();
+  }
+
+ private:
+  bool did_convert_to_i420_;
+};
+
+}  // namespace
+
+VideoFrame CreateMappableNativeFrame(int64_t ntp_time_ms,
+                                     VideoFrameBuffer::Type mappable_type,
+                                     int width,
+                                     int height) {
+  VideoFrame frame =
+      VideoFrame::Builder()
+          .set_video_frame_buffer(rtc::make_ref_counted<MappableNativeBuffer>(
+              mappable_type, width, height))
+          .set_timestamp_rtp(99)
+          .set_timestamp_ms(99)
+          .set_rotation(kVideoRotation_0)
+          .build();
+  frame.set_ntp_time_ms(ntp_time_ms);
+  return frame;
+}
+
+rtc::scoped_refptr<MappableNativeBuffer> GetMappableNativeBufferFromVideoFrame(
+    const VideoFrame& frame) {
+  return rtc::scoped_refptr<MappableNativeBuffer>(
+      static_cast<MappableNativeBuffer*>(frame.video_frame_buffer().get()));
+}
+
+MappableNativeBuffer::ScaledBuffer::ScaledBuffer(
+    rtc::scoped_refptr<MappableNativeBuffer> parent,
+    int width,
+    int height)
+    : parent_(std::move(parent)), width_(width), height_(height) {}
+
+MappableNativeBuffer::ScaledBuffer::~ScaledBuffer() {}
+
+rtc::scoped_refptr<VideoFrameBuffer>
+MappableNativeBuffer::ScaledBuffer::CropAndScale(int offset_x,
+                                                 int offset_y,
+                                                 int crop_width,
+                                                 int crop_height,
+                                                 int scaled_width,
+                                                 int scaled_height) {
+  return rtc::make_ref_counted<ScaledBuffer>(parent_, scaled_width,
+                                             scaled_height);
+}
+
+rtc::scoped_refptr<I420BufferInterface>
+MappableNativeBuffer::ScaledBuffer::ToI420() {
+  return parent_->GetOrCreateMappedBuffer(width_, height_)->ToI420();
+}
+
+rtc::scoped_refptr<VideoFrameBuffer>
+MappableNativeBuffer::ScaledBuffer::GetMappedFrameBuffer(
+    rtc::ArrayView<VideoFrameBuffer::Type> types) {
+  if (absl::c_find(types, parent_->mappable_type_) == types.end())
+    return nullptr;
+  return parent_->GetOrCreateMappedBuffer(width_, height_);
+}
+
+MappableNativeBuffer::MappableNativeBuffer(VideoFrameBuffer::Type mappable_type,
+                                           int width,
+                                           int height)
+    : mappable_type_(mappable_type), width_(width), height_(height) {
+  RTC_DCHECK(mappable_type_ == VideoFrameBuffer::Type::kI420 ||
+             mappable_type_ == VideoFrameBuffer::Type::kNV12);
+}
+
+MappableNativeBuffer::~MappableNativeBuffer() {}
+
+rtc::scoped_refptr<VideoFrameBuffer> MappableNativeBuffer::CropAndScale(
+    int offset_x,
+    int offset_y,
+    int crop_width,
+    int crop_height,
+    int scaled_width,
+    int scaled_height) {
+  return FullSizeBuffer()->CropAndScale(
+      offset_x, offset_y, crop_width, crop_height, scaled_width, scaled_height);
+}
+
+rtc::scoped_refptr<I420BufferInterface> MappableNativeBuffer::ToI420() {
+  return FullSizeBuffer()->ToI420();
+}
+
+rtc::scoped_refptr<VideoFrameBuffer> MappableNativeBuffer::GetMappedFrameBuffer(
+    rtc::ArrayView<VideoFrameBuffer::Type> types) {
+  return FullSizeBuffer()->GetMappedFrameBuffer(types);
+}
+
+std::vector<rtc::scoped_refptr<VideoFrameBuffer>>
+MappableNativeBuffer::GetMappedFramedBuffers() const {
+  MutexLock lock(&lock_);
+  return mapped_buffers_;
+}
+
+bool MappableNativeBuffer::DidConvertToI420() const {
+  if (mappable_type_ != VideoFrameBuffer::Type::kNV12)
+    return false;
+  MutexLock lock(&lock_);
+  for (auto& mapped_buffer : mapped_buffers_) {
+    if (static_cast<NV12BufferWithDidConvertToI420*>(mapped_buffer.get())
+            ->did_convert_to_i420()) {
+      return true;
+    }
+  }
+  return false;
+}
+
+rtc::scoped_refptr<MappableNativeBuffer::ScaledBuffer>
+MappableNativeBuffer::FullSizeBuffer() {
+  return rtc::make_ref_counted<ScaledBuffer>(
+      rtc::scoped_refptr<MappableNativeBuffer>(this), width_, height_);
+}
+
+rtc::scoped_refptr<VideoFrameBuffer>
+MappableNativeBuffer::GetOrCreateMappedBuffer(int width, int height) {
+  MutexLock lock(&lock_);
+  for (auto& mapped_buffer : mapped_buffers_) {
+    if (mapped_buffer->width() == width && mapped_buffer->height() == height) {
+      return mapped_buffer;
+    }
+  }
+  rtc::scoped_refptr<VideoFrameBuffer> mapped_buffer;
+  switch (mappable_type_) {
+    case VideoFrameBuffer::Type::kI420: {
+      rtc::scoped_refptr<I420Buffer> i420_buffer =
+          I420Buffer::Create(width, height);
+      I420Buffer::SetBlack(i420_buffer.get());
+      mapped_buffer = i420_buffer;
+      break;
+    }
+    case VideoFrameBuffer::Type::kNV12: {
+      auto nv12_buffer =
+          rtc::make_ref_counted<NV12BufferWithDidConvertToI420>(width, height);
+      nv12_buffer->InitializeData();
+      mapped_buffer = std::move(nv12_buffer);
+      break;
+    }
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  mapped_buffers_.push_back(mapped_buffer);
+  return mapped_buffer;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/mappable_native_buffer.h b/third_party/libwebrtc/test/mappable_native_buffer.h
new file mode 100644
index 0000000000..08f155e07f
--- /dev/null
+++ b/third_party/libwebrtc/test/mappable_native_buffer.h
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MAPPABLE_NATIVE_BUFFER_H_
+#define TEST_MAPPABLE_NATIVE_BUFFER_H_
+
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/video/video_frame.h"
+#include "common_video/include/video_frame_buffer.h"
+#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace test {
+
+class MappableNativeBuffer;
+
+VideoFrame CreateMappableNativeFrame(int64_t ntp_time_ms,
+                                     VideoFrameBuffer::Type mappable_type,
+                                     int width,
+                                     int height);
+
+rtc::scoped_refptr<MappableNativeBuffer> GetMappableNativeBufferFromVideoFrame(
+    const VideoFrame& frame);
+
+// A for-testing native buffer that is scalable and mappable. The contents of
+// the buffer is black and the pixels are created upon mapping. Mapped buffers
+// are stored inside MappableNativeBuffer, allowing tests to verify which
+// resolutions were mapped, e.g. when passing them in to an encoder or other
+// modules.
+class MappableNativeBuffer : public VideoFrameBuffer {
+ public:
+  // If `allow_i420_conversion` is false, calling ToI420() on a non-I420 buffer
+  // will DCHECK-crash. Used to ensure zero-copy in tests.
+  MappableNativeBuffer(VideoFrameBuffer::Type mappable_type,
+                       int width,
+                       int height);
+  ~MappableNativeBuffer() override;
+
+  VideoFrameBuffer::Type mappable_type() const { return mappable_type_; }
+
+  VideoFrameBuffer::Type type() const override { return Type::kNative; }
+  int width() const override { return width_; }
+  int height() const override { return height_; }
+
+  rtc::scoped_refptr<VideoFrameBuffer> CropAndScale(int offset_x,
+                                                    int offset_y,
+                                                    int crop_width,
+                                                    int crop_height,
+                                                    int scaled_width,
+                                                    int scaled_height) override;
+
+  rtc::scoped_refptr<I420BufferInterface> ToI420() override;
+  rtc::scoped_refptr<VideoFrameBuffer> GetMappedFrameBuffer(
+      rtc::ArrayView<VideoFrameBuffer::Type> types) override;
+
+  // Gets all the buffers that have been mapped so far, including mappings of
+  // cropped and scaled buffers.
+  std::vector<rtc::scoped_refptr<VideoFrameBuffer>> GetMappedFramedBuffers()
+      const;
+  bool DidConvertToI420() const;
+
+ private:
+  friend class rtc::RefCountedObject<MappableNativeBuffer>;
+
+  class ScaledBuffer : public VideoFrameBuffer {
+   public:
+    ScaledBuffer(rtc::scoped_refptr<MappableNativeBuffer> parent,
+                 int width,
+                 int height);
+    ~ScaledBuffer() override;
+
+    VideoFrameBuffer::Type type() const override { return Type::kNative; }
+    int width() const override { return width_; }
+    int height() const override { return height_; }
+
+    rtc::scoped_refptr<VideoFrameBuffer> CropAndScale(
+        int offset_x,
+        int offset_y,
+        int crop_width,
+        int crop_height,
+        int scaled_width,
+        int scaled_height) override;
+
+    rtc::scoped_refptr<I420BufferInterface> ToI420() override;
+    rtc::scoped_refptr<VideoFrameBuffer> GetMappedFrameBuffer(
+        rtc::ArrayView<VideoFrameBuffer::Type> types) override;
+
+   private:
+    friend class rtc::RefCountedObject<ScaledBuffer>;
+
+    const rtc::scoped_refptr<MappableNativeBuffer> parent_;
+    const int width_;
+    const int height_;
+  };
+
+  rtc::scoped_refptr<ScaledBuffer> FullSizeBuffer();
+  rtc::scoped_refptr<VideoFrameBuffer> GetOrCreateMappedBuffer(int width,
+                                                               int height);
+
+  const VideoFrameBuffer::Type mappable_type_;
+  const int width_;
+  const int height_;
+  mutable Mutex lock_;
+  std::vector<rtc::scoped_refptr<VideoFrameBuffer>> mapped_buffers_
+      RTC_GUARDED_BY(&lock_);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  //  TEST_MAPPABLE_NATIVE_BUFFER_H_
diff --git a/third_party/libwebrtc/test/mock_audio_decoder.cc b/third_party/libwebrtc/test/mock_audio_decoder.cc
new file mode 100644
index 0000000000..5af9f370cd
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_decoder.cc
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/mock_audio_decoder.h"
+
+namespace webrtc {
+
+MockAudioDecoder::MockAudioDecoder() = default;
+MockAudioDecoder::~MockAudioDecoder() {
+  Die();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/mock_audio_decoder.h b/third_party/libwebrtc/test/mock_audio_decoder.h
new file mode 100644
index 0000000000..8f44bf891d
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_decoder.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MOCK_AUDIO_DECODER_H_
+#define TEST_MOCK_AUDIO_DECODER_H_
+
+#include "api/audio_codecs/audio_decoder.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioDecoder : public AudioDecoder {
+ public:
+  MockAudioDecoder();
+  ~MockAudioDecoder();
+  MOCK_METHOD(void, Die, ());
+  MOCK_METHOD(int,
+              DecodeInternal,
+              (const uint8_t*, size_t, int, int16_t*, SpeechType*),
+              (override));
+  MOCK_METHOD(bool, HasDecodePlc, (), (const, override));
+  MOCK_METHOD(size_t, DecodePlc, (size_t, int16_t*), (override));
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(int, ErrorCode, (), (override));
+  MOCK_METHOD(int, PacketDuration, (const uint8_t*, size_t), (const, override));
+  MOCK_METHOD(size_t, Channels, (), (const, override));
+  MOCK_METHOD(int, SampleRateHz, (), (const, override));
+};
+
+}  // namespace webrtc
+#endif  // TEST_MOCK_AUDIO_DECODER_H_
diff --git a/third_party/libwebrtc/test/mock_audio_decoder_factory.h b/third_party/libwebrtc/test/mock_audio_decoder_factory.h
new file mode 100644
index 0000000000..425ea38f9c
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_decoder_factory.h
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MOCK_AUDIO_DECODER_FACTORY_H_
+#define TEST_MOCK_AUDIO_DECODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioDecoderFactory : public AudioDecoderFactory {
+ public:
+  MOCK_METHOD(std::vector<AudioCodecSpec>,
+              GetSupportedDecoders,
+              (),
+              (override));
+  MOCK_METHOD(bool, IsSupportedDecoder, (const SdpAudioFormat&), (override));
+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) override {
+    std::unique_ptr<AudioDecoder> return_value;
+    MakeAudioDecoderMock(format, codec_pair_id, &return_value);
+    return return_value;
+  }
+  MOCK_METHOD(void,
+              MakeAudioDecoderMock,
+              (const SdpAudioFormat& format,
+               absl::optional<AudioCodecPairId> codec_pair_id,
+               std::unique_ptr<AudioDecoder>*));
+
+  // Creates a MockAudioDecoderFactory with no formats and that may not be
+  // invoked to create a codec - useful for initializing a voice engine, for
+  // example.
+  static rtc::scoped_refptr<webrtc::MockAudioDecoderFactory>
+  CreateUnusedFactory() {
+    using ::testing::_;
+    using ::testing::AnyNumber;
+    using ::testing::Return;
+
+    rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> factory =
+        rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>();
+    ON_CALL(*factory.get(), GetSupportedDecoders())
+        .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
+    EXPECT_CALL(*factory.get(), GetSupportedDecoders()).Times(AnyNumber());
+    ON_CALL(*factory, IsSupportedDecoder(_)).WillByDefault(Return(false));
+    EXPECT_CALL(*factory, IsSupportedDecoder(_)).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), MakeAudioDecoderMock(_, _, _)).Times(0);
+    return factory;
+  }
+
+  // Creates a MockAudioDecoderFactory with no formats that may be invoked to
+  // create a codec any number of times. It will, though, return nullptr on each
+  // call, since it supports no codecs.
+  static rtc::scoped_refptr<webrtc::MockAudioDecoderFactory>
+  CreateEmptyFactory() {
+    using ::testing::_;
+    using ::testing::AnyNumber;
+    using ::testing::Return;
+    using ::testing::SetArgPointee;
+
+    rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> factory =
+        rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>();
+    ON_CALL(*factory.get(), GetSupportedDecoders())
+        .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
+    EXPECT_CALL(*factory.get(), GetSupportedDecoders()).Times(AnyNumber());
+    ON_CALL(*factory, IsSupportedDecoder(_)).WillByDefault(Return(false));
+    EXPECT_CALL(*factory, IsSupportedDecoder(_)).Times(AnyNumber());
+    ON_CALL(*factory.get(), MakeAudioDecoderMock(_, _, _))
+        .WillByDefault(SetArgPointee<2>(nullptr));
+    EXPECT_CALL(*factory.get(), MakeAudioDecoderMock(_, _, _))
+        .Times(AnyNumber());
+    return factory;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_AUDIO_DECODER_FACTORY_H_
diff --git a/third_party/libwebrtc/test/mock_audio_encoder.cc b/third_party/libwebrtc/test/mock_audio_encoder.cc
new file mode 100644
index 0000000000..36615111a5
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_encoder.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/mock_audio_encoder.h"
+
+namespace webrtc {
+
+MockAudioEncoder::MockAudioEncoder() = default;
+MockAudioEncoder::~MockAudioEncoder() = default;
+
+MockAudioEncoder::FakeEncoding::FakeEncoding(
+    const AudioEncoder::EncodedInfo& info)
+    : info_(info) {}
+
+MockAudioEncoder::FakeEncoding::FakeEncoding(size_t encoded_bytes) {
+  info_.encoded_bytes = encoded_bytes;
+}
+
+AudioEncoder::EncodedInfo MockAudioEncoder::FakeEncoding::operator()(
+    uint32_t timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  encoded->SetSize(encoded->size() + info_.encoded_bytes);
+  return info_;
+}
+
+MockAudioEncoder::CopyEncoding::~CopyEncoding() = default;
+
+MockAudioEncoder::CopyEncoding::CopyEncoding(
+    AudioEncoder::EncodedInfo info,
+    rtc::ArrayView<const uint8_t> payload)
+    : info_(info), payload_(payload) {}
+
+MockAudioEncoder::CopyEncoding::CopyEncoding(
+    rtc::ArrayView<const uint8_t> payload)
+    : payload_(payload) {
+  info_.encoded_bytes = payload_.size();
+}
+
+AudioEncoder::EncodedInfo MockAudioEncoder::CopyEncoding::operator()(
+    uint32_t timestamp,
+    rtc::ArrayView<const int16_t> audio,
+    rtc::Buffer* encoded) {
+  RTC_CHECK(encoded);
+  RTC_CHECK_LE(info_.encoded_bytes, payload_.size());
+  encoded->AppendData(payload_.data(), info_.encoded_bytes);
+  return info_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/mock_audio_encoder.h b/third_party/libwebrtc/test/mock_audio_encoder.h
new file mode 100644
index 0000000000..1f4510e885
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_encoder.h
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MOCK_AUDIO_ENCODER_H_
+#define TEST_MOCK_AUDIO_ENCODER_H_
+
+#include <string>
+
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioEncoder : public AudioEncoder {
+ public:
+  MockAudioEncoder();
+  ~MockAudioEncoder();
+  MOCK_METHOD(int, SampleRateHz, (), (const, override));
+  MOCK_METHOD(size_t, NumChannels, (), (const, override));
+  MOCK_METHOD(int, RtpTimestampRateHz, (), (const, override));
+  MOCK_METHOD(size_t, Num10MsFramesInNextPacket, (), (const, override));
+  MOCK_METHOD(size_t, Max10MsFramesInAPacket, (), (const, override));
+  MOCK_METHOD(int, GetTargetBitrate, (), (const, override));
+  MOCK_METHOD((absl::optional<std::pair<TimeDelta, TimeDelta>>),
+              GetFrameLengthRange,
+              (),
+              (const, override));
+
+  MOCK_METHOD(void, Reset, (), (override));
+  MOCK_METHOD(bool, SetFec, (bool enable), (override));
+  MOCK_METHOD(bool, SetDtx, (bool enable), (override));
+  MOCK_METHOD(bool, SetApplication, (Application application), (override));
+  MOCK_METHOD(void, SetMaxPlaybackRate, (int frequency_hz), (override));
+  MOCK_METHOD(void,
+              OnReceivedUplinkBandwidth,
+              (int target_audio_bitrate_bps,
+               absl::optional<int64_t> probing_interval_ms),
+              (override));
+  MOCK_METHOD(void,
+              OnReceivedUplinkPacketLossFraction,
+              (float uplink_packet_loss_fraction),
+              (override));
+  MOCK_METHOD(void,
+              OnReceivedOverhead,
+              (size_t overhead_bytes_per_packet),
+              (override));
+
+  MOCK_METHOD(bool,
+              EnableAudioNetworkAdaptor,
+              (const std::string& config_string, RtcEventLog*),
+              (override));
+
+  // Note, we explicitly chose not to create a mock for the Encode method.
+  MOCK_METHOD(EncodedInfo,
+              EncodeImpl,
+              (uint32_t timestamp,
+               rtc::ArrayView<const int16_t> audio,
+               rtc::Buffer*),
+              (override));
+
+  class FakeEncoding {
+   public:
+    // Creates a functor that will return `info` and adjust the rtc::Buffer
+    // given as input to it, so it is info.encoded_bytes larger.
+    explicit FakeEncoding(const AudioEncoder::EncodedInfo& info);
+
+    // Shorthand version of the constructor above, for when only setting
+    // encoded_bytes in the EncodedInfo object matters.
+    explicit FakeEncoding(size_t encoded_bytes);
+
+    AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
+                                         rtc::ArrayView<const int16_t> audio,
+                                         rtc::Buffer* encoded);
+
+   private:
+    AudioEncoder::EncodedInfo info_;
+  };
+
+  class CopyEncoding {
+   public:
+    ~CopyEncoding();
+
+    // Creates a functor that will return `info` and append the data in the
+    // payload to the buffer given as input to it. Up to info.encoded_bytes are
+    // appended - make sure the payload is big enough!  Since it uses an
+    // ArrayView, it _does not_ copy the payload. Make sure it doesn't fall out
+    // of scope!
+    CopyEncoding(AudioEncoder::EncodedInfo info,
+                 rtc::ArrayView<const uint8_t> payload);
+
+    // Shorthand version of the constructor above, for when you wish to append
+    // the whole payload and do not care about any EncodedInfo attribute other
+    // than encoded_bytes.
+    explicit CopyEncoding(rtc::ArrayView<const uint8_t> payload);
+
+    AudioEncoder::EncodedInfo operator()(uint32_t timestamp,
+                                         rtc::ArrayView<const int16_t> audio,
+                                         rtc::Buffer* encoded);
+
+   private:
+    AudioEncoder::EncodedInfo info_;
+    rtc::ArrayView<const uint8_t> payload_;
+  };
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_AUDIO_ENCODER_H_
diff --git a/third_party/libwebrtc/test/mock_audio_encoder_factory.h b/third_party/libwebrtc/test/mock_audio_encoder_factory.h
new file mode 100644
index 0000000000..eaa5b8f17d
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_audio_encoder_factory.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MOCK_AUDIO_ENCODER_FACTORY_H_
+#define TEST_MOCK_AUDIO_ENCODER_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockAudioEncoderFactory
+    : public ::testing::NiceMock<AudioEncoderFactory> {
+ public:
+  MOCK_METHOD(std::vector<AudioCodecSpec>,
+              GetSupportedEncoders,
+              (),
+              (override));
+  MOCK_METHOD(absl::optional<AudioCodecInfo>,
+              QueryAudioEncoder,
+              (const SdpAudioFormat& format),
+              (override));
+
+  std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      int payload_type,
+      const SdpAudioFormat& format,
+      absl::optional<AudioCodecPairId> codec_pair_id) override {
+    std::unique_ptr<AudioEncoder> return_value;
+    MakeAudioEncoderMock(payload_type, format, codec_pair_id, &return_value);
+    return return_value;
+  }
+  MOCK_METHOD(void,
+              MakeAudioEncoderMock,
+              (int payload_type,
+               const SdpAudioFormat& format,
+               absl::optional<AudioCodecPairId> codec_pair_id,
+               std::unique_ptr<AudioEncoder>*));
+
+  // Creates a MockAudioEncoderFactory with no formats and that may not be
+  // invoked to create a codec - useful for initializing a voice engine, for
+  // example.
+  static rtc::scoped_refptr<webrtc::MockAudioEncoderFactory>
+  CreateUnusedFactory() {
+    using ::testing::_;
+    using ::testing::AnyNumber;
+    using ::testing::Return;
+
+    auto factory = rtc::make_ref_counted<webrtc::MockAudioEncoderFactory>();
+    ON_CALL(*factory.get(), GetSupportedEncoders())
+        .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
+    ON_CALL(*factory.get(), QueryAudioEncoder(_))
+        .WillByDefault(Return(absl::nullopt));
+
+    EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _)).Times(0);
+    return factory;
+  }
+
+  // Creates a MockAudioEncoderFactory with no formats that may be invoked to
+  // create a codec any number of times. It will, though, return nullptr on each
+  // call, since it supports no codecs.
+  static rtc::scoped_refptr<webrtc::MockAudioEncoderFactory>
+  CreateEmptyFactory() {
+    using ::testing::_;
+    using ::testing::AnyNumber;
+    using ::testing::Return;
+    using ::testing::SetArgPointee;
+
+    auto factory = rtc::make_ref_counted<webrtc::MockAudioEncoderFactory>();
+    ON_CALL(*factory.get(), GetSupportedEncoders())
+        .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
+    ON_CALL(*factory.get(), QueryAudioEncoder(_))
+        .WillByDefault(Return(absl::nullopt));
+    ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
+        .WillByDefault(SetArgPointee<3>(nullptr));
+
+    EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
+    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
+        .Times(AnyNumber());
+    return factory;
+  }
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_AUDIO_ENCODER_FACTORY_H_
diff --git a/third_party/libwebrtc/test/mock_frame_transformer.h b/third_party/libwebrtc/test/mock_frame_transformer.h
new file mode 100644
index 0000000000..617cda8a43
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_frame_transformer.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MOCK_FRAME_TRANSFORMER_H_
+#define TEST_MOCK_FRAME_TRANSFORMER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/frame_transformer_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockFrameTransformer : public FrameTransformerInterface {
+ public:
+  MOCK_METHOD(void,
+              Transform,
+              (std::unique_ptr<TransformableFrameInterface>),
+              (override));
+  MOCK_METHOD(void,
+              RegisterTransformedFrameCallback,
+              (rtc::scoped_refptr<TransformedFrameCallback>),
+              (override));
+  MOCK_METHOD(void,
+              RegisterTransformedFrameSinkCallback,
+              (rtc::scoped_refptr<TransformedFrameCallback>, uint32_t),
+              (override));
+  MOCK_METHOD(void, UnregisterTransformedFrameCallback, (), (override));
+  MOCK_METHOD(void,
+              UnregisterTransformedFrameSinkCallback,
+              (uint32_t),
+              (override));
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_FRAME_TRANSFORMER_H_
diff --git a/third_party/libwebrtc/test/mock_transformable_frame.h b/third_party/libwebrtc/test/mock_transformable_frame.h
new file mode 100644
index 0000000000..039013f218
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_transformable_frame.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MOCK_TRANSFORMABLE_FRAME_H_
+#define TEST_MOCK_TRANSFORMABLE_FRAME_H_
+
+#include "api/frame_transformer_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockTransformableFrame : public TransformableFrameInterface {
+ public:
+  MOCK_METHOD(rtc::ArrayView<const uint8_t>, GetData, (), (const, override));
+  MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t>), (override));
+  MOCK_METHOD(uint8_t, GetPayloadType, (), (const, override));
+  MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));
+  MOCK_METHOD(uint32_t, GetTimestamp, (), (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_TRANSFORMABLE_FRAME_H_
diff --git a/third_party/libwebrtc/test/mock_transport.cc b/third_party/libwebrtc/test/mock_transport.cc
new file mode 100644
index 0000000000..3878b3bdbe
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_transport.cc
@@ -0,0 +1,18 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/mock_transport.h"
+
+namespace webrtc {
+
+MockTransport::MockTransport() = default;
+MockTransport::~MockTransport() = default;
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/mock_transport.h b/third_party/libwebrtc/test/mock_transport.h
new file mode 100644
index 0000000000..9c4dc4bf8d
--- /dev/null
+++ b/third_party/libwebrtc/test/mock_transport.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_MOCK_TRANSPORT_H_
+#define TEST_MOCK_TRANSPORT_H_
+
+#include "api/call/transport.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockTransport : public Transport {
+ public:
+  MockTransport();
+  ~MockTransport();
+
+  MOCK_METHOD(bool,
+              SendRtp,
+              (const uint8_t*, size_t, const PacketOptions&),
+              (override));
+  MOCK_METHOD(bool, SendRtcp, (const uint8_t*, size_t len), (override));
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_MOCK_TRANSPORT_H_
diff --git a/third_party/libwebrtc/test/network/BUILD.gn b/third_party/libwebrtc/test/network/BUILD.gn
new file mode 100644
index 0000000000..20b17bc804
--- /dev/null
+++ b/third_party/libwebrtc/test/network/BUILD.gn
@@ -0,0 +1,199 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("emulated_network") {
+  visibility = [
+    ":*",
+    "../../api:create_network_emulation_manager",
+    "../../api/test/network_emulation:create_cross_traffic",
+  ]
+  if (rtc_include_tests) {
+    visibility += [
+      "../peer_scenario:*",
+      "../scenario:*",
+    ]
+  }
+  testonly = true
+  sources = [
+    "cross_traffic.cc",
+    "cross_traffic.h",
+    "emulated_network_manager.cc",
+    "emulated_network_manager.h",
+    "emulated_turn_server.cc",
+    "emulated_turn_server.h",
+    "fake_network_socket_server.cc",
+    "fake_network_socket_server.h",
+    "network_emulation.cc",
+    "network_emulation.h",
+    "network_emulation_manager.cc",
+    "network_emulation_manager.h",
+    "traffic_route.cc",
+    "traffic_route.h",
+  ]
+  deps = [
+    "../../api:array_view",
+    "../../api:field_trials_view",
+    "../../api:network_emulation_manager_api",
+    "../../api:packet_socket_factory",
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../api:simulated_network_api",
+    "../../api:time_controller",
+    "../../api/numerics",
+    "../../api/task_queue:pending_task_safety_flag",
+    "../../api/test/network_emulation",
+    "../../api/transport:stun_types",
+    "../../api/units:data_rate",
+    "../../api/units:data_size",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../call:simulated_network",
+    "../../p2p:p2p_server_utils",
+    "../../p2p:rtc_p2p",
+    "../../rtc_base:async_packet_socket",
+    "../../rtc_base:copy_on_write_buffer",
+    "../../rtc_base:ip_address",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:network",
+    "../../rtc_base:network_constants",
+    "../../rtc_base:random",
+    "../../rtc_base:rtc_base_tests_utils",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:rtc_task_queue",
+    "../../rtc_base:safe_minmax",
+    "../../rtc_base:socket",
+    "../../rtc_base:socket_address",
+    "../../rtc_base:socket_server",
+    "../../rtc_base:stringutils",
+    "../../rtc_base:task_queue_for_test",
+    "../../rtc_base:threading",
+    "../../rtc_base/memory:always_valid_pointer",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/system:no_unique_address",
+    "../../rtc_base/task_utils:repeating_task",
+    "../../system_wrappers",
+    "../../test:scoped_key_value_config",
+    "../scenario:column_printer",
+    "../time_controller",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("network_emulation_unittest") {
+  testonly = true
+  sources = [ "network_emulation_unittest.cc" ]
+  deps = [
+    ":emulated_network",
+    "../:test_support",
+    "../../api:simulated_network_api",
+    "../../api/units:time_delta",
+    "../../call:simulated_network",
+    "../../rtc_base:gunit_helpers",
+    "../../rtc_base:logging",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base:task_queue_for_test",
+    "../../rtc_base/synchronization:mutex",
+  ]
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  rtc_library("network_emulation_pc_unittest") {
+    testonly = true
+    sources = [ "network_emulation_pc_unittest.cc" ]
+    deps = [
+      ":emulated_network",
+      "../:test_support",
+      "../../api:callfactory_api",
+      "../../api:libjingle_peerconnection_api",
+      "../../api:scoped_refptr",
+      "../../api:simulated_network_api",
+      "../../api/rtc_event_log:rtc_event_log_factory",
+      "../../api/task_queue:default_task_queue_factory",
+      "../../api/transport:field_trial_based_config",
+      "../../call:simulated_network",
+      "../../media:rtc_audio_video",
+      "../../media:rtc_media_engine_defaults",
+      "../../modules/audio_device:audio_device_impl",
+      "../../p2p:rtc_p2p",
+      "../../pc:pc_test_utils",
+      "../../pc:peerconnection_wrapper",
+      "../../rtc_base:gunit_helpers",
+      "../../rtc_base:logging",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:task_queue_for_test",
+    ]
+  }
+}
+
+rtc_library("cross_traffic_unittest") {
+  testonly = true
+  sources = [ "cross_traffic_unittest.cc" ]
+  deps = [
+    ":emulated_network",
+    "../:test_support",
+    "../../api:network_emulation_manager_api",
+    "../../api:simulated_network_api",
+    "../../call:simulated_network",
+    "../../rtc_base:logging",
+    "../../rtc_base:network_constants",
+    "../../rtc_base:rtc_event",
+    "../time_controller",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("feedback_generator") {
+    testonly = true
+    sources = [
+      "feedback_generator.cc",
+      "feedback_generator.h",
+    ]
+    deps = [
+      ":emulated_network",
+      "../../api/transport:network_control",
+      "../../api/transport:test_feedback_generator_interface",
+      "../../call:simulated_network",
+      "../../rtc_base:checks",
+      "../time_controller",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+  }
+
+  rtc_library("feedback_generator_unittest") {
+    testonly = true
+    sources = [ "feedback_generator_unittest.cc" ]
+    deps = [
+      "../:test_support",
+      "../../api/transport:test_feedback_generator",
+    ]
+  }
+
+  if (!build_with_chromium) {
+    rtc_library("network_emulation_unittests") {
+      testonly = true
+      deps = [
+        ":cross_traffic_unittest",
+        ":feedback_generator_unittest",
+        ":network_emulation_pc_unittest",
+        ":network_emulation_unittest",
+      ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/test/network/OWNERS b/third_party/libwebrtc/test/network/OWNERS
new file mode 100644
index 0000000000..b177c4eec5
--- /dev/null
+++ b/third_party/libwebrtc/test/network/OWNERS
@@ -0,0 +1 @@
+titovartem@webrtc.org
diff --git a/third_party/libwebrtc/test/network/cross_traffic.cc b/third_party/libwebrtc/test/network/cross_traffic.cc
new file mode 100644
index 0000000000..0a817a2d39
--- /dev/null
+++ b/third_party/libwebrtc/test/network/cross_traffic.cc
@@ -0,0 +1,322 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/cross_traffic.h"
+
+#include <math.h>
+
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "cross_traffic.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace test {
+
+RandomWalkCrossTraffic::RandomWalkCrossTraffic(RandomWalkConfig config,
+                                               CrossTrafficRoute* traffic_route)
+    : config_(config),
+      traffic_route_(traffic_route),
+      random_(config_.random_seed) {
+  sequence_checker_.Detach();
+}
+RandomWalkCrossTraffic::~RandomWalkCrossTraffic() = default;
+
+void RandomWalkCrossTraffic::Process(Timestamp at_time) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (last_process_time_.IsMinusInfinity()) {
+    last_process_time_ = at_time;
+  }
+  TimeDelta delta = at_time - last_process_time_;
+  last_process_time_ = at_time;
+
+  if (at_time - last_update_time_ >= config_.update_interval) {
+    intensity_ += random_.Gaussian(config_.bias, config_.variance) *
+                  sqrt((at_time - last_update_time_).seconds<double>());
+    intensity_ = rtc::SafeClamp(intensity_, 0.0, 1.0);
+    last_update_time_ = at_time;
+  }
+  pending_size_ += TrafficRate() * delta;
+
+  if (pending_size_ >= config_.min_packet_size &&
+      at_time >= last_send_time_ + config_.min_packet_interval) {
+    traffic_route_->SendPacket(pending_size_.bytes());
+    pending_size_ = DataSize::Zero();
+    last_send_time_ = at_time;
+  }
+}
+
+TimeDelta RandomWalkCrossTraffic::GetProcessInterval() const {
+  return config_.min_packet_interval;
+}
+
+DataRate RandomWalkCrossTraffic::TrafficRate() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return config_.peak_rate * intensity_;
+}
+
+ColumnPrinter RandomWalkCrossTraffic::StatsPrinter() {
+  return ColumnPrinter::Lambda(
+      "random_walk_cross_traffic_rate",
+      [this](rtc::SimpleStringBuilder& sb) {
+        sb.AppendFormat("%.0lf", TrafficRate().bps() / 8.0);
+      },
+      32);
+}
+
+PulsedPeaksCrossTraffic::PulsedPeaksCrossTraffic(
+    PulsedPeaksConfig config,
+    CrossTrafficRoute* traffic_route)
+    : config_(config), traffic_route_(traffic_route) {
+  sequence_checker_.Detach();
+}
+PulsedPeaksCrossTraffic::~PulsedPeaksCrossTraffic() = default;
+
+void PulsedPeaksCrossTraffic::Process(Timestamp at_time) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  TimeDelta time_since_toggle = at_time - last_update_time_;
+  if (time_since_toggle.IsInfinite() ||
+      (sending_ && time_since_toggle >= config_.send_duration)) {
+    sending_ = false;
+    last_update_time_ = at_time;
+  } else if (!sending_ && time_since_toggle >= config_.hold_duration) {
+    sending_ = true;
+    last_update_time_ = at_time;
+    // Start sending period.
+    last_send_time_ = at_time;
+  }
+
+  if (sending_) {
+    DataSize pending_size = config_.peak_rate * (at_time - last_send_time_);
+
+    if (pending_size >= config_.min_packet_size &&
+        at_time >= last_send_time_ + config_.min_packet_interval) {
+      traffic_route_->SendPacket(pending_size.bytes());
+      last_send_time_ = at_time;
+    }
+  }
+}
+
+TimeDelta PulsedPeaksCrossTraffic::GetProcessInterval() const {
+  return config_.min_packet_interval;
+}
+
+DataRate PulsedPeaksCrossTraffic::TrafficRate() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return sending_ ? config_.peak_rate : DataRate::Zero();
+}
+
+ColumnPrinter PulsedPeaksCrossTraffic::StatsPrinter() {
+  return ColumnPrinter::Lambda(
+      "pulsed_peaks_cross_traffic_rate",
+      [this](rtc::SimpleStringBuilder& sb) {
+        sb.AppendFormat("%.0lf", TrafficRate().bps() / 8.0);
+      },
+      32);
+}
+
+TcpMessageRouteImpl::TcpMessageRouteImpl(Clock* clock,
+                                         TaskQueueBase* task_queue,
+                                         EmulatedRoute* send_route,
+                                         EmulatedRoute* ret_route)
+    : clock_(clock),
+      task_queue_(task_queue),
+      request_route_(send_route,
+                     [this](TcpPacket packet, Timestamp) {
+                       OnRequest(std::move(packet));
+                     }),
+      response_route_(ret_route,
+                      [this](TcpPacket packet, Timestamp arrival_time) {
+                        OnResponse(std::move(packet), arrival_time);
+                      }) {}
+
+void TcpMessageRouteImpl::SendMessage(size_t size,
+                                      std::function<void()> on_received) {
+  task_queue_->PostTask(
+      [this, size, handler = std::move(on_received)] {
+        // If we are currently sending a message we won't reset the connection,
+        // we'll act as if the messages are sent in the same TCP stream. This is
+        // intended to simulate recreation of a TCP session for each message
+        // in the typical case while avoiding the complexity overhead of
+        // maintaining multiple virtual TCP sessions in parallel.
+        if (pending_.empty() && in_flight_.empty()) {
+          cwnd_ = 10;
+          ssthresh_ = INFINITY;
+        }
+        int64_t data_left = static_cast<int64_t>(size);
+        int64_t kMaxPacketSize = 1200;
+        int64_t kMinPacketSize = 4;
+        Message message{std::move(handler)};
+        while (data_left > 0) {
+          int64_t packet_size = std::min(data_left, kMaxPacketSize);
+          int fragment_id = next_fragment_id_++;
+          pending_.push_back(MessageFragment{
+              fragment_id,
+              static_cast<size_t>(std::max(kMinPacketSize, packet_size))});
+          message.pending_fragment_ids.insert(fragment_id);
+          data_left -= packet_size;
+        }
+        messages_.emplace_back(message);
+        SendPackets(clock_->CurrentTime());
+      });
+}
+
+void TcpMessageRouteImpl::OnRequest(TcpPacket packet_info) {
+  for (auto it = messages_.begin(); it != messages_.end(); ++it) {
+    if (it->pending_fragment_ids.count(packet_info.fragment.fragment_id) != 0) {
+      it->pending_fragment_ids.erase(packet_info.fragment.fragment_id);
+      if (it->pending_fragment_ids.empty()) {
+        it->handler();
+        messages_.erase(it);
+      }
+      break;
+    }
+  }
+  const size_t kAckPacketSize = 20;
+  response_route_.SendPacket(kAckPacketSize, packet_info);
+}
+
+void TcpMessageRouteImpl::OnResponse(TcpPacket packet_info, Timestamp at_time) {
+  auto it = in_flight_.find(packet_info.sequence_number);
+  if (it != in_flight_.end()) {
+    last_rtt_ = at_time - packet_info.send_time;
+    in_flight_.erase(it);
+  }
+  auto lost_end = in_flight_.lower_bound(packet_info.sequence_number);
+  for (auto lost_it = in_flight_.begin(); lost_it != lost_end;
+       lost_it = in_flight_.erase(lost_it)) {
+    pending_.push_front(lost_it->second.fragment);
+  }
+
+  if (packet_info.sequence_number - last_acked_seq_num_ > 1) {
+    HandleLoss(at_time);
+  } else if (cwnd_ <= ssthresh_) {
+    cwnd_ += 1;
+  } else {
+    cwnd_ += 1.0f / cwnd_;
+  }
+  last_acked_seq_num_ =
+      std::max(packet_info.sequence_number, last_acked_seq_num_);
+  SendPackets(at_time);
+}
+
+void TcpMessageRouteImpl::HandleLoss(Timestamp at_time) {
+  if (at_time - last_reduction_time_ < last_rtt_)
+    return;
+  last_reduction_time_ = at_time;
+  ssthresh_ = std::max(static_cast<int>(in_flight_.size() / 2), 2);
+  cwnd_ = ssthresh_;
+}
+
+void TcpMessageRouteImpl::SendPackets(Timestamp at_time) {
+  const TimeDelta kPacketTimeout = TimeDelta::Seconds(1);
+  int cwnd = std::ceil(cwnd_);
+  int packets_to_send = std::max(cwnd - static_cast<int>(in_flight_.size()), 0);
+  while (packets_to_send-- > 0 && !pending_.empty()) {
+    auto seq_num = next_sequence_number_++;
+    TcpPacket send;
+    send.sequence_number = seq_num;
+    send.send_time = at_time;
+    send.fragment = pending_.front();
+    pending_.pop_front();
+    request_route_.SendPacket(send.fragment.size, send);
+    in_flight_.insert({seq_num, send});
+    task_queue_->PostDelayedTask(
+        [this, seq_num] {
+          HandlePacketTimeout(seq_num, clock_->CurrentTime());
+        },
+        kPacketTimeout);
+  }
+}
+
+void TcpMessageRouteImpl::HandlePacketTimeout(int seq_num, Timestamp at_time) {
+  auto lost = in_flight_.find(seq_num);
+  if (lost != in_flight_.end()) {
+    pending_.push_front(lost->second.fragment);
+    in_flight_.erase(lost);
+    HandleLoss(at_time);
+    SendPackets(at_time);
+  }
+}
+
+FakeTcpCrossTraffic::FakeTcpCrossTraffic(FakeTcpConfig config,
+                                         EmulatedRoute* send_route,
+                                         EmulatedRoute* ret_route)
+    : conf_(config), route_(this, send_route, ret_route) {}
+
+TimeDelta FakeTcpCrossTraffic::GetProcessInterval() const {
+  return conf_.process_interval;
+}
+
+void FakeTcpCrossTraffic::Process(Timestamp at_time) {
+  SendPackets(at_time);
+}
+
+void FakeTcpCrossTraffic::OnRequest(int sequence_number, Timestamp at_time) {
+  const size_t kAckPacketSize = 20;
+  route_.SendResponse(kAckPacketSize, sequence_number);
+}
+
+void FakeTcpCrossTraffic::OnResponse(int sequence_number, Timestamp at_time) {
+  ack_received_ = true;
+  auto it = in_flight_.find(sequence_number);
+  if (it != in_flight_.end()) {
+    last_rtt_ = at_time - in_flight_.at(sequence_number);
+    in_flight_.erase(sequence_number);
+  }
+  if (sequence_number - last_acked_seq_num_ > 1) {
+    HandleLoss(at_time);
+  } else if (cwnd_ <= ssthresh_) {
+    cwnd_ += 1;
+  } else {
+    cwnd_ += 1.0f / cwnd_;
+  }
+  last_acked_seq_num_ = std::max(sequence_number, last_acked_seq_num_);
+  SendPackets(at_time);
+}
+
+void FakeTcpCrossTraffic::HandleLoss(Timestamp at_time) {
+  if (at_time - last_reduction_time_ < last_rtt_)
+    return;
+  last_reduction_time_ = at_time;
+  ssthresh_ = std::max(static_cast<int>(in_flight_.size() / 2), 2);
+  cwnd_ = ssthresh_;
+}
+
+void FakeTcpCrossTraffic::SendPackets(Timestamp at_time) {
+  int cwnd = std::ceil(cwnd_);
+  int packets_to_send = std::max(cwnd - static_cast<int>(in_flight_.size()), 0);
+  bool timeouts = false;
+  for (auto it = in_flight_.begin(); it != in_flight_.end();) {
+    if (it->second < at_time - conf_.packet_timeout) {
+      it = in_flight_.erase(it);
+      timeouts = true;
+    } else {
+      ++it;
+    }
+  }
+  if (timeouts)
+    HandleLoss(at_time);
+  for (int i = 0; i < packets_to_send; ++i) {
+    if ((total_sent_ + conf_.packet_size) > conf_.send_limit) {
+      break;
+    }
+    in_flight_.insert({next_sequence_number_, at_time});
+    route_.SendRequest(conf_.packet_size.bytes<size_t>(),
+                       next_sequence_number_++);
+    total_sent_ += conf_.packet_size;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/cross_traffic.h b/third_party/libwebrtc/test/network/cross_traffic.h
new file mode 100644
index 0000000000..d21e942475
--- /dev/null
+++ b/third_party/libwebrtc/test/network/cross_traffic.h
@@ -0,0 +1,174 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_NETWORK_CROSS_TRAFFIC_H_
+#define TEST_NETWORK_CROSS_TRAFFIC_H_
+
+#include <algorithm>
+#include <map>
+#include <memory>
+
+#include "api/sequence_checker.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/random.h"
+#include "test/network/network_emulation.h"
+#include "test/scenario/column_printer.h"
+
+namespace webrtc {
+namespace test {
+
+class RandomWalkCrossTraffic final : public CrossTrafficGenerator {
+ public:
+  RandomWalkCrossTraffic(RandomWalkConfig config,
+                         CrossTrafficRoute* traffic_route);
+  ~RandomWalkCrossTraffic();
+
+  void Process(Timestamp at_time) override;
+  TimeDelta GetProcessInterval() const override;
+  DataRate TrafficRate() const;
+  ColumnPrinter StatsPrinter();
+
+ private:
+  SequenceChecker sequence_checker_;
+  const RandomWalkConfig config_;
+  CrossTrafficRoute* const traffic_route_ RTC_PT_GUARDED_BY(sequence_checker_);
+  webrtc::Random random_ RTC_GUARDED_BY(sequence_checker_);
+
+  Timestamp last_process_time_ RTC_GUARDED_BY(sequence_checker_) =
+      Timestamp::MinusInfinity();
+  Timestamp last_update_time_ RTC_GUARDED_BY(sequence_checker_) =
+      Timestamp::MinusInfinity();
+  Timestamp last_send_time_ RTC_GUARDED_BY(sequence_checker_) =
+      Timestamp::MinusInfinity();
+  double intensity_ RTC_GUARDED_BY(sequence_checker_) = 0;
+  DataSize pending_size_ RTC_GUARDED_BY(sequence_checker_) = DataSize::Zero();
+};
+
+class PulsedPeaksCrossTraffic final : public CrossTrafficGenerator {
+ public:
+  PulsedPeaksCrossTraffic(PulsedPeaksConfig config,
+                          CrossTrafficRoute* traffic_route);
+  ~PulsedPeaksCrossTraffic();
+
+  void Process(Timestamp at_time) override;
+  TimeDelta GetProcessInterval() const override;
+  DataRate TrafficRate() const;
+  ColumnPrinter StatsPrinter();
+
+ private:
+  SequenceChecker sequence_checker_;
+  const PulsedPeaksConfig config_;
+  CrossTrafficRoute* const traffic_route_ RTC_PT_GUARDED_BY(sequence_checker_);
+
+  Timestamp last_update_time_ RTC_GUARDED_BY(sequence_checker_) =
+      Timestamp::MinusInfinity();
+  Timestamp last_send_time_ RTC_GUARDED_BY(sequence_checker_) =
+      Timestamp::MinusInfinity();
+  bool sending_ RTC_GUARDED_BY(sequence_checker_) = false;
+};
+
+class TcpMessageRouteImpl final : public TcpMessageRoute {
+ public:
+  TcpMessageRouteImpl(Clock* clock,
+                      TaskQueueBase* task_queue,
+                      EmulatedRoute* send_route,
+                      EmulatedRoute* ret_route);
+
+  // Sends a TCP message of the given `size` over the route, `on_received` is
+  // called when the message has been delivered. Note that the connection
+  // parameters are reset iff there's no currently pending message on the route.
+  void SendMessage(size_t size, std::function<void()> on_received) override;
+
+ private:
+  // Represents a message sent over the route. When all fragments has been
+  // delivered, the message is considered delivered and the handler is
+  // triggered. This only happen once.
+  struct Message {
+    std::function<void()> handler;
+    std::set<int> pending_fragment_ids;
+  };
+  // Represents a piece of a message that fit into a TCP packet.
+  struct MessageFragment {
+    int fragment_id;
+    size_t size;
+  };
+  // Represents a packet sent on the wire.
+  struct TcpPacket {
+    int sequence_number;
+    Timestamp send_time = Timestamp::MinusInfinity();
+    MessageFragment fragment;
+  };
+
+  void OnRequest(TcpPacket packet_info);
+  void OnResponse(TcpPacket packet_info, Timestamp at_time);
+  void HandleLoss(Timestamp at_time);
+  void SendPackets(Timestamp at_time);
+  void HandlePacketTimeout(int seq_num, Timestamp at_time);
+
+  Clock* const clock_;
+  TaskQueueBase* const task_queue_;
+  FakePacketRoute<TcpPacket> request_route_;
+  FakePacketRoute<TcpPacket> response_route_;
+
+  std::deque<MessageFragment> pending_;
+  std::map<int, TcpPacket> in_flight_;
+  std::list<Message> messages_;
+
+  double cwnd_;
+  double ssthresh_;
+
+  int last_acked_seq_num_ = 0;
+  int next_sequence_number_ = 0;
+  int next_fragment_id_ = 0;
+  Timestamp last_reduction_time_ = Timestamp::MinusInfinity();
+  TimeDelta last_rtt_ = TimeDelta::Zero();
+};
+
+class FakeTcpCrossTraffic
+    : public TwoWayFakeTrafficRoute<int, int>::TrafficHandlerInterface,
+      public CrossTrafficGenerator {
+ public:
+  FakeTcpCrossTraffic(FakeTcpConfig config,
+                      EmulatedRoute* send_route,
+                      EmulatedRoute* ret_route);
+
+  TimeDelta GetProcessInterval() const override;
+  void Process(Timestamp at_time) override;
+
+  void OnRequest(int sequence_number, Timestamp at_time) override;
+  void OnResponse(int sequence_number, Timestamp at_time) override;
+
+  void HandleLoss(Timestamp at_time);
+
+  void SendPackets(Timestamp at_time);
+
+ private:
+  const FakeTcpConfig conf_;
+  TwoWayFakeTrafficRoute<int, int> route_;
+
+  std::map<int, Timestamp> in_flight_;
+  double cwnd_ = 10;
+  double ssthresh_ = INFINITY;
+  bool ack_received_ = false;
+  int last_acked_seq_num_ = 0;
+  int next_sequence_number_ = 0;
+  Timestamp last_reduction_time_ = Timestamp::MinusInfinity();
+  TimeDelta last_rtt_ = TimeDelta::Zero();
+  DataSize total_sent_ = DataSize::Zero();
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_CROSS_TRAFFIC_H_
diff --git a/third_party/libwebrtc/test/network/cross_traffic_unittest.cc b/third_party/libwebrtc/test/network/cross_traffic_unittest.cc
new file mode 100644
index 0000000000..36aff67bb2
--- /dev/null
+++ b/third_party/libwebrtc/test/network/cross_traffic_unittest.cc
@@ -0,0 +1,163 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/cross_traffic.h"
+
+#include <atomic>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/simulated_network.h"
+#include "call/simulated_network.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/network_constants.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/network/network_emulation_manager.h"
+#include "test/network/traffic_route.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr uint32_t kTestIpAddress = 0xC0A80011;  // 192.168.0.17
+
+class CountingReceiver : public EmulatedNetworkReceiverInterface {
+ public:
+  void OnPacketReceived(EmulatedIpPacket packet) override {
+    packets_count_++;
+    total_packets_size_ += packet.size();
+  }
+
+  std::atomic<int> packets_count_{0};
+  std::atomic<uint64_t> total_packets_size_{0};
+};
+struct TrafficCounterFixture {
+  SimulatedClock clock{0};
+  CountingReceiver counter;
+  TaskQueueForTest task_queue_;
+  EmulatedEndpointImpl endpoint{EmulatedEndpointImpl::Options{
+                                    /*id=*/1,
+                                    rtc::IPAddress(kTestIpAddress),
+                                    EmulatedEndpointConfig(),
+                                    EmulatedNetworkStatsGatheringMode::kDefault,
+                                },
+                                /*is_enabled=*/true, &task_queue_, &clock};
+};
+
+}  // namespace
+
+TEST(CrossTrafficTest, TriggerPacketBurst) {
+  TrafficCounterFixture fixture;
+  CrossTrafficRouteImpl traffic(&fixture.clock, &fixture.counter,
+                                &fixture.endpoint);
+  traffic.TriggerPacketBurst(100, 1000);
+
+  EXPECT_EQ(fixture.counter.packets_count_, 100);
+  EXPECT_EQ(fixture.counter.total_packets_size_, 100 * 1000ul);
+}
+
+TEST(CrossTrafficTest, PulsedPeaksCrossTraffic) {
+  TrafficCounterFixture fixture;
+  CrossTrafficRouteImpl traffic(&fixture.clock, &fixture.counter,
+                                &fixture.endpoint);
+
+  PulsedPeaksConfig config;
+  config.peak_rate = DataRate::KilobitsPerSec(1000);
+  config.min_packet_size = DataSize::Bytes(1);
+  config.min_packet_interval = TimeDelta::Millis(25);
+  config.send_duration = TimeDelta::Millis(500);
+  config.hold_duration = TimeDelta::Millis(250);
+  PulsedPeaksCrossTraffic pulsed_peaks(config, &traffic);
+  const auto kRunTime = TimeDelta::Seconds(1);
+  while (fixture.clock.TimeInMilliseconds() < kRunTime.ms()) {
+    pulsed_peaks.Process(Timestamp::Millis(fixture.clock.TimeInMilliseconds()));
+    fixture.clock.AdvanceTimeMilliseconds(1);
+  }
+
+  RTC_LOG(LS_INFO) << fixture.counter.packets_count_ << " packets; "
+                   << fixture.counter.total_packets_size_ << " bytes";
+  // Using 50% duty cycle.
+  const auto kExpectedDataSent = kRunTime * config.peak_rate * 0.5;
+  EXPECT_NEAR(fixture.counter.total_packets_size_, kExpectedDataSent.bytes(),
+              kExpectedDataSent.bytes() * 0.1);
+}
+
+TEST(CrossTrafficTest, RandomWalkCrossTraffic) {
+  TrafficCounterFixture fixture;
+  CrossTrafficRouteImpl traffic(&fixture.clock, &fixture.counter,
+                                &fixture.endpoint);
+
+  RandomWalkConfig config;
+  config.peak_rate = DataRate::KilobitsPerSec(1000);
+  config.min_packet_size = DataSize::Bytes(1);
+  config.min_packet_interval = TimeDelta::Millis(25);
+  config.update_interval = TimeDelta::Millis(500);
+  config.variance = 0.0;
+  config.bias = 1.0;
+
+  RandomWalkCrossTraffic random_walk(config, &traffic);
+  const auto kRunTime = TimeDelta::Seconds(1);
+  while (fixture.clock.TimeInMilliseconds() < kRunTime.ms()) {
+    random_walk.Process(Timestamp::Millis(fixture.clock.TimeInMilliseconds()));
+    fixture.clock.AdvanceTimeMilliseconds(1);
+  }
+
+  RTC_LOG(LS_INFO) << fixture.counter.packets_count_ << " packets; "
+                   << fixture.counter.total_packets_size_ << " bytes";
+  // Sending at peak rate since bias = 1.
+  const auto kExpectedDataSent = kRunTime * config.peak_rate;
+  EXPECT_NEAR(fixture.counter.total_packets_size_, kExpectedDataSent.bytes(),
+              kExpectedDataSent.bytes() * 0.1);
+}
+
+TEST(TcpMessageRouteTest, DeliveredOnLossyNetwork) {
+  NetworkEmulationManagerImpl net(TimeMode::kSimulated,
+                                  EmulatedNetworkStatsGatheringMode::kDefault);
+  BuiltInNetworkBehaviorConfig send;
+  // 800 kbps means that the 100 kB message would be delivered in ca 1 second
+  // under ideal conditions and no overhead.
+  send.link_capacity_kbps = 100 * 8;
+  send.loss_percent = 50;
+  send.queue_delay_ms = 100;
+  send.delay_standard_deviation_ms = 20;
+  send.allow_reordering = true;
+  auto ret = send;
+  ret.loss_percent = 10;
+
+  auto* tcp_route =
+      net.CreateTcpRoute(net.CreateRoute({net.CreateEmulatedNode(send)}),
+                         net.CreateRoute({net.CreateEmulatedNode(ret)}));
+  int deliver_count = 0;
+  // 100 kB is more than what fits into a single packet.
+  constexpr size_t kMessageSize = 100000;
+
+  tcp_route->SendMessage(kMessageSize, [&] {
+    RTC_LOG(LS_INFO) << "Received at " << ToString(net.Now());
+    deliver_count++;
+  });
+
+  // If there was no loss, we would have delivered the message in ca 1 second,
+  // with 50% it should take much longer.
+  net.time_controller()->AdvanceTime(TimeDelta::Seconds(5));
+  ASSERT_EQ(deliver_count, 0);
+  // But given enough time the messsage will be delivered, but only once.
+  net.time_controller()->AdvanceTime(TimeDelta::Seconds(60));
+  EXPECT_EQ(deliver_count, 1);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/emulated_network_manager.cc b/third_party/libwebrtc/test/network/emulated_network_manager.cc
new file mode 100644
index 0000000000..fa4037e5db
--- /dev/null
+++ b/third_party/libwebrtc/test/network/emulated_network_manager.cc
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/emulated_network_manager.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "test/network/fake_network_socket_server.h"
+
+namespace webrtc {
+namespace test {
+
+EmulatedNetworkManager::EmulatedNetworkManager(
+    TimeController* time_controller,
+    TaskQueueForTest* task_queue,
+    EndpointsContainer* endpoints_container)
+    : task_queue_(task_queue),
+      endpoints_container_(endpoints_container),
+      sent_first_update_(false),
+      start_count_(0) {
+  auto socket_server =
+      std::make_unique<FakeNetworkSocketServer>(endpoints_container);
+  packet_socket_factory_ =
+      std::make_unique<rtc::BasicPacketSocketFactory>(socket_server.get());
+  // Since we pass ownership of the socket server to `network_thread_`, we must
+  // arrange that it outlives `packet_socket_factory_` which refers to it.
+  network_thread_ =
+      time_controller->CreateThread("net_thread", std::move(socket_server));
+}
+
+void EmulatedNetworkManager::EnableEndpoint(EmulatedEndpointImpl* endpoint) {
+  RTC_CHECK(endpoints_container_->HasEndpoint(endpoint))
+      << "No such interface: " << endpoint->GetPeerLocalAddress().ToString();
+  network_thread_->PostTask([this, endpoint]() {
+    endpoint->Enable();
+    UpdateNetworksOnce();
+  });
+}
+
+void EmulatedNetworkManager::DisableEndpoint(EmulatedEndpointImpl* endpoint) {
+  RTC_CHECK(endpoints_container_->HasEndpoint(endpoint))
+      << "No such interface: " << endpoint->GetPeerLocalAddress().ToString();
+  network_thread_->PostTask([this, endpoint]() {
+    endpoint->Disable();
+    UpdateNetworksOnce();
+  });
+}
+
+// Network manager interface. All these methods are supposed to be called from
+// the same thread.
+void EmulatedNetworkManager::StartUpdating() {
+  RTC_DCHECK_RUN_ON(network_thread_.get());
+
+  if (start_count_) {
+    // If network interfaces are already discovered and signal is sent,
+    // we should trigger network signal immediately for the new clients
+    // to start allocating ports.
+    if (sent_first_update_)
+      network_thread_->PostTask([this]() { MaybeSignalNetworksChanged(); });
+  } else {
+    network_thread_->PostTask([this]() { UpdateNetworksOnce(); });
+  }
+  ++start_count_;
+}
+
+void EmulatedNetworkManager::StopUpdating() {
+  RTC_DCHECK_RUN_ON(network_thread_.get());
+  if (!start_count_)
+    return;
+
+  --start_count_;
+  if (!start_count_) {
+    sent_first_update_ = false;
+  }
+}
+
+void EmulatedNetworkManager::GetStats(
+    std::function<void(EmulatedNetworkStats)> stats_callback) const {
+  task_queue_->PostTask([stats_callback, this]() {
+    stats_callback(endpoints_container_->GetStats());
+  });
+}
+
+void EmulatedNetworkManager::UpdateNetworksOnce() {
+  RTC_DCHECK_RUN_ON(network_thread_.get());
+
+  std::vector<std::unique_ptr<rtc::Network>> networks;
+  for (std::unique_ptr<rtc::Network>& net :
+       endpoints_container_->GetEnabledNetworks()) {
+    net->set_default_local_address_provider(this);
+    networks.push_back(std::move(net));
+  }
+
+  bool changed;
+  MergeNetworkList(std::move(networks), &changed);
+  if (changed || !sent_first_update_) {
+    MaybeSignalNetworksChanged();
+    sent_first_update_ = true;
+  }
+}
+
+void EmulatedNetworkManager::MaybeSignalNetworksChanged() {
+  RTC_DCHECK_RUN_ON(network_thread_.get());
+  // If manager is stopped we don't need to signal anything.
+  if (start_count_ == 0) {
+    return;
+  }
+  SignalNetworksChanged();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/emulated_network_manager.h b/third_party/libwebrtc/test/network/emulated_network_manager.h
new file mode 100644
index 0000000000..fb4ee1ee85
--- /dev/null
+++ b/third_party/libwebrtc/test/network/emulated_network_manager.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_NETWORK_EMULATED_NETWORK_MANAGER_H_
+#define TEST_NETWORK_EMULATED_NETWORK_MANAGER_H_
+
+#include <functional>
+#include <memory>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/time_controller.h"
+#include "rtc_base/ip_address.h"
+#include "rtc_base/network.h"
+#include "rtc_base/socket_server.h"
+#include "rtc_base/thread.h"
+#include "test/network/network_emulation.h"
+
+namespace webrtc {
+namespace test {
+
+// Framework assumes that rtc::NetworkManager is called from network thread.
+class EmulatedNetworkManager : public rtc::NetworkManagerBase,
+                               public sigslot::has_slots<>,
+                               public EmulatedNetworkManagerInterface {
+ public:
+  EmulatedNetworkManager(TimeController* time_controller,
+                         TaskQueueForTest* task_queue,
+                         EndpointsContainer* endpoints_container);
+
+  void EnableEndpoint(EmulatedEndpointImpl* endpoint);
+  void DisableEndpoint(EmulatedEndpointImpl* endpoint);
+
+  // NetworkManager interface. All these methods are supposed to be called from
+  // the same thread.
+  void StartUpdating() override;
+  void StopUpdating() override;
+
+  // We don't support any address interfaces in the network emulation framework.
+  std::vector<const rtc::Network*> GetAnyAddressNetworks() override {
+    return {};
+  }
+
+  // EmulatedNetworkManagerInterface API
+  rtc::Thread* network_thread() override { return network_thread_.get(); }
+  rtc::NetworkManager* network_manager() override { return this; }
+  rtc::PacketSocketFactory* packet_socket_factory() override {
+    return packet_socket_factory_.get();
+  }
+  std::vector<EmulatedEndpoint*> endpoints() const override {
+    return endpoints_container_->GetEndpoints();
+  }
+  void GetStats(
+      std::function<void(EmulatedNetworkStats)> stats_callback) const override;
+
+ private:
+  void UpdateNetworksOnce();
+  void MaybeSignalNetworksChanged();
+
+  TaskQueueForTest* const task_queue_;
+  const EndpointsContainer* const endpoints_container_;
+  // The `network_thread_` must outlive `packet_socket_factory_`, because they
+  // both refer to a socket server that is owned by `network_thread_`. Both
+  // pointers are assigned only in the constructor, but the way they are
+  // initialized unfortunately doesn't work with const std::unique_ptr<...>.
+  std::unique_ptr<rtc::Thread> network_thread_;
+  std::unique_ptr<rtc::PacketSocketFactory> packet_socket_factory_;
+  bool sent_first_update_ RTC_GUARDED_BY(network_thread_);
+  int start_count_ RTC_GUARDED_BY(network_thread_);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_EMULATED_NETWORK_MANAGER_H_
diff --git a/third_party/libwebrtc/test/network/emulated_turn_server.cc b/third_party/libwebrtc/test/network/emulated_turn_server.cc
new file mode 100644
index 0000000000..0bc7ec6e2a
--- /dev/null
+++ b/third_party/libwebrtc/test/network/emulated_turn_server.cc
@@ -0,0 +1,191 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/emulated_turn_server.h"
+
+#include <string>
+#include <utility>
+
+#include "api/packet_socket_factory.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/task_queue_for_test.h"
+
+namespace {
+
+static const char kTestRealm[] = "example.org";
+static const char kTestSoftware[] = "TestTurnServer";
+
+// A wrapper class for copying data between an AsyncPacketSocket and a
+// EmulatedEndpoint. This is used by the cricket::TurnServer when
+// sending data back into the emulated network.
+class AsyncPacketSocketWrapper : public rtc::AsyncPacketSocket {
+ public:
+  AsyncPacketSocketWrapper(webrtc::test::EmulatedTURNServer* turn_server,
+                           webrtc::EmulatedEndpoint* endpoint,
+                           uint16_t port)
+      : turn_server_(turn_server),
+        endpoint_(endpoint),
+        local_address_(
+            rtc::SocketAddress(endpoint_->GetPeerLocalAddress(), port)) {}
+  ~AsyncPacketSocketWrapper() { turn_server_->Unbind(local_address_); }
+
+  rtc::SocketAddress GetLocalAddress() const override { return local_address_; }
+  rtc::SocketAddress GetRemoteAddress() const override {
+    return rtc::SocketAddress();
+  }
+  int Send(const void* pv,
+           size_t cb,
+           const rtc::PacketOptions& options) override {
+    RTC_CHECK(false) << "TCP not implemented";
+    return -1;
+  }
+  int SendTo(const void* pv,
+             size_t cb,
+             const rtc::SocketAddress& addr,
+             const rtc::PacketOptions& options) override {
+    // Copy from rtc::AsyncPacketSocket to EmulatedEndpoint.
+    rtc::CopyOnWriteBuffer buf(reinterpret_cast<const char*>(pv), cb);
+    endpoint_->SendPacket(local_address_, addr, buf);
+    return cb;
+  }
+  int Close() override { return 0; }
+
+  rtc::AsyncPacketSocket::State GetState() const override {
+    return rtc::AsyncPacketSocket::STATE_BOUND;
+  }
+  int GetOption(rtc::Socket::Option opt, int* value) override { return 0; }
+  int SetOption(rtc::Socket::Option opt, int value) override { return 0; }
+  int GetError() const override { return 0; }
+  void SetError(int error) override {}
+
+ private:
+  webrtc::test::EmulatedTURNServer* const turn_server_;
+  webrtc::EmulatedEndpoint* const endpoint_;
+  const rtc::SocketAddress local_address_;
+};
+
+// A wrapper class for cricket::TurnServer to allocate sockets.
+class PacketSocketFactoryWrapper : public rtc::PacketSocketFactory {
+ public:
+  explicit PacketSocketFactoryWrapper(
+      webrtc::test::EmulatedTURNServer* turn_server)
+      : turn_server_(turn_server) {}
+  ~PacketSocketFactoryWrapper() override {}
+
+  // This method is called from TurnServer when making a TURN ALLOCATION.
+  // It will create a socket on the `peer_` endpoint.
+  rtc::AsyncPacketSocket* CreateUdpSocket(const rtc::SocketAddress& address,
+                                          uint16_t min_port,
+                                          uint16_t max_port) override {
+    return turn_server_->CreatePeerSocket();
+  }
+
+  rtc::AsyncListenSocket* CreateServerTcpSocket(
+      const rtc::SocketAddress& local_address,
+      uint16_t min_port,
+      uint16_t max_port,
+      int opts) override {
+    return nullptr;
+  }
+  rtc::AsyncPacketSocket* CreateClientTcpSocket(
+      const rtc::SocketAddress& local_address,
+      const rtc::SocketAddress& remote_address,
+      const rtc::ProxyInfo& proxy_info,
+      const std::string& user_agent,
+      const rtc::PacketSocketTcpOptions& tcp_options) override {
+    return nullptr;
+  }
+  std::unique_ptr<webrtc::AsyncDnsResolverInterface> CreateAsyncDnsResolver()
+      override {
+    return nullptr;
+  }
+
+ private:
+  webrtc::test::EmulatedTURNServer* turn_server_;
+};
+
+}  //  namespace
+
+namespace webrtc {
+namespace test {
+
+EmulatedTURNServer::EmulatedTURNServer(std::unique_ptr<rtc::Thread> thread,
+                                       EmulatedEndpoint* client,
+                                       EmulatedEndpoint* peer)
+    : thread_(std::move(thread)), client_(client), peer_(peer) {
+  ice_config_.username = "keso";
+  ice_config_.password = "keso";
+  SendTask(thread_.get(), [=]() {
+    RTC_DCHECK_RUN_ON(thread_.get());
+    turn_server_ = std::make_unique<cricket::TurnServer>(thread_.get());
+    turn_server_->set_realm(kTestRealm);
+    turn_server_->set_realm(kTestSoftware);
+    turn_server_->set_auth_hook(this);
+
+    auto client_socket = Wrap(client_);
+    turn_server_->AddInternalSocket(client_socket, cricket::PROTO_UDP);
+    turn_server_->SetExternalSocketFactory(new PacketSocketFactoryWrapper(this),
+                                           rtc::SocketAddress());
+    client_address_ = client_socket->GetLocalAddress();
+    char buf[256];
+    rtc::SimpleStringBuilder str(buf);
+    str.AppendFormat("turn:%s?transport=udp",
+                     client_address_.ToString().c_str());
+    ice_config_.url = str.str();
+  });
+}
+
+void EmulatedTURNServer::Stop() {
+  SendTask(thread_.get(), [=]() {
+    RTC_DCHECK_RUN_ON(thread_.get());
+    sockets_.clear();
+  });
+}
+
+EmulatedTURNServer::~EmulatedTURNServer() {
+  SendTask(thread_.get(), [=]() {
+    RTC_DCHECK_RUN_ON(thread_.get());
+    turn_server_.reset(nullptr);
+  });
+}
+
+rtc::AsyncPacketSocket* EmulatedTURNServer::Wrap(EmulatedEndpoint* endpoint) {
+  RTC_DCHECK_RUN_ON(thread_.get());
+  auto port = endpoint->BindReceiver(0, this).value();
+  auto socket = new AsyncPacketSocketWrapper(this, endpoint, port);
+  sockets_[rtc::SocketAddress(endpoint->GetPeerLocalAddress(), port)] = socket;
+  return socket;
+}
+
+void EmulatedTURNServer::OnPacketReceived(webrtc::EmulatedIpPacket packet) {
+  // Copy from EmulatedEndpoint to rtc::AsyncPacketSocket.
+  thread_->PostTask([this, packet(std::move(packet))]() {
+    RTC_DCHECK_RUN_ON(thread_.get());
+    auto it = sockets_.find(packet.to);
+    if (it != sockets_.end()) {
+      it->second->SignalReadPacket(
+          it->second, reinterpret_cast<const char*>(packet.cdata()),
+          packet.size(), packet.from, packet.arrival_time.ms());
+    }
+  });
+}
+
+void EmulatedTURNServer::Unbind(rtc::SocketAddress address) {
+  RTC_DCHECK_RUN_ON(thread_.get());
+  if (GetClientEndpoint()->GetPeerLocalAddress() == address.ipaddr()) {
+    GetClientEndpoint()->UnbindReceiver(address.port());
+  } else {
+    GetPeerEndpoint()->UnbindReceiver(address.port());
+  }
+  sockets_.erase(address);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/emulated_turn_server.h b/third_party/libwebrtc/test/network/emulated_turn_server.h
new file mode 100644
index 0000000000..9cb0ceabf6
--- /dev/null
+++ b/third_party/libwebrtc/test/network/emulated_turn_server.h
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_NETWORK_EMULATED_TURN_SERVER_H_
+#define TEST_NETWORK_EMULATED_TURN_SERVER_H_
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/transport/stun.h"
+#include "p2p/base/turn_server.h"
+#include "rtc_base/async_packet_socket.h"
+
+namespace webrtc {
+namespace test {
+
+// EmulatedTURNServer wraps cricket::TurnServer to be used inside
+// a emulated network.
+//
+// Packets from EmulatedEndpoint (client or peer) are received in
+// EmulatedTURNServer::OnPacketReceived which performs a map lookup
+// and delivers them into cricket::TurnServer using
+// AsyncPacketSocket::SignalReadPacket
+//
+// Packets from cricket::TurnServer to EmulatedEndpoint are sent into
+// using a wrapper around AsyncPacketSocket (no lookup required as the
+// wrapper around AsyncPacketSocket keep a pointer to the EmulatedEndpoint).
+class EmulatedTURNServer : public EmulatedTURNServerInterface,
+                           public cricket::TurnAuthInterface,
+                           public webrtc::EmulatedNetworkReceiverInterface {
+ public:
+  // Create an EmulatedTURNServer.
+  // `thread` is a thread that will be used to run cricket::TurnServer
+  // that expects all calls to be made from a single thread.
+  EmulatedTURNServer(std::unique_ptr<rtc::Thread> thread,
+                     EmulatedEndpoint* client,
+                     EmulatedEndpoint* peer);
+  ~EmulatedTURNServer() override;
+
+  IceServerConfig GetIceServerConfig() const override { return ice_config_; }
+
+  EmulatedEndpoint* GetClientEndpoint() const override { return client_; }
+
+  rtc::SocketAddress GetClientEndpointAddress() const override {
+    return client_address_;
+  }
+
+  EmulatedEndpoint* GetPeerEndpoint() const override { return peer_; }
+
+  // cricket::TurnAuthInterface
+  bool GetKey(absl::string_view username,
+              absl::string_view realm,
+              std::string* key) override {
+    return cricket::ComputeStunCredentialHash(
+        std::string(username), std::string(realm), std::string(username), key);
+  }
+
+  rtc::AsyncPacketSocket* CreatePeerSocket() { return Wrap(peer_); }
+
+  // This method is called by network emulation when a packet
+  // comes from an emulated link.
+  void OnPacketReceived(webrtc::EmulatedIpPacket packet) override;
+
+  // This is called when the TURN server deletes a socket.
+  void Unbind(rtc::SocketAddress address);
+
+  // Unbind all sockets.
+  void Stop();
+
+ private:
+  std::unique_ptr<rtc::Thread> thread_;
+  rtc::SocketAddress client_address_;
+  IceServerConfig ice_config_;
+  EmulatedEndpoint* const client_;
+  EmulatedEndpoint* const peer_;
+  std::unique_ptr<cricket::TurnServer> turn_server_ RTC_GUARDED_BY(&thread_);
+  std::map<rtc::SocketAddress, rtc::AsyncPacketSocket*> sockets_
+      RTC_GUARDED_BY(&thread_);
+
+  // Wraps a EmulatedEndpoint in a AsyncPacketSocket to bridge interaction
+  // with TurnServer. cricket::TurnServer gets ownership of the socket.
+  rtc::AsyncPacketSocket* Wrap(EmulatedEndpoint* endpoint);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_EMULATED_TURN_SERVER_H_
diff --git a/third_party/libwebrtc/test/network/fake_network_socket_server.cc b/third_party/libwebrtc/test/network/fake_network_socket_server.cc
new file mode 100644
index 0000000000..c94c4e372a
--- /dev/null
+++ b/third_party/libwebrtc/test/network/fake_network_socket_server.cc
@@ -0,0 +1,322 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/fake_network_socket_server.h"
+
+#include <algorithm>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+std::string ToString(const rtc::SocketAddress& addr) {
+  return addr.HostAsURIString() + ":" + std::to_string(addr.port());
+}
+
+}  // namespace
+
+// Represents a socket, which will operate with emulated network.
+class FakeNetworkSocket : public rtc::Socket,
+                          public EmulatedNetworkReceiverInterface {
+ public:
+  explicit FakeNetworkSocket(FakeNetworkSocketServer* scoket_manager,
+                             rtc::Thread* thread);
+  ~FakeNetworkSocket() override;
+
+  // Will be invoked by EmulatedEndpoint to deliver packets into this socket.
+  void OnPacketReceived(EmulatedIpPacket packet) override;
+
+  // rtc::Socket methods:
+  rtc::SocketAddress GetLocalAddress() const override;
+  rtc::SocketAddress GetRemoteAddress() const override;
+  int Bind(const rtc::SocketAddress& addr) override;
+  int Connect(const rtc::SocketAddress& addr) override;
+  int Close() override;
+  int Send(const void* pv, size_t cb) override;
+  int SendTo(const void* pv,
+             size_t cb,
+             const rtc::SocketAddress& addr) override;
+  int Recv(void* pv, size_t cb, int64_t* timestamp) override;
+  int RecvFrom(void* pv,
+               size_t cb,
+               rtc::SocketAddress* paddr,
+               int64_t* timestamp) override;
+  int Listen(int backlog) override;
+  rtc::Socket* Accept(rtc::SocketAddress* paddr) override;
+  int GetError() const override;
+  void SetError(int error) override;
+  ConnState GetState() const override;
+  int GetOption(Option opt, int* value) override;
+  int SetOption(Option opt, int value) override;
+
+ private:
+  FakeNetworkSocketServer* const socket_server_;
+  rtc::Thread* const thread_;
+  EmulatedEndpointImpl* endpoint_ RTC_GUARDED_BY(&thread_);
+  rtc::SocketAddress local_addr_ RTC_GUARDED_BY(&thread_);
+  rtc::SocketAddress remote_addr_ RTC_GUARDED_BY(&thread_);
+  ConnState state_ RTC_GUARDED_BY(&thread_);
+  int error_ RTC_GUARDED_BY(&thread_);
+  std::map<Option, int> options_map_ RTC_GUARDED_BY(&thread_);
+
+  absl::optional<EmulatedIpPacket> pending_ RTC_GUARDED_BY(thread_);
+  rtc::scoped_refptr<PendingTaskSafetyFlag> alive_;
+};
+
+FakeNetworkSocket::FakeNetworkSocket(FakeNetworkSocketServer* socket_server,
+                                     rtc::Thread* thread)
+    : socket_server_(socket_server),
+      thread_(thread),
+      state_(CS_CLOSED),
+      error_(0),
+      alive_(PendingTaskSafetyFlag::Create()) {}
+
+FakeNetworkSocket::~FakeNetworkSocket() {
+  // Abandon all pending packets.
+  alive_->SetNotAlive();
+
+  Close();
+  socket_server_->Unregister(this);
+}
+
+void FakeNetworkSocket::OnPacketReceived(EmulatedIpPacket packet) {
+  auto task = [this, packet = std::move(packet)]() mutable {
+    RTC_DCHECK_RUN_ON(thread_);
+    if (!endpoint_->Enabled())
+      return;
+    RTC_DCHECK(!pending_);
+    pending_ = std::move(packet);
+    // Note that we expect that this will trigger exactly one call to RecvFrom()
+    // where pending_packet will be read and reset. This call is done without
+    // any thread switch (see AsyncUDPSocket::OnReadEvent) so it's safe to
+    // assume that SignalReadEvent() will block until the packet has been read.
+    SignalReadEvent(this);
+    RTC_DCHECK(!pending_);
+  };
+  thread_->PostTask(SafeTask(alive_, std::move(task)));
+  socket_server_->WakeUp();
+}
+
+
+rtc::SocketAddress FakeNetworkSocket::GetLocalAddress() const {
+  RTC_DCHECK_RUN_ON(thread_);
+  return local_addr_;
+}
+
+rtc::SocketAddress FakeNetworkSocket::GetRemoteAddress() const {
+  RTC_DCHECK_RUN_ON(thread_);
+  return remote_addr_;
+}
+
+int FakeNetworkSocket::Bind(const rtc::SocketAddress& addr) {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_CHECK(local_addr_.IsNil())
+      << "Socket already bound to address: " << ToString(local_addr_);
+  local_addr_ = addr;
+  endpoint_ = socket_server_->GetEndpointNode(local_addr_.ipaddr());
+  if (!endpoint_) {
+    local_addr_.Clear();
+    RTC_LOG(LS_INFO) << "No endpoint for address: " << ToString(addr);
+    error_ = EADDRNOTAVAIL;
+    return 2;
+  }
+  absl::optional<uint16_t> port =
+      endpoint_->BindReceiver(local_addr_.port(), this);
+  if (!port) {
+    local_addr_.Clear();
+    RTC_LOG(LS_INFO) << "Cannot bind to in-use address: " << ToString(addr);
+    error_ = EADDRINUSE;
+    return 1;
+  }
+  local_addr_.SetPort(port.value());
+  return 0;
+}
+
+int FakeNetworkSocket::Connect(const rtc::SocketAddress& addr) {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_CHECK(remote_addr_.IsNil())
+      << "Socket already connected to address: " << ToString(remote_addr_);
+  RTC_CHECK(!local_addr_.IsNil())
+      << "Socket have to be bind to some local address";
+  remote_addr_ = addr;
+  state_ = CS_CONNECTED;
+  return 0;
+}
+
+int FakeNetworkSocket::Send(const void* pv, size_t cb) {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_CHECK(state_ == CS_CONNECTED) << "Socket cannot send: not connected";
+  return SendTo(pv, cb, remote_addr_);
+}
+
+int FakeNetworkSocket::SendTo(const void* pv,
+                              size_t cb,
+                              const rtc::SocketAddress& addr) {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_CHECK(!local_addr_.IsNil())
+      << "Socket have to be bind to some local address";
+  if (!endpoint_->Enabled()) {
+    error_ = ENETDOWN;
+    return -1;
+  }
+  rtc::CopyOnWriteBuffer packet(static_cast<const uint8_t*>(pv), cb);
+  endpoint_->SendPacket(local_addr_, addr, packet);
+  return cb;
+}
+
+int FakeNetworkSocket::Recv(void* pv, size_t cb, int64_t* timestamp) {
+  rtc::SocketAddress paddr;
+  return RecvFrom(pv, cb, &paddr, timestamp);
+}
+
+// Reads 1 packet from internal queue. Reads up to `cb` bytes into `pv`
+// and returns the length of received packet.
+int FakeNetworkSocket::RecvFrom(void* pv,
+                                size_t cb,
+                                rtc::SocketAddress* paddr,
+                                int64_t* timestamp) {
+  RTC_DCHECK_RUN_ON(thread_);
+
+  if (timestamp) {
+    *timestamp = -1;
+  }
+  RTC_CHECK(pending_);
+
+  *paddr = pending_->from;
+  size_t data_read = std::min(cb, pending_->size());
+  memcpy(pv, pending_->cdata(), data_read);
+  *timestamp = pending_->arrival_time.us();
+
+  // According to RECV(2) Linux Man page
+  // real socket will discard data, that won't fit into provided buffer,
+  // but we won't to skip such error, so we will assert here.
+  RTC_CHECK(data_read == pending_->size())
+      << "Too small buffer is provided for socket read. "
+         "Received data size: "
+      << pending_->size() << "; Provided buffer size: " << cb;
+
+  pending_.reset();
+
+  // According to RECV(2) Linux Man page
+  // real socket will return message length, not data read. In our case it is
+  // actually the same value.
+  return static_cast<int>(data_read);
+}
+
+int FakeNetworkSocket::Listen(int backlog) {
+  RTC_CHECK(false) << "Listen() isn't valid for SOCK_DGRAM";
+}
+
+rtc::Socket* FakeNetworkSocket::Accept(rtc::SocketAddress* /*paddr*/) {
+  RTC_CHECK(false) << "Accept() isn't valid for SOCK_DGRAM";
+}
+
+int FakeNetworkSocket::Close() {
+  RTC_DCHECK_RUN_ON(thread_);
+  state_ = CS_CLOSED;
+  if (!local_addr_.IsNil()) {
+    endpoint_->UnbindReceiver(local_addr_.port());
+  }
+  local_addr_.Clear();
+  remote_addr_.Clear();
+  return 0;
+}
+
+int FakeNetworkSocket::GetError() const {
+  RTC_DCHECK_RUN_ON(thread_);
+  return error_;
+}
+
+void FakeNetworkSocket::SetError(int error) {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_CHECK(error == 0);
+  error_ = error;
+}
+
+rtc::Socket::ConnState FakeNetworkSocket::GetState() const {
+  RTC_DCHECK_RUN_ON(thread_);
+  return state_;
+}
+
+int FakeNetworkSocket::GetOption(Option opt, int* value) {
+  RTC_DCHECK_RUN_ON(thread_);
+  auto it = options_map_.find(opt);
+  if (it == options_map_.end()) {
+    return -1;
+  }
+  *value = it->second;
+  return 0;
+}
+
+int FakeNetworkSocket::SetOption(Option opt, int value) {
+  RTC_DCHECK_RUN_ON(thread_);
+  options_map_[opt] = value;
+  return 0;
+}
+
+FakeNetworkSocketServer::FakeNetworkSocketServer(
+    EndpointsContainer* endpoints_container)
+    : endpoints_container_(endpoints_container),
+      wakeup_(/*manual_reset=*/false, /*initially_signaled=*/false) {}
+FakeNetworkSocketServer::~FakeNetworkSocketServer() = default;
+
+EmulatedEndpointImpl* FakeNetworkSocketServer::GetEndpointNode(
+    const rtc::IPAddress& ip) {
+  return endpoints_container_->LookupByLocalAddress(ip);
+}
+
+void FakeNetworkSocketServer::Unregister(FakeNetworkSocket* socket) {
+  MutexLock lock(&lock_);
+  sockets_.erase(absl::c_find(sockets_, socket));
+}
+
+rtc::Socket* FakeNetworkSocketServer::CreateSocket(int family, int type) {
+  RTC_DCHECK(family == AF_INET || family == AF_INET6);
+  // We support only UDP sockets for now.
+  RTC_DCHECK(type == SOCK_DGRAM) << "Only UDP sockets are supported";
+  RTC_DCHECK(thread_) << "must be attached to thread before creating sockets";
+  FakeNetworkSocket* out = new FakeNetworkSocket(this, thread_);
+  {
+    MutexLock lock(&lock_);
+    sockets_.push_back(out);
+  }
+  return out;
+}
+
+void FakeNetworkSocketServer::SetMessageQueue(rtc::Thread* thread) {
+  thread_ = thread;
+}
+
+// Always returns true (if return false, it won't be invoked again...)
+bool FakeNetworkSocketServer::Wait(webrtc::TimeDelta max_wait_duration,
+                                   bool process_io) {
+  RTC_DCHECK(thread_ == rtc::Thread::Current());
+  if (!max_wait_duration.IsZero())
+    wakeup_.Wait(max_wait_duration);
+
+  return true;
+}
+
+void FakeNetworkSocketServer::WakeUp() {
+  wakeup_.Set();
+}
+
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/fake_network_socket_server.h b/third_party/libwebrtc/test/network/fake_network_socket_server.h
new file mode 100644
index 0000000000..25c85d048a
--- /dev/null
+++ b/third_party/libwebrtc/test/network/fake_network_socket_server.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_NETWORK_FAKE_NETWORK_SOCKET_SERVER_H_
+#define TEST_NETWORK_FAKE_NETWORK_SOCKET_SERVER_H_
+
+#include <set>
+#include <vector>
+
+#include "api/units/timestamp.h"
+#include "rtc_base/event.h"
+#include "rtc_base/socket.h"
+#include "rtc_base/socket_server.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "system_wrappers/include/clock.h"
+#include "test/network/network_emulation.h"
+
+namespace webrtc {
+namespace test {
+class FakeNetworkSocket;
+
+// FakeNetworkSocketServer must outlive any sockets it creates.
+class FakeNetworkSocketServer : public rtc::SocketServer {
+ public:
+  explicit FakeNetworkSocketServer(EndpointsContainer* endpoints_controller);
+  ~FakeNetworkSocketServer() override;
+
+
+  // rtc::SocketFactory methods:
+  rtc::Socket* CreateSocket(int family, int type) override;
+
+  // rtc::SocketServer methods:
+  // Called by the network thread when this server is installed, kicking off the
+  // message handler loop.
+  void SetMessageQueue(rtc::Thread* thread) override;
+  bool Wait(webrtc::TimeDelta max_wait_duration, bool process_io) override;
+  void WakeUp() override;
+
+ protected:
+  friend class FakeNetworkSocket;
+  EmulatedEndpointImpl* GetEndpointNode(const rtc::IPAddress& ip);
+  void Unregister(FakeNetworkSocket* socket);
+
+ private:
+  const EndpointsContainer* endpoints_container_;
+  rtc::Event wakeup_;
+  rtc::Thread* thread_ = nullptr;
+
+  Mutex lock_;
+  std::vector<FakeNetworkSocket*> sockets_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_FAKE_NETWORK_SOCKET_SERVER_H_
diff --git a/third_party/libwebrtc/test/network/feedback_generator.cc b/third_party/libwebrtc/test/network/feedback_generator.cc
new file mode 100644
index 0000000000..e339fd87b0
--- /dev/null
+++ b/third_party/libwebrtc/test/network/feedback_generator.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/network/feedback_generator.h"
+
+#include "absl/memory/memory.h"
+#include "api/transport/network_types.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+FeedbackGeneratorImpl::FeedbackGeneratorImpl(
+    FeedbackGeneratorImpl::Config config)
+    : conf_(config),
+      net_(TimeMode::kSimulated, EmulatedNetworkStatsGatheringMode::kDefault),
+      send_link_{new SimulatedNetwork(conf_.send_link)},
+      ret_link_{new SimulatedNetwork(conf_.return_link)},
+      route_(this,
+             net_.CreateRoute(
+                 {net_.CreateEmulatedNode(absl::WrapUnique(send_link_))}),
+             net_.CreateRoute(
+                 {net_.CreateEmulatedNode(absl::WrapUnique(ret_link_))})) {}
+
+Timestamp FeedbackGeneratorImpl::Now() {
+  return net_.Now();
+}
+
+void FeedbackGeneratorImpl::Sleep(TimeDelta duration) {
+  net_.time_controller()->AdvanceTime(duration);
+}
+
+void FeedbackGeneratorImpl::SendPacket(size_t size) {
+  SentPacket sent;
+  sent.send_time = Now();
+  sent.size = DataSize::Bytes(size);
+  sent.sequence_number = sequence_number_++;
+  sent_packets_.push(sent);
+  route_.SendRequest(size, sent);
+}
+
+std::vector<TransportPacketsFeedback> FeedbackGeneratorImpl::PopFeedback() {
+  std::vector<TransportPacketsFeedback> ret;
+  ret.swap(feedback_);
+  return ret;
+}
+
+void FeedbackGeneratorImpl::SetSendConfig(BuiltInNetworkBehaviorConfig config) {
+  conf_.send_link = config;
+  send_link_->SetConfig(conf_.send_link);
+}
+
+void FeedbackGeneratorImpl::SetReturnConfig(
+    BuiltInNetworkBehaviorConfig config) {
+  conf_.return_link = config;
+  ret_link_->SetConfig(conf_.return_link);
+}
+
+void FeedbackGeneratorImpl::SetSendLinkCapacity(DataRate capacity) {
+  conf_.send_link.link_capacity_kbps = capacity.kbps<int>();
+  send_link_->SetConfig(conf_.send_link);
+}
+
+void FeedbackGeneratorImpl::OnRequest(SentPacket packet,
+                                      Timestamp arrival_time) {
+  PacketResult result;
+  result.sent_packet = packet;
+  result.receive_time = arrival_time;
+  received_packets_.push_back(result);
+  Timestamp first_recv = received_packets_.front().receive_time;
+  if (Now() - first_recv > conf_.feedback_interval) {
+    route_.SendResponse(conf_.feedback_packet_size.bytes<size_t>(),
+                        std::move(received_packets_));
+    received_packets_ = {};
+  }
+}
+
+void FeedbackGeneratorImpl::OnResponse(std::vector<PacketResult> packet_results,
+                                       Timestamp arrival_time) {
+  TransportPacketsFeedback feedback;
+  feedback.feedback_time = arrival_time;
+  std::vector<PacketResult>::const_iterator received_packet_iterator =
+      packet_results.begin();
+  while (received_packet_iterator != packet_results.end()) {
+    RTC_DCHECK(!sent_packets_.empty() &&
+               sent_packets_.front().sequence_number <=
+                   received_packet_iterator->sent_packet.sequence_number)
+        << "reordering not implemented";
+    if (sent_packets_.front().sequence_number <
+        received_packet_iterator->sent_packet.sequence_number) {
+      // Packet lost.
+      PacketResult lost;
+      lost.sent_packet = sent_packets_.front();
+      feedback.packet_feedbacks.push_back(lost);
+    }
+    if (sent_packets_.front().sequence_number ==
+        received_packet_iterator->sent_packet.sequence_number) {
+      feedback.packet_feedbacks.push_back(*received_packet_iterator);
+      ++received_packet_iterator;
+    }
+    sent_packets_.pop();
+  }
+  feedback_.push_back(feedback);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/feedback_generator.h b/third_party/libwebrtc/test/network/feedback_generator.h
new file mode 100644
index 0000000000..ecd4597d3f
--- /dev/null
+++ b/third_party/libwebrtc/test/network/feedback_generator.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_NETWORK_FEEDBACK_GENERATOR_H_
+#define TEST_NETWORK_FEEDBACK_GENERATOR_H_
+
+#include <cstdint>
+#include <queue>
+#include <utility>
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/transport/test/feedback_generator_interface.h"
+#include "call/simulated_network.h"
+#include "test/network/network_emulation.h"
+#include "test/network/network_emulation_manager.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+class FeedbackGeneratorImpl
+    : public FeedbackGenerator,
+      public TwoWayFakeTrafficRoute<SentPacket, std::vector<PacketResult>>::
+          TrafficHandlerInterface {
+ public:
+  explicit FeedbackGeneratorImpl(Config config);
+  Timestamp Now() override;
+  void Sleep(TimeDelta duration) override;
+  void SendPacket(size_t size) override;
+  std::vector<TransportPacketsFeedback> PopFeedback() override;
+
+  void SetSendConfig(BuiltInNetworkBehaviorConfig config) override;
+  void SetReturnConfig(BuiltInNetworkBehaviorConfig config) override;
+
+  void SetSendLinkCapacity(DataRate capacity) override;
+
+  void OnRequest(SentPacket packet, Timestamp arrival_time) override;
+  void OnResponse(std::vector<PacketResult> packet_results,
+                  Timestamp arrival_time) override;
+
+ private:
+  Config conf_;
+  ::webrtc::test::NetworkEmulationManagerImpl net_;
+  SimulatedNetwork* const send_link_;
+  SimulatedNetwork* const ret_link_;
+  TwoWayFakeTrafficRoute<SentPacket, std::vector<PacketResult>> route_;
+
+  std::queue<SentPacket> sent_packets_;
+  std::vector<PacketResult> received_packets_;
+  std::vector<TransportPacketsFeedback> feedback_;
+  int64_t sequence_number_ = 1;
+};
+}  // namespace webrtc
+#endif  // TEST_NETWORK_FEEDBACK_GENERATOR_H_
diff --git a/third_party/libwebrtc/test/network/feedback_generator_unittest.cc b/third_party/libwebrtc/test/network/feedback_generator_unittest.cc
new file mode 100644
index 0000000000..9a577bea00
--- /dev/null
+++ b/third_party/libwebrtc/test/network/feedback_generator_unittest.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/transport/test/create_feedback_generator.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+TEST(FeedbackGeneratorTest, ReportsFeedbackForSentPackets) {
+  size_t kPacketSize = 1000;
+  auto gen = CreateFeedbackGenerator(FeedbackGenerator::Config());
+  for (int i = 0; i < 10; ++i) {
+    gen->SendPacket(kPacketSize);
+    gen->Sleep(TimeDelta::Millis(50));
+  }
+  auto feedback_list = gen->PopFeedback();
+  EXPECT_GT(feedback_list.size(), 0u);
+  for (const auto& feedback : feedback_list) {
+    EXPECT_GT(feedback.packet_feedbacks.size(), 0u);
+    for (const auto& packet : feedback.packet_feedbacks) {
+      EXPECT_EQ(packet.sent_packet.size.bytes<size_t>(), kPacketSize);
+    }
+  }
+}
+
+TEST(FeedbackGeneratorTest, FeedbackIncludesLostPackets) {
+  size_t kPacketSize = 1000;
+  auto gen = CreateFeedbackGenerator(FeedbackGenerator::Config());
+  BuiltInNetworkBehaviorConfig send_config_with_loss;
+  send_config_with_loss.loss_percent = 50;
+  gen->SetSendConfig(send_config_with_loss);
+  for (int i = 0; i < 20; ++i) {
+    gen->SendPacket(kPacketSize);
+    gen->Sleep(TimeDelta::Millis(5));
+  }
+  auto feedback_list = gen->PopFeedback();
+  ASSERT_GT(feedback_list.size(), 0u);
+  EXPECT_NEAR(feedback_list[0].LostWithSendInfo().size(),
+              feedback_list[0].ReceivedWithSendInfo().size(), 2);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/g3doc/index.md b/third_party/libwebrtc/test/network/g3doc/index.md
new file mode 100644
index 0000000000..c82b56445e
--- /dev/null
+++ b/third_party/libwebrtc/test/network/g3doc/index.md
@@ -0,0 +1,137 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'titovartem' reviewed: '2021-03-01'} *-->
+
+# Network Emulation Framework
+
+[TOC]
+
+## Disclamer
+
+This documentation explain the implementation details of Network Emulation
+Framework. Framework's public APIs are located in:
+
+*   [`/api/test/network_emulation_manager.h`](https://source.chromium.org/search?q=%2Fapi%2Ftest%2Fnetwork_emulation_manager.h)
+*   [`/api/test/create_network_emulation_manager.h`](https://source.chromium.org/search?q=%2Fapi%2Ftest%2Fcreate_network_emulation_manager.h)
+*   [`/api/test/network_emulation/network_emulation_interfaces.h`](https://source.chromium.org/search?q=%2Fapi%2Ftest%2Fnetwork_emulation%2Fnetwork_emulation_interfaces.h)
+*   [`/api/test/simulated_network.h`](https://source.chromium.org/search?q=%2Fapi%2Ftest%2Fsimulated_network.h)
+
+## Overview
+
+Network Emulation Framework provides an ability to emulate network behavior
+between different clients, including a WebRTC PeerConnection client. To
+configure network behavior, the user can choose different options:
+
+*   Use predefined implementation that can be configured with parameters such as
+    packet loss, bandwidth, delay, etc.
+*   Custom implementation
+
+Conceptually the framework provides the ability to define multiple endpoints and
+routes used to connect them. All network related entities are created and
+managed by single factory class `webrtc::NetworkEmulationManager` which is
+implemented by `webrtc::test::NetworkEmulationManagerImpl` and can work in two
+modes:
+
+*   Real time
+*   Simulated time
+
+The manager has a dedicated task queue which pipes all packets through all
+network routes from senders to receivers. This task queue behaviour is
+determined by `webrtc::TimeController`, which is based on either in real time or
+simulated time mode.
+
+The network operates on IP level and supports only UDP for now.
+
+## Abstractions
+
+The framework contains the following public abstractions:
+
+*   `webrtc::NetworkBehaviorInterface` - defines how emulated network should
+    behave. It operates on packets metadata level and is responsible for telling
+    which packet at which time have to be delivered to the next receiver.
+
+*   `webrtc::EmulatedIpPacket` - represents a single packet that can be sent or
+    received via emulated network. It has source and destination address and
+    payload to transfer.
+
+*   `webrtc::EmulatedNetworkReceiverInterface` - generic packet receiver
+    interface.
+
+*   `webrtc::EmulatedEndpoint` - primary user facing abstraction of the
+    framework. It represents a network interface on client's machine. It has its
+    own unique IP address and can be used to send and receive packets.
+
+    `EmulatedEndpoint` implements `EmulatedNetworkReceiverInterface` to receive
+    packets from the network and provides an API to send packets to the network
+    and API to bind other `EmulatedNetworkReceiverInterface` which will be able
+    to receive packets from the endpoint. `EmulatedEndpoint` interface has the
+    only implementation: `webrtc::test::EmulatedEndpointImpl`.
+
+*   `webrtc::EmulatedNetworkNode` - represents single network in the real world,
+    like a 3G network between peers, or Wi-Fi for one peer and LTE for another.
+    Each `EmulatedNetworkNode` is a single direction connetion and to form
+    bidirectional connection between endpoints two nodes should be used.
+    Multiple nodes can be joined into chain emulating a network path from one
+    peer to another.
+
+    In public API this class is forward declared and fully accessible only by
+    the framework implementation.
+
+    Internally consist of two parts: `LinkEmulation`, which is responsible for
+    behavior of current `EmulatedNetworkNode` and `NetworkRouterNode` which is
+    responsible for routing packets to the next node or to the endpoint.
+
+*   `webrtc::EmulatedRoute` - represents single route from one network interface
+    on one device to another network interface on another device.
+
+    In public API this class is forward declared and fully accessible only by
+    the framework implementation.
+
+    It contains start and end endpoint and ordered list of `EmulatedNetworkNode`
+    which forms the single directional route between those endpoints.
+
+The framework has also the following private abstractions:
+
+*   `webrtc::test::NetworkRouterNode` - an `EmulatedNetworkReceiverInterface`
+    that can route incoming packets to the next receiver based on internal IP
+    routing table.
+
+*   `webrtc::test::LinkEmulation` - an `EmulatedNetworkReceiverInterface` that
+    can emulate network leg behavior via `webrtc::NetworkBehaviorInterface`
+    interface.
+
+For integrating with `webrtc::PeerConnection` there are helper abstractions:
+
+*   `webrtc::EmulatedNetworkManagerInterface` which is implemented by
+    `webrtc::test::EmulatedNetworkManager` and provides `rtc::Thread` and
+    `rtc::NetworkManager` for WebRTC to use as network thread for
+    `PeerConnection` and for `cricket::BasicPortAllocator`.
+
+    Implementation represent framework endpoints as `rtc::Network` to WebRTC.
+
+## Architecture
+
+Let's take a look on how framework's abstractions are connected to each other.
+
+When the user wants to setup emulated network, first of all, they should create
+an instance of `NetworkEmulationManager` using
+`webrtc::CreateNetworkEmulationManager(...)` API. Then user should use a manager
+to create at least one `EmulatedEndpoint` for each client. After endpoints, the
+user should create required `EmulatedNetworkNode`s and with help of manager
+chain them into `EmulatedRoute`s conecting desired endpoints.
+
+Here is a visual overview of the emulated network architecture:
+
+![Architecture](network_emulation_framework.png "Architecture")
+
+When network is hooked into `PeerConnection` it is done through network thread
+and `NetworkManager`. In the network thread the custom `rtc::SocketServer` is
+provided: `webrtc::test::FakeNetworkSocketServer`. This custom socket server
+will construct custom sockets (`webrtc::test::FakeNetworkSocket`), which
+internally bind themselves to the required endpoint. All packets processing
+inside socket have to be done on the `PeerConnection`'s network thread. When
+packet is going from `PeerConnection` to the network it's already comming from
+the network thread and when it's comming from the emulated network switch from
+the Network Emulation Framework internal task queue and `PeerConnection`'s
+network thread is done inside socket's `OnPacketReceived(...)` method.
+
+![Network Injection](network_injection_into_peer_connection.png "Network Injection")
diff --git a/third_party/libwebrtc/test/network/g3doc/network_emulation_framework.png b/third_party/libwebrtc/test/network/g3doc/network_emulation_framework.png
new file mode 100644
index 0000000000..afec47773f
--- /dev/null
+++ b/third_party/libwebrtc/test/network/g3doc/network_emulation_framework.png
diff --git a/third_party/libwebrtc/test/network/g3doc/network_injection_into_peer_connection.png b/third_party/libwebrtc/test/network/g3doc/network_injection_into_peer_connection.png
new file mode 100644
index 0000000000..c9e3bf8baf
--- /dev/null
+++ b/third_party/libwebrtc/test/network/g3doc/network_injection_into_peer_connection.png
diff --git a/third_party/libwebrtc/test/network/network_emulation.cc b/third_party/libwebrtc/test/network/network_emulation.cc
new file mode 100644
index 0000000000..f1c9ca80dd
--- /dev/null
+++ b/third_party/libwebrtc/test/network/network_emulation.cc
@@ -0,0 +1,767 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/network_emulation.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/sequence_checker.h"
+#include "api/test/network_emulation/network_emulation_interfaces.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace {
+
+EmulatedNetworkOutgoingStats GetOverallOutgoingStats(
+    const std::map<rtc::IPAddress, EmulatedNetworkOutgoingStats>&
+        outgoing_stats,
+    EmulatedNetworkStatsGatheringMode mode) {
+  EmulatedNetworkOutgoingStatsBuilder builder(mode);
+  for (const auto& entry : outgoing_stats) {
+    builder.AddOutgoingStats(entry.second);
+  }
+  return builder.Build();
+}
+
+EmulatedNetworkIncomingStats GetOverallIncomingStats(
+    const std::map<rtc::IPAddress, EmulatedNetworkIncomingStats>&
+        incoming_stats,
+    EmulatedNetworkStatsGatheringMode mode) {
+  EmulatedNetworkIncomingStatsBuilder builder(mode);
+  for (const auto& entry : incoming_stats) {
+    builder.AddIncomingStats(entry.second);
+  }
+  return builder.Build();
+}
+
+}  // namespace
+
+EmulatedNetworkOutgoingStatsBuilder::EmulatedNetworkOutgoingStatsBuilder(
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : stats_gathering_mode_(stats_gathering_mode) {
+  sequence_checker_.Detach();
+}
+
+void EmulatedNetworkOutgoingStatsBuilder::OnPacketSent(Timestamp sent_time,
+                                                       DataSize packet_size) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_CHECK_GE(packet_size, DataSize::Zero());
+  if (stats_.first_packet_sent_time.IsInfinite()) {
+    stats_.first_packet_sent_time = sent_time;
+    stats_.first_sent_packet_size = packet_size;
+  }
+  stats_.last_packet_sent_time = sent_time;
+  stats_.packets_sent++;
+  stats_.bytes_sent += packet_size;
+  if (stats_gathering_mode_ == EmulatedNetworkStatsGatheringMode::kDebug) {
+    stats_.sent_packets_size.AddSample(packet_size.bytes());
+  }
+}
+
+void EmulatedNetworkOutgoingStatsBuilder::AddOutgoingStats(
+    const EmulatedNetworkOutgoingStats& stats) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  stats_.packets_sent += stats.packets_sent;
+  stats_.bytes_sent += stats.bytes_sent;
+  stats_.sent_packets_size.AddSamples(stats.sent_packets_size);
+  if (stats_.first_packet_sent_time > stats.first_packet_sent_time) {
+    stats_.first_packet_sent_time = stats.first_packet_sent_time;
+    stats_.first_sent_packet_size = stats.first_sent_packet_size;
+  }
+  if (stats_.last_packet_sent_time < stats.last_packet_sent_time) {
+    stats_.last_packet_sent_time = stats.last_packet_sent_time;
+  }
+}
+
+EmulatedNetworkOutgoingStats EmulatedNetworkOutgoingStatsBuilder::Build()
+    const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return stats_;
+}
+
+EmulatedNetworkIncomingStatsBuilder::EmulatedNetworkIncomingStatsBuilder(
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : stats_gathering_mode_(stats_gathering_mode) {
+  sequence_checker_.Detach();
+}
+
+void EmulatedNetworkIncomingStatsBuilder::OnPacketDropped(
+    DataSize packet_size) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  stats_.packets_discarded_no_receiver++;
+  stats_.bytes_discarded_no_receiver += packet_size;
+  if (stats_gathering_mode_ == EmulatedNetworkStatsGatheringMode::kDebug) {
+    stats_.packets_discarded_no_receiver_size.AddSample(packet_size.bytes());
+  }
+}
+
+void EmulatedNetworkIncomingStatsBuilder::OnPacketReceived(
+    Timestamp received_time,
+    DataSize packet_size) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  RTC_CHECK_GE(packet_size, DataSize::Zero());
+  if (stats_.first_packet_received_time.IsInfinite()) {
+    stats_.first_packet_received_time = received_time;
+    stats_.first_received_packet_size = packet_size;
+  }
+  stats_.last_packet_received_time = received_time;
+  stats_.packets_received++;
+  stats_.bytes_received += packet_size;
+  if (stats_gathering_mode_ == EmulatedNetworkStatsGatheringMode::kDebug) {
+    stats_.received_packets_size.AddSample(packet_size.bytes());
+  }
+}
+
+void EmulatedNetworkIncomingStatsBuilder::AddIncomingStats(
+    const EmulatedNetworkIncomingStats& stats) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  stats_.packets_received += stats.packets_received;
+  stats_.bytes_received += stats.bytes_received;
+  stats_.received_packets_size.AddSamples(stats.received_packets_size);
+  stats_.packets_discarded_no_receiver += stats.packets_discarded_no_receiver;
+  stats_.bytes_discarded_no_receiver += stats.bytes_discarded_no_receiver;
+  stats_.packets_discarded_no_receiver_size.AddSamples(
+      stats.packets_discarded_no_receiver_size);
+  if (stats_.first_packet_received_time > stats.first_packet_received_time) {
+    stats_.first_packet_received_time = stats.first_packet_received_time;
+    stats_.first_received_packet_size = stats.first_received_packet_size;
+  }
+  if (stats_.last_packet_received_time < stats.last_packet_received_time) {
+    stats_.last_packet_received_time = stats.last_packet_received_time;
+  }
+}
+
+EmulatedNetworkIncomingStats EmulatedNetworkIncomingStatsBuilder::Build()
+    const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return stats_;
+}
+
+EmulatedNetworkStatsBuilder::EmulatedNetworkStatsBuilder(
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : stats_gathering_mode_(stats_gathering_mode) {
+  sequence_checker_.Detach();
+}
+
+EmulatedNetworkStatsBuilder::EmulatedNetworkStatsBuilder(
+    rtc::IPAddress local_ip,
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : stats_gathering_mode_(stats_gathering_mode) {
+  local_addresses_.push_back(local_ip);
+  sequence_checker_.Detach();
+}
+
+void EmulatedNetworkStatsBuilder::OnPacketSent(Timestamp queued_time,
+                                               Timestamp sent_time,
+                                               rtc::IPAddress destination_ip,
+                                               DataSize packet_size) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (stats_gathering_mode_ == EmulatedNetworkStatsGatheringMode::kDebug) {
+    sent_packets_queue_wait_time_us_.AddSample((sent_time - queued_time).us());
+  }
+  auto it = outgoing_stats_per_destination_.find(destination_ip);
+  if (it == outgoing_stats_per_destination_.end()) {
+    outgoing_stats_per_destination_
+        .emplace(destination_ip,
+                 std::make_unique<EmulatedNetworkOutgoingStatsBuilder>(
+                     stats_gathering_mode_))
+        .first->second->OnPacketSent(sent_time, packet_size);
+  } else {
+    it->second->OnPacketSent(sent_time, packet_size);
+  }
+}
+
+void EmulatedNetworkStatsBuilder::OnPacketDropped(rtc::IPAddress source_ip,
+                                                  DataSize packet_size) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  auto it = incoming_stats_per_source_.find(source_ip);
+  if (it == incoming_stats_per_source_.end()) {
+    incoming_stats_per_source_
+        .emplace(source_ip,
+                 std::make_unique<EmulatedNetworkIncomingStatsBuilder>(
+                     stats_gathering_mode_))
+        .first->second->OnPacketDropped(packet_size);
+  } else {
+    it->second->OnPacketDropped(packet_size);
+  }
+}
+
+void EmulatedNetworkStatsBuilder::OnPacketReceived(Timestamp received_time,
+                                                   rtc::IPAddress source_ip,
+                                                   DataSize packet_size) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  auto it = incoming_stats_per_source_.find(source_ip);
+  if (it == incoming_stats_per_source_.end()) {
+    incoming_stats_per_source_
+        .emplace(source_ip,
+                 std::make_unique<EmulatedNetworkIncomingStatsBuilder>(
+                     stats_gathering_mode_))
+        .first->second->OnPacketReceived(received_time, packet_size);
+  } else {
+    it->second->OnPacketReceived(received_time, packet_size);
+  }
+}
+
+void EmulatedNetworkStatsBuilder::AddEmulatedNetworkStats(
+    const EmulatedNetworkStats& stats) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+
+  // Append IPs from other endpoints stats to the builder.
+  for (const rtc::IPAddress& addr : stats.local_addresses) {
+    local_addresses_.push_back(addr);
+  }
+
+  sent_packets_queue_wait_time_us_.AddSamples(
+      stats.sent_packets_queue_wait_time_us);
+
+  // Add outgoing stats from other endpoints to the builder.
+  for (const auto& entry : stats.outgoing_stats_per_destination) {
+    auto it = outgoing_stats_per_destination_.find(entry.first);
+    if (it == outgoing_stats_per_destination_.end()) {
+      outgoing_stats_per_destination_
+          .emplace(entry.first,
+                   std::make_unique<EmulatedNetworkOutgoingStatsBuilder>(
+                       stats_gathering_mode_))
+          .first->second->AddOutgoingStats(entry.second);
+    } else {
+      it->second->AddOutgoingStats(entry.second);
+    }
+  }
+
+  // Add incoming stats from other endpoints to the builder.
+  for (const auto& entry : stats.incoming_stats_per_source) {
+    auto it = incoming_stats_per_source_.find(entry.first);
+    if (it == incoming_stats_per_source_.end()) {
+      incoming_stats_per_source_
+          .emplace(entry.first,
+                   std::make_unique<EmulatedNetworkIncomingStatsBuilder>(
+                       stats_gathering_mode_))
+          .first->second->AddIncomingStats(entry.second);
+    } else {
+      it->second->AddIncomingStats(entry.second);
+    }
+  }
+}
+
+EmulatedNetworkStats EmulatedNetworkStatsBuilder::Build() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  std::map<rtc::IPAddress, EmulatedNetworkOutgoingStats> outgoing_stats;
+  for (const auto& entry : outgoing_stats_per_destination_) {
+    outgoing_stats.emplace(entry.first, entry.second->Build());
+  }
+  std::map<rtc::IPAddress, EmulatedNetworkIncomingStats> incoming_stats;
+  for (const auto& entry : incoming_stats_per_source_) {
+    incoming_stats.emplace(entry.first, entry.second->Build());
+  }
+  return EmulatedNetworkStats{
+      .local_addresses = local_addresses_,
+      .overall_outgoing_stats =
+          GetOverallOutgoingStats(outgoing_stats, stats_gathering_mode_),
+      .overall_incoming_stats =
+          GetOverallIncomingStats(incoming_stats, stats_gathering_mode_),
+      .outgoing_stats_per_destination = std::move(outgoing_stats),
+      .incoming_stats_per_source = std::move(incoming_stats),
+      .sent_packets_queue_wait_time_us = sent_packets_queue_wait_time_us_};
+}
+
+EmulatedNetworkNodeStatsBuilder::EmulatedNetworkNodeStatsBuilder(
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : stats_gathering_mode_(stats_gathering_mode) {
+  sequence_checker_.Detach();
+}
+
+void EmulatedNetworkNodeStatsBuilder::AddPacketTransportTime(
+    TimeDelta time,
+    size_t packet_size) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (stats_gathering_mode_ == EmulatedNetworkStatsGatheringMode::kDebug) {
+    stats_.packet_transport_time.AddSample(time.ms<double>());
+    stats_.size_to_packet_transport_time.AddSample(packet_size /
+                                                   time.ms<double>());
+  }
+}
+
+void EmulatedNetworkNodeStatsBuilder::AddEmulatedNetworkNodeStats(
+    const EmulatedNetworkNodeStats& stats) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  stats_.packet_transport_time.AddSamples(stats.packet_transport_time);
+  stats_.size_to_packet_transport_time.AddSamples(
+      stats.size_to_packet_transport_time);
+}
+
+EmulatedNetworkNodeStats EmulatedNetworkNodeStatsBuilder::Build() const {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  return stats_;
+}
+
+void LinkEmulation::OnPacketReceived(EmulatedIpPacket packet) {
+  task_queue_->PostTask([this, packet = std::move(packet)]() mutable {
+    RTC_DCHECK_RUN_ON(task_queue_);
+
+    uint64_t packet_id = next_packet_id_++;
+    bool sent = network_behavior_->EnqueuePacket(PacketInFlightInfo(
+        packet.ip_packet_size(), packet.arrival_time.us(), packet_id));
+    if (sent) {
+      packets_.emplace_back(StoredPacket{.id = packet_id,
+                                         .sent_time = clock_->CurrentTime(),
+                                         .packet = std::move(packet),
+                                         .removed = false});
+    }
+    if (process_task_.Running())
+      return;
+    absl::optional<int64_t> next_time_us =
+        network_behavior_->NextDeliveryTimeUs();
+    if (!next_time_us)
+      return;
+    Timestamp current_time = clock_->CurrentTime();
+    process_task_ = RepeatingTaskHandle::DelayedStart(
+        task_queue_->Get(),
+        std::max(TimeDelta::Zero(),
+                 Timestamp::Micros(*next_time_us) - current_time),
+        [this]() {
+          RTC_DCHECK_RUN_ON(task_queue_);
+          Timestamp current_time = clock_->CurrentTime();
+          Process(current_time);
+          absl::optional<int64_t> next_time_us =
+              network_behavior_->NextDeliveryTimeUs();
+          if (!next_time_us) {
+            process_task_.Stop();
+            return TimeDelta::Zero();  // This is ignored.
+          }
+          RTC_DCHECK_GE(*next_time_us, current_time.us());
+          return Timestamp::Micros(*next_time_us) - current_time;
+        });
+  });
+}
+
+EmulatedNetworkNodeStats LinkEmulation::stats() const {
+  RTC_DCHECK_RUN_ON(task_queue_);
+  return stats_builder_.Build();
+}
+
+void LinkEmulation::Process(Timestamp at_time) {
+  std::vector<PacketDeliveryInfo> delivery_infos =
+      network_behavior_->DequeueDeliverablePackets(at_time.us());
+  for (PacketDeliveryInfo& delivery_info : delivery_infos) {
+    StoredPacket* packet = nullptr;
+    for (auto& stored_packet : packets_) {
+      if (stored_packet.id == delivery_info.packet_id) {
+        packet = &stored_packet;
+        break;
+      }
+    }
+    RTC_CHECK(packet);
+    RTC_DCHECK(!packet->removed);
+    packet->removed = true;
+    stats_builder_.AddPacketTransportTime(
+        clock_->CurrentTime() - packet->sent_time,
+        packet->packet.ip_packet_size());
+
+    if (delivery_info.receive_time_us != PacketDeliveryInfo::kNotReceived) {
+      packet->packet.arrival_time =
+          Timestamp::Micros(delivery_info.receive_time_us);
+      receiver_->OnPacketReceived(std::move(packet->packet));
+    }
+    while (!packets_.empty() && packets_.front().removed) {
+      packets_.pop_front();
+    }
+  }
+}
+
+NetworkRouterNode::NetworkRouterNode(rtc::TaskQueue* task_queue)
+    : task_queue_(task_queue) {}
+
+void NetworkRouterNode::OnPacketReceived(EmulatedIpPacket packet) {
+  RTC_DCHECK_RUN_ON(task_queue_);
+  if (watcher_) {
+    watcher_(packet);
+  }
+  if (filter_) {
+    if (!filter_(packet))
+      return;
+  }
+  auto receiver_it = routing_.find(packet.to.ipaddr());
+  if (receiver_it == routing_.end()) {
+    if (default_receiver_.has_value()) {
+      (*default_receiver_)->OnPacketReceived(std::move(packet));
+    }
+    return;
+  }
+  RTC_CHECK(receiver_it != routing_.end());
+
+  receiver_it->second->OnPacketReceived(std::move(packet));
+}
+
+void NetworkRouterNode::SetReceiver(
+    const rtc::IPAddress& dest_ip,
+    EmulatedNetworkReceiverInterface* receiver) {
+  task_queue_->PostTask([=] {
+    RTC_DCHECK_RUN_ON(task_queue_);
+    EmulatedNetworkReceiverInterface* cur_receiver = routing_[dest_ip];
+    RTC_CHECK(cur_receiver == nullptr || cur_receiver == receiver)
+        << "Routing for dest_ip=" << dest_ip.ToString() << " already exists";
+    routing_[dest_ip] = receiver;
+  });
+}
+
+void NetworkRouterNode::RemoveReceiver(const rtc::IPAddress& dest_ip) {
+  RTC_DCHECK_RUN_ON(task_queue_);
+  routing_.erase(dest_ip);
+}
+
+void NetworkRouterNode::SetDefaultReceiver(
+    EmulatedNetworkReceiverInterface* receiver) {
+  task_queue_->PostTask([=] {
+    RTC_DCHECK_RUN_ON(task_queue_);
+    if (default_receiver_.has_value()) {
+      RTC_CHECK_EQ(*default_receiver_, receiver)
+          << "Router already default receiver";
+    }
+    default_receiver_ = receiver;
+  });
+}
+
+void NetworkRouterNode::RemoveDefaultReceiver() {
+  RTC_DCHECK_RUN_ON(task_queue_);
+  default_receiver_ = absl::nullopt;
+}
+
+void NetworkRouterNode::SetWatcher(
+    std::function<void(const EmulatedIpPacket&)> watcher) {
+  task_queue_->PostTask([=] {
+    RTC_DCHECK_RUN_ON(task_queue_);
+    watcher_ = watcher;
+  });
+}
+
+void NetworkRouterNode::SetFilter(
+    std::function<bool(const EmulatedIpPacket&)> filter) {
+  task_queue_->PostTask([=] {
+    RTC_DCHECK_RUN_ON(task_queue_);
+    filter_ = filter;
+  });
+}
+
+EmulatedNetworkNode::EmulatedNetworkNode(
+    Clock* clock,
+    rtc::TaskQueue* task_queue,
+    std::unique_ptr<NetworkBehaviorInterface> network_behavior,
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : router_(task_queue),
+      link_(clock,
+            task_queue,
+            std::move(network_behavior),
+            &router_,
+            stats_gathering_mode) {}
+
+void EmulatedNetworkNode::OnPacketReceived(EmulatedIpPacket packet) {
+  link_.OnPacketReceived(std::move(packet));
+}
+
+EmulatedNetworkNodeStats EmulatedNetworkNode::stats() const {
+  return link_.stats();
+}
+
+void EmulatedNetworkNode::CreateRoute(
+    const rtc::IPAddress& receiver_ip,
+    std::vector<EmulatedNetworkNode*> nodes,
+    EmulatedNetworkReceiverInterface* receiver) {
+  RTC_CHECK(!nodes.empty());
+  for (size_t i = 0; i + 1 < nodes.size(); ++i)
+    nodes[i]->router()->SetReceiver(receiver_ip, nodes[i + 1]);
+  nodes.back()->router()->SetReceiver(receiver_ip, receiver);
+}
+
+void EmulatedNetworkNode::ClearRoute(const rtc::IPAddress& receiver_ip,
+                                     std::vector<EmulatedNetworkNode*> nodes) {
+  for (EmulatedNetworkNode* node : nodes)
+    node->router()->RemoveReceiver(receiver_ip);
+}
+
+EmulatedNetworkNode::~EmulatedNetworkNode() = default;
+
+EmulatedEndpointImpl::Options::Options(
+    uint64_t id,
+    const rtc::IPAddress& ip,
+    const EmulatedEndpointConfig& config,
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : id(id),
+      ip(ip),
+      stats_gathering_mode(stats_gathering_mode),
+      type(config.type),
+      allow_send_packet_with_different_source_ip(
+          config.allow_send_packet_with_different_source_ip),
+      allow_receive_packets_with_different_dest_ip(
+          config.allow_receive_packets_with_different_dest_ip),
+      log_name(ip.ToString() + " (" + config.name.value_or("") + ")") {}
+
+EmulatedEndpointImpl::EmulatedEndpointImpl(const Options& options,
+                                           bool is_enabled,
+                                           rtc::TaskQueue* task_queue,
+                                           Clock* clock)
+    : options_(options),
+      is_enabled_(is_enabled),
+      clock_(clock),
+      task_queue_(task_queue),
+      router_(task_queue_),
+      next_port_(kFirstEphemeralPort),
+      stats_builder_(options_.ip, options_.stats_gathering_mode) {
+  constexpr int kIPv4NetworkPrefixLength = 24;
+  constexpr int kIPv6NetworkPrefixLength = 64;
+
+  int prefix_length = 0;
+  if (options_.ip.family() == AF_INET) {
+    prefix_length = kIPv4NetworkPrefixLength;
+  } else if (options_.ip.family() == AF_INET6) {
+    prefix_length = kIPv6NetworkPrefixLength;
+  }
+  rtc::IPAddress prefix = TruncateIP(options_.ip, prefix_length);
+  network_ = std::make_unique<rtc::Network>(
+      options_.ip.ToString(), "Endpoint id=" + std::to_string(options_.id),
+      prefix, prefix_length, options_.type);
+  network_->AddIP(options_.ip);
+
+  enabled_state_checker_.Detach();
+  RTC_LOG(LS_INFO) << "Created emulated endpoint " << options_.log_name
+                   << "; id=" << options_.id;
+}
+EmulatedEndpointImpl::~EmulatedEndpointImpl() = default;
+
+uint64_t EmulatedEndpointImpl::GetId() const {
+  return options_.id;
+}
+
+void EmulatedEndpointImpl::SendPacket(const rtc::SocketAddress& from,
+                                      const rtc::SocketAddress& to,
+                                      rtc::CopyOnWriteBuffer packet_data,
+                                      uint16_t application_overhead) {
+  if (!options_.allow_send_packet_with_different_source_ip) {
+    RTC_CHECK(from.ipaddr() == options_.ip);
+  }
+  EmulatedIpPacket packet(from, to, std::move(packet_data),
+                          clock_->CurrentTime(), application_overhead);
+  task_queue_->PostTask([this, packet = std::move(packet)]() mutable {
+    RTC_DCHECK_RUN_ON(task_queue_);
+    stats_builder_.OnPacketSent(packet.arrival_time, clock_->CurrentTime(),
+                                packet.to.ipaddr(),
+                                DataSize::Bytes(packet.ip_packet_size()));
+
+    if (packet.to.ipaddr() == options_.ip) {
+      OnPacketReceived(std::move(packet));
+    } else {
+      router_.OnPacketReceived(std::move(packet));
+    }
+  });
+}
+
+absl::optional<uint16_t> EmulatedEndpointImpl::BindReceiver(
+    uint16_t desired_port,
+    EmulatedNetworkReceiverInterface* receiver) {
+  return BindReceiverInternal(desired_port, receiver, /*is_one_shot=*/false);
+}
+
+absl::optional<uint16_t> EmulatedEndpointImpl::BindOneShotReceiver(
+    uint16_t desired_port,
+    EmulatedNetworkReceiverInterface* receiver) {
+  return BindReceiverInternal(desired_port, receiver, /*is_one_shot=*/true);
+}
+
+absl::optional<uint16_t> EmulatedEndpointImpl::BindReceiverInternal(
+    uint16_t desired_port,
+    EmulatedNetworkReceiverInterface* receiver,
+    bool is_one_shot) {
+  MutexLock lock(&receiver_lock_);
+  uint16_t port = desired_port;
+  if (port == 0) {
+    // Because client can specify its own port, next_port_ can be already in
+    // use, so we need to find next available port.
+    int ports_pool_size =
+        std::numeric_limits<uint16_t>::max() - kFirstEphemeralPort + 1;
+    for (int i = 0; i < ports_pool_size; ++i) {
+      uint16_t next_port = NextPort();
+      if (port_to_receiver_.find(next_port) == port_to_receiver_.end()) {
+        port = next_port;
+        break;
+      }
+    }
+  }
+  RTC_CHECK(port != 0) << "Can't find free port for receiver in endpoint "
+                       << options_.log_name << "; id=" << options_.id;
+  bool result =
+      port_to_receiver_.insert({port, {receiver, is_one_shot}}).second;
+  if (!result) {
+    RTC_LOG(LS_INFO) << "Can't bind receiver to used port " << desired_port
+                     << " in endpoint " << options_.log_name
+                     << "; id=" << options_.id;
+    return absl::nullopt;
+  }
+  RTC_LOG(LS_INFO) << "New receiver is binded to endpoint " << options_.log_name
+                   << "; id=" << options_.id << " on port " << port;
+  return port;
+}
+
+uint16_t EmulatedEndpointImpl::NextPort() {
+  uint16_t out = next_port_;
+  if (next_port_ == std::numeric_limits<uint16_t>::max()) {
+    next_port_ = kFirstEphemeralPort;
+  } else {
+    next_port_++;
+  }
+  return out;
+}
+
+void EmulatedEndpointImpl::UnbindReceiver(uint16_t port) {
+  MutexLock lock(&receiver_lock_);
+  RTC_LOG(LS_INFO) << "Receiver is removed on port " << port
+                   << " from endpoint " << options_.log_name
+                   << "; id=" << options_.id;
+  port_to_receiver_.erase(port);
+}
+
+void EmulatedEndpointImpl::BindDefaultReceiver(
+    EmulatedNetworkReceiverInterface* receiver) {
+  MutexLock lock(&receiver_lock_);
+  RTC_CHECK(!default_receiver_.has_value())
+      << "Endpoint " << options_.log_name << "; id=" << options_.id
+      << " already has default receiver";
+  RTC_LOG(LS_INFO) << "Default receiver is binded to endpoint "
+                   << options_.log_name << "; id=" << options_.id;
+  default_receiver_ = receiver;
+}
+
+void EmulatedEndpointImpl::UnbindDefaultReceiver() {
+  MutexLock lock(&receiver_lock_);
+  RTC_LOG(LS_INFO) << "Default receiver is removed from endpoint "
+                   << options_.log_name << "; id=" << options_.id;
+  default_receiver_ = absl::nullopt;
+}
+
+rtc::IPAddress EmulatedEndpointImpl::GetPeerLocalAddress() const {
+  return options_.ip;
+}
+
+void EmulatedEndpointImpl::OnPacketReceived(EmulatedIpPacket packet) {
+  RTC_DCHECK_RUN_ON(task_queue_);
+  if (!options_.allow_receive_packets_with_different_dest_ip) {
+    RTC_CHECK(packet.to.ipaddr() == options_.ip)
+        << "Routing error: wrong destination endpoint. Packet.to.ipaddr()=: "
+        << packet.to.ipaddr().ToString()
+        << "; Receiver options_.ip=" << options_.ip.ToString();
+  }
+  MutexLock lock(&receiver_lock_);
+  stats_builder_.OnPacketReceived(clock_->CurrentTime(), packet.from.ipaddr(),
+                                  DataSize::Bytes(packet.ip_packet_size()));
+  auto it = port_to_receiver_.find(packet.to.port());
+  if (it == port_to_receiver_.end()) {
+    if (default_receiver_.has_value()) {
+      (*default_receiver_)->OnPacketReceived(std::move(packet));
+      return;
+    }
+    // It can happen, that remote peer closed connection, but there still some
+    // packets, that are going to it. It can happen during peer connection close
+    // process: one peer closed connection, second still sending data.
+    RTC_LOG(LS_INFO) << "Drop packet: no receiver registered in "
+                     << options_.log_name << "; id=" << options_.id
+                     << " on port " << packet.to.port()
+                     << ". Packet source: " << packet.from.ToString();
+    stats_builder_.OnPacketDropped(packet.from.ipaddr(),
+                                   DataSize::Bytes(packet.ip_packet_size()));
+    return;
+  }
+  // Endpoint holds lock during packet processing to ensure that a call to
+  // UnbindReceiver followed by a delete of the receiver cannot race with this
+  // call to OnPacketReceived.
+  it->second.receiver->OnPacketReceived(std::move(packet));
+
+  if (it->second.is_one_shot) {
+    port_to_receiver_.erase(it);
+  }
+}
+
+void EmulatedEndpointImpl::Enable() {
+  RTC_DCHECK_RUN_ON(&enabled_state_checker_);
+  RTC_CHECK(!is_enabled_);
+  is_enabled_ = true;
+}
+
+void EmulatedEndpointImpl::Disable() {
+  RTC_DCHECK_RUN_ON(&enabled_state_checker_);
+  RTC_CHECK(is_enabled_);
+  is_enabled_ = false;
+}
+
+bool EmulatedEndpointImpl::Enabled() const {
+  RTC_DCHECK_RUN_ON(&enabled_state_checker_);
+  return is_enabled_;
+}
+
+EmulatedNetworkStats EmulatedEndpointImpl::stats() const {
+  RTC_DCHECK_RUN_ON(task_queue_);
+  return stats_builder_.Build();
+}
+
+EmulatedEndpointImpl* EndpointsContainer::LookupByLocalAddress(
+    const rtc::IPAddress& local_ip) const {
+  for (auto* endpoint : endpoints_) {
+    rtc::IPAddress peer_local_address = endpoint->GetPeerLocalAddress();
+    if (peer_local_address == local_ip) {
+      return endpoint;
+    }
+  }
+  RTC_CHECK(false) << "No network found for address" << local_ip.ToString();
+}
+
+EndpointsContainer::EndpointsContainer(
+    const std::vector<EmulatedEndpointImpl*>& endpoints,
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : endpoints_(endpoints), stats_gathering_mode_(stats_gathering_mode) {}
+
+bool EndpointsContainer::HasEndpoint(EmulatedEndpointImpl* endpoint) const {
+  for (auto* e : endpoints_) {
+    if (e->GetId() == endpoint->GetId()) {
+      return true;
+    }
+  }
+  return false;
+}
+
+std::vector<std::unique_ptr<rtc::Network>>
+EndpointsContainer::GetEnabledNetworks() const {
+  std::vector<std::unique_ptr<rtc::Network>> networks;
+  for (auto* endpoint : endpoints_) {
+    if (endpoint->Enabled()) {
+      networks.emplace_back(
+          std::make_unique<rtc::Network>(endpoint->network()));
+    }
+  }
+  return networks;
+}
+
+std::vector<EmulatedEndpoint*> EndpointsContainer::GetEndpoints() const {
+  return std::vector<EmulatedEndpoint*>(endpoints_.begin(), endpoints_.end());
+}
+
+EmulatedNetworkStats EndpointsContainer::GetStats() const {
+  EmulatedNetworkStatsBuilder stats_builder(stats_gathering_mode_);
+  for (auto* endpoint : endpoints_) {
+    stats_builder.AddEmulatedNetworkStats(endpoint->stats());
+  }
+  return stats_builder.Build();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/network_emulation.h b/third_party/libwebrtc/test/network/network_emulation.h
new file mode 100644
index 0000000000..dffabafa7c
--- /dev/null
+++ b/third_party/libwebrtc/test/network/network_emulation.h
@@ -0,0 +1,467 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_NETWORK_NETWORK_EMULATION_H_
+#define TEST_NETWORK_NETWORK_EMULATION_H_
+
+#include <cstdint>
+#include <deque>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/sequence_checker.h"
+#include "api/test/network_emulation/network_emulation_interfaces.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/simulated_network.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/network.h"
+#include "rtc_base/network_constants.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+// All methods of EmulatedNetworkOutgoingStatsBuilder have to be used on a
+// single thread. It may be created on another thread.
+class EmulatedNetworkOutgoingStatsBuilder {
+ public:
+  explicit EmulatedNetworkOutgoingStatsBuilder(
+      EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+
+  void OnPacketSent(Timestamp sent_time, DataSize packet_size);
+
+  void AddOutgoingStats(const EmulatedNetworkOutgoingStats& stats);
+
+  EmulatedNetworkOutgoingStats Build() const;
+
+ private:
+  const EmulatedNetworkStatsGatheringMode stats_gathering_mode_;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  EmulatedNetworkOutgoingStats stats_ RTC_GUARDED_BY(sequence_checker_);
+};
+
+// All methods of EmulatedNetworkIncomingStatsBuilder have to be used on a
+// single thread. It may be created on another thread.
+class EmulatedNetworkIncomingStatsBuilder {
+ public:
+  explicit EmulatedNetworkIncomingStatsBuilder(
+      EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+
+  void OnPacketDropped(DataSize packet_size);
+
+  void OnPacketReceived(Timestamp received_time, DataSize packet_size);
+
+  // Adds stats collected from another endpoints to the builder.
+  void AddIncomingStats(const EmulatedNetworkIncomingStats& stats);
+
+  EmulatedNetworkIncomingStats Build() const;
+
+ private:
+  const EmulatedNetworkStatsGatheringMode stats_gathering_mode_;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  EmulatedNetworkIncomingStats stats_ RTC_GUARDED_BY(sequence_checker_);
+};
+
+// All methods of EmulatedNetworkStatsBuilder have to be used on a single
+// thread. It may be created on another thread.
+class EmulatedNetworkStatsBuilder {
+ public:
+  explicit EmulatedNetworkStatsBuilder(
+      EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+  explicit EmulatedNetworkStatsBuilder(
+      rtc::IPAddress local_ip,
+      EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+
+  void OnPacketSent(Timestamp queued_time,
+                    Timestamp sent_time,
+                    rtc::IPAddress destination_ip,
+                    DataSize packet_size);
+
+  void OnPacketDropped(rtc::IPAddress source_ip, DataSize packet_size);
+
+  void OnPacketReceived(Timestamp received_time,
+                        rtc::IPAddress source_ip,
+                        DataSize packet_size);
+
+  void AddEmulatedNetworkStats(const EmulatedNetworkStats& stats);
+
+  EmulatedNetworkStats Build() const;
+
+ private:
+  const EmulatedNetworkStatsGatheringMode stats_gathering_mode_;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  std::vector<rtc::IPAddress> local_addresses_
+      RTC_GUARDED_BY(sequence_checker_);
+  SamplesStatsCounter sent_packets_queue_wait_time_us_;
+  std::map<rtc::IPAddress, std::unique_ptr<EmulatedNetworkOutgoingStatsBuilder>>
+      outgoing_stats_per_destination_ RTC_GUARDED_BY(sequence_checker_);
+  std::map<rtc::IPAddress, std::unique_ptr<EmulatedNetworkIncomingStatsBuilder>>
+      incoming_stats_per_source_ RTC_GUARDED_BY(sequence_checker_);
+};
+
+// All methods of EmulatedNetworkNodeStatsBuilder have to be used on a
+// single thread. It may be created on another thread.
+class EmulatedNetworkNodeStatsBuilder {
+ public:
+  explicit EmulatedNetworkNodeStatsBuilder(
+      EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+
+  void AddPacketTransportTime(TimeDelta time, size_t packet_size);
+
+  void AddEmulatedNetworkNodeStats(const EmulatedNetworkNodeStats& stats);
+
+  EmulatedNetworkNodeStats Build() const;
+
+ private:
+  const EmulatedNetworkStatsGatheringMode stats_gathering_mode_;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  EmulatedNetworkNodeStats stats_ RTC_GUARDED_BY(sequence_checker_);
+};
+
+class LinkEmulation : public EmulatedNetworkReceiverInterface {
+ public:
+  LinkEmulation(Clock* clock,
+                rtc::TaskQueue* task_queue,
+                std::unique_ptr<NetworkBehaviorInterface> network_behavior,
+                EmulatedNetworkReceiverInterface* receiver,
+                EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+      : clock_(clock),
+        task_queue_(task_queue),
+        network_behavior_(std::move(network_behavior)),
+        receiver_(receiver),
+        stats_builder_(stats_gathering_mode) {}
+  void OnPacketReceived(EmulatedIpPacket packet) override;
+
+  EmulatedNetworkNodeStats stats() const;
+
+ private:
+  struct StoredPacket {
+    uint64_t id;
+    Timestamp sent_time;
+    EmulatedIpPacket packet;
+    bool removed;
+  };
+  void Process(Timestamp at_time) RTC_RUN_ON(task_queue_);
+
+  Clock* const clock_;
+  rtc::TaskQueue* const task_queue_;
+  const std::unique_ptr<NetworkBehaviorInterface> network_behavior_
+      RTC_GUARDED_BY(task_queue_);
+  EmulatedNetworkReceiverInterface* const receiver_;
+
+  RepeatingTaskHandle process_task_ RTC_GUARDED_BY(task_queue_);
+  std::deque<StoredPacket> packets_ RTC_GUARDED_BY(task_queue_);
+  uint64_t next_packet_id_ RTC_GUARDED_BY(task_queue_) = 1;
+
+  EmulatedNetworkNodeStatsBuilder stats_builder_ RTC_GUARDED_BY(task_queue_);
+};
+
+// Represents a component responsible for routing packets based on their IP
+// address. All possible routes have to be set explicitly before packet for
+// desired destination will be seen for the first time. If route is unknown
+// the packet will be silently dropped.
+class NetworkRouterNode : public EmulatedNetworkReceiverInterface {
+ public:
+  explicit NetworkRouterNode(rtc::TaskQueue* task_queue);
+
+  void OnPacketReceived(EmulatedIpPacket packet) override;
+  void SetReceiver(const rtc::IPAddress& dest_ip,
+                   EmulatedNetworkReceiverInterface* receiver);
+  void RemoveReceiver(const rtc::IPAddress& dest_ip);
+  // Sets a default receive that will be used for all incoming packets for which
+  // there is no specific receiver binded to their destination port.
+  void SetDefaultReceiver(EmulatedNetworkReceiverInterface* receiver);
+  void RemoveDefaultReceiver();
+  void SetWatcher(std::function<void(const EmulatedIpPacket&)> watcher);
+  void SetFilter(std::function<bool(const EmulatedIpPacket&)> filter);
+
+ private:
+  rtc::TaskQueue* const task_queue_;
+  absl::optional<EmulatedNetworkReceiverInterface*> default_receiver_
+      RTC_GUARDED_BY(task_queue_);
+  std::map<rtc::IPAddress, EmulatedNetworkReceiverInterface*> routing_
+      RTC_GUARDED_BY(task_queue_);
+  std::function<void(const EmulatedIpPacket&)> watcher_
+      RTC_GUARDED_BY(task_queue_);
+  std::function<bool(const EmulatedIpPacket&)> filter_
+      RTC_GUARDED_BY(task_queue_);
+};
+
+// Represents node in the emulated network. Nodes can be connected with each
+// other to form different networks with different behavior. The behavior of
+// the node itself is determined by a concrete implementation of
+// NetworkBehaviorInterface that is provided on construction.
+class EmulatedNetworkNode : public EmulatedNetworkReceiverInterface {
+ public:
+  // Creates node based on `network_behavior`. The specified `packet_overhead`
+  // is added to the size of each packet in the information provided to
+  // `network_behavior`.
+  // `task_queue` is used to process packets and to forward the packets when
+  // they are ready.
+  EmulatedNetworkNode(
+      Clock* clock,
+      rtc::TaskQueue* task_queue,
+      std::unique_ptr<NetworkBehaviorInterface> network_behavior,
+      EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+  ~EmulatedNetworkNode() override;
+
+  EmulatedNetworkNode(const EmulatedNetworkNode&) = delete;
+  EmulatedNetworkNode& operator=(const EmulatedNetworkNode&) = delete;
+
+  void OnPacketReceived(EmulatedIpPacket packet) override;
+
+  LinkEmulation* link() { return &link_; }
+  NetworkRouterNode* router() { return &router_; }
+  EmulatedNetworkNodeStats stats() const;
+
+  // Creates a route for the given receiver_ip over all the given nodes to the
+  // given receiver.
+  static void CreateRoute(const rtc::IPAddress& receiver_ip,
+                          std::vector<EmulatedNetworkNode*> nodes,
+                          EmulatedNetworkReceiverInterface* receiver);
+  static void ClearRoute(const rtc::IPAddress& receiver_ip,
+                         std::vector<EmulatedNetworkNode*> nodes);
+
+ private:
+  NetworkRouterNode router_;
+  LinkEmulation link_;
+};
+
+// Represents single network interface on the device.
+// It will be used as sender from socket side to send data to the network and
+// will act as packet receiver from emulated network side to receive packets
+// from other EmulatedNetworkNodes.
+class EmulatedEndpointImpl : public EmulatedEndpoint {
+ public:
+  struct Options {
+    Options(uint64_t id,
+            const rtc::IPAddress& ip,
+            const EmulatedEndpointConfig& config,
+            EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+
+    // TODO(titovartem) check if we can remove id.
+    uint64_t id;
+    // Endpoint local IP address.
+    rtc::IPAddress ip;
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode;
+    rtc::AdapterType type;
+    // Allow endpoint to send packets specifying source IP address different to
+    // the current endpoint IP address. If false endpoint will crash if attempt
+    // to send such packet will be done.
+    bool allow_send_packet_with_different_source_ip;
+    // Allow endpoint to receive packet with destination IP address different to
+    // the current endpoint IP address. If false endpoint will crash if such
+    // packet will arrive.
+    bool allow_receive_packets_with_different_dest_ip;
+    // Name of the endpoint used for logging purposes.
+    std::string log_name;
+  };
+
+  EmulatedEndpointImpl(const Options& options,
+                       bool is_enabled,
+                       rtc::TaskQueue* task_queue,
+                       Clock* clock);
+  ~EmulatedEndpointImpl() override;
+
+  uint64_t GetId() const;
+
+  NetworkRouterNode* router() { return &router_; }
+
+  void SendPacket(const rtc::SocketAddress& from,
+                  const rtc::SocketAddress& to,
+                  rtc::CopyOnWriteBuffer packet_data,
+                  uint16_t application_overhead = 0) override;
+
+  absl::optional<uint16_t> BindReceiver(
+      uint16_t desired_port,
+      EmulatedNetworkReceiverInterface* receiver) override;
+  // Binds a receiver, and automatically removes the binding after first call to
+  // OnPacketReceived.
+  absl::optional<uint16_t> BindOneShotReceiver(
+      uint16_t desired_port,
+      EmulatedNetworkReceiverInterface* receiver);
+  void UnbindReceiver(uint16_t port) override;
+  void BindDefaultReceiver(EmulatedNetworkReceiverInterface* receiver) override;
+  void UnbindDefaultReceiver() override;
+
+  rtc::IPAddress GetPeerLocalAddress() const override;
+
+  // Will be called to deliver packet into endpoint from network node.
+  void OnPacketReceived(EmulatedIpPacket packet) override;
+
+  void Enable();
+  void Disable();
+  bool Enabled() const;
+
+  const rtc::Network& network() const { return *network_.get(); }
+
+  EmulatedNetworkStats stats() const;
+
+ private:
+  struct ReceiverBinding {
+    EmulatedNetworkReceiverInterface* receiver;
+    bool is_one_shot;
+  };
+
+  absl::optional<uint16_t> BindReceiverInternal(
+      uint16_t desired_port,
+      EmulatedNetworkReceiverInterface* receiver,
+      bool is_one_shot);
+
+  static constexpr uint16_t kFirstEphemeralPort = 49152;
+  uint16_t NextPort() RTC_EXCLUSIVE_LOCKS_REQUIRED(receiver_lock_);
+
+  Mutex receiver_lock_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker enabled_state_checker_;
+
+  const Options options_;
+  bool is_enabled_ RTC_GUARDED_BY(enabled_state_checker_);
+  Clock* const clock_;
+  rtc::TaskQueue* const task_queue_;
+  std::unique_ptr<rtc::Network> network_;
+  NetworkRouterNode router_;
+
+  uint16_t next_port_ RTC_GUARDED_BY(receiver_lock_);
+  absl::optional<EmulatedNetworkReceiverInterface*> default_receiver_
+      RTC_GUARDED_BY(receiver_lock_);
+  std::map<uint16_t, ReceiverBinding> port_to_receiver_
+      RTC_GUARDED_BY(receiver_lock_);
+
+  EmulatedNetworkStatsBuilder stats_builder_ RTC_GUARDED_BY(task_queue_);
+};
+
+class EmulatedRoute {
+ public:
+  EmulatedRoute(EmulatedEndpointImpl* from,
+                std::vector<EmulatedNetworkNode*> via_nodes,
+                EmulatedEndpointImpl* to,
+                bool is_default)
+      : from(from),
+        via_nodes(std::move(via_nodes)),
+        to(to),
+        active(true),
+        is_default(is_default) {}
+
+  EmulatedEndpointImpl* from;
+  std::vector<EmulatedNetworkNode*> via_nodes;
+  EmulatedEndpointImpl* to;
+  bool active;
+  bool is_default;
+};
+
+// This object is immutable and so thread safe.
+class EndpointsContainer {
+ public:
+  EndpointsContainer(const std::vector<EmulatedEndpointImpl*>& endpoints,
+                     EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+
+  EmulatedEndpointImpl* LookupByLocalAddress(
+      const rtc::IPAddress& local_ip) const;
+  bool HasEndpoint(EmulatedEndpointImpl* endpoint) const;
+  // Returns list of networks for enabled endpoints. Caller takes ownership of
+  // returned rtc::Network objects.
+  std::vector<std::unique_ptr<rtc::Network>> GetEnabledNetworks() const;
+  std::vector<EmulatedEndpoint*> GetEndpoints() const;
+  EmulatedNetworkStats GetStats() const;
+
+ private:
+  const std::vector<EmulatedEndpointImpl*> endpoints_;
+  const EmulatedNetworkStatsGatheringMode stats_gathering_mode_;
+};
+
+template <typename FakePacketType>
+class FakePacketRoute : public EmulatedNetworkReceiverInterface {
+ public:
+  FakePacketRoute(EmulatedRoute* route,
+                  std::function<void(FakePacketType, Timestamp)> action)
+      : route_(route),
+        action_(std::move(action)),
+        send_addr_(route_->from->GetPeerLocalAddress(), 0),
+        recv_addr_(route_->to->GetPeerLocalAddress(),
+                   *route_->to->BindReceiver(0, this)) {}
+
+  ~FakePacketRoute() { route_->to->UnbindReceiver(recv_addr_.port()); }
+
+  void SendPacket(size_t size, FakePacketType packet) {
+    RTC_CHECK_GE(size, sizeof(int));
+    sent_.emplace(next_packet_id_, packet);
+    rtc::CopyOnWriteBuffer buf(size);
+    reinterpret_cast<int*>(buf.MutableData())[0] = next_packet_id_++;
+    route_->from->SendPacket(send_addr_, recv_addr_, buf);
+  }
+
+  void OnPacketReceived(EmulatedIpPacket packet) override {
+    int packet_id = reinterpret_cast<const int*>(packet.data.data())[0];
+    action_(std::move(sent_[packet_id]), packet.arrival_time);
+    sent_.erase(packet_id);
+  }
+
+ private:
+  EmulatedRoute* const route_;
+  const std::function<void(FakePacketType, Timestamp)> action_;
+  const rtc::SocketAddress send_addr_;
+  const rtc::SocketAddress recv_addr_;
+  int next_packet_id_ = 0;
+  std::map<int, FakePacketType> sent_;
+};
+
+template <typename RequestPacketType, typename ResponsePacketType>
+class TwoWayFakeTrafficRoute {
+ public:
+  class TrafficHandlerInterface {
+   public:
+    virtual void OnRequest(RequestPacketType, Timestamp) = 0;
+    virtual void OnResponse(ResponsePacketType, Timestamp) = 0;
+    virtual ~TrafficHandlerInterface() = default;
+  };
+  TwoWayFakeTrafficRoute(TrafficHandlerInterface* handler,
+                         EmulatedRoute* send_route,
+                         EmulatedRoute* ret_route)
+      : handler_(handler),
+        request_handler_{send_route,
+                         [&](RequestPacketType packet, Timestamp arrival_time) {
+                           handler_->OnRequest(std::move(packet), arrival_time);
+                         }},
+        response_handler_{
+            ret_route, [&](ResponsePacketType packet, Timestamp arrival_time) {
+              handler_->OnResponse(std::move(packet), arrival_time);
+            }} {}
+  void SendRequest(size_t size, RequestPacketType packet) {
+    request_handler_.SendPacket(size, std::move(packet));
+  }
+  void SendResponse(size_t size, ResponsePacketType packet) {
+    response_handler_.SendPacket(size, std::move(packet));
+  }
+
+ private:
+  TrafficHandlerInterface* handler_;
+  FakePacketRoute<RequestPacketType> request_handler_;
+  FakePacketRoute<ResponsePacketType> response_handler_;
+};
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_NETWORK_EMULATION_H_
diff --git a/third_party/libwebrtc/test/network/network_emulation_manager.cc b/third_party/libwebrtc/test/network/network_emulation_manager.cc
new file mode 100644
index 0000000000..97c0bc1ba8
--- /dev/null
+++ b/third_party/libwebrtc/test/network/network_emulation_manager.cc
@@ -0,0 +1,373 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/network_emulation_manager.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "call/simulated_network.h"
+#include "test/network/emulated_turn_server.h"
+#include "test/network/traffic_route.h"
+#include "test/time_controller/real_time_controller.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+// uint32_t representation of 192.168.0.0 address
+constexpr uint32_t kMinIPv4Address = 0xC0A80000;
+// uint32_t representation of 192.168.255.255 address
+constexpr uint32_t kMaxIPv4Address = 0xC0A8FFFF;
+
+std::unique_ptr<TimeController> CreateTimeController(TimeMode mode) {
+  switch (mode) {
+    case TimeMode::kRealTime:
+      return std::make_unique<RealTimeController>();
+    case TimeMode::kSimulated:
+      // Using an offset of 100000 to get nice fixed width and readable
+      // timestamps in typical test scenarios.
+      const Timestamp kSimulatedStartTime = Timestamp::Seconds(100000);
+      return std::make_unique<GlobalSimulatedTimeController>(
+          kSimulatedStartTime);
+  }
+}
+}  // namespace
+
+NetworkEmulationManagerImpl::NetworkEmulationManagerImpl(
+    TimeMode mode,
+    EmulatedNetworkStatsGatheringMode stats_gathering_mode)
+    : time_mode_(mode),
+      stats_gathering_mode_(stats_gathering_mode),
+      time_controller_(CreateTimeController(mode)),
+      clock_(time_controller_->GetClock()),
+      next_node_id_(1),
+      next_ip4_address_(kMinIPv4Address),
+      task_queue_(time_controller_->GetTaskQueueFactory()->CreateTaskQueue(
+          "NetworkEmulation",
+          TaskQueueFactory::Priority::NORMAL)) {}
+
+// TODO(srte): Ensure that any pending task that must be run for consistency
+// (such as stats collection tasks) are not cancelled when the task queue is
+// destroyed.
+NetworkEmulationManagerImpl::~NetworkEmulationManagerImpl() {
+  for (auto& turn_server : turn_servers_) {
+    turn_server->Stop();
+  }
+}
+
+EmulatedNetworkNode* NetworkEmulationManagerImpl::CreateEmulatedNode(
+    BuiltInNetworkBehaviorConfig config,
+    uint64_t random_seed) {
+  return CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(config, random_seed));
+}
+
+EmulatedNetworkNode* NetworkEmulationManagerImpl::CreateEmulatedNode(
+    std::unique_ptr<NetworkBehaviorInterface> network_behavior) {
+  auto node = std::make_unique<EmulatedNetworkNode>(
+      clock_, &task_queue_, std::move(network_behavior), stats_gathering_mode_);
+  EmulatedNetworkNode* out = node.get();
+  task_queue_.PostTask([this, node = std::move(node)]() mutable {
+    network_nodes_.push_back(std::move(node));
+  });
+  return out;
+}
+
+NetworkEmulationManager::SimulatedNetworkNode::Builder
+NetworkEmulationManagerImpl::NodeBuilder() {
+  return SimulatedNetworkNode::Builder(this);
+}
+
+EmulatedEndpointImpl* NetworkEmulationManagerImpl::CreateEndpoint(
+    EmulatedEndpointConfig config) {
+  absl::optional<rtc::IPAddress> ip = config.ip;
+  if (!ip) {
+    switch (config.generated_ip_family) {
+      case EmulatedEndpointConfig::IpAddressFamily::kIpv4:
+        ip = GetNextIPv4Address();
+        RTC_CHECK(ip) << "All auto generated IPv4 addresses exhausted";
+        break;
+      case EmulatedEndpointConfig::IpAddressFamily::kIpv6:
+        ip = GetNextIPv4Address();
+        RTC_CHECK(ip) << "All auto generated IPv6 addresses exhausted";
+        ip = ip->AsIPv6Address();
+        break;
+    }
+  }
+
+  bool res = used_ip_addresses_.insert(*ip).second;
+  RTC_CHECK(res) << "IP=" << ip->ToString() << " already in use";
+  auto node = std::make_unique<EmulatedEndpointImpl>(
+      EmulatedEndpointImpl::Options(next_node_id_++, *ip, config,
+                                    stats_gathering_mode_),
+      config.start_as_enabled, &task_queue_, clock_);
+  EmulatedEndpointImpl* out = node.get();
+  endpoints_.push_back(std::move(node));
+  return out;
+}
+
+void NetworkEmulationManagerImpl::EnableEndpoint(EmulatedEndpoint* endpoint) {
+  EmulatedNetworkManager* network_manager =
+      endpoint_to_network_manager_[endpoint];
+  RTC_CHECK(network_manager);
+  network_manager->EnableEndpoint(static_cast<EmulatedEndpointImpl*>(endpoint));
+}
+
+void NetworkEmulationManagerImpl::DisableEndpoint(EmulatedEndpoint* endpoint) {
+  EmulatedNetworkManager* network_manager =
+      endpoint_to_network_manager_[endpoint];
+  RTC_CHECK(network_manager);
+  network_manager->DisableEndpoint(
+      static_cast<EmulatedEndpointImpl*>(endpoint));
+}
+
+EmulatedRoute* NetworkEmulationManagerImpl::CreateRoute(
+    EmulatedEndpoint* from,
+    const std::vector<EmulatedNetworkNode*>& via_nodes,
+    EmulatedEndpoint* to) {
+  // Because endpoint has no send node by default at least one should be
+  // provided here.
+  RTC_CHECK(!via_nodes.empty());
+
+  static_cast<EmulatedEndpointImpl*>(from)->router()->SetReceiver(
+      to->GetPeerLocalAddress(), via_nodes[0]);
+  EmulatedNetworkNode* cur_node = via_nodes[0];
+  for (size_t i = 1; i < via_nodes.size(); ++i) {
+    cur_node->router()->SetReceiver(to->GetPeerLocalAddress(), via_nodes[i]);
+    cur_node = via_nodes[i];
+  }
+  cur_node->router()->SetReceiver(to->GetPeerLocalAddress(), to);
+
+  std::unique_ptr<EmulatedRoute> route = std::make_unique<EmulatedRoute>(
+      static_cast<EmulatedEndpointImpl*>(from), std::move(via_nodes),
+      static_cast<EmulatedEndpointImpl*>(to), /*is_default=*/false);
+  EmulatedRoute* out = route.get();
+  routes_.push_back(std::move(route));
+  return out;
+}
+
+EmulatedRoute* NetworkEmulationManagerImpl::CreateRoute(
+    const std::vector<EmulatedNetworkNode*>& via_nodes) {
+  EmulatedEndpoint* from = CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* to = CreateEndpoint(EmulatedEndpointConfig());
+  return CreateRoute(from, via_nodes, to);
+}
+
+EmulatedRoute* NetworkEmulationManagerImpl::CreateDefaultRoute(
+    EmulatedEndpoint* from,
+    const std::vector<EmulatedNetworkNode*>& via_nodes,
+    EmulatedEndpoint* to) {
+  // Because endpoint has no send node by default at least one should be
+  // provided here.
+  RTC_CHECK(!via_nodes.empty());
+
+  static_cast<EmulatedEndpointImpl*>(from)->router()->SetDefaultReceiver(
+      via_nodes[0]);
+  EmulatedNetworkNode* cur_node = via_nodes[0];
+  for (size_t i = 1; i < via_nodes.size(); ++i) {
+    cur_node->router()->SetDefaultReceiver(via_nodes[i]);
+    cur_node = via_nodes[i];
+  }
+  cur_node->router()->SetDefaultReceiver(to);
+
+  std::unique_ptr<EmulatedRoute> route = std::make_unique<EmulatedRoute>(
+      static_cast<EmulatedEndpointImpl*>(from), std::move(via_nodes),
+      static_cast<EmulatedEndpointImpl*>(to), /*is_default=*/true);
+  EmulatedRoute* out = route.get();
+  routes_.push_back(std::move(route));
+  return out;
+}
+
+void NetworkEmulationManagerImpl::ClearRoute(EmulatedRoute* route) {
+  RTC_CHECK(route->active) << "Route already cleared";
+  task_queue_.SendTask([route]() {
+    // Remove receiver from intermediate nodes.
+    for (auto* node : route->via_nodes) {
+      if (route->is_default) {
+        node->router()->RemoveDefaultReceiver();
+      } else {
+        node->router()->RemoveReceiver(route->to->GetPeerLocalAddress());
+      }
+    }
+    // Remove destination endpoint from source endpoint's router.
+    if (route->is_default) {
+      route->from->router()->RemoveDefaultReceiver();
+    } else {
+      route->from->router()->RemoveReceiver(route->to->GetPeerLocalAddress());
+    }
+
+    route->active = false;
+  });
+}
+
+TcpMessageRoute* NetworkEmulationManagerImpl::CreateTcpRoute(
+    EmulatedRoute* send_route,
+    EmulatedRoute* ret_route) {
+  auto tcp_route = std::make_unique<TcpMessageRouteImpl>(
+      clock_, task_queue_.Get(), send_route, ret_route);
+  auto* route_ptr = tcp_route.get();
+  task_queue_.PostTask([this, tcp_route = std::move(tcp_route)]() mutable {
+    tcp_message_routes_.push_back(std::move(tcp_route));
+  });
+  return route_ptr;
+}
+
+CrossTrafficRoute* NetworkEmulationManagerImpl::CreateCrossTrafficRoute(
+    const std::vector<EmulatedNetworkNode*>& via_nodes) {
+  RTC_CHECK(!via_nodes.empty());
+  EmulatedEndpointImpl* endpoint = CreateEndpoint(EmulatedEndpointConfig());
+
+  // Setup a route via specified nodes.
+  EmulatedNetworkNode* cur_node = via_nodes[0];
+  for (size_t i = 1; i < via_nodes.size(); ++i) {
+    cur_node->router()->SetReceiver(endpoint->GetPeerLocalAddress(),
+                                    via_nodes[i]);
+    cur_node = via_nodes[i];
+  }
+  cur_node->router()->SetReceiver(endpoint->GetPeerLocalAddress(), endpoint);
+
+  std::unique_ptr<CrossTrafficRoute> traffic_route =
+      std::make_unique<CrossTrafficRouteImpl>(clock_, via_nodes[0], endpoint);
+  CrossTrafficRoute* out = traffic_route.get();
+  traffic_routes_.push_back(std::move(traffic_route));
+  return out;
+}
+
+CrossTrafficGenerator* NetworkEmulationManagerImpl::StartCrossTraffic(
+    std::unique_ptr<CrossTrafficGenerator> generator) {
+  CrossTrafficGenerator* out = generator.get();
+  task_queue_.PostTask([this, generator = std::move(generator)]() mutable {
+    auto* generator_ptr = generator.get();
+
+    auto repeating_task_handle =
+        RepeatingTaskHandle::Start(task_queue_.Get(), [this, generator_ptr] {
+          generator_ptr->Process(Now());
+          return generator_ptr->GetProcessInterval();
+        });
+
+    cross_traffics_.push_back(CrossTrafficSource(
+        std::move(generator), std::move(repeating_task_handle)));
+  });
+  return out;
+}
+
+void NetworkEmulationManagerImpl::StopCrossTraffic(
+    CrossTrafficGenerator* generator) {
+  task_queue_.PostTask([=]() {
+    auto it = std::find_if(cross_traffics_.begin(), cross_traffics_.end(),
+                           [=](const CrossTrafficSource& el) {
+                             return el.first.get() == generator;
+                           });
+    it->second.Stop();
+    cross_traffics_.erase(it);
+  });
+}
+
+EmulatedNetworkManagerInterface*
+NetworkEmulationManagerImpl::CreateEmulatedNetworkManagerInterface(
+    const std::vector<EmulatedEndpoint*>& endpoints) {
+  std::vector<EmulatedEndpointImpl*> endpoint_impls;
+  endpoint_impls.reserve(endpoints.size());
+  for (EmulatedEndpoint* endpoint : endpoints) {
+    endpoint_impls.push_back(static_cast<EmulatedEndpointImpl*>(endpoint));
+  }
+  auto endpoints_container = std::make_unique<EndpointsContainer>(
+      endpoint_impls, stats_gathering_mode_);
+  auto network_manager = std::make_unique<EmulatedNetworkManager>(
+      time_controller_.get(), &task_queue_, endpoints_container.get());
+  for (auto* endpoint : endpoints) {
+    // Associate endpoint with network manager.
+    bool insertion_result =
+        endpoint_to_network_manager_.insert({endpoint, network_manager.get()})
+            .second;
+    RTC_CHECK(insertion_result)
+        << "Endpoint ip=" << endpoint->GetPeerLocalAddress().ToString()
+        << " is already used for another network";
+  }
+
+  EmulatedNetworkManagerInterface* out = network_manager.get();
+
+  endpoints_containers_.push_back(std::move(endpoints_container));
+  network_managers_.push_back(std::move(network_manager));
+  return out;
+}
+
+void NetworkEmulationManagerImpl::GetStats(
+    rtc::ArrayView<EmulatedEndpoint* const> endpoints,
+    std::function<void(EmulatedNetworkStats)> stats_callback) {
+  task_queue_.PostTask([endpoints, stats_callback,
+                        stats_gathering_mode = stats_gathering_mode_]() {
+    EmulatedNetworkStatsBuilder stats_builder(stats_gathering_mode);
+    for (auto* endpoint : endpoints) {
+      // It's safe to cast here because EmulatedEndpointImpl can be the only
+      // implementation of EmulatedEndpoint, because only it has access to
+      // EmulatedEndpoint constructor.
+      auto endpoint_impl = static_cast<EmulatedEndpointImpl*>(endpoint);
+      stats_builder.AddEmulatedNetworkStats(endpoint_impl->stats());
+    }
+    stats_callback(stats_builder.Build());
+  });
+}
+
+void NetworkEmulationManagerImpl::GetStats(
+    rtc::ArrayView<EmulatedNetworkNode* const> nodes,
+    std::function<void(EmulatedNetworkNodeStats)> stats_callback) {
+  task_queue_.PostTask(
+      [nodes, stats_callback, stats_gathering_mode = stats_gathering_mode_]() {
+        EmulatedNetworkNodeStatsBuilder stats_builder(stats_gathering_mode);
+        for (auto* node : nodes) {
+          stats_builder.AddEmulatedNetworkNodeStats(node->stats());
+        }
+        stats_callback(stats_builder.Build());
+      });
+}
+
+absl::optional<rtc::IPAddress>
+NetworkEmulationManagerImpl::GetNextIPv4Address() {
+  uint32_t addresses_count = kMaxIPv4Address - kMinIPv4Address;
+  for (uint32_t i = 0; i < addresses_count; i++) {
+    rtc::IPAddress ip(next_ip4_address_);
+    if (next_ip4_address_ == kMaxIPv4Address) {
+      next_ip4_address_ = kMinIPv4Address;
+    } else {
+      next_ip4_address_++;
+    }
+    if (used_ip_addresses_.find(ip) == used_ip_addresses_.end()) {
+      return ip;
+    }
+  }
+  return absl::nullopt;
+}
+
+Timestamp NetworkEmulationManagerImpl::Now() const {
+  return clock_->CurrentTime();
+}
+
+EmulatedTURNServerInterface* NetworkEmulationManagerImpl::CreateTURNServer(
+    EmulatedTURNServerConfig config) {
+  auto* client = CreateEndpoint(config.client_config);
+  auto* peer = CreateEndpoint(config.client_config);
+  char buf[128];
+  rtc::SimpleStringBuilder str(buf);
+  str.AppendFormat("turn_server_%u",
+                   static_cast<unsigned>(turn_servers_.size()));
+  auto turn = std::make_unique<EmulatedTURNServer>(
+      time_controller_->CreateThread(str.str()), client, peer);
+  auto out = turn.get();
+  turn_servers_.push_back(std::move(turn));
+  return out;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/network_emulation_manager.h b/third_party/libwebrtc/test/network/network_emulation_manager.h
new file mode 100644
index 0000000000..29debca693
--- /dev/null
+++ b/third_party/libwebrtc/test/network/network_emulation_manager.h
@@ -0,0 +1,138 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_NETWORK_NETWORK_EMULATION_MANAGER_H_
+#define TEST_NETWORK_NETWORK_EMULATION_MANAGER_H_
+
+#include <map>
+#include <memory>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/simulated_network.h"
+#include "api/test/time_controller.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "system_wrappers/include/clock.h"
+#include "test/network/cross_traffic.h"
+#include "test/network/emulated_network_manager.h"
+#include "test/network/emulated_turn_server.h"
+#include "test/network/network_emulation.h"
+
+namespace webrtc {
+namespace test {
+
+class NetworkEmulationManagerImpl : public NetworkEmulationManager {
+ public:
+  NetworkEmulationManagerImpl(
+      TimeMode mode,
+      EmulatedNetworkStatsGatheringMode stats_gathering_mode);
+  ~NetworkEmulationManagerImpl();
+
+  EmulatedNetworkNode* CreateEmulatedNode(BuiltInNetworkBehaviorConfig config,
+                                          uint64_t random_seed = 1) override;
+  EmulatedNetworkNode* CreateEmulatedNode(
+      std::unique_ptr<NetworkBehaviorInterface> network_behavior) override;
+
+  SimulatedNetworkNode::Builder NodeBuilder() override;
+
+  EmulatedEndpointImpl* CreateEndpoint(EmulatedEndpointConfig config) override;
+  void EnableEndpoint(EmulatedEndpoint* endpoint) override;
+  void DisableEndpoint(EmulatedEndpoint* endpoint) override;
+
+  EmulatedRoute* CreateRoute(EmulatedEndpoint* from,
+                             const std::vector<EmulatedNetworkNode*>& via_nodes,
+                             EmulatedEndpoint* to) override;
+
+  EmulatedRoute* CreateRoute(
+      const std::vector<EmulatedNetworkNode*>& via_nodes) override;
+
+  EmulatedRoute* CreateDefaultRoute(
+      EmulatedEndpoint* from,
+      const std::vector<EmulatedNetworkNode*>& via_nodes,
+      EmulatedEndpoint* to) override;
+
+  void ClearRoute(EmulatedRoute* route) override;
+
+  TcpMessageRoute* CreateTcpRoute(EmulatedRoute* send_route,
+                                  EmulatedRoute* ret_route) override;
+
+  CrossTrafficRoute* CreateCrossTrafficRoute(
+      const std::vector<EmulatedNetworkNode*>& via_nodes) override;
+
+  CrossTrafficGenerator* StartCrossTraffic(
+      std::unique_ptr<CrossTrafficGenerator> generator) override;
+  void StopCrossTraffic(CrossTrafficGenerator* generator) override;
+
+  EmulatedNetworkManagerInterface* CreateEmulatedNetworkManagerInterface(
+      const std::vector<EmulatedEndpoint*>& endpoints) override;
+
+  void GetStats(
+      rtc::ArrayView<EmulatedEndpoint* const> endpoints,
+      std::function<void(EmulatedNetworkStats)> stats_callback) override;
+
+  void GetStats(
+      rtc::ArrayView<EmulatedNetworkNode* const> nodes,
+      std::function<void(EmulatedNetworkNodeStats)> stats_callback) override;
+
+  TimeController* time_controller() override { return time_controller_.get(); }
+
+  TimeMode time_mode() const override { return time_mode_; }
+
+  Timestamp Now() const;
+
+  EmulatedTURNServerInterface* CreateTURNServer(
+      EmulatedTURNServerConfig config) override;
+
+ private:
+  using CrossTrafficSource =
+      std::pair<std::unique_ptr<CrossTrafficGenerator>, RepeatingTaskHandle>;
+
+  absl::optional<rtc::IPAddress> GetNextIPv4Address();
+
+  const TimeMode time_mode_;
+  const EmulatedNetworkStatsGatheringMode stats_gathering_mode_;
+  const std::unique_ptr<TimeController> time_controller_;
+  Clock* const clock_;
+  int next_node_id_;
+
+  RepeatingTaskHandle process_task_handle_;
+
+  uint32_t next_ip4_address_;
+  std::set<rtc::IPAddress> used_ip_addresses_;
+
+  // All objects can be added to the manager only when it is idle.
+  std::vector<std::unique_ptr<EmulatedEndpoint>> endpoints_;
+  std::vector<std::unique_ptr<EmulatedNetworkNode>> network_nodes_;
+  std::vector<std::unique_ptr<EmulatedRoute>> routes_;
+  std::vector<std::unique_ptr<CrossTrafficRoute>> traffic_routes_;
+  std::vector<CrossTrafficSource> cross_traffics_;
+  std::list<std::unique_ptr<TcpMessageRouteImpl>> tcp_message_routes_;
+  std::vector<std::unique_ptr<EndpointsContainer>> endpoints_containers_;
+  std::vector<std::unique_ptr<EmulatedNetworkManager>> network_managers_;
+  std::vector<std::unique_ptr<EmulatedTURNServer>> turn_servers_;
+
+  std::map<EmulatedEndpoint*, EmulatedNetworkManager*>
+      endpoint_to_network_manager_;
+
+  // Must be the last field, so it will be deleted first, because tasks
+  // in the TaskQueue can access other fields of the instance of this class.
+  TaskQueueForTest task_queue_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_NETWORK_EMULATION_MANAGER_H_
diff --git a/third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc b/third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc
new file mode 100644
index 0000000000..51a45a8234
--- /dev/null
+++ b/third_party/libwebrtc/test/network/network_emulation_pc_unittest.cc
@@ -0,0 +1,319 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdint>
+#include <memory>
+
+#include "api/call/call_factory_interface.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/scoped_refptr.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/field_trial_based_config.h"
+#include "call/simulated_network.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "media/engine/webrtc_media_engine_defaults.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "p2p/base/basic_packet_socket_factory.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "pc/peer_connection_wrapper.h"
+#include "pc/test/mock_peer_connection_observers.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/network/network_emulation.h"
+#include "test/network/network_emulation_manager.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kDefaultTimeoutMs = 1000;
+constexpr int kMaxAptitude = 32000;
+constexpr int kSamplingFrequency = 48000;
+constexpr char kSignalThreadName[] = "signaling_thread";
+
+bool AddIceCandidates(PeerConnectionWrapper* peer,
+                      std::vector<const IceCandidateInterface*> candidates) {
+  bool success = true;
+  for (const auto candidate : candidates) {
+    if (!peer->pc()->AddIceCandidate(candidate)) {
+      success = false;
+    }
+  }
+  return success;
+}
+
+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
+    rtc::Thread* signaling_thread,
+    rtc::Thread* network_thread) {
+  PeerConnectionFactoryDependencies pcf_deps;
+  pcf_deps.task_queue_factory = CreateDefaultTaskQueueFactory();
+  pcf_deps.call_factory = CreateCallFactory();
+  pcf_deps.event_log_factory =
+      std::make_unique<RtcEventLogFactory>(pcf_deps.task_queue_factory.get());
+  pcf_deps.network_thread = network_thread;
+  pcf_deps.signaling_thread = signaling_thread;
+  pcf_deps.trials = std::make_unique<FieldTrialBasedConfig>();
+  cricket::MediaEngineDependencies media_deps;
+  media_deps.task_queue_factory = pcf_deps.task_queue_factory.get();
+  media_deps.adm = TestAudioDeviceModule::Create(
+      media_deps.task_queue_factory,
+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(kMaxAptitude,
+                                                       kSamplingFrequency),
+      TestAudioDeviceModule::CreateDiscardRenderer(kSamplingFrequency),
+      /*speed=*/1.f);
+  media_deps.trials = pcf_deps.trials.get();
+  SetMediaEngineDefaults(&media_deps);
+  pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+  return CreateModularPeerConnectionFactory(std::move(pcf_deps));
+}
+
+rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
+    const rtc::scoped_refptr<PeerConnectionFactoryInterface>& pcf,
+    PeerConnectionObserver* observer,
+    rtc::PacketSocketFactory* packet_socket_factory,
+    rtc::NetworkManager* network_manager,
+    EmulatedTURNServerInterface* turn_server = nullptr) {
+  PeerConnectionDependencies pc_deps(observer);
+  auto port_allocator = std::make_unique<cricket::BasicPortAllocator>(
+      network_manager, packet_socket_factory);
+
+  // This test does not support TCP
+  int flags = cricket::PORTALLOCATOR_DISABLE_TCP;
+  port_allocator->set_flags(port_allocator->flags() | flags);
+
+  pc_deps.allocator = std::move(port_allocator);
+  PeerConnectionInterface::RTCConfiguration rtc_configuration;
+  rtc_configuration.sdp_semantics = SdpSemantics::kUnifiedPlan;
+  if (turn_server != nullptr) {
+    webrtc::PeerConnectionInterface::IceServer server;
+    server.username = turn_server->GetIceServerConfig().username;
+    server.password = turn_server->GetIceServerConfig().username;
+    server.urls.push_back(turn_server->GetIceServerConfig().url);
+    rtc_configuration.servers.push_back(server);
+  }
+
+  auto result =
+      pcf->CreatePeerConnectionOrError(rtc_configuration, std::move(pc_deps));
+  if (!result.ok()) {
+    return nullptr;
+  }
+  return result.MoveValue();
+}
+
+}  // namespace
+
+TEST(NetworkEmulationManagerPCTest, Run) {
+  std::unique_ptr<rtc::Thread> signaling_thread = rtc::Thread::Create();
+  signaling_thread->SetName(kSignalThreadName, nullptr);
+  signaling_thread->Start();
+
+  // Setup emulated network
+  NetworkEmulationManagerImpl emulation(
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault);
+
+  EmulatedNetworkNode* alice_node = emulation.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedNetworkNode* bob_node = emulation.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedEndpoint* alice_endpoint =
+      emulation.CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      emulation.CreateEndpoint(EmulatedEndpointConfig());
+  emulation.CreateRoute(alice_endpoint, {alice_node}, bob_endpoint);
+  emulation.CreateRoute(bob_endpoint, {bob_node}, alice_endpoint);
+
+  EmulatedNetworkManagerInterface* alice_network =
+      emulation.CreateEmulatedNetworkManagerInterface({alice_endpoint});
+  EmulatedNetworkManagerInterface* bob_network =
+      emulation.CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  // Setup peer connections.
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> alice_pcf;
+  rtc::scoped_refptr<PeerConnectionInterface> alice_pc;
+  std::unique_ptr<MockPeerConnectionObserver> alice_observer =
+      std::make_unique<MockPeerConnectionObserver>();
+
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> bob_pcf;
+  rtc::scoped_refptr<PeerConnectionInterface> bob_pc;
+  std::unique_ptr<MockPeerConnectionObserver> bob_observer =
+      std::make_unique<MockPeerConnectionObserver>();
+
+  SendTask(signaling_thread.get(), [&]() {
+    alice_pcf = CreatePeerConnectionFactory(signaling_thread.get(),
+                                            alice_network->network_thread());
+    alice_pc = CreatePeerConnection(alice_pcf, alice_observer.get(),
+                                    alice_network->packet_socket_factory(),
+                                    alice_network->network_manager());
+
+    bob_pcf = CreatePeerConnectionFactory(signaling_thread.get(),
+                                          bob_network->network_thread());
+    bob_pc = CreatePeerConnection(bob_pcf, bob_observer.get(),
+                                  bob_network->packet_socket_factory(),
+                                  bob_network->network_manager());
+  });
+
+  std::unique_ptr<PeerConnectionWrapper> alice =
+      std::make_unique<PeerConnectionWrapper>(alice_pcf, alice_pc,
+                                              std::move(alice_observer));
+  std::unique_ptr<PeerConnectionWrapper> bob =
+      std::make_unique<PeerConnectionWrapper>(bob_pcf, bob_pc,
+                                              std::move(bob_observer));
+
+  SendTask(signaling_thread.get(), [&]() {
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+        alice_pcf->CreateAudioSource(cricket::AudioOptions());
+    rtc::scoped_refptr<AudioTrackInterface> track =
+        alice_pcf->CreateAudioTrack("audio", source.get());
+    alice->AddTransceiver(track);
+
+    // Connect peers.
+    ASSERT_TRUE(alice->ExchangeOfferAnswerWith(bob.get()));
+    // Do the SDP negotiation, and also exchange ice candidates.
+    ASSERT_TRUE_WAIT(
+        alice->signaling_state() == PeerConnectionInterface::kStable,
+        kDefaultTimeoutMs);
+    ASSERT_TRUE_WAIT(alice->IsIceGatheringDone(), kDefaultTimeoutMs);
+    ASSERT_TRUE_WAIT(bob->IsIceGatheringDone(), kDefaultTimeoutMs);
+
+    // Connect an ICE candidate pairs.
+    ASSERT_TRUE(
+        AddIceCandidates(bob.get(), alice->observer()->GetAllCandidates()));
+    ASSERT_TRUE(
+        AddIceCandidates(alice.get(), bob->observer()->GetAllCandidates()));
+    // This means that ICE and DTLS are connected.
+    ASSERT_TRUE_WAIT(bob->IsIceConnected(), kDefaultTimeoutMs);
+    ASSERT_TRUE_WAIT(alice->IsIceConnected(), kDefaultTimeoutMs);
+
+    // Close peer connections
+    alice->pc()->Close();
+    bob->pc()->Close();
+
+    // Delete peers.
+    alice.reset();
+    bob.reset();
+  });
+}
+
+TEST(NetworkEmulationManagerPCTest, RunTURN) {
+  std::unique_ptr<rtc::Thread> signaling_thread = rtc::Thread::Create();
+  signaling_thread->SetName(kSignalThreadName, nullptr);
+  signaling_thread->Start();
+
+  // Setup emulated network
+  NetworkEmulationManagerImpl emulation(
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault);
+
+  EmulatedNetworkNode* alice_node = emulation.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedNetworkNode* bob_node = emulation.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedNetworkNode* turn_node = emulation.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedEndpoint* alice_endpoint =
+      emulation.CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      emulation.CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedTURNServerInterface* alice_turn =
+      emulation.CreateTURNServer(EmulatedTURNServerConfig());
+  EmulatedTURNServerInterface* bob_turn =
+      emulation.CreateTURNServer(EmulatedTURNServerConfig());
+
+  emulation.CreateRoute(alice_endpoint, {alice_node},
+                        alice_turn->GetClientEndpoint());
+  emulation.CreateRoute(alice_turn->GetClientEndpoint(), {alice_node},
+                        alice_endpoint);
+
+  emulation.CreateRoute(bob_endpoint, {bob_node},
+                        bob_turn->GetClientEndpoint());
+  emulation.CreateRoute(bob_turn->GetClientEndpoint(), {bob_node},
+                        bob_endpoint);
+
+  emulation.CreateRoute(alice_turn->GetPeerEndpoint(), {turn_node},
+                        bob_turn->GetPeerEndpoint());
+  emulation.CreateRoute(bob_turn->GetPeerEndpoint(), {turn_node},
+                        alice_turn->GetPeerEndpoint());
+
+  EmulatedNetworkManagerInterface* alice_network =
+      emulation.CreateEmulatedNetworkManagerInterface({alice_endpoint});
+  EmulatedNetworkManagerInterface* bob_network =
+      emulation.CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  // Setup peer connections.
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> alice_pcf;
+  rtc::scoped_refptr<PeerConnectionInterface> alice_pc;
+  std::unique_ptr<MockPeerConnectionObserver> alice_observer =
+      std::make_unique<MockPeerConnectionObserver>();
+
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> bob_pcf;
+  rtc::scoped_refptr<PeerConnectionInterface> bob_pc;
+  std::unique_ptr<MockPeerConnectionObserver> bob_observer =
+      std::make_unique<MockPeerConnectionObserver>();
+
+  SendTask(signaling_thread.get(), [&]() {
+    alice_pcf = CreatePeerConnectionFactory(signaling_thread.get(),
+                                            alice_network->network_thread());
+    alice_pc = CreatePeerConnection(
+        alice_pcf, alice_observer.get(), alice_network->packet_socket_factory(),
+        alice_network->network_manager(), alice_turn);
+
+    bob_pcf = CreatePeerConnectionFactory(signaling_thread.get(),
+                                          bob_network->network_thread());
+    bob_pc = CreatePeerConnection(bob_pcf, bob_observer.get(),
+                                  bob_network->packet_socket_factory(),
+                                  bob_network->network_manager(), bob_turn);
+  });
+
+  std::unique_ptr<PeerConnectionWrapper> alice =
+      std::make_unique<PeerConnectionWrapper>(alice_pcf, alice_pc,
+                                              std::move(alice_observer));
+  std::unique_ptr<PeerConnectionWrapper> bob =
+      std::make_unique<PeerConnectionWrapper>(bob_pcf, bob_pc,
+                                              std::move(bob_observer));
+
+  SendTask(signaling_thread.get(), [&]() {
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+        alice_pcf->CreateAudioSource(cricket::AudioOptions());
+    rtc::scoped_refptr<AudioTrackInterface> track =
+        alice_pcf->CreateAudioTrack("audio", source.get());
+    alice->AddTransceiver(track);
+
+    // Connect peers.
+    ASSERT_TRUE(alice->ExchangeOfferAnswerWith(bob.get()));
+    // Do the SDP negotiation, and also exchange ice candidates.
+    ASSERT_TRUE_WAIT(
+        alice->signaling_state() == PeerConnectionInterface::kStable,
+        kDefaultTimeoutMs);
+    ASSERT_TRUE_WAIT(alice->IsIceGatheringDone(), kDefaultTimeoutMs);
+    ASSERT_TRUE_WAIT(bob->IsIceGatheringDone(), kDefaultTimeoutMs);
+
+    // Connect an ICE candidate pairs.
+    ASSERT_TRUE(
+        AddIceCandidates(bob.get(), alice->observer()->GetAllCandidates()));
+    ASSERT_TRUE(
+        AddIceCandidates(alice.get(), bob->observer()->GetAllCandidates()));
+    // This means that ICE and DTLS are connected.
+    ASSERT_TRUE_WAIT(bob->IsIceConnected(), kDefaultTimeoutMs);
+    ASSERT_TRUE_WAIT(alice->IsIceConnected(), kDefaultTimeoutMs);
+
+    // Close peer connections
+    alice->pc()->Close();
+    bob->pc()->Close();
+
+    // Delete peers.
+    alice.reset();
+    bob.reset();
+  });
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/network_emulation_unittest.cc b/third_party/libwebrtc/test/network/network_emulation_unittest.cc
new file mode 100644
index 0000000000..2e67a5a00a
--- /dev/null
+++ b/third_party/libwebrtc/test/network/network_emulation_unittest.cc
@@ -0,0 +1,676 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/network_emulation.h"
+
+#include <atomic>
+#include <memory>
+#include <set>
+
+#include "api/test/simulated_network.h"
+#include "api/units/time_delta.h"
+#include "call/simulated_network.h"
+#include "rtc_base/event.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/network/network_emulation_manager.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+using ::testing::ElementsAreArray;
+
+constexpr TimeDelta kNetworkPacketWaitTimeout = TimeDelta::Millis(100);
+constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1);
+constexpr int kOverheadIpv4Udp = 20 + 8;
+
+class SocketReader : public sigslot::has_slots<> {
+ public:
+  explicit SocketReader(rtc::Socket* socket, rtc::Thread* network_thread)
+      : socket_(socket), network_thread_(network_thread) {
+    socket_->SignalReadEvent.connect(this, &SocketReader::OnReadEvent);
+    size_ = 128 * 1024;
+    buf_ = new char[size_];
+  }
+  ~SocketReader() override { delete[] buf_; }
+
+  void OnReadEvent(rtc::Socket* socket) {
+    RTC_DCHECK(socket_ == socket);
+    RTC_DCHECK(network_thread_->IsCurrent());
+    int64_t timestamp;
+    len_ = socket_->Recv(buf_, size_, &timestamp);
+
+    MutexLock lock(&lock_);
+    received_count_++;
+  }
+
+  int ReceivedCount() {
+    MutexLock lock(&lock_);
+    return received_count_;
+  }
+
+ private:
+  rtc::Socket* const socket_;
+  rtc::Thread* const network_thread_;
+  char* buf_;
+  size_t size_;
+  int len_;
+
+  Mutex lock_;
+  int received_count_ RTC_GUARDED_BY(lock_) = 0;
+};
+
+class MockReceiver : public EmulatedNetworkReceiverInterface {
+ public:
+  MOCK_METHOD(void, OnPacketReceived, (EmulatedIpPacket packet), (override));
+};
+
+class NetworkEmulationManagerThreeNodesRoutingTest : public ::testing::Test {
+ public:
+  NetworkEmulationManagerThreeNodesRoutingTest() {
+    e1_ = emulation_.CreateEndpoint(EmulatedEndpointConfig());
+    e2_ = emulation_.CreateEndpoint(EmulatedEndpointConfig());
+    e3_ = emulation_.CreateEndpoint(EmulatedEndpointConfig());
+  }
+
+  void SetupRouting(
+      std::function<void(EmulatedEndpoint*,
+                         EmulatedEndpoint*,
+                         EmulatedEndpoint*,
+                         NetworkEmulationManager*)> create_routing_func) {
+    create_routing_func(e1_, e2_, e3_, &emulation_);
+  }
+
+  void SendPacketsAndValidateDelivery() {
+    EXPECT_CALL(r_e1_e2_, OnPacketReceived(::testing::_)).Times(1);
+    EXPECT_CALL(r_e2_e1_, OnPacketReceived(::testing::_)).Times(1);
+    EXPECT_CALL(r_e1_e3_, OnPacketReceived(::testing::_)).Times(1);
+    EXPECT_CALL(r_e3_e1_, OnPacketReceived(::testing::_)).Times(1);
+
+    uint16_t common_send_port = 80;
+    uint16_t r_e1_e2_port = e2_->BindReceiver(0, &r_e1_e2_).value();
+    uint16_t r_e2_e1_port = e1_->BindReceiver(0, &r_e2_e1_).value();
+    uint16_t r_e1_e3_port = e3_->BindReceiver(0, &r_e1_e3_).value();
+    uint16_t r_e3_e1_port = e1_->BindReceiver(0, &r_e3_e1_).value();
+
+    // Next code is using API of EmulatedEndpoint, that is visible only for
+    // internals of network emulation layer. Don't use this API in other tests.
+    // Send packet from e1 to e2.
+    e1_->SendPacket(
+        rtc::SocketAddress(e1_->GetPeerLocalAddress(), common_send_port),
+        rtc::SocketAddress(e2_->GetPeerLocalAddress(), r_e1_e2_port),
+        rtc::CopyOnWriteBuffer(10));
+
+    // Send packet from e2 to e1.
+    e2_->SendPacket(
+        rtc::SocketAddress(e2_->GetPeerLocalAddress(), common_send_port),
+        rtc::SocketAddress(e1_->GetPeerLocalAddress(), r_e2_e1_port),
+        rtc::CopyOnWriteBuffer(10));
+
+    // Send packet from e1 to e3.
+    e1_->SendPacket(
+        rtc::SocketAddress(e1_->GetPeerLocalAddress(), common_send_port),
+        rtc::SocketAddress(e3_->GetPeerLocalAddress(), r_e1_e3_port),
+        rtc::CopyOnWriteBuffer(10));
+
+    // Send packet from e3 to e1.
+    e3_->SendPacket(
+        rtc::SocketAddress(e3_->GetPeerLocalAddress(), common_send_port),
+        rtc::SocketAddress(e1_->GetPeerLocalAddress(), r_e3_e1_port),
+        rtc::CopyOnWriteBuffer(10));
+
+    // Sleep at the end to wait for async packets delivery.
+    emulation_.time_controller()->AdvanceTime(kNetworkPacketWaitTimeout);
+  }
+
+ private:
+  // Receivers: r_<source endpoint>_<destination endpoint>
+  // They must be destroyed after emulation, so they should be declared before.
+  MockReceiver r_e1_e2_;
+  MockReceiver r_e2_e1_;
+  MockReceiver r_e1_e3_;
+  MockReceiver r_e3_e1_;
+
+  NetworkEmulationManagerImpl emulation_{
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault};
+  EmulatedEndpoint* e1_;
+  EmulatedEndpoint* e2_;
+  EmulatedEndpoint* e3_;
+};
+
+EmulatedNetworkNode* CreateEmulatedNodeWithDefaultBuiltInConfig(
+    NetworkEmulationManager* emulation) {
+  return emulation->CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+}
+
+}  // namespace
+
+using ::testing::_;
+
+TEST(NetworkEmulationManagerTest, GeneratedIpv4AddressDoesNotCollide) {
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault);
+  std::set<rtc::IPAddress> ips;
+  EmulatedEndpointConfig config;
+  config.generated_ip_family = EmulatedEndpointConfig::IpAddressFamily::kIpv4;
+  for (int i = 0; i < 1000; i++) {
+    EmulatedEndpoint* endpoint = network_manager.CreateEndpoint(config);
+    ASSERT_EQ(endpoint->GetPeerLocalAddress().family(), AF_INET);
+    bool result = ips.insert(endpoint->GetPeerLocalAddress()).second;
+    ASSERT_TRUE(result);
+  }
+}
+
+TEST(NetworkEmulationManagerTest, GeneratedIpv6AddressDoesNotCollide) {
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault);
+  std::set<rtc::IPAddress> ips;
+  EmulatedEndpointConfig config;
+  config.generated_ip_family = EmulatedEndpointConfig::IpAddressFamily::kIpv6;
+  for (int i = 0; i < 1000; i++) {
+    EmulatedEndpoint* endpoint = network_manager.CreateEndpoint(config);
+    ASSERT_EQ(endpoint->GetPeerLocalAddress().family(), AF_INET6);
+    bool result = ips.insert(endpoint->GetPeerLocalAddress()).second;
+    ASSERT_TRUE(result);
+  }
+}
+
+TEST(NetworkEmulationManagerTest, Run) {
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault);
+
+  EmulatedNetworkNode* alice_node = network_manager.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedNetworkNode* bob_node = network_manager.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedEndpoint* alice_endpoint =
+      network_manager.CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_manager.CreateEndpoint(EmulatedEndpointConfig());
+  network_manager.CreateRoute(alice_endpoint, {alice_node}, bob_endpoint);
+  network_manager.CreateRoute(bob_endpoint, {bob_node}, alice_endpoint);
+
+  EmulatedNetworkManagerInterface* nt1 =
+      network_manager.CreateEmulatedNetworkManagerInterface({alice_endpoint});
+  EmulatedNetworkManagerInterface* nt2 =
+      network_manager.CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  rtc::Thread* t1 = nt1->network_thread();
+  rtc::Thread* t2 = nt2->network_thread();
+
+  rtc::CopyOnWriteBuffer data("Hello");
+  for (uint64_t j = 0; j < 2; j++) {
+    rtc::Socket* s1 = nullptr;
+    rtc::Socket* s2 = nullptr;
+    SendTask(t1, [&] {
+      s1 = t1->socketserver()->CreateSocket(AF_INET, SOCK_DGRAM);
+    });
+    SendTask(t2, [&] {
+      s2 = t2->socketserver()->CreateSocket(AF_INET, SOCK_DGRAM);
+    });
+
+    SocketReader r1(s1, t1);
+    SocketReader r2(s2, t2);
+
+    rtc::SocketAddress a1(alice_endpoint->GetPeerLocalAddress(), 0);
+    rtc::SocketAddress a2(bob_endpoint->GetPeerLocalAddress(), 0);
+
+    SendTask(t1, [&] {
+      s1->Bind(a1);
+      a1 = s1->GetLocalAddress();
+    });
+    SendTask(t2, [&] {
+      s2->Bind(a2);
+      a2 = s2->GetLocalAddress();
+    });
+
+    SendTask(t1, [&] { s1->Connect(a2); });
+    SendTask(t2, [&] { s2->Connect(a1); });
+
+    for (uint64_t i = 0; i < 1000; i++) {
+      t1->PostTask([&]() { s1->Send(data.data(), data.size()); });
+      t2->PostTask([&]() { s2->Send(data.data(), data.size()); });
+    }
+
+    network_manager.time_controller()->AdvanceTime(TimeDelta::Seconds(1));
+
+    EXPECT_EQ(r1.ReceivedCount(), 1000);
+    EXPECT_EQ(r2.ReceivedCount(), 1000);
+
+    SendTask(t1, [&] { delete s1; });
+    SendTask(t2, [&] { delete s2; });
+  }
+
+  const int64_t single_packet_size = data.size() + kOverheadIpv4Udp;
+  std::atomic<int> received_stats_count{0};
+  nt1->GetStats([&](EmulatedNetworkStats st) {
+    EXPECT_EQ(st.PacketsSent(), 2000l);
+    EXPECT_EQ(st.BytesSent().bytes(), single_packet_size * 2000l);
+    EXPECT_THAT(st.local_addresses,
+                ElementsAreArray({alice_endpoint->GetPeerLocalAddress()}));
+    EXPECT_EQ(st.PacketsReceived(), 2000l);
+    EXPECT_EQ(st.BytesReceived().bytes(), single_packet_size * 2000l);
+    EXPECT_EQ(st.PacketsDiscardedNoReceiver(), 0l);
+    EXPECT_EQ(st.BytesDiscardedNoReceiver().bytes(), 0l);
+
+    rtc::IPAddress bob_ip = bob_endpoint->GetPeerLocalAddress();
+    std::map<rtc::IPAddress, EmulatedNetworkIncomingStats> source_st =
+        st.incoming_stats_per_source;
+    ASSERT_EQ(source_st.size(), 1lu);
+    EXPECT_EQ(source_st.at(bob_ip).packets_received, 2000l);
+    EXPECT_EQ(source_st.at(bob_ip).bytes_received.bytes(),
+              single_packet_size * 2000l);
+    EXPECT_EQ(source_st.at(bob_ip).packets_discarded_no_receiver, 0l);
+    EXPECT_EQ(source_st.at(bob_ip).bytes_discarded_no_receiver.bytes(), 0l);
+
+    std::map<rtc::IPAddress, EmulatedNetworkOutgoingStats> dest_st =
+        st.outgoing_stats_per_destination;
+    ASSERT_EQ(dest_st.size(), 1lu);
+    EXPECT_EQ(dest_st.at(bob_ip).packets_sent, 2000l);
+    EXPECT_EQ(dest_st.at(bob_ip).bytes_sent.bytes(),
+              single_packet_size * 2000l);
+
+    // No debug stats are collected by default.
+    EXPECT_TRUE(st.SentPacketsSizeCounter().IsEmpty());
+    EXPECT_TRUE(st.sent_packets_queue_wait_time_us.IsEmpty());
+    EXPECT_TRUE(st.ReceivedPacketsSizeCounter().IsEmpty());
+    EXPECT_TRUE(st.PacketsDiscardedNoReceiverSizeCounter().IsEmpty());
+    EXPECT_TRUE(dest_st.at(bob_ip).sent_packets_size.IsEmpty());
+    EXPECT_TRUE(source_st.at(bob_ip).received_packets_size.IsEmpty());
+    EXPECT_TRUE(
+        source_st.at(bob_ip).packets_discarded_no_receiver_size.IsEmpty());
+
+    received_stats_count++;
+  });
+  nt2->GetStats([&](EmulatedNetworkStats st) {
+    EXPECT_EQ(st.PacketsSent(), 2000l);
+    EXPECT_EQ(st.BytesSent().bytes(), single_packet_size * 2000l);
+    EXPECT_THAT(st.local_addresses,
+                ElementsAreArray({bob_endpoint->GetPeerLocalAddress()}));
+    EXPECT_EQ(st.PacketsReceived(), 2000l);
+    EXPECT_EQ(st.BytesReceived().bytes(), single_packet_size * 2000l);
+    EXPECT_EQ(st.PacketsDiscardedNoReceiver(), 0l);
+    EXPECT_EQ(st.BytesDiscardedNoReceiver().bytes(), 0l);
+    EXPECT_GT(st.FirstReceivedPacketSize(), DataSize::Zero());
+    EXPECT_TRUE(st.FirstPacketReceivedTime().IsFinite());
+    EXPECT_TRUE(st.LastPacketReceivedTime().IsFinite());
+
+    rtc::IPAddress alice_ip = alice_endpoint->GetPeerLocalAddress();
+    std::map<rtc::IPAddress, EmulatedNetworkIncomingStats> source_st =
+        st.incoming_stats_per_source;
+    ASSERT_EQ(source_st.size(), 1lu);
+    EXPECT_EQ(source_st.at(alice_ip).packets_received, 2000l);
+    EXPECT_EQ(source_st.at(alice_ip).bytes_received.bytes(),
+              single_packet_size * 2000l);
+    EXPECT_EQ(source_st.at(alice_ip).packets_discarded_no_receiver, 0l);
+    EXPECT_EQ(source_st.at(alice_ip).bytes_discarded_no_receiver.bytes(), 0l);
+
+    std::map<rtc::IPAddress, EmulatedNetworkOutgoingStats> dest_st =
+        st.outgoing_stats_per_destination;
+    ASSERT_EQ(dest_st.size(), 1lu);
+    EXPECT_EQ(dest_st.at(alice_ip).packets_sent, 2000l);
+    EXPECT_EQ(dest_st.at(alice_ip).bytes_sent.bytes(),
+              single_packet_size * 2000l);
+
+    // No debug stats are collected by default.
+    EXPECT_TRUE(st.SentPacketsSizeCounter().IsEmpty());
+    EXPECT_TRUE(st.sent_packets_queue_wait_time_us.IsEmpty());
+    EXPECT_TRUE(st.ReceivedPacketsSizeCounter().IsEmpty());
+    EXPECT_TRUE(st.PacketsDiscardedNoReceiverSizeCounter().IsEmpty());
+    EXPECT_TRUE(dest_st.at(alice_ip).sent_packets_size.IsEmpty());
+    EXPECT_TRUE(source_st.at(alice_ip).received_packets_size.IsEmpty());
+    EXPECT_TRUE(
+        source_st.at(alice_ip).packets_discarded_no_receiver_size.IsEmpty());
+
+    received_stats_count++;
+  });
+  ASSERT_EQ_SIMULATED_WAIT(received_stats_count.load(), 2,
+                           kStatsWaitTimeout.ms(),
+                           *network_manager.time_controller());
+}
+
+TEST(NetworkEmulationManagerTest, DebugStatsCollectedInDebugMode) {
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kSimulated, EmulatedNetworkStatsGatheringMode::kDebug);
+
+  EmulatedNetworkNode* alice_node = network_manager.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedNetworkNode* bob_node = network_manager.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedEndpoint* alice_endpoint =
+      network_manager.CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_manager.CreateEndpoint(EmulatedEndpointConfig());
+  network_manager.CreateRoute(alice_endpoint, {alice_node}, bob_endpoint);
+  network_manager.CreateRoute(bob_endpoint, {bob_node}, alice_endpoint);
+
+  EmulatedNetworkManagerInterface* nt1 =
+      network_manager.CreateEmulatedNetworkManagerInterface({alice_endpoint});
+  EmulatedNetworkManagerInterface* nt2 =
+      network_manager.CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  rtc::Thread* t1 = nt1->network_thread();
+  rtc::Thread* t2 = nt2->network_thread();
+
+  rtc::CopyOnWriteBuffer data("Hello");
+  for (uint64_t j = 0; j < 2; j++) {
+    rtc::Socket* s1 = nullptr;
+    rtc::Socket* s2 = nullptr;
+    SendTask(t1, [&] {
+      s1 = t1->socketserver()->CreateSocket(AF_INET, SOCK_DGRAM);
+    });
+    SendTask(t2, [&] {
+      s2 = t2->socketserver()->CreateSocket(AF_INET, SOCK_DGRAM);
+    });
+
+    SocketReader r1(s1, t1);
+    SocketReader r2(s2, t2);
+
+    rtc::SocketAddress a1(alice_endpoint->GetPeerLocalAddress(), 0);
+    rtc::SocketAddress a2(bob_endpoint->GetPeerLocalAddress(), 0);
+
+    SendTask(t1, [&] {
+      s1->Bind(a1);
+      a1 = s1->GetLocalAddress();
+    });
+    SendTask(t2, [&] {
+      s2->Bind(a2);
+      a2 = s2->GetLocalAddress();
+    });
+
+    SendTask(t1, [&] { s1->Connect(a2); });
+    SendTask(t2, [&] { s2->Connect(a1); });
+
+    for (uint64_t i = 0; i < 1000; i++) {
+      t1->PostTask([&]() { s1->Send(data.data(), data.size()); });
+      t2->PostTask([&]() { s2->Send(data.data(), data.size()); });
+    }
+
+    network_manager.time_controller()->AdvanceTime(TimeDelta::Seconds(1));
+
+    EXPECT_EQ(r1.ReceivedCount(), 1000);
+    EXPECT_EQ(r2.ReceivedCount(), 1000);
+
+    SendTask(t1, [&] { delete s1; });
+    SendTask(t2, [&] { delete s2; });
+  }
+
+  const int64_t single_packet_size = data.size() + kOverheadIpv4Udp;
+  std::atomic<int> received_stats_count{0};
+  nt1->GetStats([&](EmulatedNetworkStats st) {
+    rtc::IPAddress bob_ip = bob_endpoint->GetPeerLocalAddress();
+    std::map<rtc::IPAddress, EmulatedNetworkIncomingStats> source_st =
+        st.incoming_stats_per_source;
+    ASSERT_EQ(source_st.size(), 1lu);
+
+    std::map<rtc::IPAddress, EmulatedNetworkOutgoingStats> dest_st =
+        st.outgoing_stats_per_destination;
+    ASSERT_EQ(dest_st.size(), 1lu);
+
+    // No debug stats are collected by default.
+    EXPECT_EQ(st.SentPacketsSizeCounter().NumSamples(), 2000l);
+    EXPECT_EQ(st.ReceivedPacketsSizeCounter().GetAverage(), single_packet_size);
+    EXPECT_EQ(st.sent_packets_queue_wait_time_us.NumSamples(), 2000l);
+    EXPECT_LT(st.sent_packets_queue_wait_time_us.GetMax(), 1);
+    EXPECT_TRUE(st.PacketsDiscardedNoReceiverSizeCounter().IsEmpty());
+    EXPECT_EQ(dest_st.at(bob_ip).sent_packets_size.NumSamples(), 2000l);
+    EXPECT_EQ(dest_st.at(bob_ip).sent_packets_size.GetAverage(),
+              single_packet_size);
+    EXPECT_EQ(source_st.at(bob_ip).received_packets_size.NumSamples(), 2000l);
+    EXPECT_EQ(source_st.at(bob_ip).received_packets_size.GetAverage(),
+              single_packet_size);
+    EXPECT_TRUE(
+        source_st.at(bob_ip).packets_discarded_no_receiver_size.IsEmpty());
+
+    received_stats_count++;
+  });
+  ASSERT_EQ_SIMULATED_WAIT(received_stats_count.load(), 1,
+                           kStatsWaitTimeout.ms(),
+                           *network_manager.time_controller());
+}
+
+TEST(NetworkEmulationManagerTest, ThroughputStats) {
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault);
+
+  EmulatedNetworkNode* alice_node = network_manager.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedNetworkNode* bob_node = network_manager.CreateEmulatedNode(
+      std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+  EmulatedEndpoint* alice_endpoint =
+      network_manager.CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_manager.CreateEndpoint(EmulatedEndpointConfig());
+  network_manager.CreateRoute(alice_endpoint, {alice_node}, bob_endpoint);
+  network_manager.CreateRoute(bob_endpoint, {bob_node}, alice_endpoint);
+
+  EmulatedNetworkManagerInterface* nt1 =
+      network_manager.CreateEmulatedNetworkManagerInterface({alice_endpoint});
+  EmulatedNetworkManagerInterface* nt2 =
+      network_manager.CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  rtc::Thread* t1 = nt1->network_thread();
+  rtc::Thread* t2 = nt2->network_thread();
+
+  constexpr int64_t kUdpPayloadSize = 100;
+  constexpr int64_t kSinglePacketSize = kUdpPayloadSize + kOverheadIpv4Udp;
+  rtc::CopyOnWriteBuffer data(kUdpPayloadSize);
+
+  rtc::Socket* s1 = nullptr;
+  rtc::Socket* s2 = nullptr;
+  SendTask(t1,
+           [&] { s1 = t1->socketserver()->CreateSocket(AF_INET, SOCK_DGRAM); });
+  SendTask(t2,
+           [&] { s2 = t2->socketserver()->CreateSocket(AF_INET, SOCK_DGRAM); });
+
+  SocketReader r1(s1, t1);
+  SocketReader r2(s2, t2);
+
+  rtc::SocketAddress a1(alice_endpoint->GetPeerLocalAddress(), 0);
+  rtc::SocketAddress a2(bob_endpoint->GetPeerLocalAddress(), 0);
+
+  SendTask(t1, [&] {
+    s1->Bind(a1);
+    a1 = s1->GetLocalAddress();
+  });
+  SendTask(t2, [&] {
+    s2->Bind(a2);
+    a2 = s2->GetLocalAddress();
+  });
+
+  SendTask(t1, [&] { s1->Connect(a2); });
+  SendTask(t2, [&] { s2->Connect(a1); });
+
+  // Send 11 packets, totalizing 1 second between the first and the last->
+  const int kNumPacketsSent = 11;
+  const TimeDelta kDelay = TimeDelta::Millis(100);
+  for (int i = 0; i < kNumPacketsSent; i++) {
+    t1->PostTask([&]() { s1->Send(data.data(), data.size()); });
+    t2->PostTask([&]() { s2->Send(data.data(), data.size()); });
+    network_manager.time_controller()->AdvanceTime(kDelay);
+  }
+
+  std::atomic<int> received_stats_count{0};
+  nt1->GetStats([&](EmulatedNetworkStats st) {
+    EXPECT_EQ(st.PacketsSent(), kNumPacketsSent);
+    EXPECT_EQ(st.BytesSent().bytes(), kSinglePacketSize * kNumPacketsSent);
+
+    const double tolerance = 0.95;  // Accept 5% tolerance for timing.
+    EXPECT_GE(st.LastPacketSentTime() - st.FirstPacketSentTime(),
+              (kNumPacketsSent - 1) * kDelay * tolerance);
+    EXPECT_GT(st.AverageSendRate().bps(), 0);
+    received_stats_count++;
+  });
+
+  ASSERT_EQ_SIMULATED_WAIT(received_stats_count.load(), 1,
+                           kStatsWaitTimeout.ms(),
+                           *network_manager.time_controller());
+
+  EXPECT_EQ(r1.ReceivedCount(), 11);
+  EXPECT_EQ(r2.ReceivedCount(), 11);
+
+  SendTask(t1, [&] { delete s1; });
+  SendTask(t2, [&] { delete s2; });
+}
+
+// Testing that packets are delivered via all routes using a routing scheme as
+// follows:
+//  * e1 -> n1 -> e2
+//  * e2 -> n2 -> e1
+//  * e1 -> n3 -> e3
+//  * e3 -> n4 -> e1
+TEST_F(NetworkEmulationManagerThreeNodesRoutingTest,
+       PacketsAreDeliveredInBothWaysWhenConnectedToTwoPeers) {
+  SetupRouting([](EmulatedEndpoint* e1, EmulatedEndpoint* e2,
+                  EmulatedEndpoint* e3, NetworkEmulationManager* emulation) {
+    auto* node1 = CreateEmulatedNodeWithDefaultBuiltInConfig(emulation);
+    auto* node2 = CreateEmulatedNodeWithDefaultBuiltInConfig(emulation);
+    auto* node3 = CreateEmulatedNodeWithDefaultBuiltInConfig(emulation);
+    auto* node4 = CreateEmulatedNodeWithDefaultBuiltInConfig(emulation);
+
+    emulation->CreateRoute(e1, {node1}, e2);
+    emulation->CreateRoute(e2, {node2}, e1);
+
+    emulation->CreateRoute(e1, {node3}, e3);
+    emulation->CreateRoute(e3, {node4}, e1);
+  });
+  SendPacketsAndValidateDelivery();
+}
+
+// Testing that packets are delivered via all routes using a routing scheme as
+// follows:
+//  * e1 -> n1 -> e2
+//  * e2 -> n2 -> e1
+//  * e1 -> n1 -> e3
+//  * e3 -> n4 -> e1
+TEST_F(NetworkEmulationManagerThreeNodesRoutingTest,
+       PacketsAreDeliveredInBothWaysWhenConnectedToTwoPeersOverSameSendLink) {
+  SetupRouting([](EmulatedEndpoint* e1, EmulatedEndpoint* e2,
+                  EmulatedEndpoint* e3, NetworkEmulationManager* emulation) {
+    auto* node1 = CreateEmulatedNodeWithDefaultBuiltInConfig(emulation);
+    auto* node2 = CreateEmulatedNodeWithDefaultBuiltInConfig(emulation);
+    auto* node3 = CreateEmulatedNodeWithDefaultBuiltInConfig(emulation);
+
+    emulation->CreateRoute(e1, {node1}, e2);
+    emulation->CreateRoute(e2, {node2}, e1);
+
+    emulation->CreateRoute(e1, {node1}, e3);
+    emulation->CreateRoute(e3, {node3}, e1);
+  });
+  SendPacketsAndValidateDelivery();
+}
+
+TEST(NetworkEmulationManagerTest, EndpointLoopback) {
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kSimulated, EmulatedNetworkStatsGatheringMode::kDefault);
+  auto endpoint = network_manager.CreateEndpoint(EmulatedEndpointConfig());
+
+  MockReceiver receiver;
+  EXPECT_CALL(receiver, OnPacketReceived(::testing::_)).Times(1);
+  ASSERT_EQ(endpoint->BindReceiver(80, &receiver), 80);
+
+  endpoint->SendPacket(rtc::SocketAddress(endpoint->GetPeerLocalAddress(), 80),
+                       rtc::SocketAddress(endpoint->GetPeerLocalAddress(), 80),
+                       "Hello");
+  network_manager.time_controller()->AdvanceTime(TimeDelta::Seconds(1));
+}
+
+TEST(NetworkEmulationManagerTest, EndpointCanSendWithDifferentSourceIp) {
+  constexpr uint32_t kEndpointIp = 0xC0A80011;  // 192.168.0.17
+  constexpr uint32_t kSourceIp = 0xC0A80012;    // 192.168.0.18
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kSimulated, EmulatedNetworkStatsGatheringMode::kDefault);
+  EmulatedEndpointConfig endpoint_config;
+  endpoint_config.ip = rtc::IPAddress(kEndpointIp);
+  endpoint_config.allow_send_packet_with_different_source_ip = true;
+  auto endpoint = network_manager.CreateEndpoint(endpoint_config);
+
+  MockReceiver receiver;
+  EXPECT_CALL(receiver, OnPacketReceived(::testing::_)).Times(1);
+  ASSERT_EQ(endpoint->BindReceiver(80, &receiver), 80);
+
+  endpoint->SendPacket(rtc::SocketAddress(kSourceIp, 80),
+                       rtc::SocketAddress(endpoint->GetPeerLocalAddress(), 80),
+                       "Hello");
+  network_manager.time_controller()->AdvanceTime(TimeDelta::Seconds(1));
+}
+
+TEST(NetworkEmulationManagerTest,
+     EndpointCanReceiveWithDifferentDestIpThroughDefaultRoute) {
+  constexpr uint32_t kDestEndpointIp = 0xC0A80011;  // 192.168.0.17
+  constexpr uint32_t kDestIp = 0xC0A80012;          // 192.168.0.18
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kSimulated, EmulatedNetworkStatsGatheringMode::kDefault);
+  auto sender_endpoint =
+      network_manager.CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpointConfig endpoint_config;
+  endpoint_config.ip = rtc::IPAddress(kDestEndpointIp);
+  endpoint_config.allow_receive_packets_with_different_dest_ip = true;
+  auto receiver_endpoint = network_manager.CreateEndpoint(endpoint_config);
+
+  MockReceiver receiver;
+  EXPECT_CALL(receiver, OnPacketReceived(::testing::_)).Times(1);
+  ASSERT_EQ(receiver_endpoint->BindReceiver(80, &receiver), 80);
+
+  network_manager.CreateDefaultRoute(
+      sender_endpoint, {network_manager.NodeBuilder().Build().node},
+      receiver_endpoint);
+
+  sender_endpoint->SendPacket(
+      rtc::SocketAddress(sender_endpoint->GetPeerLocalAddress(), 80),
+      rtc::SocketAddress(kDestIp, 80), "Hello");
+  network_manager.time_controller()->AdvanceTime(TimeDelta::Seconds(1));
+}
+
+TEST(NetworkEmulationManagerTURNTest, GetIceServerConfig) {
+  NetworkEmulationManagerImpl network_manager(
+      TimeMode::kRealTime, EmulatedNetworkStatsGatheringMode::kDefault);
+  auto turn = network_manager.CreateTURNServer(EmulatedTURNServerConfig());
+
+  EXPECT_GT(turn->GetIceServerConfig().username.size(), 0u);
+  EXPECT_GT(turn->GetIceServerConfig().password.size(), 0u);
+  EXPECT_NE(turn->GetIceServerConfig().url.find(
+                turn->GetClientEndpoint()->GetPeerLocalAddress().ToString()),
+            std::string::npos);
+}
+
+TEST(NetworkEmulationManagerTURNTest, ClientTraffic) {
+  NetworkEmulationManagerImpl emulation(
+      TimeMode::kSimulated, EmulatedNetworkStatsGatheringMode::kDefault);
+  auto* ep = emulation.CreateEndpoint(EmulatedEndpointConfig());
+  auto* turn = emulation.CreateTURNServer(EmulatedTURNServerConfig());
+  auto* node = CreateEmulatedNodeWithDefaultBuiltInConfig(&emulation);
+  emulation.CreateRoute(ep, {node}, turn->GetClientEndpoint());
+  emulation.CreateRoute(turn->GetClientEndpoint(), {node}, ep);
+
+  MockReceiver recv;
+  int port = ep->BindReceiver(0, &recv).value();
+
+  // Construct a STUN BINDING.
+  cricket::StunMessage ping(cricket::STUN_BINDING_REQUEST);
+  rtc::ByteBufferWriter buf;
+  ping.Write(&buf);
+  rtc::CopyOnWriteBuffer packet(buf.Data(), buf.Length());
+
+  // We expect to get a ping reply.
+  EXPECT_CALL(recv, OnPacketReceived(::testing::_)).Times(1);
+
+  ep->SendPacket(rtc::SocketAddress(ep->GetPeerLocalAddress(), port),
+                 turn->GetClientEndpointAddress(), packet);
+  emulation.time_controller()->AdvanceTime(TimeDelta::Seconds(1));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/traffic_route.cc b/third_party/libwebrtc/test/network/traffic_route.cc
new file mode 100644
index 0000000000..81bb8ca514
--- /dev/null
+++ b/third_party/libwebrtc/test/network/traffic_route.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/network/traffic_route.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+class NullReceiver : public EmulatedNetworkReceiverInterface {
+ public:
+  void OnPacketReceived(EmulatedIpPacket packet) override {}
+};
+
+class ActionReceiver : public EmulatedNetworkReceiverInterface {
+ public:
+  explicit ActionReceiver(std::function<void()> action) : action_(action) {}
+  ~ActionReceiver() override = default;
+
+  void OnPacketReceived(EmulatedIpPacket packet) override {
+    action_();
+  }
+
+ private:
+  std::function<void()> action_;
+};
+
+}  // namespace
+
+CrossTrafficRouteImpl::CrossTrafficRouteImpl(
+    Clock* clock,
+    EmulatedNetworkReceiverInterface* receiver,
+    EmulatedEndpointImpl* endpoint)
+    : clock_(clock), receiver_(receiver), endpoint_(endpoint) {
+  null_receiver_ = std::make_unique<NullReceiver>();
+  absl::optional<uint16_t> port =
+      endpoint_->BindReceiver(0, null_receiver_.get());
+  RTC_DCHECK(port);
+  null_receiver_port_ = port.value();
+}
+CrossTrafficRouteImpl::~CrossTrafficRouteImpl() = default;
+
+void CrossTrafficRouteImpl::TriggerPacketBurst(size_t num_packets,
+                                               size_t packet_size) {
+  for (size_t i = 0; i < num_packets; ++i) {
+    SendPacket(packet_size);
+  }
+}
+
+void CrossTrafficRouteImpl::NetworkDelayedAction(size_t packet_size,
+                                                 std::function<void()> action) {
+  auto action_receiver = std::make_unique<ActionReceiver>(action);
+  // BindOneShotReceiver arranges to free the port in the endpoint after the
+  // action is done.
+  absl::optional<uint16_t> port =
+      endpoint_->BindOneShotReceiver(0, action_receiver.get());
+  RTC_DCHECK(port);
+  actions_.push_back(std::move(action_receiver));
+  SendPacket(packet_size, port.value());
+}
+
+void CrossTrafficRouteImpl::SendPacket(size_t packet_size) {
+  SendPacket(packet_size, null_receiver_port_);
+}
+
+void CrossTrafficRouteImpl::SendPacket(size_t packet_size, uint16_t dest_port) {
+  rtc::CopyOnWriteBuffer data(packet_size);
+  std::fill_n(data.MutableData(), data.size(), 0);
+  receiver_->OnPacketReceived(EmulatedIpPacket(
+      /*from=*/rtc::SocketAddress(),
+      rtc::SocketAddress(endpoint_->GetPeerLocalAddress(), dest_port), data,
+      clock_->CurrentTime()));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/network/traffic_route.h b/third_party/libwebrtc/test/network/traffic_route.h
new file mode 100644
index 0000000000..dbc41a694f
--- /dev/null
+++ b/third_party/libwebrtc/test/network/traffic_route.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_NETWORK_TRAFFIC_ROUTE_H_
+#define TEST_NETWORK_TRAFFIC_ROUTE_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/test/network_emulation_manager.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "system_wrappers/include/clock.h"
+#include "test/network/network_emulation.h"
+
+namespace webrtc {
+namespace test {
+
+// Represents the endpoint for cross traffic that is going through the network.
+// It can be used to emulate unexpected network load.
+class CrossTrafficRouteImpl final : public CrossTrafficRoute {
+ public:
+  CrossTrafficRouteImpl(Clock* clock,
+                        EmulatedNetworkReceiverInterface* receiver,
+                        EmulatedEndpointImpl* endpoint);
+  ~CrossTrafficRouteImpl();
+
+  // Triggers sending of dummy packets with size `packet_size` bytes.
+  void TriggerPacketBurst(size_t num_packets, size_t packet_size) override;
+  // Sends a packet over the nodes and runs `action` when it has been delivered.
+  void NetworkDelayedAction(size_t packet_size,
+                            std::function<void()> action) override;
+
+  void SendPacket(size_t packet_size) override;
+
+ private:
+  void SendPacket(size_t packet_size, uint16_t dest_port);
+
+  Clock* const clock_;
+  EmulatedNetworkReceiverInterface* const receiver_;
+  EmulatedEndpointImpl* const endpoint_;
+
+  uint16_t null_receiver_port_;
+  std::unique_ptr<EmulatedNetworkReceiverInterface> null_receiver_;
+  std::vector<std::unique_ptr<EmulatedNetworkReceiverInterface>> actions_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_NETWORK_TRAFFIC_ROUTE_H_
diff --git a/third_party/libwebrtc/test/null_platform_renderer.cc b/third_party/libwebrtc/test/null_platform_renderer.cc
new file mode 100644
index 0000000000..7ea604ead8
--- /dev/null
+++ b/third_party/libwebrtc/test/null_platform_renderer.cc
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace test {
+
+VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title,
+                                                     size_t width,
+                                                     size_t height) {
+  return NULL;
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/null_transport.cc b/third_party/libwebrtc/test/null_transport.cc
new file mode 100644
index 0000000000..efbd9499d8
--- /dev/null
+++ b/third_party/libwebrtc/test/null_transport.cc
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/null_transport.h"
+
+namespace webrtc {
+namespace test {
+
+bool NullTransport::SendRtp(const uint8_t* packet,
+                            size_t length,
+                            const PacketOptions& options) {
+  return true;
+}
+
+bool NullTransport::SendRtcp(const uint8_t* packet, size_t length) {
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/null_transport.h b/third_party/libwebrtc/test/null_transport.h
new file mode 100644
index 0000000000..f264e7b45a
--- /dev/null
+++ b/third_party/libwebrtc/test/null_transport.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_NULL_TRANSPORT_H_
+#define TEST_NULL_TRANSPORT_H_
+
+#include "api/call/transport.h"
+
+namespace webrtc {
+
+class PacketReceiver;
+
+namespace test {
+class NullTransport : public Transport {
+ public:
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_NULL_TRANSPORT_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/BUILD.gn b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
new file mode 100644
index 0000000000..7354aa8ba4
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
@@ -0,0 +1,573 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("metric_metadata_keys") {
+  testonly = true
+  sources = [ "metric_metadata_keys.h" ]
+}
+
+if (!build_with_chromium) {
+  group("e2e") {
+    testonly = true
+
+    deps = [ ":metric_metadata_keys" ]
+    if (rtc_include_tests) {
+      deps += [
+        ":peerconnection_quality_test",
+        ":test_peer",
+      ]
+    }
+  }
+
+  if (rtc_include_tests) {
+    group("e2e_unittests") {
+      testonly = true
+
+      deps = [
+        ":peer_connection_e2e_smoke_test",
+        ":peer_connection_quality_test_metric_names_test",
+        ":peer_connection_quality_test_test",
+        ":stats_based_network_quality_metrics_reporter_test",
+        ":stats_poller_test",
+      ]
+    }
+  }
+
+  if (rtc_include_tests) {
+    rtc_library("echo_emulation") {
+      testonly = true
+      sources = [
+        "echo/echo_emulation.cc",
+        "echo/echo_emulation.h",
+      ]
+      deps = [
+        "../../../api/test/pclf:media_configuration",
+        "../../../modules/audio_device:audio_device_impl",
+        "../../../rtc_base:swap_queue",
+      ]
+    }
+
+    rtc_library("test_peer") {
+      testonly = true
+      sources = [
+        "test_peer.cc",
+        "test_peer.h",
+      ]
+      deps = [
+        ":stats_provider",
+        "../../../api:frame_generator_api",
+        "../../../api:function_view",
+        "../../../api:libjingle_peerconnection_api",
+        "../../../api:scoped_refptr",
+        "../../../api:sequence_checker",
+        "../../../api/task_queue:pending_task_safety_flag",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../modules/audio_processing:api",
+        "../../../pc:peerconnection_wrapper",
+        "../../../rtc_base:logging",
+        "../../../rtc_base:refcount",
+        "../../../rtc_base/synchronization:mutex",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/memory",
+        "//third_party/abseil-cpp/absl/strings",
+        "//third_party/abseil-cpp/absl/types:variant",
+      ]
+    }
+
+    rtc_library("test_peer_factory") {
+      testonly = true
+      sources = [
+        "test_peer_factory.cc",
+        "test_peer_factory.h",
+      ]
+      deps = [
+        ":echo_emulation",
+        ":test_peer",
+        "../..:copy_to_file_audio_capturer",
+        "../../../api:create_time_controller",
+        "../../../api:time_controller",
+        "../../../api/rtc_event_log:rtc_event_log_factory",
+        "../../../api/task_queue:default_task_queue_factory",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../api/transport:field_trial_based_config",
+        "../../../api/video_codecs:builtin_video_decoder_factory",
+        "../../../api/video_codecs:builtin_video_encoder_factory",
+        "../../../media:rtc_audio_video",
+        "../../../media:rtc_media_engine_defaults",
+        "../../../modules/audio_device:audio_device_impl",
+        "../../../modules/audio_processing/aec_dump",
+        "../../../p2p:rtc_p2p",
+        "../../../rtc_base:rtc_task_queue",
+        "../../../rtc_base:threading",
+        "analyzer/video:quality_analyzing_video_encoder",
+        "analyzer/video:video_quality_analyzer_injection_helper",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/memory",
+        "//third_party/abseil-cpp/absl/strings",
+      ]
+    }
+
+    rtc_library("media_helper") {
+      testonly = true
+      sources = [
+        "media/media_helper.cc",
+        "media/media_helper.h",
+        "media/test_video_capturer_video_track_source.h",
+      ]
+      deps = [
+        ":test_peer",
+        "../..:fileutils",
+        "../..:platform_video_capturer",
+        "../..:video_test_common",
+        "../../../api:create_frame_generator",
+        "../../../api:frame_generator_api",
+        "../../../api:media_stream_interface",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../api/video:video_frame",
+        "../../../pc:session_description",
+        "../../../pc:video_track_source",
+        "analyzer/video:video_quality_analyzer_injection_helper",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:variant" ]
+    }
+
+    rtc_library("peer_params_preprocessor") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [
+        "peer_params_preprocessor.cc",
+        "peer_params_preprocessor.h",
+      ]
+      deps = [
+        "../..:fileutils",
+        "../../../api:peer_network_dependencies",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../modules/video_coding/svc:scalability_mode_util",
+        "../../../modules/video_coding/svc:scalability_structures",
+        "../../../rtc_base:macromagic",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    }
+
+    rtc_library("test_activities_executor") {
+      testonly = true
+      sources = [
+        "test_activities_executor.cc",
+        "test_activities_executor.h",
+      ]
+      deps = [
+        "../../../api/task_queue",
+        "../../../api/units:time_delta",
+        "../../../api/units:timestamp",
+        "../../../rtc_base:checks",
+        "../../../rtc_base:criticalsection",
+        "../../../rtc_base:logging",
+        "../../../rtc_base:task_queue_for_test",
+        "../../../rtc_base/synchronization:mutex",
+        "../../../rtc_base/task_utils:repeating_task",
+        "../../../system_wrappers",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/memory",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+    }
+
+    rtc_library("peerconnection_quality_test") {
+      testonly = true
+
+      sources = [
+        "peer_connection_quality_test.cc",
+        "peer_connection_quality_test.h",
+      ]
+      deps = [
+        ":analyzer_helper",
+        ":cross_media_metrics_reporter",
+        ":default_audio_quality_analyzer",
+        ":media_helper",
+        ":metric_metadata_keys",
+        ":peer_params_preprocessor",
+        ":sdp_changer",
+        ":stats_poller",
+        ":test_activities_executor",
+        ":test_peer",
+        ":test_peer_factory",
+        "../..:field_trial",
+        "../..:fileutils",
+        "../..:perf_test",
+        "../../../api:audio_quality_analyzer_api",
+        "../../../api:libjingle_peerconnection_api",
+        "../../../api:media_stream_interface",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api:rtc_event_log_output_file",
+        "../../../api:scoped_refptr",
+        "../../../api:time_controller",
+        "../../../api:video_quality_analyzer_api",
+        "../../../api/rtc_event_log",
+        "../../../api/task_queue",
+        "../../../api/test/metrics:metric",
+        "../../../api/test/metrics:metrics_logger",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../api/units:time_delta",
+        "../../../api/units:timestamp",
+        "../../../pc:pc_test_utils",
+        "../../../pc:sdp_utils",
+        "../../../rtc_base:gunit_helpers",
+        "../../../rtc_base:macromagic",
+        "../../../rtc_base:safe_conversions",
+        "../../../rtc_base:stringutils",
+        "../../../rtc_base:task_queue_for_test",
+        "../../../rtc_base:threading",
+        "../../../rtc_base/synchronization:mutex",
+        "../../../system_wrappers",
+        "../../../system_wrappers:field_trial",
+        "analyzer/video:default_video_quality_analyzer",
+        "analyzer/video:single_process_encoded_image_data_injector",
+        "analyzer/video:video_frame_tracking_id_injector",
+        "analyzer/video:video_quality_analyzer_injection_helper",
+        "analyzer/video:video_quality_metrics_reporter",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    }
+
+    peer_connection_e2e_smoke_test_resources = [
+      "../../../resources/pc_quality_smoke_test_alice_source.wav",
+      "../../../resources/pc_quality_smoke_test_bob_source.wav",
+    ]
+    if (is_ios) {
+      bundle_data("peer_connection_e2e_smoke_test_resources_bundle_data") {
+        testonly = true
+        sources = peer_connection_e2e_smoke_test_resources
+        outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+      }
+    }
+
+    rtc_library("peer_connection_e2e_smoke_test") {
+      testonly = true
+
+      sources = [ "peer_connection_e2e_smoke_test.cc" ]
+      deps = [
+        ":default_audio_quality_analyzer",
+        ":network_quality_metrics_reporter",
+        ":stats_based_network_quality_metrics_reporter",
+        "../../../api:callfactory_api",
+        "../../../api:create_network_emulation_manager",
+        "../../../api:create_peer_connection_quality_test_frame_generator",
+        "../../../api:create_peerconnection_quality_test_fixture",
+        "../../../api:libjingle_peerconnection_api",
+        "../../../api:media_stream_interface",
+        "../../../api:network_emulation_manager_api",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api:scoped_refptr",
+        "../../../api:simulated_network_api",
+        "../../../api/audio_codecs:builtin_audio_decoder_factory",
+        "../../../api/audio_codecs:builtin_audio_encoder_factory",
+        "../../../api/test/metrics:global_metrics_logger_and_exporter",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../api/video_codecs:builtin_video_decoder_factory",
+        "../../../api/video_codecs:builtin_video_encoder_factory",
+        "../../../call:simulated_network",
+        "../../../media:rtc_audio_video",
+        "../../../modules/audio_device:audio_device_impl",
+        "../../../p2p:rtc_p2p",
+        "../../../pc:pc_test_utils",
+        "../../../pc:peerconnection_wrapper",
+        "../../../rtc_base:gunit_helpers",
+        "../../../rtc_base:logging",
+        "../../../rtc_base:rtc_event",
+        "../../../system_wrappers:field_trial",
+        "../../../test:field_trial",
+        "../../../test:fileutils",
+        "../../../test:test_support",
+        "analyzer/video:default_video_quality_analyzer",
+        "analyzer/video:default_video_quality_analyzer_shared",
+      ]
+      data = peer_connection_e2e_smoke_test_resources
+      if (is_mac || is_ios) {
+        deps += [ "../../../modules/video_coding:objc_codec_factory_helper" ]
+      }
+      if (is_ios) {
+        deps += [ ":peer_connection_e2e_smoke_test_resources_bundle_data" ]
+      }
+    }
+
+    rtc_library("peer_connection_quality_test_metric_names_test") {
+      testonly = true
+      sources = [ "peer_connection_quality_test_metric_names_test.cc" ]
+      deps = [
+        ":metric_metadata_keys",
+        ":peerconnection_quality_test",
+        ":stats_based_network_quality_metrics_reporter",
+        "../..:test_support",
+        "../../../api:create_network_emulation_manager",
+        "../../../api:create_peer_connection_quality_test_frame_generator",
+        "../../../api:network_emulation_manager_api",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api/test/metrics:metrics_logger",
+        "../../../api/test/metrics:stdout_metrics_exporter",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../api/units:time_delta",
+      ]
+    }
+
+    rtc_library("stats_based_network_quality_metrics_reporter_test") {
+      testonly = true
+      sources = [ "stats_based_network_quality_metrics_reporter_test.cc" ]
+      deps = [
+        ":metric_metadata_keys",
+        ":peerconnection_quality_test",
+        ":stats_based_network_quality_metrics_reporter",
+        "../..:test_support",
+        "../../../api:array_view",
+        "../../../api:create_network_emulation_manager",
+        "../../../api:create_peer_connection_quality_test_frame_generator",
+        "../../../api:network_emulation_manager_api",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api/test/metrics:metrics_logger",
+        "../../../api/test/metrics:stdout_metrics_exporter",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../api/units:time_delta",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/strings",
+        "//third_party/abseil-cpp/absl/types:optional",
+      ]
+    }
+
+    rtc_library("peer_connection_quality_test_test") {
+      testonly = true
+      sources = [ "peer_connection_quality_test_test.cc" ]
+      deps = [
+        ":peerconnection_quality_test",
+        "../..:fileutils",
+        "../..:test_support",
+        "../..:video_test_support",
+        "../../../api:create_network_emulation_manager",
+        "../../../api:network_emulation_manager_api",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api/test/metrics:global_metrics_logger_and_exporter",
+        "../../../api/test/pclf:media_configuration",
+        "../../../api/test/pclf:media_quality_test_params",
+        "../../../api/test/pclf:peer_configurer",
+        "../../../api/units:time_delta",
+        "../../../rtc_base:timeutils",
+      ]
+    }
+
+    rtc_library("stats_provider") {
+      testonly = true
+      sources = [ "stats_provider.h" ]
+      deps = [ "../../../api:rtc_stats_api" ]
+    }
+
+    rtc_library("stats_poller") {
+      testonly = true
+      sources = [
+        "stats_poller.cc",
+        "stats_poller.h",
+      ]
+      deps = [
+        ":stats_provider",
+        ":test_peer",
+        "../../../api:libjingle_peerconnection_api",
+        "../../../api:rtc_stats_api",
+        "../../../api:stats_observer_interface",
+        "../../../rtc_base:logging",
+        "../../../rtc_base:macromagic",
+        "../../../rtc_base/synchronization:mutex",
+      ]
+    }
+
+    rtc_library("stats_poller_test") {
+      testonly = true
+      sources = [ "stats_poller_test.cc" ]
+      deps = [
+        ":stats_poller",
+        "../..:test_support",
+        "../../../api:rtc_stats_api",
+      ]
+    }
+  }
+
+  rtc_library("analyzer_helper") {
+    sources = [
+      "analyzer_helper.cc",
+      "analyzer_helper.h",
+    ]
+    deps = [
+      "../../../api:sequence_checker",
+      "../../../api:track_id_stream_info_map",
+      "../../../rtc_base:macromagic",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_library("default_audio_quality_analyzer") {
+    testonly = true
+    sources = [
+      "analyzer/audio/default_audio_quality_analyzer.cc",
+      "analyzer/audio/default_audio_quality_analyzer.h",
+    ]
+
+    deps = [
+      ":metric_metadata_keys",
+      "../..:perf_test",
+      "../../../api:audio_quality_analyzer_api",
+      "../../../api:rtc_stats_api",
+      "../../../api:stats_observer_interface",
+      "../../../api:track_id_stream_info_map",
+      "../../../api/numerics",
+      "../../../api/test/metrics:metric",
+      "../../../api/test/metrics:metrics_logger",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:rtc_numerics",
+      "../../../rtc_base/synchronization:mutex",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("network_quality_metrics_reporter") {
+    testonly = true
+    sources = [
+      "network_quality_metrics_reporter.cc",
+      "network_quality_metrics_reporter.h",
+    ]
+    deps = [
+      "../..:perf_test",
+      "../../../api:network_emulation_manager_api",
+      "../../../api:peer_connection_quality_test_fixture_api",
+      "../../../api:rtc_stats_api",
+      "../../../api:track_id_stream_info_map",
+      "../../../api/test/metrics:metric",
+      "../../../api/test/metrics:metrics_logger",
+      "../../../api/units:data_size",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:rtc_event",
+      "../../../rtc_base/synchronization:mutex",
+      "../../../system_wrappers:field_trial",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("stats_based_network_quality_metrics_reporter") {
+    testonly = true
+    sources = [
+      "stats_based_network_quality_metrics_reporter.cc",
+      "stats_based_network_quality_metrics_reporter.h",
+    ]
+    deps = [
+      ":metric_metadata_keys",
+      "../..:perf_test",
+      "../../../api:array_view",
+      "../../../api:network_emulation_manager_api",
+      "../../../api:peer_connection_quality_test_fixture_api",
+      "../../../api:rtc_stats_api",
+      "../../../api:scoped_refptr",
+      "../../../api:sequence_checker",
+      "../../../api/numerics",
+      "../../../api/test/metrics:metric",
+      "../../../api/test/metrics:metrics_logger",
+      "../../../api/test/network_emulation",
+      "../../../api/units:data_rate",
+      "../../../api/units:data_size",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:ip_address",
+      "../../../rtc_base:rtc_event",
+      "../../../rtc_base:stringutils",
+      "../../../rtc_base/synchronization:mutex",
+      "../../../rtc_base/system:no_unique_address",
+      "../../../system_wrappers:field_trial",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("cross_media_metrics_reporter") {
+    testonly = true
+    sources = [
+      "cross_media_metrics_reporter.cc",
+      "cross_media_metrics_reporter.h",
+    ]
+    deps = [
+      ":metric_metadata_keys",
+      "../..:perf_test",
+      "../../../api:network_emulation_manager_api",
+      "../../../api:peer_connection_quality_test_fixture_api",
+      "../../../api:rtc_stats_api",
+      "../../../api:track_id_stream_info_map",
+      "../../../api/numerics",
+      "../../../api/test/metrics:metric",
+      "../../../api/test/metrics:metrics_logger",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:rtc_event",
+      "../../../rtc_base:rtc_numerics",
+      "../../../rtc_base/synchronization:mutex",
+      "../../../system_wrappers:field_trial",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_library("sdp_changer") {
+    testonly = true
+    sources = [
+      "sdp/sdp_changer.cc",
+      "sdp/sdp_changer.h",
+    ]
+    deps = [
+      "../../../api:array_view",
+      "../../../api:libjingle_peerconnection_api",
+      "../../../api:rtp_parameters",
+      "../../../api/test/pclf:media_configuration",
+      "../../../media:media_constants",
+      "../../../media:rid_description",
+      "../../../media:rtc_media_base",
+      "../../../p2p:rtc_p2p",
+      "../../../pc:sdp_utils",
+      "../../../pc:session_description",
+      "../../../pc:simulcast_description",
+      "../../../rtc_base:stringutils",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/strings:strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
new file mode 100644
index 0000000000..98d0c533c2
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
@@ -0,0 +1,175 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h"
+
+#include "api/stats/rtc_stats.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/test/metrics/metric.h"
+#include "api/test/track_id_stream_info_map.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+DefaultAudioQualityAnalyzer::DefaultAudioQualityAnalyzer(
+    test::MetricsLogger* const metrics_logger)
+    : metrics_logger_(metrics_logger) {
+  RTC_CHECK(metrics_logger_);
+}
+
+void DefaultAudioQualityAnalyzer::Start(std::string test_case_name,
+                                        TrackIdStreamInfoMap* analyzer_helper) {
+  test_case_name_ = std::move(test_case_name);
+  analyzer_helper_ = analyzer_helper;
+}
+
+void DefaultAudioQualityAnalyzer::OnStatsReports(
+    absl::string_view pc_label,
+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
+  auto stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
+
+  for (auto& stat : stats) {
+    if (!stat->kind.is_defined() ||
+        !(*stat->kind == RTCMediaStreamTrackKind::kAudio)) {
+      continue;
+    }
+
+    StatsSample sample;
+    sample.total_samples_received =
+        stat->total_samples_received.ValueOrDefault(0ul);
+    sample.concealed_samples = stat->concealed_samples.ValueOrDefault(0ul);
+    sample.removed_samples_for_acceleration =
+        stat->removed_samples_for_acceleration.ValueOrDefault(0ul);
+    sample.inserted_samples_for_deceleration =
+        stat->inserted_samples_for_deceleration.ValueOrDefault(0ul);
+    sample.silent_concealed_samples =
+        stat->silent_concealed_samples.ValueOrDefault(0ul);
+    sample.jitter_buffer_delay =
+        TimeDelta::Seconds(stat->jitter_buffer_delay.ValueOrDefault(0.));
+    sample.jitter_buffer_target_delay =
+        TimeDelta::Seconds(stat->jitter_buffer_target_delay.ValueOrDefault(0.));
+    sample.jitter_buffer_emitted_count =
+        stat->jitter_buffer_emitted_count.ValueOrDefault(0ul);
+
+    TrackIdStreamInfoMap::StreamInfo stream_info =
+        analyzer_helper_->GetStreamInfoFromTrackId(*stat->track_identifier);
+
+    MutexLock lock(&lock_);
+    stream_info_.emplace(stream_info.stream_label, stream_info);
+    StatsSample prev_sample = last_stats_sample_[stream_info.stream_label];
+    RTC_CHECK_GE(sample.total_samples_received,
+                 prev_sample.total_samples_received);
+    double total_samples_diff = static_cast<double>(
+        sample.total_samples_received - prev_sample.total_samples_received);
+    if (total_samples_diff == 0) {
+      return;
+    }
+
+    AudioStreamStats& audio_stream_stats =
+        streams_stats_[stream_info.stream_label];
+    audio_stream_stats.expand_rate.AddSample(
+        (sample.concealed_samples - prev_sample.concealed_samples) /
+        total_samples_diff);
+    audio_stream_stats.accelerate_rate.AddSample(
+        (sample.removed_samples_for_acceleration -
+         prev_sample.removed_samples_for_acceleration) /
+        total_samples_diff);
+    audio_stream_stats.preemptive_rate.AddSample(
+        (sample.inserted_samples_for_deceleration -
+         prev_sample.inserted_samples_for_deceleration) /
+        total_samples_diff);
+
+    int64_t speech_concealed_samples =
+        sample.concealed_samples - sample.silent_concealed_samples;
+    int64_t prev_speech_concealed_samples =
+        prev_sample.concealed_samples - prev_sample.silent_concealed_samples;
+    audio_stream_stats.speech_expand_rate.AddSample(
+        (speech_concealed_samples - prev_speech_concealed_samples) /
+        total_samples_diff);
+
+    int64_t jitter_buffer_emitted_count_diff =
+        sample.jitter_buffer_emitted_count -
+        prev_sample.jitter_buffer_emitted_count;
+    if (jitter_buffer_emitted_count_diff > 0) {
+      TimeDelta jitter_buffer_delay_diff =
+          sample.jitter_buffer_delay - prev_sample.jitter_buffer_delay;
+      TimeDelta jitter_buffer_target_delay_diff =
+          sample.jitter_buffer_target_delay -
+          prev_sample.jitter_buffer_target_delay;
+      audio_stream_stats.average_jitter_buffer_delay_ms.AddSample(
+          jitter_buffer_delay_diff.ms<double>() /
+          jitter_buffer_emitted_count_diff);
+      audio_stream_stats.preferred_buffer_size_ms.AddSample(
+          jitter_buffer_target_delay_diff.ms<double>() /
+          jitter_buffer_emitted_count_diff);
+    }
+
+    last_stats_sample_[stream_info.stream_label] = sample;
+  }
+}
+
+std::string DefaultAudioQualityAnalyzer::GetTestCaseName(
+    const std::string& stream_label) const {
+  return test_case_name_ + "/" + stream_label;
+}
+
+void DefaultAudioQualityAnalyzer::Stop() {
+  MutexLock lock(&lock_);
+  for (auto& item : streams_stats_) {
+    const TrackIdStreamInfoMap::StreamInfo& stream_info =
+        stream_info_[item.first];
+    // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+    std::map<std::string, std::string> metric_metadata{
+        {MetricMetadataKey::kAudioStreamMetadataKey, item.first},
+        {MetricMetadataKey::kPeerMetadataKey, stream_info.receiver_peer},
+        {MetricMetadataKey::kReceiverMetadataKey, stream_info.receiver_peer},
+        {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}};
+
+    metrics_logger_->LogMetric("expand_rate", GetTestCaseName(item.first),
+                               item.second.expand_rate, Unit::kUnitless,
+                               ImprovementDirection::kSmallerIsBetter,
+                               metric_metadata);
+    metrics_logger_->LogMetric("accelerate_rate", GetTestCaseName(item.first),
+                               item.second.accelerate_rate, Unit::kUnitless,
+                               ImprovementDirection::kSmallerIsBetter,
+                               metric_metadata);
+    metrics_logger_->LogMetric("preemptive_rate", GetTestCaseName(item.first),
+                               item.second.preemptive_rate, Unit::kUnitless,
+                               ImprovementDirection::kSmallerIsBetter,
+                               metric_metadata);
+    metrics_logger_->LogMetric(
+        "speech_expand_rate", GetTestCaseName(item.first),
+        item.second.speech_expand_rate, Unit::kUnitless,
+        ImprovementDirection::kSmallerIsBetter, metric_metadata);
+    metrics_logger_->LogMetric(
+        "average_jitter_buffer_delay_ms", GetTestCaseName(item.first),
+        item.second.average_jitter_buffer_delay_ms, Unit::kMilliseconds,
+        ImprovementDirection::kNeitherIsBetter, metric_metadata);
+    metrics_logger_->LogMetric(
+        "preferred_buffer_size_ms", GetTestCaseName(item.first),
+        item.second.preferred_buffer_size_ms, Unit::kMilliseconds,
+        ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+}
+
+std::map<std::string, AudioStreamStats>
+DefaultAudioQualityAnalyzer::GetAudioStreamsStats() const {
+  MutexLock lock(&lock_);
+  return streams_stats_;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h
new file mode 100644
index 0000000000..9e427afed8
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
+#define TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/audio_quality_analyzer_interface.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/track_id_stream_info_map.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+struct AudioStreamStats {
+  SamplesStatsCounter expand_rate;
+  SamplesStatsCounter accelerate_rate;
+  SamplesStatsCounter preemptive_rate;
+  SamplesStatsCounter speech_expand_rate;
+  SamplesStatsCounter average_jitter_buffer_delay_ms;
+  SamplesStatsCounter preferred_buffer_size_ms;
+};
+
+class DefaultAudioQualityAnalyzer : public AudioQualityAnalyzerInterface {
+ public:
+  explicit DefaultAudioQualityAnalyzer(
+      test::MetricsLogger* const metrics_logger);
+
+  void Start(std::string test_case_name,
+             TrackIdStreamInfoMap* analyzer_helper) override;
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override;
+  void Stop() override;
+
+  // Returns audio quality stats per stream label.
+  std::map<std::string, AudioStreamStats> GetAudioStreamsStats() const;
+
+ private:
+  struct StatsSample {
+    uint64_t total_samples_received = 0;
+    uint64_t concealed_samples = 0;
+    uint64_t removed_samples_for_acceleration = 0;
+    uint64_t inserted_samples_for_deceleration = 0;
+    uint64_t silent_concealed_samples = 0;
+    TimeDelta jitter_buffer_delay = TimeDelta::Zero();
+    TimeDelta jitter_buffer_target_delay = TimeDelta::Zero();
+    uint64_t jitter_buffer_emitted_count = 0;
+  };
+
+  std::string GetTestCaseName(const std::string& stream_label) const;
+
+  test::MetricsLogger* const metrics_logger_;
+
+  std::string test_case_name_;
+  TrackIdStreamInfoMap* analyzer_helper_;
+
+  mutable Mutex lock_;
+  std::map<std::string, AudioStreamStats> streams_stats_ RTC_GUARDED_BY(lock_);
+  std::map<std::string, TrackIdStreamInfoMap::StreamInfo> stream_info_
+      RTC_GUARDED_BY(lock_);
+  std::map<std::string, StatsSample> last_stats_sample_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_AUDIO_DEFAULT_AUDIO_QUALITY_ANALYZER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn b/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
new file mode 100644
index 0000000000..cbb4c078f3
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/BUILD.gn
@@ -0,0 +1,573 @@
+# Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../../../webrtc.gni")
+
+if (!build_with_chromium) {
+  group("video_analyzer") {
+    testonly = true
+
+    deps = [
+      ":analyzing_video_sinks_helper",
+      ":default_video_quality_analyzer_internal",
+      ":encoded_image_data_injector_api",
+      ":example_video_quality_analyzer",
+      ":multi_reader_queue",
+      ":quality_analyzing_video_decoder",
+      ":quality_analyzing_video_encoder",
+      ":simulcast_dummy_buffer_helper",
+      ":single_process_encoded_image_data_injector",
+      ":video_dumping",
+      ":video_frame_tracking_id_injector",
+      ":video_quality_metrics_reporter",
+    ]
+    if (rtc_include_tests) {
+      deps += [
+        ":analyzing_video_sink",
+        ":video_quality_analyzer_injection_helper",
+      ]
+    }
+  }
+
+  if (rtc_include_tests) {
+    group("video_analyzer_unittests") {
+      testonly = true
+
+      deps = [
+        ":analyzing_video_sink_test",
+        ":analyzing_video_sinks_helper_test",
+        ":default_video_quality_analyzer_frames_comparator_test",
+        ":default_video_quality_analyzer_metric_names_test",
+        ":default_video_quality_analyzer_stream_state_test",
+        ":default_video_quality_analyzer_test",
+        ":multi_reader_queue_test",
+        ":names_collection_test",
+        ":simulcast_dummy_buffer_helper_test",
+        ":single_process_encoded_image_data_injector_unittest",
+        ":video_dumping_test",
+        ":video_frame_tracking_id_injector_unittest",
+      ]
+    }
+  }
+}
+
+rtc_library("video_dumping") {
+  testonly = true
+  sources = [
+    "video_dumping.cc",
+    "video_dumping.h",
+  ]
+  deps = [
+    "../../../..:video_test_support",
+    "../../../../../api/test/video:video_frame_writer",
+    "../../../../../api/video:video_frame",
+    "../../../../../rtc_base:logging",
+    "../../../../../system_wrappers",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("encoded_image_data_injector_api") {
+  testonly = true
+  sources = [ "encoded_image_data_injector.h" ]
+
+  deps = [ "../../../../../api/video:encoded_image" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("single_process_encoded_image_data_injector") {
+  testonly = true
+  sources = [
+    "single_process_encoded_image_data_injector.cc",
+    "single_process_encoded_image_data_injector.h",
+  ]
+
+  deps = [
+    ":encoded_image_data_injector_api",
+    "../../../../../api/video:encoded_image",
+    "../../../../../rtc_base:checks",
+    "../../../../../rtc_base/synchronization:mutex",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+}
+
+rtc_library("video_frame_tracking_id_injector") {
+  testonly = true
+  sources = [
+    "video_frame_tracking_id_injector.cc",
+    "video_frame_tracking_id_injector.h",
+  ]
+
+  deps = [
+    ":encoded_image_data_injector_api",
+    "../../../../../api/video:encoded_image",
+    "../../../../../rtc_base:checks",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+}
+
+rtc_library("simulcast_dummy_buffer_helper") {
+  testonly = true
+  sources = [
+    "simulcast_dummy_buffer_helper.cc",
+    "simulcast_dummy_buffer_helper.h",
+  ]
+  deps = [ "../../../../../api/video:video_frame" ]
+}
+
+rtc_library("quality_analyzing_video_decoder") {
+  testonly = true
+  sources = [
+    "quality_analyzing_video_decoder.cc",
+    "quality_analyzing_video_decoder.h",
+  ]
+  deps = [
+    ":encoded_image_data_injector_api",
+    ":simulcast_dummy_buffer_helper",
+    "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/video:encoded_image",
+    "../../../../../api/video:video_frame",
+    "../../../../../api/video_codecs:video_codecs_api",
+    "../../../../../modules/video_coding:video_codec_interface",
+    "../../../../../rtc_base:logging",
+    "../../../../../rtc_base/synchronization:mutex",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("quality_analyzing_video_encoder") {
+  testonly = true
+  sources = [
+    "quality_analyzing_video_encoder.cc",
+    "quality_analyzing_video_encoder.h",
+  ]
+  deps = [
+    ":encoded_image_data_injector_api",
+    "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/test/pclf:media_configuration",
+    "../../../../../api/video:video_frame",
+    "../../../../../api/video_codecs:video_codecs_api",
+    "../../../../../modules/video_coding:video_codec_interface",
+    "../../../../../modules/video_coding/svc:scalability_mode_util",
+    "../../../../../rtc_base:logging",
+    "../../../../../rtc_base/synchronization:mutex",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("analyzing_video_sinks_helper") {
+  testonly = true
+  sources = [
+    "analyzing_video_sinks_helper.cc",
+    "analyzing_video_sinks_helper.h",
+  ]
+  deps = [
+    "../../../../../api/test/pclf:media_configuration",
+    "../../../../../api/test/video:video_frame_writer",
+    "../../../../../rtc_base:macromagic",
+    "../../../../../rtc_base/synchronization:mutex",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("example_video_quality_analyzer") {
+  testonly = true
+  sources = [
+    "example_video_quality_analyzer.cc",
+    "example_video_quality_analyzer.h",
+  ]
+
+  deps = [
+    "../../../../../api:array_view",
+    "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/video:encoded_image",
+    "../../../../../api/video:video_frame",
+    "../../../../../rtc_base:logging",
+    "../../../../../rtc_base/synchronization:mutex",
+  ]
+}
+
+# This target contains implementation details of DefaultVideoQualityAnalyzer,
+# so headers exported by it shouldn't be used in other places.
+rtc_library("default_video_quality_analyzer_internal") {
+  visibility = [
+    ":default_video_quality_analyzer",
+    ":default_video_quality_analyzer_frames_comparator_test",
+    ":default_video_quality_analyzer_stream_state_test",
+    ":names_collection_test",
+    ":video_analyzer",
+  ]
+
+  testonly = true
+  sources = [
+    "default_video_quality_analyzer_cpu_measurer.cc",
+    "default_video_quality_analyzer_cpu_measurer.h",
+    "default_video_quality_analyzer_frame_in_flight.cc",
+    "default_video_quality_analyzer_frame_in_flight.h",
+    "default_video_quality_analyzer_frames_comparator.cc",
+    "default_video_quality_analyzer_frames_comparator.h",
+    "default_video_quality_analyzer_internal_shared_objects.cc",
+    "default_video_quality_analyzer_internal_shared_objects.h",
+    "default_video_quality_analyzer_stream_state.cc",
+    "default_video_quality_analyzer_stream_state.h",
+    "names_collection.cc",
+    "names_collection.h",
+  ]
+
+  deps = [
+    ":default_video_quality_analyzer_shared",
+    ":multi_reader_queue",
+    "../..:metric_metadata_keys",
+    "../../../../../api:array_view",
+    "../../../../../api:scoped_refptr",
+    "../../../../../api/numerics",
+    "../../../../../api/units:data_size",
+    "../../../../../api/units:timestamp",
+    "../../../../../api/video:video_frame",
+    "../../../../../api/video:video_frame_type",
+    "../../../../../common_video",
+    "../../../../../rtc_base:checks",
+    "../../../../../rtc_base:platform_thread",
+    "../../../../../rtc_base:rtc_base_tests_utils",
+    "../../../../../rtc_base:rtc_event",
+    "../../../../../rtc_base:stringutils",
+    "../../../../../rtc_base:timeutils",
+    "../../../../../rtc_base/synchronization:mutex",
+    "../../../../../rtc_tools:video_quality_analysis",
+    "../../../../../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings:strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("multi_reader_queue") {
+  testonly = true
+  sources = [ "multi_reader_queue.h" ]
+  deps = [ "../../../../../rtc_base:checks" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("video_quality_metrics_reporter") {
+  testonly = true
+  sources = [
+    "video_quality_metrics_reporter.cc",
+    "video_quality_metrics_reporter.h",
+  ]
+  deps = [
+    "../..:metric_metadata_keys",
+    "../../../../../api:peer_connection_quality_test_fixture_api",
+    "../../../../../api:rtc_stats_api",
+    "../../../../../api:track_id_stream_info_map",
+    "../../../../../api/numerics",
+    "../../../../../api/test/metrics:metric",
+    "../../../../../api/test/metrics:metrics_logger",
+    "../../../../../api/units:data_rate",
+    "../../../../../api/units:data_size",
+    "../../../../../api/units:time_delta",
+    "../../../../../api/units:timestamp",
+    "../../../../../rtc_base:checks",
+    "../../../../../rtc_base/synchronization:mutex",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("default_video_quality_analyzer") {
+  testonly = true
+  sources = [
+    "default_video_quality_analyzer.cc",
+    "default_video_quality_analyzer.h",
+  ]
+
+  deps = [
+    ":default_video_quality_analyzer_internal",
+    ":default_video_quality_analyzer_shared",
+    "../..:metric_metadata_keys",
+    "../../../../../api:array_view",
+    "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/numerics",
+    "../../../../../api/test/metrics:metric",
+    "../../../../../api/test/metrics:metrics_logger",
+    "../../../../../api/units:data_size",
+    "../../../../../api/units:time_delta",
+    "../../../../../api/units:timestamp",
+    "../../../../../api/video:encoded_image",
+    "../../../../../api/video:video_frame",
+    "../../../../../rtc_base:checks",
+    "../../../../../rtc_base:logging",
+    "../../../../../rtc_base:macromagic",
+    "../../../../../rtc_base:stringutils",
+    "../../../../../rtc_base/synchronization:mutex",
+    "../../../../../system_wrappers",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("default_video_quality_analyzer_shared") {
+  testonly = true
+  sources = [
+    "default_video_quality_analyzer_shared_objects.cc",
+    "default_video_quality_analyzer_shared_objects.h",
+  ]
+
+  deps = [
+    "../../../../../api/numerics",
+    "../../../../../api/units:timestamp",
+    "../../../../../rtc_base:checks",
+    "../../../../../rtc_base:stringutils",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("analyzing_video_sink") {
+  testonly = true
+  sources = [
+    "analyzing_video_sink.cc",
+    "analyzing_video_sink.h",
+  ]
+  deps = [
+    ":analyzing_video_sinks_helper",
+    ":simulcast_dummy_buffer_helper",
+    ":video_dumping",
+    "../../../..:fixed_fps_video_frame_writer_adapter",
+    "../../../..:test_renderer",
+    "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/numerics",
+    "../../../../../api/test/pclf:media_configuration",
+    "../../../../../api/test/video:video_frame_writer",
+    "../../../../../api/units:timestamp",
+    "../../../../../api/video:video_frame",
+    "../../../../../rtc_base:checks",
+    "../../../../../rtc_base:logging",
+    "../../../../../rtc_base:macromagic",
+    "../../../../../rtc_base/synchronization:mutex",
+    "../../../../../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory:memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("video_quality_analyzer_injection_helper") {
+  testonly = true
+  sources = [
+    "video_quality_analyzer_injection_helper.cc",
+    "video_quality_analyzer_injection_helper.h",
+  ]
+  deps = [
+    ":analyzing_video_sink",
+    ":analyzing_video_sinks_helper",
+    ":encoded_image_data_injector_api",
+    ":quality_analyzing_video_decoder",
+    ":quality_analyzing_video_encoder",
+    ":simulcast_dummy_buffer_helper",
+    ":video_dumping",
+    "../../../..:fixed_fps_video_frame_writer_adapter",
+    "../../../..:test_renderer",
+    "../../../..:video_test_common",
+    "../../../..:video_test_support",
+    "../../../../../api:array_view",
+    "../../../../../api:stats_observer_interface",
+    "../../../../../api:video_quality_analyzer_api",
+    "../../../../../api/test/pclf:media_configuration",
+    "../../../../../api/video:video_frame",
+    "../../../../../api/video_codecs:video_codecs_api",
+    "../../../../../rtc_base:checks",
+    "../../../../../rtc_base:logging",
+    "../../../../../rtc_base:stringutils",
+    "../../../../../rtc_base/synchronization:mutex",
+    "../../../../../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("simulcast_dummy_buffer_helper_test") {
+    testonly = true
+    sources = [ "simulcast_dummy_buffer_helper_test.cc" ]
+    deps = [
+      ":simulcast_dummy_buffer_helper",
+      "../../../..:test_support",
+      "../../../../../api/video:video_frame",
+      "../../../../../rtc_base:random",
+    ]
+  }
+
+  rtc_library("analyzing_video_sink_test") {
+    testonly = true
+    sources = [ "analyzing_video_sink_test.cc" ]
+    deps = [
+      ":analyzing_video_sink",
+      ":example_video_quality_analyzer",
+      "../../../..:fileutils",
+      "../../../..:test_support",
+      "../../../..:video_test_support",
+      "../../../../../api:create_frame_generator",
+      "../../../../../api:frame_generator_api",
+      "../../../../../api:scoped_refptr",
+      "../../../../../api/test/pclf:media_configuration",
+      "../../../../../api/units:time_delta",
+      "../../../../../api/units:timestamp",
+      "../../../../../api/video:video_frame",
+      "../../../../../common_video",
+      "../../../../../rtc_base:timeutils",
+      "../../../../../system_wrappers",
+      "../../../../time_controller",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_library("analyzing_video_sinks_helper_test") {
+    testonly = true
+    sources = [ "analyzing_video_sinks_helper_test.cc" ]
+    deps = [
+      ":analyzing_video_sinks_helper",
+      "../../../..:test_support",
+      "../../../../../api/test/pclf:media_configuration",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  rtc_library("default_video_quality_analyzer_frames_comparator_test") {
+    testonly = true
+    sources = [ "default_video_quality_analyzer_frames_comparator_test.cc" ]
+    deps = [
+      ":default_video_quality_analyzer_internal",
+      ":default_video_quality_analyzer_shared",
+      "../../../..:test_support",
+      "../../../../../api:create_frame_generator",
+      "../../../../../api/units:timestamp",
+      "../../../../../rtc_base:stringutils",
+      "../../../../../system_wrappers",
+    ]
+  }
+
+  rtc_library("names_collection_test") {
+    testonly = true
+    sources = [ "names_collection_test.cc" ]
+    deps = [
+      ":default_video_quality_analyzer_internal",
+      "../../../..:test_support",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings:strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_library("multi_reader_queue_test") {
+    testonly = true
+    sources = [ "multi_reader_queue_test.cc" ]
+    deps = [
+      ":multi_reader_queue",
+      "../../../..:test_support",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  rtc_library("default_video_quality_analyzer_stream_state_test") {
+    testonly = true
+    sources = [ "default_video_quality_analyzer_stream_state_test.cc" ]
+    deps = [
+      ":default_video_quality_analyzer_internal",
+      "../../../..:test_support",
+      "../../../../../api/units:timestamp",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  rtc_library("default_video_quality_analyzer_test") {
+    testonly = true
+    sources = [ "default_video_quality_analyzer_test.cc" ]
+    deps = [
+      ":default_video_quality_analyzer",
+      ":default_video_quality_analyzer_shared",
+      "../../../..:test_support",
+      "../../../../../api:create_frame_generator",
+      "../../../../../api:rtp_packet_info",
+      "../../../../../api/test/metrics:global_metrics_logger_and_exporter",
+      "../../../../../api/video:encoded_image",
+      "../../../../../api/video:video_frame",
+      "../../../../../common_video",
+      "../../../../../rtc_base:stringutils",
+      "../../../../../rtc_tools:video_quality_analysis",
+      "../../../../../system_wrappers",
+    ]
+  }
+
+  rtc_library("default_video_quality_analyzer_metric_names_test") {
+    testonly = true
+    sources = [ "default_video_quality_analyzer_metric_names_test.cc" ]
+    deps = [
+      ":default_video_quality_analyzer",
+      "../../../..:test_support",
+      "../../../../../api:create_frame_generator",
+      "../../../../../api:rtp_packet_info",
+      "../../../../../api/test/metrics:metric",
+      "../../../../../api/test/metrics:metrics_logger",
+      "../../../../../api/test/metrics:stdout_metrics_exporter",
+      "../../../../../api/video:encoded_image",
+      "../../../../../api/video:video_frame",
+      "../../../../../common_video",
+      "../../../../../rtc_tools:video_quality_analysis",
+      "../../../../../system_wrappers",
+    ]
+  }
+
+  rtc_library("video_dumping_test") {
+    testonly = true
+    sources = [ "video_dumping_test.cc" ]
+    deps = [
+      ":video_dumping",
+      "../../../..:fileutils",
+      "../../../..:test_support",
+      "../../../..:video_test_support",
+      "../../../../../api:scoped_refptr",
+      "../../../../../api/video:video_frame",
+      "../../../../../rtc_base:random",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  rtc_library("single_process_encoded_image_data_injector_unittest") {
+    testonly = true
+    sources = [ "single_process_encoded_image_data_injector_unittest.cc" ]
+    deps = [
+      ":single_process_encoded_image_data_injector",
+      "../../../..:test_support",
+      "../../../../../api/video:encoded_image",
+      "../../../../../rtc_base:buffer",
+    ]
+  }
+
+  rtc_library("video_frame_tracking_id_injector_unittest") {
+    testonly = true
+    sources = [ "video_frame_tracking_id_injector_unittest.cc" ]
+    deps = [
+      ":video_frame_tracking_id_injector",
+      "../../../..:test_support",
+      "../../../../../api/video:encoded_image",
+      "../../../../../rtc_base:buffer",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink.cc
new file mode 100644
index 0000000000..fb221e6797
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink.cc
@@ -0,0 +1,220 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/analyzing_video_sink.h"
+
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/video/video_frame_writer.h"
+#include "api/units/timestamp.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h"
+#include "test/pc/e2e/analyzer/video/video_dumping.h"
+#include "test/testsupport/fixed_fps_video_frame_writer_adapter.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+AnalyzingVideoSink::AnalyzingVideoSink(absl::string_view peer_name,
+                                       Clock* clock,
+                                       VideoQualityAnalyzerInterface& analyzer,
+                                       AnalyzingVideoSinksHelper& sinks_helper,
+                                       const VideoSubscription& subscription,
+                                       bool report_infra_stats)
+    : peer_name_(peer_name),
+      report_infra_stats_(report_infra_stats),
+      clock_(clock),
+      analyzer_(&analyzer),
+      sinks_helper_(&sinks_helper),
+      subscription_(subscription) {}
+
+void AnalyzingVideoSink::UpdateSubscription(
+    const VideoSubscription& subscription) {
+  // For peers with changed resolutions we need to close current writers and
+  // open new ones. This is done by removing existing sinks, which will force
+  // creation of the new sinks when next frame will be received.
+  std::set<test::VideoFrameWriter*> writers_to_close;
+  {
+    MutexLock lock(&mutex_);
+    subscription_ = subscription;
+    for (auto it = stream_sinks_.cbegin(); it != stream_sinks_.cend();) {
+      absl::optional<VideoResolution> new_requested_resolution =
+          subscription_.GetResolutionForPeer(it->second.sender_peer_name);
+      if (!new_requested_resolution.has_value() ||
+          (*new_requested_resolution != it->second.resolution)) {
+        RTC_LOG(LS_INFO) << peer_name_ << ": Subscribed resolution for stream "
+                         << it->first << " from " << it->second.sender_peer_name
+                         << " was updated from "
+                         << it->second.resolution.ToString() << " to "
+                         << new_requested_resolution->ToString()
+                         << ". Repopulating all video sinks and recreating "
+                         << "requested video writers";
+        writers_to_close.insert(it->second.video_frame_writer);
+        it = stream_sinks_.erase(it);
+      } else {
+        ++it;
+      }
+    }
+  }
+  sinks_helper_->CloseAndRemoveVideoWriters(writers_to_close);
+}
+
+void AnalyzingVideoSink::OnFrame(const VideoFrame& frame) {
+  if (IsDummyFrame(frame)) {
+    // This is dummy frame, so we  don't need to process it further.
+    return;
+  }
+
+  if (frame.id() == VideoFrame::kNotSetId) {
+    // If frame ID is unknown we can't get required render resolution, so pass
+    // to the analyzer in the actual resolution of the frame.
+    AnalyzeFrame(frame);
+  } else {
+    std::string stream_label = analyzer_->GetStreamLabel(frame.id());
+    MutexLock lock(&mutex_);
+    Timestamp processing_started = clock_->CurrentTime();
+    SinksDescriptor* sinks_descriptor = PopulateSinks(stream_label);
+    RTC_CHECK(sinks_descriptor != nullptr);
+
+    VideoFrame scaled_frame =
+        ScaleVideoFrame(frame, sinks_descriptor->resolution);
+    AnalyzeFrame(scaled_frame);
+    for (auto& sink : sinks_descriptor->sinks) {
+      sink->OnFrame(scaled_frame);
+    }
+    Timestamp processing_finished = clock_->CurrentTime();
+
+    if (report_infra_stats_) {
+      stats_.analyzing_sink_processing_time_ms.AddSample(
+          (processing_finished - processing_started).ms<double>());
+    }
+  }
+}
+
+AnalyzingVideoSink::Stats AnalyzingVideoSink::stats() const {
+  MutexLock lock(&mutex_);
+  return stats_;
+}
+
+VideoFrame AnalyzingVideoSink::ScaleVideoFrame(
+    const VideoFrame& frame,
+    const VideoResolution& required_resolution) {
+  Timestamp processing_started = clock_->CurrentTime();
+  if (required_resolution.width() == static_cast<size_t>(frame.width()) &&
+      required_resolution.height() == static_cast<size_t>(frame.height())) {
+    if (report_infra_stats_) {
+      stats_.scaling_tims_ms.AddSample(
+          (clock_->CurrentTime() - processing_started).ms<double>());
+    }
+    return frame;
+  }
+
+  // We allow some difference in the aspect ration because when decoder
+  // downscales video stream it may round up some dimensions to make them even,
+  // ex: 960x540 -> 480x270 -> 240x136 instead of 240x135.
+  RTC_CHECK_LE(std::abs(static_cast<double>(required_resolution.width()) /
+                            required_resolution.height() -
+                        static_cast<double>(frame.width()) / frame.height()),
+               0.1)
+      << peer_name_
+      << ": Received frame has too different aspect ratio compared to "
+      << "requested video resolution: required resolution="
+      << required_resolution.ToString()
+      << "; actual resolution=" << frame.width() << "x" << frame.height();
+
+  rtc::scoped_refptr<I420Buffer> scaled_buffer(I420Buffer::Create(
+      required_resolution.width(), required_resolution.height()));
+  scaled_buffer->ScaleFrom(*frame.video_frame_buffer()->ToI420());
+
+  VideoFrame scaled_frame = frame;
+  scaled_frame.set_video_frame_buffer(scaled_buffer);
+  if (report_infra_stats_) {
+    stats_.scaling_tims_ms.AddSample(
+        (clock_->CurrentTime() - processing_started).ms<double>());
+  }
+  return scaled_frame;
+}
+
+void AnalyzingVideoSink::AnalyzeFrame(const VideoFrame& frame) {
+  VideoFrame frame_copy = frame;
+  frame_copy.set_video_frame_buffer(
+      I420Buffer::Copy(*frame.video_frame_buffer()->ToI420()));
+  analyzer_->OnFrameRendered(peer_name_, frame_copy);
+}
+
+AnalyzingVideoSink::SinksDescriptor* AnalyzingVideoSink::PopulateSinks(
+    absl::string_view stream_label) {
+  // Fast pass: sinks already exists.
+  auto sinks_it = stream_sinks_.find(std::string(stream_label));
+  if (sinks_it != stream_sinks_.end()) {
+    return &sinks_it->second;
+  }
+
+  // Slow pass: we need to create and save sinks
+  absl::optional<std::pair<std::string, VideoConfig>> peer_and_config =
+      sinks_helper_->GetPeerAndConfig(stream_label);
+  RTC_CHECK(peer_and_config.has_value())
+      << "No video config for stream " << stream_label;
+  const std::string& sender_peer_name = peer_and_config->first;
+  const VideoConfig& config = peer_and_config->second;
+
+  absl::optional<VideoResolution> resolution =
+      subscription_.GetResolutionForPeer(sender_peer_name);
+  if (!resolution.has_value()) {
+    RTC_LOG(LS_ERROR) << peer_name_ << " received stream " << stream_label
+                      << " from " << sender_peer_name
+                      << " for which they were not subscribed";
+    resolution = config.GetResolution();
+  }
+  if (!resolution->IsRegular()) {
+    RTC_LOG(LS_ERROR) << peer_name_ << " received stream " << stream_label
+                      << " from " << sender_peer_name
+                      << " for which resolution wasn't resolved";
+    resolution = config.GetResolution();
+  }
+
+  RTC_CHECK(resolution.has_value());
+
+  SinksDescriptor sinks_descriptor(sender_peer_name, *resolution);
+  if (config.output_dump_options.has_value()) {
+    std::unique_ptr<test::VideoFrameWriter> writer =
+        config.output_dump_options->CreateOutputDumpVideoFrameWriter(
+            stream_label, peer_name_, *resolution);
+    if (config.output_dump_use_fixed_framerate) {
+      writer = std::make_unique<test::FixedFpsVideoFrameWriterAdapter>(
+          resolution->fps(), clock_, std::move(writer));
+    }
+    sinks_descriptor.sinks.push_back(std::make_unique<VideoWriter>(
+        writer.get(), config.output_dump_options->sampling_modulo()));
+    sinks_descriptor.video_frame_writer =
+        sinks_helper_->AddVideoWriter(std::move(writer));
+  }
+  if (config.show_on_screen) {
+    sinks_descriptor.sinks.push_back(
+        absl::WrapUnique(test::VideoRenderer::Create(
+            (*config.stream_label + "-render").c_str(), resolution->width(),
+            resolution->height())));
+  }
+  return &stream_sinks_.emplace(stream_label, std::move(sinks_descriptor))
+              .first->second;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink.h
new file mode 100644
index 0000000000..1834bbe469
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink.h
@@ -0,0 +1,106 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINK_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINK_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/video/video_frame_writer.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+#include "test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// A sink to inject video quality analyzer as a sink into WebRTC.
+class AnalyzingVideoSink : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  struct Stats {
+    // Time required to scale video frame to the requested rendered resolution.
+    // Collected only for frames with ID set and iff `report_infra_stats` is
+    // true.
+    SamplesStatsCounter scaling_tims_ms;
+    // Time required to process single video frame. Collected only for frames
+    // with ID set and iff `report_infra_stats` is true.
+    SamplesStatsCounter analyzing_sink_processing_time_ms;
+  };
+
+  AnalyzingVideoSink(absl::string_view peer_name,
+                     Clock* clock,
+                     VideoQualityAnalyzerInterface& analyzer,
+                     AnalyzingVideoSinksHelper& sinks_helper,
+                     const VideoSubscription& subscription,
+                     bool report_infra_stats);
+
+  // Updates subscription used by this peer to render received video.
+  void UpdateSubscription(const VideoSubscription& subscription);
+
+  void OnFrame(const VideoFrame& frame) override;
+
+  Stats stats() const;
+
+ private:
+  struct SinksDescriptor {
+    SinksDescriptor(absl::string_view sender_peer_name,
+                    const VideoResolution& resolution)
+        : sender_peer_name(sender_peer_name), resolution(resolution) {}
+
+    // Required to be able to resolve resolutions on new subscription and
+    // understand if we need to recreate `video_frame_writer` and `sinks`.
+    std::string sender_peer_name;
+    // Resolution which was used to create `video_frame_writer` and `sinks`.
+    VideoResolution resolution;
+
+    // Is set if dumping of output video was requested;
+    test::VideoFrameWriter* video_frame_writer = nullptr;
+    std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
+  };
+
+  // Scales video frame to `required_resolution` if necessary. Crashes if video
+  // frame and `required_resolution` have different aspect ratio.
+  VideoFrame ScaleVideoFrame(const VideoFrame& frame,
+                             const VideoResolution& required_resolution)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Creates full copy of the frame to free any frame owned internal buffers
+  // and passes created copy to analyzer. Uses `I420Buffer` to represent
+  // frame content.
+  void AnalyzeFrame(const VideoFrame& frame);
+  // Populates sink for specified stream and caches them in `stream_sinks_`.
+  SinksDescriptor* PopulateSinks(absl::string_view stream_label)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  const std::string peer_name_;
+  const bool report_infra_stats_;
+  Clock* const clock_;
+  VideoQualityAnalyzerInterface* const analyzer_;
+  AnalyzingVideoSinksHelper* const sinks_helper_;
+
+  mutable Mutex mutex_;
+  VideoSubscription subscription_ RTC_GUARDED_BY(mutex_);
+  std::map<std::string, SinksDescriptor> stream_sinks_ RTC_GUARDED_BY(mutex_);
+  Stats stats_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINK_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink_test.cc
new file mode 100644
index 0000000000..6cd89551ea
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sink_test.cc
@@ -0,0 +1,598 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/analyzing_video_sink.h"
+
+#include <stdio.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/test/create_frame_generator.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/analyzer/video/example_video_quality_analyzer.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::Ge;
+using ::testing::Test;
+
+// Remove files and directories in a directory non-recursively.
+void CleanDir(absl::string_view dir, size_t expected_output_files_count) {
+  absl::optional<std::vector<std::string>> dir_content =
+      test::ReadDirectory(dir);
+  if (expected_output_files_count == 0) {
+    ASSERT_TRUE(!dir_content.has_value() || dir_content->empty())
+        << "Empty directory is expected";
+  } else {
+    ASSERT_TRUE(dir_content.has_value()) << "Test directory is empty!";
+    EXPECT_EQ(dir_content->size(), expected_output_files_count);
+    for (const auto& entry : *dir_content) {
+      if (test::DirExists(entry)) {
+        EXPECT_TRUE(test::RemoveDir(entry))
+            << "Failed to remove sub directory: " << entry;
+      } else if (test::FileExists(entry)) {
+        EXPECT_TRUE(test::RemoveFile(entry))
+            << "Failed to remove file: " << entry;
+      } else {
+        FAIL() << "Can't remove unknown file type: " << entry;
+      }
+    }
+  }
+  EXPECT_TRUE(test::RemoveDir(dir)) << "Failed to remove directory: " << dir;
+}
+
+VideoFrame CreateFrame(test::FrameGeneratorInterface& frame_generator) {
+  test::FrameGeneratorInterface::VideoFrameData frame_data =
+      frame_generator.NextFrame();
+  return VideoFrame::Builder()
+      .set_video_frame_buffer(frame_data.buffer)
+      .set_update_rect(frame_data.update_rect)
+      .build();
+}
+
+std::unique_ptr<test::FrameGeneratorInterface> CreateFrameGenerator(
+    size_t width,
+    size_t height) {
+  return test::CreateSquareFrameGenerator(width, height,
+                                          /*type=*/absl::nullopt,
+                                          /*num_squares=*/absl::nullopt);
+}
+
+void AssertFrameIdsAre(const std::string& filename,
+                       std::vector<std::string> expected_ids) {
+  FILE* file = fopen(filename.c_str(), "r");
+  ASSERT_TRUE(file != nullptr) << "Failed to open frame ids file: " << filename;
+  std::vector<std::string> actual_ids;
+  char buffer[8];
+  while (fgets(buffer, sizeof buffer, file) != nullptr) {
+    std::string current_id(buffer);
+    EXPECT_GE(current_id.size(), 2lu)
+        << "Found invalid frame id: [" << current_id << "]";
+    if (current_id.size() < 2) {
+      continue;
+    }
+    // Trim "\n" at the end.
+    actual_ids.push_back(current_id.substr(0, current_id.size() - 1));
+  }
+  fclose(file);
+  EXPECT_THAT(actual_ids, ElementsAreArray(expected_ids));
+}
+
+class AnalyzingVideoSinkTest : public Test {
+ protected:
+  ~AnalyzingVideoSinkTest() override = default;
+
+  void SetUp() override {
+    // Create an empty temporary directory for this test.
+    test_directory_ = test::JoinFilename(
+        test::OutputPath(),
+        "TestDir_AnalyzingVideoSinkTest_" +
+            std::string(
+                testing::UnitTest::GetInstance()->current_test_info()->name()));
+    test::CreateDir(test_directory_);
+  }
+
+  void TearDown() override {
+    CleanDir(test_directory_, expected_output_files_count_);
+  }
+
+  void ExpectOutputFilesCount(size_t count) {
+    expected_output_files_count_ = count;
+  }
+
+  std::string test_directory_;
+  size_t expected_output_files_count_ = 0;
+};
+
+TEST_F(AnalyzingVideoSinkTest, VideoFramesAreDumpedCorrectly) {
+  VideoSubscription subscription;
+  subscription.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/640, /*height=*/360, /*fps=*/30));
+  VideoConfig video_config("alice_video", /*width=*/1280, /*height=*/720,
+                           /*fps=*/30);
+  video_config.output_dump_options = VideoDumpOptions(test_directory_);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/1280, /*height=*/720);
+  VideoFrame frame = CreateFrame(*frame_generator);
+  frame.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame));
+
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                            subscription, /*report_infra_stats=*/false);
+    sink.OnFrame(frame);
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(1)));
+
+  auto frame_reader = test::CreateY4mFrameReader(
+      test::JoinFilename(test_directory_, "alice_video_bob_640x360_30.y4m"));
+  EXPECT_THAT(frame_reader->num_frames(), Eq(1));
+  rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+  rtc::scoped_refptr<I420BufferInterface> expected_frame =
+      frame.video_frame_buffer()->ToI420();
+  double psnr = I420PSNR(*expected_frame, *actual_frame);
+  double ssim = I420SSIM(*expected_frame, *actual_frame);
+  // Actual should be downscaled version of expected.
+  EXPECT_GT(ssim, 0.98);
+  EXPECT_GT(psnr, 38);
+
+  ExpectOutputFilesCount(1);
+}
+
+TEST_F(AnalyzingVideoSinkTest,
+       FallbackOnConfigResolutionIfNoSubscriptionProvided) {
+  VideoSubscription subscription;
+  VideoConfig video_config("alice_video", /*width=*/320, /*height=*/240,
+                           /*fps=*/30);
+  video_config.output_dump_options = VideoDumpOptions(test_directory_);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/320, /*height=*/240);
+  VideoFrame frame = CreateFrame(*frame_generator);
+  frame.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame));
+
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                            subscription, /*report_infra_stats=*/false);
+    sink.OnFrame(frame);
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(1)));
+
+  auto frame_reader = test::CreateY4mFrameReader(
+      test::JoinFilename(test_directory_, "alice_video_bob_320x240_30.y4m"));
+  EXPECT_THAT(frame_reader->num_frames(), Eq(1));
+  rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+  rtc::scoped_refptr<I420BufferInterface> expected_frame =
+      frame.video_frame_buffer()->ToI420();
+  double psnr = I420PSNR(*expected_frame, *actual_frame);
+  double ssim = I420SSIM(*expected_frame, *actual_frame);
+  // Frames should be equal.
+  EXPECT_DOUBLE_EQ(ssim, 1.00);
+  EXPECT_DOUBLE_EQ(psnr, 48);
+
+  ExpectOutputFilesCount(1);
+}
+
+TEST_F(AnalyzingVideoSinkTest,
+       FallbackOnConfigResolutionIfNoSubscriptionIsNotResolved) {
+  VideoSubscription subscription;
+  subscription.SubscribeToAllPeers(
+      VideoResolution(VideoResolution::Spec::kMaxFromSender));
+  VideoConfig video_config("alice_video", /*width=*/320, /*height=*/240,
+                           /*fps=*/30);
+  video_config.output_dump_options = VideoDumpOptions(test_directory_);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/320, /*height=*/240);
+  VideoFrame frame = CreateFrame(*frame_generator);
+  frame.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame));
+
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                            subscription, /*report_infra_stats=*/false);
+    sink.OnFrame(frame);
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(1)));
+
+  auto frame_reader = test::CreateY4mFrameReader(
+      test::JoinFilename(test_directory_, "alice_video_bob_320x240_30.y4m"));
+  EXPECT_THAT(frame_reader->num_frames(), Eq(1));
+  rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+  rtc::scoped_refptr<I420BufferInterface> expected_frame =
+      frame.video_frame_buffer()->ToI420();
+  double psnr = I420PSNR(*expected_frame, *actual_frame);
+  double ssim = I420SSIM(*expected_frame, *actual_frame);
+  // Frames should be equal.
+  EXPECT_DOUBLE_EQ(ssim, 1.00);
+  EXPECT_DOUBLE_EQ(psnr, 48);
+
+  ExpectOutputFilesCount(1);
+}
+
+TEST_F(AnalyzingVideoSinkTest,
+       VideoFramesAreDumpedCorrectlyWhenSubscriptionChanged) {
+  VideoSubscription subscription_before;
+  subscription_before.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/1280, /*height=*/720, /*fps=*/30));
+  VideoSubscription subscription_after;
+  subscription_after.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/640, /*height=*/360, /*fps=*/30));
+  VideoConfig video_config("alice_video", /*width=*/1280, /*height=*/720,
+                           /*fps=*/30);
+  video_config.output_dump_options = VideoDumpOptions(test_directory_);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/1280, /*height=*/720);
+  VideoFrame frame_before = CreateFrame(*frame_generator);
+  frame_before.set_id(
+      analyzer.OnFrameCaptured("alice", "alice_video", frame_before));
+  VideoFrame frame_after = CreateFrame(*frame_generator);
+  frame_after.set_id(
+      analyzer.OnFrameCaptured("alice", "alice_video", frame_after));
+
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                            subscription_before, /*report_infra_stats=*/false);
+    sink.OnFrame(frame_before);
+
+    sink.UpdateSubscription(subscription_after);
+    sink.OnFrame(frame_after);
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(2)));
+
+  {
+    auto frame_reader = test::CreateY4mFrameReader(
+        test::JoinFilename(test_directory_, "alice_video_bob_1280x720_30.y4m"));
+    EXPECT_THAT(frame_reader->num_frames(), Eq(1));
+    rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+    rtc::scoped_refptr<I420BufferInterface> expected_frame =
+        frame_before.video_frame_buffer()->ToI420();
+    double psnr = I420PSNR(*expected_frame, *actual_frame);
+    double ssim = I420SSIM(*expected_frame, *actual_frame);
+    // Frames should be equal.
+    EXPECT_DOUBLE_EQ(ssim, 1.00);
+    EXPECT_DOUBLE_EQ(psnr, 48);
+  }
+  {
+    auto frame_reader = test::CreateY4mFrameReader(
+        test::JoinFilename(test_directory_, "alice_video_bob_640x360_30.y4m"));
+    EXPECT_THAT(frame_reader->num_frames(), Eq(1));
+    rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+    rtc::scoped_refptr<I420BufferInterface> expected_frame =
+        frame_after.video_frame_buffer()->ToI420();
+    double psnr = I420PSNR(*expected_frame, *actual_frame);
+    double ssim = I420SSIM(*expected_frame, *actual_frame);
+    // Actual should be downscaled version of expected.
+    EXPECT_GT(ssim, 0.98);
+    EXPECT_GT(psnr, 38);
+  }
+
+  ExpectOutputFilesCount(2);
+}
+
+TEST_F(AnalyzingVideoSinkTest,
+       VideoFramesAreDumpedCorrectlyWhenSubscriptionChangedOnTheSameOne) {
+  VideoSubscription subscription_before;
+  subscription_before.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/640, /*height=*/360, /*fps=*/30));
+  VideoSubscription subscription_after;
+  subscription_after.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/640, /*height=*/360, /*fps=*/30));
+  VideoConfig video_config("alice_video", /*width=*/640, /*height=*/360,
+                           /*fps=*/30);
+  video_config.output_dump_options = VideoDumpOptions(test_directory_);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/640, /*height=*/360);
+  VideoFrame frame_before = CreateFrame(*frame_generator);
+  frame_before.set_id(
+      analyzer.OnFrameCaptured("alice", "alice_video", frame_before));
+  VideoFrame frame_after = CreateFrame(*frame_generator);
+  frame_after.set_id(
+      analyzer.OnFrameCaptured("alice", "alice_video", frame_after));
+
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                            subscription_before, /*report_infra_stats=*/false);
+    sink.OnFrame(frame_before);
+
+    sink.UpdateSubscription(subscription_after);
+    sink.OnFrame(frame_after);
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(2)));
+
+  {
+    auto frame_reader = test::CreateY4mFrameReader(
+        test::JoinFilename(test_directory_, "alice_video_bob_640x360_30.y4m"));
+    EXPECT_THAT(frame_reader->num_frames(), Eq(2));
+    // Read the first frame.
+    rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+    rtc::scoped_refptr<I420BufferInterface> expected_frame =
+        frame_before.video_frame_buffer()->ToI420();
+    // Frames should be equal.
+    EXPECT_DOUBLE_EQ(I420SSIM(*expected_frame, *actual_frame), 1.00);
+    EXPECT_DOUBLE_EQ(I420PSNR(*expected_frame, *actual_frame), 48);
+    // Read the second frame.
+    actual_frame = frame_reader->PullFrame();
+    expected_frame = frame_after.video_frame_buffer()->ToI420();
+    // Frames should be equal.
+    EXPECT_DOUBLE_EQ(I420SSIM(*expected_frame, *actual_frame), 1.00);
+    EXPECT_DOUBLE_EQ(I420PSNR(*expected_frame, *actual_frame), 48);
+  }
+
+  ExpectOutputFilesCount(1);
+}
+
+TEST_F(AnalyzingVideoSinkTest, SmallDiviationsInAspectRationAreAllowed) {
+  VideoSubscription subscription;
+  subscription.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/480, /*height=*/270, /*fps=*/30));
+  VideoConfig video_config("alice_video", /*width=*/480, /*height=*/270,
+                           /*fps=*/30);
+  video_config.output_dump_options = VideoDumpOptions(test_directory_);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  // Generator produces downscaled frames with a bit different aspect ration.
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/240, /*height=*/136);
+  VideoFrame frame = CreateFrame(*frame_generator);
+  frame.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame));
+
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                            subscription, /*report_infra_stats=*/false);
+    sink.OnFrame(frame);
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(1)));
+
+  {
+    auto frame_reader = test::CreateY4mFrameReader(
+        test::JoinFilename(test_directory_, "alice_video_bob_480x270_30.y4m"));
+    EXPECT_THAT(frame_reader->num_frames(), Eq(1));
+    // Read the first frame.
+    rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+    rtc::scoped_refptr<I420BufferInterface> expected_frame =
+        frame.video_frame_buffer()->ToI420();
+    // Actual frame is upscaled version of the expected. But because rendered
+    // resolution is equal to the actual frame size we need to upscale expected
+    // during comparison and then they have to be the same.
+    EXPECT_DOUBLE_EQ(I420SSIM(*actual_frame, *expected_frame), 1);
+    EXPECT_DOUBLE_EQ(I420PSNR(*actual_frame, *expected_frame), 48);
+  }
+
+  ExpectOutputFilesCount(1);
+}
+
+TEST_F(AnalyzingVideoSinkTest, VideoFramesIdsAreDumpedWhenRequested) {
+  VideoSubscription subscription;
+  subscription.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/320, /*height=*/240, /*fps=*/30));
+  VideoConfig video_config("alice_video", /*width=*/320, /*height=*/240,
+                           /*fps=*/30);
+  video_config.output_dump_options =
+      VideoDumpOptions(test_directory_, /*export_frame_ids=*/true);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/320, /*height=*/240);
+
+  std::vector<std::string> expected_frame_ids;
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                            subscription, /*report_infra_stats=*/false);
+    for (int i = 0; i < 10; ++i) {
+      VideoFrame frame = CreateFrame(*frame_generator);
+      frame.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame));
+      expected_frame_ids.push_back(std::to_string(frame.id()));
+      sink.OnFrame(frame);
+    }
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(10)));
+
+  AssertFrameIdsAre(
+      test::JoinFilename(test_directory_,
+                         "alice_video_bob_320x240_30.frame_ids.txt"),
+      expected_frame_ids);
+
+  ExpectOutputFilesCount(2);
+}
+
+TEST_F(AnalyzingVideoSinkTest,
+       VideoFramesAndIdsAreDumpedWithFixedFpsWhenRequested) {
+  GlobalSimulatedTimeController simulated_time(Timestamp::Seconds(100000));
+
+  VideoSubscription subscription;
+  subscription.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/320, /*height=*/240, /*fps=*/10));
+  VideoConfig video_config("alice_video", /*width=*/320, /*height=*/240,
+                           /*fps=*/10);
+  video_config.output_dump_options =
+      VideoDumpOptions(test_directory_, /*export_frame_ids=*/true);
+  video_config.output_dump_use_fixed_framerate = true;
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/320, /*height=*/240);
+
+  VideoFrame frame1 = CreateFrame(*frame_generator);
+  frame1.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame1));
+  VideoFrame frame2 = CreateFrame(*frame_generator);
+  frame2.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame2));
+
+  {
+    // `helper` and `sink` has to be destroyed so all frames will be written
+    // to the disk.
+    AnalyzingVideoSinksHelper helper;
+    helper.AddConfig("alice", video_config);
+    AnalyzingVideoSink sink("bob", simulated_time.GetClock(), analyzer, helper,
+                            subscription, /*report_infra_stats=*/false);
+    sink.OnFrame(frame1);
+    // Advance almost 1 second, so the first frame has to be repeated 9 time
+    // more.
+    simulated_time.AdvanceTime(TimeDelta::Millis(990));
+    sink.OnFrame(frame2);
+    simulated_time.AdvanceTime(TimeDelta::Millis(100));
+  }
+
+  EXPECT_THAT(analyzer.frames_rendered(), Eq(static_cast<uint64_t>(2)));
+
+  auto frame_reader = test::CreateY4mFrameReader(
+      test::JoinFilename(test_directory_, "alice_video_bob_320x240_10.y4m"));
+  EXPECT_THAT(frame_reader->num_frames(), Eq(11));
+  for (int i = 0; i < 10; ++i) {
+    rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+    rtc::scoped_refptr<I420BufferInterface> expected_frame =
+        frame1.video_frame_buffer()->ToI420();
+    double psnr = I420PSNR(*expected_frame, *actual_frame);
+    double ssim = I420SSIM(*expected_frame, *actual_frame);
+    // Frames should be equal.
+    EXPECT_DOUBLE_EQ(ssim, 1.00);
+    EXPECT_DOUBLE_EQ(psnr, 48);
+  }
+  rtc::scoped_refptr<I420Buffer> actual_frame = frame_reader->PullFrame();
+  rtc::scoped_refptr<I420BufferInterface> expected_frame =
+      frame2.video_frame_buffer()->ToI420();
+  double psnr = I420PSNR(*expected_frame, *actual_frame);
+  double ssim = I420SSIM(*expected_frame, *actual_frame);
+  // Frames should be equal.
+  EXPECT_DOUBLE_EQ(ssim, 1.00);
+  EXPECT_DOUBLE_EQ(psnr, 48);
+
+  AssertFrameIdsAre(
+      test::JoinFilename(test_directory_,
+                         "alice_video_bob_320x240_10.frame_ids.txt"),
+      {std::to_string(frame1.id()), std::to_string(frame1.id()),
+       std::to_string(frame1.id()), std::to_string(frame1.id()),
+       std::to_string(frame1.id()), std::to_string(frame1.id()),
+       std::to_string(frame1.id()), std::to_string(frame1.id()),
+       std::to_string(frame1.id()), std::to_string(frame1.id()),
+       std::to_string(frame2.id())});
+
+  ExpectOutputFilesCount(2);
+}
+
+TEST_F(AnalyzingVideoSinkTest, InfraMetricsCollectedWhenRequested) {
+  VideoSubscription subscription;
+  subscription.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/1280, /*height=*/720, /*fps=*/30));
+  VideoConfig video_config("alice_video", /*width=*/640, /*height=*/360,
+                           /*fps=*/30);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/640, /*height=*/360);
+  VideoFrame frame = CreateFrame(*frame_generator);
+  frame.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame));
+
+  AnalyzingVideoSinksHelper helper;
+  helper.AddConfig("alice", video_config);
+  AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                          subscription, /*report_infra_stats=*/true);
+  sink.OnFrame(frame);
+
+  AnalyzingVideoSink::Stats stats = sink.stats();
+  EXPECT_THAT(stats.scaling_tims_ms.NumSamples(), Eq(1));
+  EXPECT_THAT(stats.scaling_tims_ms.GetAverage(), Ge(0));
+  EXPECT_THAT(stats.analyzing_sink_processing_time_ms.NumSamples(), Eq(1));
+  EXPECT_THAT(stats.analyzing_sink_processing_time_ms.GetAverage(),
+              Ge(stats.scaling_tims_ms.GetAverage()));
+
+  ExpectOutputFilesCount(0);
+}
+
+TEST_F(AnalyzingVideoSinkTest, InfraMetricsNotCollectedWhenNotRequested) {
+  VideoSubscription subscription;
+  subscription.SubscribeToPeer(
+      "alice", VideoResolution(/*width=*/1280, /*height=*/720, /*fps=*/30));
+  VideoConfig video_config("alice_video", /*width=*/640, /*height=*/360,
+                           /*fps=*/30);
+
+  ExampleVideoQualityAnalyzer analyzer;
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      CreateFrameGenerator(/*width=*/640, /*height=*/360);
+  VideoFrame frame = CreateFrame(*frame_generator);
+  frame.set_id(analyzer.OnFrameCaptured("alice", "alice_video", frame));
+
+  AnalyzingVideoSinksHelper helper;
+  helper.AddConfig("alice", video_config);
+  AnalyzingVideoSink sink("bob", Clock::GetRealTimeClock(), analyzer, helper,
+                          subscription, /*report_infra_stats=*/false);
+  sink.OnFrame(frame);
+
+  AnalyzingVideoSink::Stats stats = sink.stats();
+  EXPECT_THAT(stats.scaling_tims_ms.NumSamples(), Eq(0));
+  EXPECT_THAT(stats.analyzing_sink_processing_time_ms.NumSamples(), Eq(0));
+
+  ExpectOutputFilesCount(0);
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.cc
new file mode 100644
index 0000000000..70dc4b00b5
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.cc
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h"
+
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/video/video_frame_writer.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+void AnalyzingVideoSinksHelper::AddConfig(absl::string_view sender_peer_name,
+                                          VideoConfig config) {
+  MutexLock lock(&mutex_);
+  auto it = video_configs_.find(*config.stream_label);
+  if (it == video_configs_.end()) {
+    std::string stream_label = *config.stream_label;
+    video_configs_.emplace(
+        std::move(stream_label),
+        std::pair{std::string(sender_peer_name), std::move(config)});
+  } else {
+    it->second = std::pair{std::string(sender_peer_name), std::move(config)};
+  }
+}
+
+absl::optional<std::pair<std::string, VideoConfig>>
+AnalyzingVideoSinksHelper::GetPeerAndConfig(absl::string_view stream_label) {
+  MutexLock lock(&mutex_);
+  auto it = video_configs_.find(std::string(stream_label));
+  if (it == video_configs_.end()) {
+    return absl::nullopt;
+  }
+  return it->second;
+}
+
+void AnalyzingVideoSinksHelper::RemoveConfig(absl::string_view stream_label) {
+  MutexLock lock(&mutex_);
+  video_configs_.erase(std::string(stream_label));
+}
+
+test::VideoFrameWriter* AnalyzingVideoSinksHelper::AddVideoWriter(
+    std::unique_ptr<test::VideoFrameWriter> video_writer) {
+  MutexLock lock(&mutex_);
+  test::VideoFrameWriter* out = video_writer.get();
+  video_writers_.push_back(std::move(video_writer));
+  return out;
+}
+
+void AnalyzingVideoSinksHelper::CloseAndRemoveVideoWriters(
+    std::set<test::VideoFrameWriter*> writers_to_close) {
+  MutexLock lock(&mutex_);
+  for (auto it = video_writers_.cbegin(); it != video_writers_.cend();) {
+    if (writers_to_close.find(it->get()) != writers_to_close.end()) {
+      (*it)->Close();
+      it = video_writers_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+
+void AnalyzingVideoSinksHelper::Clear() {
+  MutexLock lock(&mutex_);
+  video_configs_.clear();
+  for (const auto& video_writer : video_writers_) {
+    video_writer->Close();
+  }
+  video_writers_.clear();
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h
new file mode 100644
index 0000000000..5f38c5a40e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINKS_HELPER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINKS_HELPER_H_
+
+#include <list>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/video/video_frame_writer.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Registry of known video configs and video writers.
+// This class is thread safe.
+class AnalyzingVideoSinksHelper {
+ public:
+  // Adds config in the registry. If config with such stream label was
+  // registered before, the new value will override the old one.
+  void AddConfig(absl::string_view sender_peer_name, VideoConfig config);
+  absl::optional<std::pair<std::string, VideoConfig>> GetPeerAndConfig(
+      absl::string_view stream_label);
+  // Removes video config for specified stream label. If there are no know video
+  // config for such stream label - does nothing.
+  void RemoveConfig(absl::string_view stream_label);
+
+  // Takes ownership of the provided video writer. All video writers owned by
+  // this class will be closed during `AnalyzingVideoSinksHelper` destruction
+  // and guaranteed to be alive either until explicitly removed by
+  // `CloseAndRemoveVideoWriters` or until `AnalyzingVideoSinksHelper` is
+  // destroyed.
+  //
+  // Returns pointer to the added writer. Ownership is maintained by
+  // `AnalyzingVideoSinksHelper`.
+  test::VideoFrameWriter* AddVideoWriter(
+      std::unique_ptr<test::VideoFrameWriter> video_writer);
+  // For each provided `writers_to_close`, if it is known, will close and
+  // destroy it, otherwise does nothing with it.
+  void CloseAndRemoveVideoWriters(
+      std::set<test::VideoFrameWriter*> writers_to_close);
+
+  // Removes all added configs and close and removes all added writers.
+  void Clear();
+
+ private:
+  Mutex mutex_;
+  std::map<std::string, std::pair<std::string, VideoConfig>> video_configs_
+      RTC_GUARDED_BY(mutex_);
+  std::list<std::unique_ptr<test::VideoFrameWriter>> video_writers_
+      RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_ANALYZING_VIDEO_SINKS_HELPER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper_test.cc
new file mode 100644
index 0000000000..1a820a5229
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/analyzing_video_sinks_helper_test.cc
@@ -0,0 +1,160 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h"
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/test/pclf/media_configuration.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::testing::Eq;
+
+// Asserts equality of the main fields of the video config. We don't compare
+// the full config due to the lack of equality definition for a lot of subtypes.
+void AssertConfigsAreEquals(const VideoConfig& actual,
+                            const VideoConfig& expected) {
+  EXPECT_THAT(actual.stream_label, Eq(expected.stream_label));
+  EXPECT_THAT(actual.width, Eq(expected.width));
+  EXPECT_THAT(actual.height, Eq(expected.height));
+  EXPECT_THAT(actual.fps, Eq(expected.fps));
+}
+
+TEST(AnalyzingVideoSinksHelperTest, ConfigsCanBeAdded) {
+  VideoConfig config("alice_video", /*width=*/1280, /*height=*/720, /*fps=*/30);
+
+  AnalyzingVideoSinksHelper helper;
+  helper.AddConfig("alice", config);
+
+  absl::optional<std::pair<std::string, VideoConfig>> registred_config =
+      helper.GetPeerAndConfig("alice_video");
+  ASSERT_TRUE(registred_config.has_value());
+  EXPECT_THAT(registred_config->first, Eq("alice"));
+  AssertConfigsAreEquals(registred_config->second, config);
+}
+
+TEST(AnalyzingVideoSinksHelperTest, AddingForExistingLabelWillOverwriteValue) {
+  VideoConfig config_before("alice_video", /*width=*/1280, /*height=*/720,
+                            /*fps=*/30);
+  VideoConfig config_after("alice_video", /*width=*/640, /*height=*/360,
+                           /*fps=*/15);
+
+  AnalyzingVideoSinksHelper helper;
+  helper.AddConfig("alice", config_before);
+
+  absl::optional<std::pair<std::string, VideoConfig>> registred_config =
+      helper.GetPeerAndConfig("alice_video");
+  ASSERT_TRUE(registred_config.has_value());
+  EXPECT_THAT(registred_config->first, Eq("alice"));
+  AssertConfigsAreEquals(registred_config->second, config_before);
+
+  helper.AddConfig("alice", config_after);
+
+  registred_config = helper.GetPeerAndConfig("alice_video");
+  ASSERT_TRUE(registred_config.has_value());
+  EXPECT_THAT(registred_config->first, Eq("alice"));
+  AssertConfigsAreEquals(registred_config->second, config_after);
+}
+
+TEST(AnalyzingVideoSinksHelperTest, ConfigsCanBeRemoved) {
+  VideoConfig config("alice_video", /*width=*/1280, /*height=*/720, /*fps=*/30);
+
+  AnalyzingVideoSinksHelper helper;
+  helper.AddConfig("alice", config);
+
+  ASSERT_TRUE(helper.GetPeerAndConfig("alice_video").has_value());
+
+  helper.RemoveConfig("alice_video");
+  ASSERT_FALSE(helper.GetPeerAndConfig("alice_video").has_value());
+}
+
+TEST(AnalyzingVideoSinksHelperTest, RemoveOfNonExistingConfigDontCrash) {
+  AnalyzingVideoSinksHelper helper;
+  helper.RemoveConfig("alice_video");
+}
+
+TEST(AnalyzingVideoSinksHelperTest, ClearRemovesAllConfigs) {
+  VideoConfig config1("alice_video", /*width=*/640, /*height=*/360, /*fps=*/30);
+  VideoConfig config2("bob_video", /*width=*/640, /*height=*/360, /*fps=*/30);
+
+  AnalyzingVideoSinksHelper helper;
+  helper.AddConfig("alice", config1);
+  helper.AddConfig("bob", config2);
+
+  ASSERT_TRUE(helper.GetPeerAndConfig("alice_video").has_value());
+  ASSERT_TRUE(helper.GetPeerAndConfig("bob_video").has_value());
+
+  helper.Clear();
+  ASSERT_FALSE(helper.GetPeerAndConfig("alice_video").has_value());
+  ASSERT_FALSE(helper.GetPeerAndConfig("bob_video").has_value());
+}
+
+struct TestVideoFrameWriterFactory {
+  int closed_writers_count = 0;
+  int deleted_writers_count = 0;
+
+  std::unique_ptr<test::VideoFrameWriter> CreateWriter() {
+    return std::make_unique<TestVideoFrameWriter>(this);
+  }
+
+ private:
+  class TestVideoFrameWriter : public test::VideoFrameWriter {
+   public:
+    explicit TestVideoFrameWriter(TestVideoFrameWriterFactory* factory)
+        : factory_(factory) {}
+    ~TestVideoFrameWriter() override { factory_->deleted_writers_count++; }
+
+    bool WriteFrame(const VideoFrame& frame) override { return true; }
+
+    void Close() override { factory_->closed_writers_count++; }
+
+   private:
+    TestVideoFrameWriterFactory* factory_;
+  };
+};
+
+TEST(AnalyzingVideoSinksHelperTest, RemovingWritersCloseAndDestroyAllOfThem) {
+  TestVideoFrameWriterFactory factory;
+
+  AnalyzingVideoSinksHelper helper;
+  test::VideoFrameWriter* writer1 =
+      helper.AddVideoWriter(factory.CreateWriter());
+  test::VideoFrameWriter* writer2 =
+      helper.AddVideoWriter(factory.CreateWriter());
+
+  helper.CloseAndRemoveVideoWriters({writer1, writer2});
+
+  EXPECT_THAT(factory.closed_writers_count, Eq(2));
+  EXPECT_THAT(factory.deleted_writers_count, Eq(2));
+}
+
+TEST(AnalyzingVideoSinksHelperTest, ClearCloseAndDestroyAllWriters) {
+  TestVideoFrameWriterFactory factory;
+
+  AnalyzingVideoSinksHelper helper;
+  helper.AddVideoWriter(factory.CreateWriter());
+  helper.AddVideoWriter(factory.CreateWriter());
+
+  helper.Clear();
+
+  EXPECT_THAT(factory.closed_writers_count, Eq(2));
+  EXPECT_THAT(factory.deleted_writers_count, Eq(2));
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
new file mode 100644
index 0000000000..59144589fc
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -0,0 +1,1228 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
+
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/metric.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+using ::webrtc::webrtc_pc_e2e::MetricMetadataKey;
+
+constexpr int kBitsInByte = 8;
+constexpr absl::string_view kSkipRenderedFrameReasonProcessed = "processed";
+constexpr absl::string_view kSkipRenderedFrameReasonRendered = "rendered";
+constexpr absl::string_view kSkipRenderedFrameReasonDropped =
+    "considered dropped";
+
+void LogFrameCounters(const std::string& name, const FrameCounters& counters) {
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Captured         : " << counters.captured;
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Pre encoded      : " << counters.pre_encoded;
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Encoded          : " << counters.encoded;
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Received         : " << counters.received;
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Decoded          : " << counters.decoded;
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Rendered         : " << counters.rendered;
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Dropped          : " << counters.dropped;
+  RTC_LOG(LS_INFO) << "[" << name
+                   << "] Failed to decode : " << counters.failed_to_decode;
+}
+
+void LogStreamInternalStats(const std::string& name,
+                            const StreamStats& stats,
+                            Timestamp start_time) {
+  for (const auto& entry : stats.dropped_by_phase) {
+    RTC_LOG(LS_INFO) << "[" << name << "] Dropped at " << ToString(entry.first)
+                     << ": " << entry.second;
+  }
+  Timestamp first_encoded_frame_time = Timestamp::PlusInfinity();
+  for (const StreamCodecInfo& encoder : stats.encoders) {
+    RTC_DCHECK(encoder.switched_on_at.IsFinite());
+    RTC_DCHECK(encoder.switched_from_at.IsFinite());
+    if (first_encoded_frame_time.IsInfinite()) {
+      first_encoded_frame_time = encoder.switched_on_at;
+    }
+    RTC_LOG(LS_INFO)
+        << "[" << name << "] Used encoder: \"" << encoder.codec_name
+        << "\" used from (frame_id=" << encoder.first_frame_id
+        << "; from_stream_start="
+        << (encoder.switched_on_at - stats.stream_started_time).ms()
+        << "ms, from_call_start=" << (encoder.switched_on_at - start_time).ms()
+        << "ms) until (frame_id=" << encoder.last_frame_id
+        << "; from_stream_start="
+        << (encoder.switched_from_at - stats.stream_started_time).ms()
+        << "ms, from_call_start="
+        << (encoder.switched_from_at - start_time).ms() << "ms)";
+  }
+  for (const StreamCodecInfo& decoder : stats.decoders) {
+    RTC_DCHECK(decoder.switched_on_at.IsFinite());
+    RTC_DCHECK(decoder.switched_from_at.IsFinite());
+    RTC_LOG(LS_INFO)
+        << "[" << name << "] Used decoder: \"" << decoder.codec_name
+        << "\" used from (frame_id=" << decoder.first_frame_id
+        << "; from_stream_start="
+        << (decoder.switched_on_at - stats.stream_started_time).ms()
+        << "ms, from_call_start=" << (decoder.switched_on_at - start_time).ms()
+        << "ms) until (frame_id=" << decoder.last_frame_id
+        << "; from_stream_start="
+        << (decoder.switched_from_at - stats.stream_started_time).ms()
+        << "ms, from_call_start="
+        << (decoder.switched_from_at - start_time).ms() << "ms)";
+  }
+}
+
+template <typename T>
+absl::optional<T> MaybeGetValue(const std::map<size_t, T>& map, size_t key) {
+  auto it = map.find(key);
+  if (it == map.end()) {
+    return absl::nullopt;
+  }
+  return it->second;
+}
+
+SamplesStatsCounter::StatsSample StatsSample(double value,
+                                             Timestamp sampling_time) {
+  return SamplesStatsCounter::StatsSample{value, sampling_time};
+}
+
+}  // namespace
+
+DefaultVideoQualityAnalyzer::DefaultVideoQualityAnalyzer(
+    webrtc::Clock* clock,
+    test::MetricsLogger* metrics_logger,
+    DefaultVideoQualityAnalyzerOptions options)
+    : options_(options),
+      clock_(clock),
+      metrics_logger_(metrics_logger),
+      frames_comparator_(clock, cpu_measurer_, options) {
+  RTC_CHECK(metrics_logger_);
+}
+
+DefaultVideoQualityAnalyzer::~DefaultVideoQualityAnalyzer() {
+  Stop();
+}
+
+void DefaultVideoQualityAnalyzer::Start(
+    std::string test_case_name,
+    rtc::ArrayView<const std::string> peer_names,
+    int max_threads_count) {
+  test_label_ = std::move(test_case_name);
+  frames_comparator_.Start(max_threads_count);
+  {
+    MutexLock lock(&mutex_);
+    peers_ = std::make_unique<NamesCollection>(peer_names);
+    RTC_CHECK(start_time_.IsMinusInfinity());
+
+    RTC_CHECK_EQ(state_, State::kNew)
+        << "DefaultVideoQualityAnalyzer is already started";
+    state_ = State::kActive;
+    start_time_ = Now();
+  }
+}
+
+uint16_t DefaultVideoQualityAnalyzer::OnFrameCaptured(
+    absl::string_view peer_name,
+    const std::string& stream_label,
+    const webrtc::VideoFrame& frame) {
+  // `next_frame_id` is atomic, so we needn't lock here.
+  Timestamp captured_time = Now();
+  Timestamp start_time = Timestamp::MinusInfinity();
+  size_t peer_index = -1;
+  size_t peers_count = -1;
+  size_t stream_index;
+  uint16_t frame_id = VideoFrame::kNotSetId;
+  {
+    MutexLock lock(&mutex_);
+    frame_id = GetNextFrameId();
+    RTC_CHECK_EQ(state_, State::kActive)
+        << "DefaultVideoQualityAnalyzer has to be started before use";
+    // Create a local copy of `start_time_`, peer's index and total peers count
+    // to access it without holding a `mutex_` during access to
+    // `frames_comparator_`.
+    start_time = start_time_;
+    peer_index = peers_->index(peer_name);
+    peers_count = peers_->size();
+    stream_index = streams_.AddIfAbsent(stream_label);
+  }
+  // Ensure stats for this stream exists.
+  frames_comparator_.EnsureStatsForStream(stream_index, peer_index, peers_count,
+                                          captured_time, start_time);
+  {
+    MutexLock lock(&mutex_);
+    stream_to_sender_[stream_index] = peer_index;
+    frame_counters_.captured++;
+    for (size_t i : peers_->GetAllIndexes()) {
+      if (i != peer_index || options_.enable_receive_own_stream) {
+        InternalStatsKey key(stream_index, peer_index, i);
+        stream_frame_counters_[key].captured++;
+      }
+    }
+
+    std::set<size_t> frame_receivers_indexes = peers_->GetPresentIndexes();
+    if (!options_.enable_receive_own_stream) {
+      frame_receivers_indexes.erase(peer_index);
+    }
+
+    auto state_it = stream_states_.find(stream_index);
+    if (state_it == stream_states_.end()) {
+      stream_states_.emplace(
+          stream_index,
+          StreamState(peer_index, frame_receivers_indexes, captured_time));
+    }
+    StreamState* state = &stream_states_.at(stream_index);
+    state->PushBack(frame_id);
+    // Update frames in flight info.
+    auto it = captured_frames_in_flight_.find(frame_id);
+    if (it != captured_frames_in_flight_.end()) {
+      // If we overflow uint16_t and hit previous frame id and this frame is
+      // still in flight, it means that this stream wasn't rendered for long
+      // time and we need to process existing frame as dropped.
+      for (size_t i : peers_->GetPresentIndexes()) {
+        if (i == peer_index && !options_.enable_receive_own_stream) {
+          continue;
+        }
+
+        uint16_t oldest_frame_id = state->PopFront(i);
+        RTC_DCHECK_EQ(frame_id, oldest_frame_id);
+        frame_counters_.dropped++;
+        InternalStatsKey key(stream_index, peer_index, i);
+        stream_frame_counters_.at(key).dropped++;
+
+        analyzer_stats_.frames_in_flight_left_count.AddSample(
+            StatsSample(captured_frames_in_flight_.size(), Now()));
+        frames_comparator_.AddComparison(
+            InternalStatsKey(stream_index, peer_index, i),
+            /*captured=*/absl::nullopt,
+            /*rendered=*/absl::nullopt, FrameComparisonType::kDroppedFrame,
+            it->second.GetStatsForPeer(i));
+      }
+
+      captured_frames_in_flight_.erase(it);
+    }
+    captured_frames_in_flight_.emplace(
+        frame_id, FrameInFlight(stream_index, frame, captured_time,
+                                std::move(frame_receivers_indexes)));
+    // Set frame id on local copy of the frame
+    captured_frames_in_flight_.at(frame_id).SetFrameId(frame_id);
+
+    // Update history stream<->frame mapping
+    for (auto it = stream_to_frame_id_history_.begin();
+         it != stream_to_frame_id_history_.end(); ++it) {
+      it->second.erase(frame_id);
+    }
+    stream_to_frame_id_history_[stream_index].insert(frame_id);
+    stream_to_frame_id_full_history_[stream_index].push_back(frame_id);
+
+    // If state has too many frames that are in flight => remove the oldest
+    // queued frame in order to avoid to use too much memory.
+    if (state->GetAliveFramesCount() >
+        options_.max_frames_in_flight_per_stream_count) {
+      uint16_t frame_id_to_remove = state->MarkNextAliveFrameAsDead();
+      auto it = captured_frames_in_flight_.find(frame_id_to_remove);
+      RTC_CHECK(it != captured_frames_in_flight_.end())
+          << "Frame with ID " << frame_id_to_remove
+          << " is expected to be in flight, but hasn't been found in "
+          << "|captured_frames_in_flight_|";
+      bool is_removed = it->second.RemoveFrame();
+      RTC_DCHECK(is_removed)
+          << "Invalid stream state: alive frame is removed already";
+    }
+    if (options_.report_infra_metrics) {
+      analyzer_stats_.on_frame_captured_processing_time_ms.AddSample(
+          (Now() - captured_time).ms<double>());
+    }
+  }
+  return frame_id;
+}
+
+void DefaultVideoQualityAnalyzer::OnFramePreEncode(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame) {
+  Timestamp processing_started = Now();
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "DefaultVideoQualityAnalyzer has to be started before use";
+
+  auto it = captured_frames_in_flight_.find(frame.id());
+  RTC_CHECK(it != captured_frames_in_flight_.end())
+      << "Frame id=" << frame.id() << " not found";
+  FrameInFlight& frame_in_flight = it->second;
+  frame_counters_.pre_encoded++;
+  size_t peer_index = peers_->index(peer_name);
+  for (size_t i : peers_->GetAllIndexes()) {
+    if (i != peer_index || options_.enable_receive_own_stream) {
+      InternalStatsKey key(frame_in_flight.stream(), peer_index, i);
+      stream_frame_counters_.at(key).pre_encoded++;
+    }
+  }
+  frame_in_flight.SetPreEncodeTime(Now());
+
+  if (options_.report_infra_metrics) {
+    analyzer_stats_.on_frame_pre_encode_processing_time_ms.AddSample(
+        (Now() - processing_started).ms<double>());
+  }
+}
+
+void DefaultVideoQualityAnalyzer::OnFrameEncoded(
+    absl::string_view peer_name,
+    uint16_t frame_id,
+    const webrtc::EncodedImage& encoded_image,
+    const EncoderStats& stats,
+    bool discarded) {
+  if (discarded)
+    return;
+
+  Timestamp processing_started = Now();
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "DefaultVideoQualityAnalyzer has to be started before use";
+
+  auto it = captured_frames_in_flight_.find(frame_id);
+  if (it == captured_frames_in_flight_.end()) {
+    RTC_LOG(LS_WARNING)
+        << "The encoding of video frame with id [" << frame_id << "] for peer ["
+        << peer_name << "] finished after all receivers rendered this frame or "
+        << "were removed. It can be OK for simulcast/SVC if higher quality "
+        << "stream is not required or the last receiver was unregistered "
+        << "between encoding of different layers, but it may indicate an ERROR "
+        << "for singlecast or if it happens often.";
+    return;
+  }
+  FrameInFlight& frame_in_flight = it->second;
+  // For SVC we can receive multiple encoded images for one frame, so to cover
+  // all cases we have to pick the last encode time.
+  if (!frame_in_flight.HasEncodedTime()) {
+    // Increase counters only when we meet this frame first time.
+    frame_counters_.encoded++;
+    size_t peer_index = peers_->index(peer_name);
+    for (size_t i : peers_->GetAllIndexes()) {
+      if (i != peer_index || options_.enable_receive_own_stream) {
+        InternalStatsKey key(frame_in_flight.stream(), peer_index, i);
+        stream_frame_counters_.at(key).encoded++;
+      }
+    }
+  }
+  Timestamp now = Now();
+  StreamCodecInfo used_encoder;
+  used_encoder.codec_name = stats.encoder_name;
+  used_encoder.first_frame_id = frame_id;
+  used_encoder.last_frame_id = frame_id;
+  used_encoder.switched_on_at = now;
+  used_encoder.switched_from_at = now;
+  frame_in_flight.OnFrameEncoded(
+      now, encoded_image._frameType, DataSize::Bytes(encoded_image.size()),
+      stats.target_encode_bitrate, encoded_image.SpatialIndex().value_or(0),
+      stats.qp, used_encoder);
+
+  if (options_.report_infra_metrics) {
+    analyzer_stats_.on_frame_encoded_processing_time_ms.AddSample(
+        (Now() - processing_started).ms<double>());
+  }
+}
+
+void DefaultVideoQualityAnalyzer::OnFrameDropped(
+    absl::string_view peer_name,
+    webrtc::EncodedImageCallback::DropReason reason) {
+  // Here we do nothing, because we will see this drop on renderer side.
+}
+
+void DefaultVideoQualityAnalyzer::OnFramePreDecode(
+    absl::string_view peer_name,
+    uint16_t frame_id,
+    const webrtc::EncodedImage& input_image) {
+  Timestamp processing_started = Now();
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "DefaultVideoQualityAnalyzer has to be started before use";
+
+  size_t peer_index = peers_->index(peer_name);
+
+  if (frame_id == VideoFrame::kNotSetId) {
+    frame_counters_.received++;
+    unknown_sender_frame_counters_[std::string(peer_name)].received++;
+    return;
+  }
+
+  auto it = captured_frames_in_flight_.find(frame_id);
+  if (it == captured_frames_in_flight_.end() ||
+      it->second.HasReceivedTime(peer_index)) {
+    // It means this frame was predecoded before, so we can skip it. It may
+    // happen when we have multiple simulcast streams in one track and received
+    // the same picture from two different streams because SFU can't reliably
+    // correlate two simulcast streams and started relaying the second stream
+    // from the same frame it has relayed right before for the first stream.
+    return;
+  }
+
+  frame_counters_.received++;
+  InternalStatsKey key(it->second.stream(),
+                       stream_to_sender_.at(it->second.stream()), peer_index);
+  stream_frame_counters_.at(key).received++;
+  // Determine the time of the last received packet of this video frame.
+  RTC_DCHECK(!input_image.PacketInfos().empty());
+  Timestamp last_receive_time =
+      std::max_element(input_image.PacketInfos().cbegin(),
+                       input_image.PacketInfos().cend(),
+                       [](const RtpPacketInfo& a, const RtpPacketInfo& b) {
+                         return a.receive_time() < b.receive_time();
+                       })
+          ->receive_time();
+  it->second.OnFramePreDecode(peer_index,
+                              /*received_time=*/last_receive_time,
+                              /*decode_start_time=*/Now(),
+                              input_image._frameType,
+                              DataSize::Bytes(input_image.size()));
+
+  if (options_.report_infra_metrics) {
+    analyzer_stats_.on_frame_pre_decode_processing_time_ms.AddSample(
+        (Now() - processing_started).ms<double>());
+  }
+}
+
+void DefaultVideoQualityAnalyzer::OnFrameDecoded(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame,
+    const DecoderStats& stats) {
+  Timestamp processing_started = Now();
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "DefaultVideoQualityAnalyzer has to be started before use";
+
+  size_t peer_index = peers_->index(peer_name);
+
+  if (frame.id() == VideoFrame::kNotSetId) {
+    frame_counters_.decoded++;
+    unknown_sender_frame_counters_[std::string(peer_name)].decoded++;
+    return;
+  }
+
+  auto it = captured_frames_in_flight_.find(frame.id());
+  if (it == captured_frames_in_flight_.end() ||
+      it->second.HasDecodeEndTime(peer_index)) {
+    // It means this frame was decoded before, so we can skip it. It may happen
+    // when we have multiple simulcast streams in one track and received
+    // the same frame from two different streams because SFU can't reliably
+    // correlate two simulcast streams and started relaying the second stream
+    // from the same frame it has relayed right before for the first stream.
+    return;
+  }
+  frame_counters_.decoded++;
+  InternalStatsKey key(it->second.stream(),
+                       stream_to_sender_.at(it->second.stream()), peer_index);
+  stream_frame_counters_.at(key).decoded++;
+  Timestamp now = Now();
+  StreamCodecInfo used_decoder;
+  used_decoder.codec_name = stats.decoder_name;
+  used_decoder.first_frame_id = frame.id();
+  used_decoder.last_frame_id = frame.id();
+  used_decoder.switched_on_at = now;
+  used_decoder.switched_from_at = now;
+  it->second.OnFrameDecoded(peer_index, now, frame.width(), frame.height(),
+                            used_decoder);
+
+  if (options_.report_infra_metrics) {
+    analyzer_stats_.on_frame_decoded_processing_time_ms.AddSample(
+        (Now() - processing_started).ms<double>());
+  }
+}
+
+void DefaultVideoQualityAnalyzer::OnFrameRendered(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame) {
+  Timestamp processing_started = Now();
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "DefaultVideoQualityAnalyzer has to be started before use";
+
+  size_t peer_index = peers_->index(peer_name);
+
+  if (frame.id() == VideoFrame::kNotSetId) {
+    frame_counters_.rendered++;
+    unknown_sender_frame_counters_[std::string(peer_name)].rendered++;
+    return;
+  }
+
+  auto frame_it = captured_frames_in_flight_.find(frame.id());
+  if (frame_it == captured_frames_in_flight_.end() ||
+      frame_it->second.HasRenderedTime(peer_index) ||
+      frame_it->second.IsDropped(peer_index)) {
+    // It means this frame was rendered or dropped before, so we can skip it.
+    // It may happen when we have multiple simulcast streams in one track and
+    // received the same frame from two different streams because SFU can't
+    // reliably correlate two simulcast streams and started relaying the second
+    // stream from the same frame it has relayed right before for the first
+    // stream.
+    absl::string_view reason = kSkipRenderedFrameReasonProcessed;
+    if (frame_it != captured_frames_in_flight_.end()) {
+      if (frame_it->second.HasRenderedTime(peer_index)) {
+        reason = kSkipRenderedFrameReasonRendered;
+      } else if (frame_it->second.IsDropped(peer_index)) {
+        reason = kSkipRenderedFrameReasonDropped;
+      }
+    }
+    RTC_LOG(LS_WARNING)
+        << "Peer " << peer_name
+        << "; Received frame out of order: received frame with id "
+        << frame.id() << " which was " << reason << " before";
+    return;
+  }
+
+  // Find corresponding captured frame.
+  FrameInFlight* frame_in_flight = &frame_it->second;
+  absl::optional<VideoFrame> captured_frame = frame_in_flight->frame();
+
+  const size_t stream_index = frame_in_flight->stream();
+  StreamState* state = &stream_states_.at(stream_index);
+  const InternalStatsKey stats_key(stream_index, state->sender(), peer_index);
+
+  // Update frames counters.
+  frame_counters_.rendered++;
+  stream_frame_counters_.at(stats_key).rendered++;
+
+  // Update current frame stats.
+  frame_in_flight->OnFrameRendered(peer_index, Now());
+
+  // After we received frame here we need to check if there are any dropped
+  // frames between this one and last one, that was rendered for this video
+  // stream.
+  int dropped_count = 0;
+  while (!state->IsEmpty(peer_index) &&
+         state->Front(peer_index) != frame.id()) {
+    dropped_count++;
+    uint16_t dropped_frame_id = state->PopFront(peer_index);
+    // Frame with id `dropped_frame_id` was dropped. We need:
+    // 1. Update global and stream frame counters
+    // 2. Extract corresponding frame from `captured_frames_in_flight_`
+    // 3. Send extracted frame to comparison with dropped=true
+    // 4. Cleanup dropped frame
+    frame_counters_.dropped++;
+    stream_frame_counters_.at(stats_key).dropped++;
+
+    auto dropped_frame_it = captured_frames_in_flight_.find(dropped_frame_id);
+    RTC_DCHECK(dropped_frame_it != captured_frames_in_flight_.end());
+    dropped_frame_it->second.MarkDropped(peer_index);
+
+    analyzer_stats_.frames_in_flight_left_count.AddSample(
+        StatsSample(captured_frames_in_flight_.size(), Now()));
+    frames_comparator_.AddComparison(
+        stats_key, /*captured=*/absl::nullopt, /*rendered=*/absl::nullopt,
+        FrameComparisonType::kDroppedFrame,
+        dropped_frame_it->second.GetStatsForPeer(peer_index));
+
+    if (dropped_frame_it->second.HaveAllPeersReceived()) {
+      captured_frames_in_flight_.erase(dropped_frame_it);
+    }
+  }
+  RTC_DCHECK(!state->IsEmpty(peer_index));
+  state->PopFront(peer_index);
+
+  if (state->last_rendered_frame_time(peer_index)) {
+    frame_in_flight->SetPrevFrameRenderedTime(
+        peer_index, state->last_rendered_frame_time(peer_index).value());
+  }
+  state->SetLastRenderedFrameTime(peer_index,
+                                  frame_in_flight->rendered_time(peer_index));
+  analyzer_stats_.frames_in_flight_left_count.AddSample(
+      StatsSample(captured_frames_in_flight_.size(), Now()));
+  frames_comparator_.AddComparison(
+      stats_key, dropped_count, captured_frame, /*rendered=*/frame,
+      FrameComparisonType::kRegular,
+      frame_in_flight->GetStatsForPeer(peer_index));
+
+  if (frame_it->second.HaveAllPeersReceived()) {
+    captured_frames_in_flight_.erase(frame_it);
+  }
+
+  if (options_.report_infra_metrics) {
+    analyzer_stats_.on_frame_rendered_processing_time_ms.AddSample(
+        (Now() - processing_started).ms<double>());
+  }
+}
+
+void DefaultVideoQualityAnalyzer::OnEncoderError(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame,
+    int32_t error_code) {
+  RTC_LOG(LS_ERROR) << "Encoder error for frame.id=" << frame.id()
+                    << ", code=" << error_code;
+}
+
+void DefaultVideoQualityAnalyzer::OnDecoderError(absl::string_view peer_name,
+                                                 uint16_t frame_id,
+                                                 int32_t error_code,
+                                                 const DecoderStats& stats) {
+  RTC_LOG(LS_ERROR) << "Decoder error for frame_id=" << frame_id
+                    << ", code=" << error_code;
+
+  Timestamp processing_started = Now();
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "DefaultVideoQualityAnalyzer has to be started before use";
+
+  size_t peer_index = peers_->index(peer_name);
+
+  if (frame_id == VideoFrame::kNotSetId) {
+    frame_counters_.failed_to_decode++;
+    unknown_sender_frame_counters_[std::string(peer_name)].failed_to_decode++;
+    return;
+  }
+
+  auto it = captured_frames_in_flight_.find(frame_id);
+  if (it == captured_frames_in_flight_.end() ||
+      it->second.HasDecodeEndTime(peer_index)) {
+    // It means this frame was decoded before, so we can skip it. It may happen
+    // when we have multiple simulcast streams in one track and received
+    // the same frame from two different streams because SFU can't reliably
+    // correlate two simulcast streams and started relaying the second stream
+    // from the same frame it has relayed right before for the first stream.
+    return;
+  }
+  frame_counters_.failed_to_decode++;
+  InternalStatsKey key(it->second.stream(),
+                       stream_to_sender_.at(it->second.stream()), peer_index);
+  stream_frame_counters_.at(key).failed_to_decode++;
+  Timestamp now = Now();
+  StreamCodecInfo used_decoder;
+  used_decoder.codec_name = stats.decoder_name;
+  used_decoder.first_frame_id = frame_id;
+  used_decoder.last_frame_id = frame_id;
+  used_decoder.switched_on_at = now;
+  used_decoder.switched_from_at = now;
+  it->second.OnDecoderError(peer_index, used_decoder);
+
+  if (options_.report_infra_metrics) {
+    analyzer_stats_.on_decoder_error_processing_time_ms.AddSample(
+        (Now() - processing_started).ms<double>());
+  }
+}
+
+void DefaultVideoQualityAnalyzer::RegisterParticipantInCall(
+    absl::string_view peer_name) {
+  MutexLock lock(&mutex_);
+  RTC_CHECK(!peers_->HasName(peer_name));
+  size_t new_peer_index = peers_->AddIfAbsent(peer_name);
+
+  // Ensure stats for receiving (for frames from other peers to this one)
+  // streams exists. Since in flight frames will be sent to the new peer
+  // as well. Sending stats (from this peer to others) will be added by
+  // DefaultVideoQualityAnalyzer::OnFrameCaptured.
+  std::vector<std::pair<InternalStatsKey, Timestamp>> stream_started_time;
+  for (auto [stream_index, sender_peer_index] : stream_to_sender_) {
+    InternalStatsKey key(stream_index, sender_peer_index, new_peer_index);
+
+    // To initiate `FrameCounters` for the stream we should pick frame
+    // counters with the same stream index and the same sender's peer index
+    // and any receiver's peer index and copy from its sender side
+    // counters.
+    FrameCounters counters;
+    for (size_t i : peers_->GetPresentIndexes()) {
+      InternalStatsKey prototype_key(stream_index, sender_peer_index, i);
+      auto it = stream_frame_counters_.find(prototype_key);
+      if (it != stream_frame_counters_.end()) {
+        counters.captured = it->second.captured;
+        counters.pre_encoded = it->second.pre_encoded;
+        counters.encoded = it->second.encoded;
+        break;
+      }
+    }
+    // It may happen if we had only one peer before this method was invoked,
+    // then `counters` will be empty. In such case empty `counters` are ok.
+    stream_frame_counters_.insert({key, std::move(counters)});
+
+    stream_started_time.push_back(
+        {key, stream_states_.at(stream_index).stream_started_time()});
+  }
+  frames_comparator_.RegisterParticipantInCall(stream_started_time,
+                                               start_time_);
+  // Ensure, that frames states are handled correctly
+  // (e.g. dropped frames tracking).
+  for (auto& [stream_index, stream_state] : stream_states_) {
+    stream_state.AddPeer(new_peer_index);
+  }
+  // Register new peer for every frame in flight.
+  // It is guaranteed, that no garbage FrameInFlight objects will stay in
+  // memory because of adding new peer. Even if the new peer won't receive the
+  // frame, the frame will be removed by OnFrameRendered after next frame comes
+  // for the new peer. It is important because FrameInFlight is a large object.
+  for (auto& [frame_id, frame_in_flight] : captured_frames_in_flight_) {
+    frame_in_flight.AddExpectedReceiver(new_peer_index);
+  }
+}
+
+void DefaultVideoQualityAnalyzer::UnregisterParticipantInCall(
+    absl::string_view peer_name) {
+  MutexLock lock(&mutex_);
+  RTC_CHECK(peers_->HasName(peer_name));
+  absl::optional<size_t> peer_index = peers_->RemoveIfPresent(peer_name);
+  RTC_CHECK(peer_index.has_value());
+
+  for (auto& [stream_index, stream_state] : stream_states_) {
+    if (!options_.enable_receive_own_stream &&
+        peer_index == stream_state.sender()) {
+      continue;
+    }
+
+    AddExistingFramesInFlightForStreamToComparator(stream_index, stream_state,
+                                                   *peer_index);
+
+    stream_state.RemovePeer(*peer_index);
+  }
+
+  // Remove peer from every frame in flight. If we removed that last expected
+  // receiver for the frame, then we should removed this frame if it was
+  // already encoded. If frame wasn't encoded, it still will be used by sender
+  // side pipeline, so we can't delete it yet.
+  for (auto it = captured_frames_in_flight_.begin();
+       it != captured_frames_in_flight_.end();) {
+    FrameInFlight& frame_in_flight = it->second;
+    frame_in_flight.RemoveExpectedReceiver(*peer_index);
+    // If frame was fully sent and all receivers received it, then erase it.
+    // It may happen that when we remove FrameInFlight only some Simulcast/SVC
+    // layers were encoded and frame has encoded time, but more layers might be
+    // encoded after removal. In such case it's safe to still remove a frame,
+    // because OnFrameEncoded method will correctly handle the case when there
+    // is no FrameInFlight for the received encoded image.
+    if (frame_in_flight.HasEncodedTime() &&
+        frame_in_flight.HaveAllPeersReceived()) {
+      it = captured_frames_in_flight_.erase(it);
+    } else {
+      it++;
+    }
+  }
+}
+
+void DefaultVideoQualityAnalyzer::Stop() {
+  std::map<InternalStatsKey, Timestamp> last_rendered_frame_times;
+  {
+    MutexLock lock(&mutex_);
+    if (state_ == State::kStopped) {
+      return;
+    }
+    RTC_CHECK_EQ(state_, State::kActive)
+        << "DefaultVideoQualityAnalyzer has to be started before use";
+
+    state_ = State::kStopped;
+
+    // Add the amount of frames in flight to the analyzer stats before all left
+    // frames in flight will be sent to the `frames_compartor_`.
+    analyzer_stats_.frames_in_flight_left_count.AddSample(
+        StatsSample(captured_frames_in_flight_.size(), Now()));
+
+    for (auto& state_entry : stream_states_) {
+      const size_t stream_index = state_entry.first;
+      StreamState& stream_state = state_entry.second;
+
+      // Populate `last_rendered_frame_times` map for all peers that were met in
+      // call, not only for the currently presented ones.
+      for (size_t peer_index : peers_->GetAllIndexes()) {
+        if (peer_index == stream_state.sender() &&
+            !options_.enable_receive_own_stream) {
+          continue;
+        }
+
+        InternalStatsKey stats_key(stream_index, stream_state.sender(),
+                                   peer_index);
+
+        // If there are no freezes in the call we have to report
+        // time_between_freezes_ms as call duration and in such case
+        // `stream_last_freeze_end_time` for this stream will be `start_time_`.
+        // If there is freeze, then we need add time from last rendered frame
+        // to last freeze end as time between freezes.
+        if (stream_state.last_rendered_frame_time(peer_index)) {
+          last_rendered_frame_times.emplace(
+              stats_key,
+              stream_state.last_rendered_frame_time(peer_index).value());
+        }
+      }
+
+      // Push left frame in flight for analysis for the peers that are still in
+      // the call.
+      for (size_t peer_index : peers_->GetPresentIndexes()) {
+        if (peer_index == stream_state.sender() &&
+            !options_.enable_receive_own_stream) {
+          continue;
+        }
+
+        AddExistingFramesInFlightForStreamToComparator(
+            stream_index, stream_state, peer_index);
+      }
+    }
+  }
+  frames_comparator_.Stop(last_rendered_frame_times);
+
+  // Perform final Metrics update. On this place analyzer is stopped and no one
+  // holds any locks.
+  {
+    MutexLock lock(&mutex_);
+    FramesComparatorStats frames_comparator_stats =
+        frames_comparator_.frames_comparator_stats();
+    analyzer_stats_.comparisons_queue_size =
+        std::move(frames_comparator_stats.comparisons_queue_size);
+    analyzer_stats_.comparisons_done = frames_comparator_stats.comparisons_done;
+    analyzer_stats_.cpu_overloaded_comparisons_done =
+        frames_comparator_stats.cpu_overloaded_comparisons_done;
+    analyzer_stats_.memory_overloaded_comparisons_done =
+        frames_comparator_stats.memory_overloaded_comparisons_done;
+  }
+  ReportResults();
+}
+
+std::string DefaultVideoQualityAnalyzer::GetStreamLabel(uint16_t frame_id) {
+  MutexLock lock1(&mutex_);
+  auto it = captured_frames_in_flight_.find(frame_id);
+  if (it != captured_frames_in_flight_.end()) {
+    return streams_.name(it->second.stream());
+  }
+  for (auto hist_it = stream_to_frame_id_history_.begin();
+       hist_it != stream_to_frame_id_history_.end(); ++hist_it) {
+    auto hist_set_it = hist_it->second.find(frame_id);
+    if (hist_set_it != hist_it->second.end()) {
+      return streams_.name(hist_it->first);
+    }
+  }
+  RTC_CHECK(false) << "Unknown frame_id=" << frame_id;
+}
+
+std::set<StatsKey> DefaultVideoQualityAnalyzer::GetKnownVideoStreams() const {
+  MutexLock lock(&mutex_);
+  std::set<StatsKey> out;
+  for (auto& item : frames_comparator_.stream_stats()) {
+    RTC_LOG(LS_INFO) << item.first.ToString() << " ==> "
+                     << ToStatsKey(item.first).ToString();
+    out.insert(ToStatsKey(item.first));
+  }
+  return out;
+}
+
+VideoStreamsInfo DefaultVideoQualityAnalyzer::GetKnownStreams() const {
+  MutexLock lock(&mutex_);
+  std::map<std::string, std::string> stream_to_sender;
+  std::map<std::string, std::set<std::string>> sender_to_streams;
+  std::map<std::string, std::set<std::string>> stream_to_receivers;
+
+  for (auto& item : frames_comparator_.stream_stats()) {
+    const std::string& stream_label = streams_.name(item.first.stream);
+    const std::string& sender = peers_->name(item.first.sender);
+    const std::string& receiver = peers_->name(item.first.receiver);
+    RTC_LOG(LS_INFO) << item.first.ToString() << " ==> "
+                     << "stream=" << stream_label << "; sender=" << sender
+                     << "; receiver=" << receiver;
+    stream_to_sender.emplace(stream_label, sender);
+    auto streams_it = sender_to_streams.find(sender);
+    if (streams_it != sender_to_streams.end()) {
+      streams_it->second.emplace(stream_label);
+    } else {
+      sender_to_streams.emplace(sender, std::set<std::string>{stream_label});
+    }
+    auto receivers_it = stream_to_receivers.find(stream_label);
+    if (receivers_it != stream_to_receivers.end()) {
+      receivers_it->second.emplace(receiver);
+    } else {
+      stream_to_receivers.emplace(stream_label,
+                                  std::set<std::string>{receiver});
+    }
+  }
+
+  return VideoStreamsInfo(std::move(stream_to_sender),
+                          std::move(sender_to_streams),
+                          std::move(stream_to_receivers));
+}
+
+FrameCounters DefaultVideoQualityAnalyzer::GetGlobalCounters() const {
+  MutexLock lock(&mutex_);
+  return frame_counters_;
+}
+
+std::map<std::string, FrameCounters>
+DefaultVideoQualityAnalyzer::GetUnknownSenderFrameCounters() const {
+  MutexLock lock(&mutex_);
+  return unknown_sender_frame_counters_;
+}
+
+std::map<StatsKey, FrameCounters>
+DefaultVideoQualityAnalyzer::GetPerStreamCounters() const {
+  MutexLock lock(&mutex_);
+  std::map<StatsKey, FrameCounters> out;
+  for (auto& item : stream_frame_counters_) {
+    out.emplace(ToStatsKey(item.first), item.second);
+  }
+  return out;
+}
+
+std::map<StatsKey, StreamStats> DefaultVideoQualityAnalyzer::GetStats() const {
+  MutexLock lock1(&mutex_);
+  std::map<StatsKey, StreamStats> out;
+  for (auto& item : frames_comparator_.stream_stats()) {
+    out.emplace(ToStatsKey(item.first), item.second);
+  }
+  return out;
+}
+
+AnalyzerStats DefaultVideoQualityAnalyzer::GetAnalyzerStats() const {
+  MutexLock lock(&mutex_);
+  return analyzer_stats_;
+}
+
+uint16_t DefaultVideoQualityAnalyzer::GetNextFrameId() {
+  uint16_t frame_id = next_frame_id_++;
+  if (next_frame_id_ == VideoFrame::kNotSetId) {
+    next_frame_id_ = 1;
+  }
+  return frame_id;
+}
+
+void DefaultVideoQualityAnalyzer::
+    AddExistingFramesInFlightForStreamToComparator(size_t stream_index,
+                                                   StreamState& stream_state,
+                                                   size_t peer_index) {
+  InternalStatsKey stats_key(stream_index, stream_state.sender(), peer_index);
+
+  // Add frames in flight for this stream into frames comparator.
+  // Frames in flight were not rendered, so they won't affect stream's
+  // last rendered frame time.
+  while (!stream_state.IsEmpty(peer_index)) {
+    uint16_t frame_id = stream_state.PopFront(peer_index);
+    auto it = captured_frames_in_flight_.find(frame_id);
+    RTC_DCHECK(it != captured_frames_in_flight_.end());
+    FrameInFlight& frame = it->second;
+
+    frames_comparator_.AddComparison(stats_key, /*captured=*/absl::nullopt,
+                                     /*rendered=*/absl::nullopt,
+                                     FrameComparisonType::kFrameInFlight,
+                                     frame.GetStatsForPeer(peer_index));
+  }
+}
+
+void DefaultVideoQualityAnalyzer::ReportResults() {
+  MutexLock lock(&mutex_);
+  for (auto& item : frames_comparator_.stream_stats()) {
+    ReportResults(item.first, item.second,
+                  stream_frame_counters_.at(item.first));
+  }
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  metrics_logger_->LogSingleValueMetric(
+      "cpu_usage_%", test_label_, GetCpuUsagePercent(), Unit::kUnitless,
+      ImprovementDirection::kSmallerIsBetter,
+      {{MetricMetadataKey::kExperimentalTestNameMetadataKey, test_label_}});
+  LogFrameCounters("Global", frame_counters_);
+  if (!unknown_sender_frame_counters_.empty()) {
+    RTC_LOG(LS_INFO) << "Received frame counters with unknown frame id:";
+    for (const auto& [peer_name, frame_counters] :
+         unknown_sender_frame_counters_) {
+      LogFrameCounters(peer_name, frame_counters);
+    }
+  }
+  RTC_LOG(LS_INFO) << "Received frame counters per stream:";
+  for (const auto& [stats_key, stream_stats] :
+       frames_comparator_.stream_stats()) {
+    LogFrameCounters(ToStatsKey(stats_key).ToString(),
+                     stream_frame_counters_.at(stats_key));
+    LogStreamInternalStats(ToStatsKey(stats_key).ToString(), stream_stats,
+                           start_time_);
+  }
+  if (!analyzer_stats_.comparisons_queue_size.IsEmpty()) {
+    RTC_LOG(LS_INFO) << "comparisons_queue_size min="
+                     << analyzer_stats_.comparisons_queue_size.GetMin()
+                     << "; max="
+                     << analyzer_stats_.comparisons_queue_size.GetMax()
+                     << "; 99%="
+                     << analyzer_stats_.comparisons_queue_size.GetPercentile(
+                            0.99);
+  }
+  RTC_LOG(LS_INFO) << "comparisons_done=" << analyzer_stats_.comparisons_done;
+  RTC_LOG(LS_INFO) << "cpu_overloaded_comparisons_done="
+                   << analyzer_stats_.cpu_overloaded_comparisons_done;
+  RTC_LOG(LS_INFO) << "memory_overloaded_comparisons_done="
+                   << analyzer_stats_.memory_overloaded_comparisons_done;
+  if (options_.report_infra_metrics) {
+    metrics_logger_->LogMetric("comparisons_queue_size", test_label_,
+                               analyzer_stats_.comparisons_queue_size,
+                               Unit::kCount,
+                               ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogMetric("frames_in_flight_left_count", test_label_,
+                               analyzer_stats_.frames_in_flight_left_count,
+                               Unit::kCount,
+                               ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogSingleValueMetric(
+        "comparisons_done", test_label_, analyzer_stats_.comparisons_done,
+        Unit::kCount, ImprovementDirection::kNeitherIsBetter);
+    metrics_logger_->LogSingleValueMetric(
+        "cpu_overloaded_comparisons_done", test_label_,
+        analyzer_stats_.cpu_overloaded_comparisons_done, Unit::kCount,
+        ImprovementDirection::kNeitherIsBetter);
+    metrics_logger_->LogSingleValueMetric(
+        "memory_overloaded_comparisons_done", test_label_,
+        analyzer_stats_.memory_overloaded_comparisons_done, Unit::kCount,
+        ImprovementDirection::kNeitherIsBetter);
+    metrics_logger_->LogSingleValueMetric(
+        "test_duration", test_label_, (Now() - start_time_).ms(),
+        Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter);
+
+    metrics_logger_->LogMetric(
+        "on_frame_captured_processing_time_ms", test_label_,
+        analyzer_stats_.on_frame_captured_processing_time_ms,
+        Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogMetric(
+        "on_frame_pre_encode_processing_time_ms", test_label_,
+        analyzer_stats_.on_frame_pre_encode_processing_time_ms,
+        Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogMetric(
+        "on_frame_encoded_processing_time_ms", test_label_,
+        analyzer_stats_.on_frame_encoded_processing_time_ms,
+        Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogMetric(
+        "on_frame_pre_decode_processing_time_ms", test_label_,
+        analyzer_stats_.on_frame_pre_decode_processing_time_ms,
+        Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogMetric(
+        "on_frame_decoded_processing_time_ms", test_label_,
+        analyzer_stats_.on_frame_decoded_processing_time_ms,
+        Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogMetric(
+        "on_frame_rendered_processing_time_ms", test_label_,
+        analyzer_stats_.on_frame_rendered_processing_time_ms,
+        Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
+    metrics_logger_->LogMetric(
+        "on_decoder_error_processing_time_ms", test_label_,
+        analyzer_stats_.on_decoder_error_processing_time_ms,
+        Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter);
+  }
+}
+
+void DefaultVideoQualityAnalyzer::ReportResults(
+    const InternalStatsKey& key,
+    const StreamStats& stats,
+    const FrameCounters& frame_counters) {
+  TimeDelta test_duration = Now() - start_time_;
+  std::string test_case_name = GetTestCaseName(ToMetricName(key));
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  std::map<std::string, std::string> metric_metadata{
+      {MetricMetadataKey::kPeerMetadataKey, peers_->name(key.sender)},
+      {MetricMetadataKey::kVideoStreamMetadataKey, streams_.name(key.stream)},
+      {MetricMetadataKey::kSenderMetadataKey, peers_->name(key.sender)},
+      {MetricMetadataKey::kReceiverMetadataKey, peers_->name(key.receiver)},
+      {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_label_}};
+
+  double sum_squared_interframe_delays_secs = 0;
+  Timestamp video_start_time = Timestamp::PlusInfinity();
+  Timestamp video_end_time = Timestamp::MinusInfinity();
+  for (const SamplesStatsCounter::StatsSample& sample :
+       stats.time_between_rendered_frames_ms.GetTimedSamples()) {
+    double interframe_delay_ms = sample.value;
+    const double interframe_delays_secs = interframe_delay_ms / 1000.0;
+    // Sum of squared inter frame intervals is used to calculate the harmonic
+    // frame rate metric. The metric aims to reflect overall experience related
+    // to smoothness of video playback and includes both freezes and pauses.
+    sum_squared_interframe_delays_secs +=
+        interframe_delays_secs * interframe_delays_secs;
+    if (sample.time < video_start_time) {
+      video_start_time = sample.time;
+    }
+    if (sample.time > video_end_time) {
+      video_end_time = sample.time;
+    }
+  }
+  double harmonic_framerate_fps = 0;
+  TimeDelta video_duration = video_end_time - video_start_time;
+  if (sum_squared_interframe_delays_secs > 0.0 && video_duration.IsFinite()) {
+    harmonic_framerate_fps =
+        video_duration.seconds<double>() / sum_squared_interframe_delays_secs;
+  }
+
+  metrics_logger_->LogMetric(
+      "psnr_dB", test_case_name, stats.psnr, Unit::kUnitless,
+      ImprovementDirection::kBiggerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric(
+      "ssim", test_case_name, stats.ssim, Unit::kUnitless,
+      ImprovementDirection::kBiggerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric("transport_time", test_case_name,
+                             stats.transport_time_ms, Unit::kMilliseconds,
+                             ImprovementDirection::kSmallerIsBetter,
+                             metric_metadata);
+  metrics_logger_->LogMetric(
+      "total_delay_incl_transport", test_case_name,
+      stats.total_delay_incl_transport_ms, Unit::kMilliseconds,
+      ImprovementDirection::kSmallerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric(
+      "time_between_rendered_frames", test_case_name,
+      stats.time_between_rendered_frames_ms, Unit::kMilliseconds,
+      ImprovementDirection::kSmallerIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "harmonic_framerate", test_case_name, harmonic_framerate_fps,
+      Unit::kHertz, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "encode_frame_rate", test_case_name,
+      stats.encode_frame_rate.IsEmpty()
+          ? 0
+          : stats.encode_frame_rate.GetEventsPerSecond(),
+      Unit::kHertz, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric(
+      "encode_time", test_case_name, stats.encode_time_ms, Unit::kMilliseconds,
+      ImprovementDirection::kSmallerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric("time_between_freezes", test_case_name,
+                             stats.time_between_freezes_ms, Unit::kMilliseconds,
+                             ImprovementDirection::kBiggerIsBetter,
+                             metric_metadata);
+  metrics_logger_->LogMetric("freeze_time_ms", test_case_name,
+                             stats.freeze_time_ms, Unit::kMilliseconds,
+                             ImprovementDirection::kSmallerIsBetter,
+                             metric_metadata);
+  metrics_logger_->LogMetric(
+      "pixels_per_frame", test_case_name, stats.resolution_of_decoded_frame,
+      Unit::kCount, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "min_psnr_dB", test_case_name,
+      stats.psnr.IsEmpty() ? 0 : stats.psnr.GetMin(), Unit::kUnitless,
+      ImprovementDirection::kBiggerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric(
+      "decode_time", test_case_name, stats.decode_time_ms, Unit::kMilliseconds,
+      ImprovementDirection::kSmallerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric(
+      "receive_to_render_time", test_case_name, stats.receive_to_render_time_ms,
+      Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter,
+      metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "dropped_frames", test_case_name, frame_counters.dropped, Unit::kCount,
+      ImprovementDirection::kSmallerIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "frames_in_flight", test_case_name,
+      frame_counters.captured - frame_counters.rendered -
+          frame_counters.dropped,
+      Unit::kCount, ImprovementDirection::kSmallerIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "rendered_frames", test_case_name, frame_counters.rendered, Unit::kCount,
+      ImprovementDirection::kBiggerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric(
+      "max_skipped", test_case_name, stats.skipped_between_rendered,
+      Unit::kCount, ImprovementDirection::kSmallerIsBetter, metric_metadata);
+  metrics_logger_->LogMetric(
+      "target_encode_bitrate", test_case_name,
+      stats.target_encode_bitrate / 1000, Unit::kKilobitsPerSecond,
+      ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  for (const auto& [spatial_layer, qp] : stats.spatial_layers_qp) {
+    std::map<std::string, std::string> qp_metadata = metric_metadata;
+    qp_metadata[MetricMetadataKey::kSpatialLayerMetadataKey] =
+        std::to_string(spatial_layer);
+    metrics_logger_->LogMetric("qp_sl" + std::to_string(spatial_layer),
+                               test_case_name, qp, Unit::kUnitless,
+                               ImprovementDirection::kSmallerIsBetter,
+                               std::move(qp_metadata));
+  }
+  metrics_logger_->LogSingleValueMetric(
+      "actual_encode_bitrate", test_case_name,
+      static_cast<double>(stats.total_encoded_images_payload) /
+          test_duration.seconds<double>() * kBitsInByte / 1000,
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter,
+      metric_metadata);
+
+  if (options_.report_detailed_frame_stats) {
+    metrics_logger_->LogSingleValueMetric(
+        "capture_frame_rate", test_case_name,
+        stats.capture_frame_rate.IsEmpty()
+            ? 0
+            : stats.capture_frame_rate.GetEventsPerSecond(),
+        Unit::kHertz, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+    metrics_logger_->LogSingleValueMetric(
+        "num_encoded_frames", test_case_name, frame_counters.encoded,
+        Unit::kCount, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+    metrics_logger_->LogSingleValueMetric(
+        "num_decoded_frames", test_case_name, frame_counters.decoded,
+        Unit::kCount, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+    metrics_logger_->LogSingleValueMetric(
+        "num_send_key_frames", test_case_name, stats.num_send_key_frames,
+        Unit::kCount, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+    metrics_logger_->LogSingleValueMetric(
+        "num_recv_key_frames", test_case_name, stats.num_recv_key_frames,
+        Unit::kCount, ImprovementDirection::kBiggerIsBetter, metric_metadata);
+
+    metrics_logger_->LogMetric("recv_key_frame_size_bytes", test_case_name,
+                               stats.recv_key_frame_size_bytes, Unit::kCount,
+                               ImprovementDirection::kBiggerIsBetter,
+                               metric_metadata);
+    metrics_logger_->LogMetric("recv_delta_frame_size_bytes", test_case_name,
+                               stats.recv_delta_frame_size_bytes, Unit::kCount,
+                               ImprovementDirection::kBiggerIsBetter,
+                               metric_metadata);
+  }
+}
+
+std::string DefaultVideoQualityAnalyzer::GetTestCaseName(
+    const std::string& stream_label) const {
+  return test_label_ + "/" + stream_label;
+}
+
+Timestamp DefaultVideoQualityAnalyzer::Now() {
+  return clock_->CurrentTime();
+}
+
+StatsKey DefaultVideoQualityAnalyzer::ToStatsKey(
+    const InternalStatsKey& key) const {
+  return StatsKey(streams_.name(key.stream), peers_->name(key.receiver));
+}
+
+std::string DefaultVideoQualityAnalyzer::ToMetricName(
+    const InternalStatsKey& key) const {
+  const std::string& stream_label = streams_.name(key.stream);
+  if (peers_->GetKnownSize() <= 2 && key.sender != key.receiver) {
+    // TODO(titovartem): remove this special case.
+    return stream_label;
+  }
+  rtc::StringBuilder out;
+  out << stream_label << "_" << peers_->name(key.sender) << "_"
+      << peers_->name(key.receiver);
+  return out.str();
+}
+
+double DefaultVideoQualityAnalyzer::GetCpuUsagePercent() {
+  return cpu_measurer_.GetCpuUsagePercent();
+}
+
+std::map<std::string, std::vector<uint16_t>>
+DefaultVideoQualityAnalyzer::GetStreamFrames() const {
+  MutexLock lock(&mutex_);
+  std::map<std::string, std::vector<uint16_t>> out;
+  for (auto entry_it : stream_to_frame_id_full_history_) {
+    out.insert({streams_.name(entry_it.first), entry_it.second});
+  }
+  return out;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
new file mode 100644
index 0000000000..b67e5a0147
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
@@ -0,0 +1,197 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_H_
+
+#include <atomic>
+#include <cstdint>
+#include <deque>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h"
+#include "test/pc/e2e/analyzer/video/names_collection.h"
+
+namespace webrtc {
+
+class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
+ public:
+  DefaultVideoQualityAnalyzer(webrtc::Clock* clock,
+                              test::MetricsLogger* metrics_logger,
+                              DefaultVideoQualityAnalyzerOptions options = {});
+  ~DefaultVideoQualityAnalyzer() override;
+
+  void Start(std::string test_case_name,
+             rtc::ArrayView<const std::string> peer_names,
+             int max_threads_count) override;
+  uint16_t OnFrameCaptured(absl::string_view peer_name,
+                           const std::string& stream_label,
+                           const VideoFrame& frame) override;
+  void OnFramePreEncode(absl::string_view peer_name,
+                        const VideoFrame& frame) override;
+  void OnFrameEncoded(absl::string_view peer_name,
+                      uint16_t frame_id,
+                      const EncodedImage& encoded_image,
+                      const EncoderStats& stats,
+                      bool discarded) override;
+  void OnFrameDropped(absl::string_view peer_name,
+                      EncodedImageCallback::DropReason reason) override;
+  void OnFramePreDecode(absl::string_view peer_name,
+                        uint16_t frame_id,
+                        const EncodedImage& input_image) override;
+  void OnFrameDecoded(absl::string_view peer_name,
+                      const VideoFrame& frame,
+                      const DecoderStats& stats) override;
+  void OnFrameRendered(absl::string_view peer_name,
+                       const VideoFrame& frame) override;
+  void OnEncoderError(absl::string_view peer_name,
+                      const VideoFrame& frame,
+                      int32_t error_code) override;
+  void OnDecoderError(absl::string_view peer_name,
+                      uint16_t frame_id,
+                      int32_t error_code,
+                      const DecoderStats& stats) override;
+
+  void RegisterParticipantInCall(absl::string_view peer_name) override;
+  void UnregisterParticipantInCall(absl::string_view peer_name) override;
+
+  void Stop() override;
+  std::string GetStreamLabel(uint16_t frame_id) override;
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override {}
+
+  // Returns set of stream labels, that were met during test call.
+  std::set<StatsKey> GetKnownVideoStreams() const;
+  VideoStreamsInfo GetKnownStreams() const;
+  FrameCounters GetGlobalCounters() const;
+  // Returns frame counter for frames received without frame id set.
+  std::map<std::string, FrameCounters> GetUnknownSenderFrameCounters() const;
+  // Returns frame counter per stream label. Valid stream labels can be obtained
+  // by calling GetKnownVideoStreams()
+  std::map<StatsKey, FrameCounters> GetPerStreamCounters() const;
+  // Returns video quality stats per stream label. Valid stream labels can be
+  // obtained by calling GetKnownVideoStreams()
+  std::map<StatsKey, StreamStats> GetStats() const;
+  AnalyzerStats GetAnalyzerStats() const;
+  double GetCpuUsagePercent();
+
+  // Returns mapping from the stream label to the history of frames that were
+  // met in this stream in the order as they were captured.
+  std::map<std::string, std::vector<uint16_t>> GetStreamFrames() const;
+
+ private:
+  enum State { kNew, kActive, kStopped };
+
+  // Returns next frame id to use. Frame ID can't be `VideoFrame::kNotSetId`,
+  // because this value is reserved by `VideoFrame` as "ID not set".
+  uint16_t GetNextFrameId() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  void AddExistingFramesInFlightForStreamToComparator(size_t stream_index,
+                                                      StreamState& stream_state,
+                                                      size_t peer_index)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  // Report results for all metrics for all streams.
+  void ReportResults();
+  void ReportResults(const InternalStatsKey& key,
+                     const StreamStats& stats,
+                     const FrameCounters& frame_counters)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Returns name of current test case for reporting.
+  std::string GetTestCaseName(const std::string& stream_label) const;
+  Timestamp Now();
+  StatsKey ToStatsKey(const InternalStatsKey& key) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Returns string representation of stats key for metrics naming. Used for
+  // backward compatibility by metrics naming for 2 peers cases.
+  std::string ToMetricName(const InternalStatsKey& key) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  static const uint16_t kStartingFrameId = 1;
+
+  const DefaultVideoQualityAnalyzerOptions options_;
+  webrtc::Clock* const clock_;
+  test::MetricsLogger* const metrics_logger_;
+
+  std::string test_label_;
+
+  mutable Mutex mutex_;
+  uint16_t next_frame_id_ RTC_GUARDED_BY(mutex_) = kStartingFrameId;
+  std::unique_ptr<NamesCollection> peers_ RTC_GUARDED_BY(mutex_);
+  State state_ RTC_GUARDED_BY(mutex_) = State::kNew;
+  Timestamp start_time_ RTC_GUARDED_BY(mutex_) = Timestamp::MinusInfinity();
+  // Mapping from stream label to unique size_t value to use in stats and avoid
+  // extra string copying.
+  NamesCollection streams_ RTC_GUARDED_BY(mutex_);
+  // Frames that were captured by all streams and still aren't rendered on
+  // receivers or deemed dropped. Frame with id X can be removed from this map
+  // if:
+  // 1. The frame with id X was received in OnFrameRendered by all expected
+  //    receivers.
+  // 2. The frame with id Y > X was received in OnFrameRendered by all expected
+  //    receivers.
+  // 3. Next available frame id for newly captured frame is X
+  // 4. There too many frames in flight for current video stream and X is the
+  //    oldest frame id in this stream. In such case only the frame content
+  //    will be removed, but the map entry will be preserved.
+  std::map<uint16_t, FrameInFlight> captured_frames_in_flight_
+      RTC_GUARDED_BY(mutex_);
+  // Global frames count for all video streams.
+  FrameCounters frame_counters_ RTC_GUARDED_BY(mutex_);
+  // Frame counters for received frames without video frame id set.
+  // Map from peer name to the frame counters.
+  std::map<std::string, FrameCounters> unknown_sender_frame_counters_
+      RTC_GUARDED_BY(mutex_);
+  // Frame counters per each stream per each receiver.
+  std::map<InternalStatsKey, FrameCounters> stream_frame_counters_
+      RTC_GUARDED_BY(mutex_);
+  // Map from stream index in `streams_` to its StreamState.
+  std::map<size_t, StreamState> stream_states_ RTC_GUARDED_BY(mutex_);
+  // Map from stream index in `streams_` to sender peer index in `peers_`.
+  std::map<size_t, size_t> stream_to_sender_ RTC_GUARDED_BY(mutex_);
+
+  // Stores history mapping between stream index in `streams_` and frame ids.
+  // Updated when frame id overlap. It required to properly return stream label
+  // after 1st frame from simulcast streams was already rendered and last is
+  // still encoding.
+  std::map<size_t, std::set<uint16_t>> stream_to_frame_id_history_
+      RTC_GUARDED_BY(mutex_);
+  // Map from stream index to the list of frames as they were met in the stream.
+  std::map<size_t, std::vector<uint16_t>> stream_to_frame_id_full_history_
+      RTC_GUARDED_BY(mutex_);
+  AnalyzerStats analyzer_stats_ RTC_GUARDED_BY(mutex_);
+
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer_;
+  DefaultVideoQualityAnalyzerFramesComparator frames_comparator_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.cc
new file mode 100644
index 0000000000..847c9f09a6
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.h"
+
+#include "rtc_base/cpu_time.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system_time.h"
+
+namespace webrtc {
+
+void DefaultVideoQualityAnalyzerCpuMeasurer::StartMeasuringCpuProcessTime() {
+  MutexLock lock(&mutex_);
+  cpu_time_ -= rtc::GetProcessCpuTimeNanos();
+  wallclock_time_ -= rtc::SystemTimeNanos();
+}
+
+void DefaultVideoQualityAnalyzerCpuMeasurer::StopMeasuringCpuProcessTime() {
+  MutexLock lock(&mutex_);
+  cpu_time_ += rtc::GetProcessCpuTimeNanos();
+  wallclock_time_ += rtc::SystemTimeNanos();
+}
+
+void DefaultVideoQualityAnalyzerCpuMeasurer::StartExcludingCpuThreadTime() {
+  MutexLock lock(&mutex_);
+  cpu_time_ += rtc::GetThreadCpuTimeNanos();
+}
+
+void DefaultVideoQualityAnalyzerCpuMeasurer::StopExcludingCpuThreadTime() {
+  MutexLock lock(&mutex_);
+  cpu_time_ -= rtc::GetThreadCpuTimeNanos();
+}
+
+double DefaultVideoQualityAnalyzerCpuMeasurer::GetCpuUsagePercent() {
+  MutexLock lock(&mutex_);
+  return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.h
new file mode 100644
index 0000000000..dd9fa07af2
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_CPU_MEASURER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_CPU_MEASURER_H_
+
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// This class is thread safe.
+class DefaultVideoQualityAnalyzerCpuMeasurer {
+ public:
+  double GetCpuUsagePercent();
+
+  void StartMeasuringCpuProcessTime();
+  void StopMeasuringCpuProcessTime();
+  void StartExcludingCpuThreadTime();
+  void StopExcludingCpuThreadTime();
+
+ private:
+  Mutex mutex_;
+  int64_t cpu_time_ RTC_GUARDED_BY(mutex_) = 0;
+  int64_t wallclock_time_ RTC_GUARDED_BY(mutex_) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_CPU_MEASURER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc
new file mode 100644
index 0000000000..df34dadaf0
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc
@@ -0,0 +1,209 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h"
+
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_type.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h"
+
+namespace webrtc {
+namespace {
+
+template <typename T>
+absl::optional<T> MaybeGetValue(const std::map<size_t, T>& map, size_t key) {
+  auto it = map.find(key);
+  if (it == map.end()) {
+    return absl::nullopt;
+  }
+  return it->second;
+}
+
+}  // namespace
+
+FrameInFlight::FrameInFlight(size_t stream,
+                             VideoFrame frame,
+                             Timestamp captured_time,
+                             std::set<size_t> expected_receivers)
+    : stream_(stream),
+      expected_receivers_(std::move(expected_receivers)),
+      frame_(std::move(frame)),
+      captured_time_(captured_time) {}
+
+bool FrameInFlight::RemoveFrame() {
+  if (!frame_) {
+    return false;
+  }
+  frame_ = absl::nullopt;
+  return true;
+}
+
+void FrameInFlight::SetFrameId(uint16_t id) {
+  if (frame_) {
+    frame_->set_id(id);
+  }
+  frame_id_ = id;
+}
+
+std::vector<size_t> FrameInFlight::GetPeersWhichDidntReceive() const {
+  std::vector<size_t> out;
+  for (size_t peer : expected_receivers_) {
+    auto it = receiver_stats_.find(peer);
+    if (it == receiver_stats_.end() ||
+        (!it->second.dropped && it->second.rendered_time.IsInfinite())) {
+      out.push_back(peer);
+    }
+  }
+  return out;
+}
+
+bool FrameInFlight::HaveAllPeersReceived() const {
+  for (size_t peer : expected_receivers_) {
+    auto it = receiver_stats_.find(peer);
+    if (it == receiver_stats_.end()) {
+      return false;
+    }
+
+    if (!it->second.dropped && it->second.rendered_time.IsInfinite()) {
+      return false;
+    }
+  }
+  return true;
+}
+
+void FrameInFlight::OnFrameEncoded(webrtc::Timestamp time,
+                                   VideoFrameType frame_type,
+                                   DataSize encoded_image_size,
+                                   uint32_t target_encode_bitrate,
+                                   int spatial_layer,
+                                   int qp,
+                                   StreamCodecInfo used_encoder) {
+  encoded_time_ = time;
+  frame_type_ = frame_type;
+  encoded_image_size_ = encoded_image_size;
+  target_encode_bitrate_ += target_encode_bitrate;
+  spatial_layers_qp_[spatial_layer].AddSample(SamplesStatsCounter::StatsSample{
+      .value = static_cast<double>(qp), .time = time});
+  // Update used encoder info. If simulcast/SVC is used, this method can
+  // be called multiple times, in such case we should preserve the value
+  // of `used_encoder_.switched_on_at` from the first invocation as the
+  // smallest one.
+  Timestamp encoder_switched_on_at = used_encoder_.has_value()
+                                         ? used_encoder_->switched_on_at
+                                         : Timestamp::PlusInfinity();
+  RTC_DCHECK(used_encoder.switched_on_at.IsFinite());
+  RTC_DCHECK(used_encoder.switched_from_at.IsFinite());
+  used_encoder_ = used_encoder;
+  if (encoder_switched_on_at < used_encoder_->switched_on_at) {
+    used_encoder_->switched_on_at = encoder_switched_on_at;
+  }
+}
+
+void FrameInFlight::OnFramePreDecode(size_t peer,
+                                     webrtc::Timestamp received_time,
+                                     webrtc::Timestamp decode_start_time,
+                                     VideoFrameType frame_type,
+                                     DataSize encoded_image_size) {
+  receiver_stats_[peer].received_time = received_time;
+  receiver_stats_[peer].decode_start_time = decode_start_time;
+  receiver_stats_[peer].frame_type = frame_type;
+  receiver_stats_[peer].encoded_image_size = encoded_image_size;
+}
+
+bool FrameInFlight::HasReceivedTime(size_t peer) const {
+  auto it = receiver_stats_.find(peer);
+  if (it == receiver_stats_.end()) {
+    return false;
+  }
+  return it->second.received_time.IsFinite();
+}
+
+void FrameInFlight::OnFrameDecoded(size_t peer,
+                                   webrtc::Timestamp time,
+                                   int width,
+                                   int height,
+                                   const StreamCodecInfo& used_decoder) {
+  receiver_stats_[peer].decode_end_time = time;
+  receiver_stats_[peer].used_decoder = used_decoder;
+  receiver_stats_[peer].decoded_frame_width = width;
+  receiver_stats_[peer].decoded_frame_height = height;
+}
+
+void FrameInFlight::OnDecoderError(size_t peer,
+                                   const StreamCodecInfo& used_decoder) {
+  receiver_stats_[peer].decoder_failed = true;
+  receiver_stats_[peer].used_decoder = used_decoder;
+}
+
+bool FrameInFlight::HasDecodeEndTime(size_t peer) const {
+  auto it = receiver_stats_.find(peer);
+  if (it == receiver_stats_.end()) {
+    return false;
+  }
+  return it->second.decode_end_time.IsFinite();
+}
+
+void FrameInFlight::OnFrameRendered(size_t peer, webrtc::Timestamp time) {
+  receiver_stats_[peer].rendered_time = time;
+}
+
+bool FrameInFlight::HasRenderedTime(size_t peer) const {
+  auto it = receiver_stats_.find(peer);
+  if (it == receiver_stats_.end()) {
+    return false;
+  }
+  return it->second.rendered_time.IsFinite();
+}
+
+bool FrameInFlight::IsDropped(size_t peer) const {
+  auto it = receiver_stats_.find(peer);
+  if (it == receiver_stats_.end()) {
+    return false;
+  }
+  return it->second.dropped;
+}
+
+FrameStats FrameInFlight::GetStatsForPeer(size_t peer) const {
+  RTC_DCHECK_NE(frame_id_, VideoFrame::kNotSetId)
+      << "Frame id isn't initialized";
+  FrameStats stats(frame_id_, captured_time_);
+  stats.pre_encode_time = pre_encode_time_;
+  stats.encoded_time = encoded_time_;
+  stats.target_encode_bitrate = target_encode_bitrate_;
+  stats.encoded_frame_type = frame_type_;
+  stats.encoded_image_size = encoded_image_size_;
+  stats.used_encoder = used_encoder_;
+  stats.spatial_layers_qp = spatial_layers_qp_;
+
+  absl::optional<ReceiverFrameStats> receiver_stats =
+      MaybeGetValue<ReceiverFrameStats>(receiver_stats_, peer);
+  if (receiver_stats.has_value()) {
+    stats.received_time = receiver_stats->received_time;
+    stats.decode_start_time = receiver_stats->decode_start_time;
+    stats.decode_end_time = receiver_stats->decode_end_time;
+    stats.rendered_time = receiver_stats->rendered_time;
+    stats.prev_frame_rendered_time = receiver_stats->prev_frame_rendered_time;
+    stats.decoded_frame_width = receiver_stats->decoded_frame_width;
+    stats.decoded_frame_height = receiver_stats->decoded_frame_height;
+    stats.used_decoder = receiver_stats->used_decoder;
+    stats.pre_decoded_frame_type = receiver_stats->frame_type;
+    stats.pre_decoded_image_size = receiver_stats->encoded_image_size;
+    stats.decoder_failed = receiver_stats->decoder_failed;
+  }
+  return stats;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h
new file mode 100644
index 0000000000..52a526d09b
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h
@@ -0,0 +1,169 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_FRAME_IN_FLIGHT_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_FRAME_IN_FLIGHT_H_
+
+#include <map>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_type.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h"
+
+namespace webrtc {
+
+struct ReceiverFrameStats {
+  // Time when last packet of a frame was received.
+  Timestamp received_time = Timestamp::MinusInfinity();
+  Timestamp decode_start_time = Timestamp::MinusInfinity();
+  Timestamp decode_end_time = Timestamp::MinusInfinity();
+  Timestamp rendered_time = Timestamp::MinusInfinity();
+  Timestamp prev_frame_rendered_time = Timestamp::MinusInfinity();
+
+  // Type and encoded size of received frame.
+  VideoFrameType frame_type = VideoFrameType::kEmptyFrame;
+  DataSize encoded_image_size = DataSize::Bytes(0);
+
+  absl::optional<int> decoded_frame_width = absl::nullopt;
+  absl::optional<int> decoded_frame_height = absl::nullopt;
+
+  // Can be not set if frame was dropped in the network.
+  absl::optional<StreamCodecInfo> used_decoder = absl::nullopt;
+
+  bool dropped = false;
+  bool decoder_failed = false;
+};
+
+// Represents a frame which was sent by sender and is currently on the way to
+// multiple receivers. Some receivers may receive this frame and some don't.
+//
+// Contains all statistic associated with the frame and gathered in multiple
+// points of the video pipeline.
+//
+// Internally may store the copy of the source frame which was sent. In such
+// case this frame is "alive".
+class FrameInFlight {
+ public:
+  FrameInFlight(size_t stream,
+                VideoFrame frame,
+                Timestamp captured_time,
+                std::set<size_t> expected_receivers);
+
+  size_t stream() const { return stream_; }
+  // Returns internal copy of source `VideoFrame` or `absl::nullopt` if it was
+  // removed before.
+  const absl::optional<VideoFrame>& frame() const { return frame_; }
+  // Removes internal copy of the source `VideoFrame` to free up extra memory.
+  // Returns was frame removed or not.
+  bool RemoveFrame();
+  void SetFrameId(uint16_t id);
+
+  void AddExpectedReceiver(size_t peer) { expected_receivers_.insert(peer); }
+
+  void RemoveExpectedReceiver(size_t peer) { expected_receivers_.erase(peer); }
+
+  std::vector<size_t> GetPeersWhichDidntReceive() const;
+
+  // Returns if all peers which were expected to receive this frame actually
+  // received it or not.
+  bool HaveAllPeersReceived() const;
+
+  void SetPreEncodeTime(webrtc::Timestamp time) { pre_encode_time_ = time; }
+
+  void OnFrameEncoded(webrtc::Timestamp time,
+                      VideoFrameType frame_type,
+                      DataSize encoded_image_size,
+                      uint32_t target_encode_bitrate,
+                      int spatial_layer,
+                      int qp,
+                      StreamCodecInfo used_encoder);
+
+  bool HasEncodedTime() const { return encoded_time_.IsFinite(); }
+
+  void OnFramePreDecode(size_t peer,
+                        webrtc::Timestamp received_time,
+                        webrtc::Timestamp decode_start_time,
+                        VideoFrameType frame_type,
+                        DataSize encoded_image_size);
+
+  bool HasReceivedTime(size_t peer) const;
+
+  void OnFrameDecoded(size_t peer,
+                      webrtc::Timestamp time,
+                      int width,
+                      int height,
+                      const StreamCodecInfo& used_decoder);
+  void OnDecoderError(size_t peer, const StreamCodecInfo& used_decoder);
+
+  bool HasDecodeEndTime(size_t peer) const;
+
+  void OnFrameRendered(size_t peer, webrtc::Timestamp time);
+
+  bool HasRenderedTime(size_t peer) const;
+
+  // Crash if rendered time is not set for specified `peer`.
+  webrtc::Timestamp rendered_time(size_t peer) const {
+    return receiver_stats_.at(peer).rendered_time;
+  }
+
+  // Marks that frame was dropped and wasn't seen by particular `peer`.
+  void MarkDropped(size_t peer) { receiver_stats_[peer].dropped = true; }
+  bool IsDropped(size_t peer) const;
+
+  void SetPrevFrameRenderedTime(size_t peer, webrtc::Timestamp time) {
+    receiver_stats_[peer].prev_frame_rendered_time = time;
+  }
+
+  FrameStats GetStatsForPeer(size_t peer) const;
+
+ private:
+  const size_t stream_;
+  // Set of peer's indexes who are expected to receive this frame. This is not
+  // the set of peer's indexes that received the frame. For example, if peer A
+  // was among expected receivers, it received frame and then left the call, A
+  // will be removed from this set, but the Stats for peer A still will be
+  // preserved in the FrameInFlight.
+  //
+  // This set is used to determine if this frame is expected to be received by
+  // any peer or can be safely deleted. It is responsibility of the user of this
+  // object to decide when it should be deleted.
+  std::set<size_t> expected_receivers_;
+  absl::optional<VideoFrame> frame_;
+  // Store frame id separately because `frame_` can be removed when we have too
+  // much memory consuption.
+  uint16_t frame_id_ = VideoFrame::kNotSetId;
+
+  // Frame events timestamp.
+  Timestamp captured_time_;
+  Timestamp pre_encode_time_ = Timestamp::MinusInfinity();
+  Timestamp encoded_time_ = Timestamp::MinusInfinity();
+  // Type and encoded size of sent frame.
+  VideoFrameType frame_type_ = VideoFrameType::kEmptyFrame;
+  DataSize encoded_image_size_ = DataSize::Bytes(0);
+  uint32_t target_encode_bitrate_ = 0;
+  // Sender side qp values per spatial layer. In case when spatial layer is not
+  // set for `webrtc::EncodedImage`, 0 is used as default.
+  std::map<int, SamplesStatsCounter> spatial_layers_qp_;
+  // Can be not set if frame was dropped by encoder.
+  absl::optional<StreamCodecInfo> used_encoder_ = absl::nullopt;
+  // Map from the receiver peer's index to frame stats for that peer.
+  std::map<size_t, ReceiverFrameStats> receiver_stats_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_FRAME_IN_FLIGHT_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
new file mode 100644
index 0000000000..cbc0b7e8f3
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
@@ -0,0 +1,575 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h"
+
+#include <algorithm>
+#include <map>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_type.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_tools/frame_analyzer/video_geometry_aligner.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::webrtc_pc_e2e::SampleMetadataKey;
+
+constexpr TimeDelta kFreezeThreshold = TimeDelta::Millis(150);
+constexpr int kMaxActiveComparisons = 10;
+
+SamplesStatsCounter::StatsSample StatsSample(
+    double value,
+    Timestamp sampling_time,
+    std::map<std::string, std::string> metadata) {
+  return SamplesStatsCounter::StatsSample{value, sampling_time,
+                                          std::move(metadata)};
+}
+
+SamplesStatsCounter::StatsSample StatsSample(
+    TimeDelta duration,
+    Timestamp sampling_time,
+    std::map<std::string, std::string> metadata) {
+  return SamplesStatsCounter::StatsSample{duration.ms<double>(), sampling_time,
+                                          std::move(metadata)};
+}
+
+FrameComparison ValidateFrameComparison(FrameComparison comparison) {
+  RTC_DCHECK(comparison.frame_stats.captured_time.IsFinite())
+      << "Any comparison has to have finite captured_time";
+  switch (comparison.type) {
+    case FrameComparisonType::kRegular:
+      // Regular comparison has to have all FrameStats filled in.
+      RTC_DCHECK(comparison.captured.has_value() ||
+                 comparison.overload_reason != OverloadReason::kNone)
+          << "Regular comparison has to have captured frame if it's not "
+          << "overloaded comparison";
+      RTC_DCHECK(comparison.rendered.has_value() ||
+                 comparison.overload_reason != OverloadReason::kNone)
+          << "rendered frame has to be presented if it's not overloaded "
+          << "comparison";
+      RTC_DCHECK(comparison.frame_stats.pre_encode_time.IsFinite())
+          << "Regular comparison has to have finite pre_encode_time";
+      RTC_DCHECK(comparison.frame_stats.encoded_time.IsFinite())
+          << "Regular comparison has to have finite encoded_time";
+      RTC_DCHECK(comparison.frame_stats.received_time.IsFinite())
+          << "Regular comparison has to have finite received_time";
+      RTC_DCHECK(comparison.frame_stats.decode_start_time.IsFinite())
+          << "Regular comparison has to have finite decode_start_time";
+      RTC_DCHECK(comparison.frame_stats.decode_end_time.IsFinite())
+          << "Regular comparison has to have finite decode_end_time";
+      RTC_DCHECK(comparison.frame_stats.rendered_time.IsFinite())
+          << "Regular comparison has to have finite rendered_time";
+      RTC_DCHECK(comparison.frame_stats.decoded_frame_width.has_value())
+          << "Regular comparison has to have decoded_frame_width";
+      RTC_DCHECK(comparison.frame_stats.decoded_frame_height.has_value())
+          << "Regular comparison has to have decoded_frame_height";
+      RTC_DCHECK(comparison.frame_stats.used_encoder.has_value())
+          << "Regular comparison has to have used_encoder";
+      RTC_DCHECK(comparison.frame_stats.used_decoder.has_value())
+          << "Regular comparison has to have used_decoder";
+      RTC_DCHECK(!comparison.frame_stats.decoder_failed)
+          << "Regular comparison can't have decoder failure";
+      break;
+    case FrameComparisonType::kDroppedFrame:
+      // Frame can be dropped before encoder, by encoder, inside network or
+      // after decoder.
+      RTC_DCHECK(!comparison.captured.has_value())
+          << "Dropped frame comparison can't have captured frame";
+      RTC_DCHECK(!comparison.rendered.has_value())
+          << "Dropped frame comparison can't have rendered frame";
+
+      if (comparison.frame_stats.encoded_time.IsFinite()) {
+        RTC_DCHECK(comparison.frame_stats.used_encoder.has_value())
+            << "Dropped frame comparison has to have used_encoder when "
+            << "encoded_time is set";
+        RTC_DCHECK(comparison.frame_stats.pre_encode_time.IsFinite())
+            << "Dropped frame comparison has to have finite pre_encode_time "
+            << "when encoded_time is finite.";
+      }
+
+      if (comparison.frame_stats.decode_end_time.IsFinite() ||
+          comparison.frame_stats.decoder_failed) {
+        RTC_DCHECK(comparison.frame_stats.received_time.IsFinite())
+            << "Dropped frame comparison has to have received_time when "
+            << "decode_end_time is set or decoder_failed is true";
+        RTC_DCHECK(comparison.frame_stats.decode_start_time.IsFinite())
+            << "Dropped frame comparison has to have decode_start_time when "
+            << "decode_end_time is set or decoder_failed is true";
+        RTC_DCHECK(comparison.frame_stats.used_decoder.has_value())
+            << "Dropped frame comparison has to have used_decoder when "
+            << "decode_end_time is set or decoder_failed is true";
+      } else if (comparison.frame_stats.decode_end_time.IsFinite()) {
+        RTC_DCHECK(comparison.frame_stats.decoded_frame_width.has_value())
+            << "Dropped frame comparison has to have decoded_frame_width when "
+            << "decode_end_time is set";
+        RTC_DCHECK(comparison.frame_stats.decoded_frame_height.has_value())
+            << "Dropped frame comparison has to have decoded_frame_height when "
+            << "decode_end_time is set";
+      }
+      RTC_DCHECK(!comparison.frame_stats.rendered_time.IsFinite())
+          << "Dropped frame comparison can't have rendered_time";
+      break;
+    case FrameComparisonType::kFrameInFlight:
+      // Frame in flight comparison may miss almost any FrameStats, but if
+      // stats for stage X are set, then stats for stage X - 1 also has to be
+      // set. Also these frames were never rendered.
+      RTC_DCHECK(!comparison.captured.has_value())
+          << "Frame in flight comparison can't have captured frame";
+      RTC_DCHECK(!comparison.rendered.has_value())
+          << "Frame in flight comparison can't have rendered frame";
+      RTC_DCHECK(!comparison.frame_stats.rendered_time.IsFinite())
+          << "Frame in flight comparison can't have rendered_time";
+
+      if (comparison.frame_stats.decode_end_time.IsFinite() ||
+          comparison.frame_stats.decoder_failed) {
+        RTC_DCHECK(comparison.frame_stats.used_decoder.has_value())
+            << "Frame in flight comparison has to have used_decoder when "
+            << "decode_end_time is set or decoder_failed is true.";
+        RTC_DCHECK(comparison.frame_stats.decode_start_time.IsFinite())
+            << "Frame in flight comparison has to have finite "
+            << "decode_start_time when decode_end_time is finite or "
+            << "decoder_failed is true.";
+      }
+      if (comparison.frame_stats.decode_end_time.IsFinite()) {
+        RTC_DCHECK(comparison.frame_stats.decoded_frame_width.has_value())
+            << "Frame in flight comparison has to have decoded_frame_width "
+            << "when decode_end_time is set.";
+        RTC_DCHECK(comparison.frame_stats.decoded_frame_height.has_value())
+            << "Frame in flight comparison has to have decoded_frame_height "
+            << "when decode_end_time is set.";
+      }
+      if (comparison.frame_stats.decode_start_time.IsFinite()) {
+        RTC_DCHECK(comparison.frame_stats.received_time.IsFinite())
+            << "Frame in flight comparison has to have finite received_time "
+            << "when decode_start_time is finite.";
+      }
+      if (comparison.frame_stats.received_time.IsFinite()) {
+        RTC_DCHECK(comparison.frame_stats.encoded_time.IsFinite())
+            << "Frame in flight comparison has to have finite encoded_time "
+            << "when received_time is finite.";
+      }
+      if (comparison.frame_stats.encoded_time.IsFinite()) {
+        RTC_DCHECK(comparison.frame_stats.used_encoder.has_value())
+            << "Frame in flight comparison has to have used_encoder when "
+            << "encoded_time is set";
+        RTC_DCHECK(comparison.frame_stats.pre_encode_time.IsFinite())
+            << "Frame in flight comparison has to have finite pre_encode_time "
+            << "when encoded_time is finite.";
+      }
+      break;
+  }
+  return comparison;
+}
+
+}  // namespace
+
+void DefaultVideoQualityAnalyzerFramesComparator::Start(int max_threads_count) {
+  for (int i = 0; i < max_threads_count; i++) {
+    thread_pool_.push_back(rtc::PlatformThread::SpawnJoinable(
+        [this] { ProcessComparisons(); },
+        "DefaultVideoQualityAnalyzerFramesComparator-" + std::to_string(i)));
+  }
+  {
+    MutexLock lock(&mutex_);
+    RTC_CHECK_EQ(state_, State::kNew) << "Frames comparator is already started";
+    state_ = State::kActive;
+  }
+  cpu_measurer_.StartMeasuringCpuProcessTime();
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::Stop(
+    const std::map<InternalStatsKey, Timestamp>& last_rendered_frame_times) {
+  {
+    MutexLock lock(&mutex_);
+    if (state_ == State::kStopped) {
+      return;
+    }
+    RTC_CHECK_EQ(state_, State::kActive)
+        << "Frames comparator has to be started before it will be used";
+    state_ = State::kStopped;
+  }
+  cpu_measurer_.StopMeasuringCpuProcessTime();
+  comparison_available_event_.Set();
+  thread_pool_.clear();
+
+  {
+    MutexLock lock(&mutex_);
+    // Perform final Metrics update. On this place analyzer is stopped and no
+    // one holds any locks.
+
+    // Time between freezes.
+    // Count time since the last freeze to the end of the call as time
+    // between freezes.
+    for (auto& entry : last_rendered_frame_times) {
+      const InternalStatsKey& stats_key = entry.first;
+      const Timestamp& last_rendered_frame_time = entry.second;
+
+      // If there are no freezes in the call we have to report
+      // time_between_freezes_ms as call duration and in such case
+      // `last_rendered_frame_time` for this stream will be stream start time.
+      // If there is freeze, then we need add time from last rendered frame
+      // to last freeze end as time between freezes.
+      stream_stats_.at(stats_key).time_between_freezes_ms.AddSample(StatsSample(
+          last_rendered_frame_time - stream_last_freeze_end_time_.at(stats_key),
+          Now(), /*metadata=*/{}));
+    }
+
+    // Freeze Time:
+    // If there were no freezes on a video stream, add only one sample with
+    // value 0 (0ms freezes time).
+    for (auto& [key, stream_stats] : stream_stats_) {
+      if (stream_stats.freeze_time_ms.IsEmpty()) {
+        stream_stats.freeze_time_ms.AddSample(0);
+      }
+    }
+  }
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::EnsureStatsForStream(
+    size_t stream_index,
+    size_t sender_peer_index,
+    size_t peers_count,
+    Timestamp captured_time,
+    Timestamp start_time) {
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "Frames comparator has to be started before it will be used";
+
+  for (size_t i = 0; i < peers_count; ++i) {
+    if (i == sender_peer_index && !options_.enable_receive_own_stream) {
+      continue;
+    }
+    InternalStatsKey stats_key(stream_index, sender_peer_index, i);
+    if (stream_stats_.find(stats_key) == stream_stats_.end()) {
+      stream_stats_.insert({stats_key, StreamStats(captured_time)});
+      // Assume that the first freeze was before first stream frame captured.
+      // This way time before the first freeze would be counted as time
+      // between freezes.
+      stream_last_freeze_end_time_.insert({stats_key, start_time});
+    } else {
+      // When we see some `stream_label` for the first time we need to create
+      // stream stats object for it and set up some states, but we need to do
+      // it only once and for all receivers, so on the next frame on the same
+      // `stream_label` we can be sure, that it's already done and we needn't
+      // to scan though all peers again.
+      break;
+    }
+  }
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::RegisterParticipantInCall(
+    rtc::ArrayView<std::pair<InternalStatsKey, Timestamp>> stream_started_time,
+    Timestamp start_time) {
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "Frames comparator has to be started before it will be used";
+
+  for (const std::pair<InternalStatsKey, Timestamp>& pair :
+       stream_started_time) {
+    stream_stats_.insert({pair.first, StreamStats(pair.second)});
+    stream_last_freeze_end_time_.insert({pair.first, start_time});
+  }
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::AddComparison(
+    InternalStatsKey stats_key,
+    absl::optional<VideoFrame> captured,
+    absl::optional<VideoFrame> rendered,
+    FrameComparisonType type,
+    FrameStats frame_stats) {
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "Frames comparator has to be started before it will be used";
+  AddComparisonInternal(std::move(stats_key), std::move(captured),
+                        std::move(rendered), type, std::move(frame_stats));
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::AddComparison(
+    InternalStatsKey stats_key,
+    int skipped_between_rendered,
+    absl::optional<VideoFrame> captured,
+    absl::optional<VideoFrame> rendered,
+    FrameComparisonType type,
+    FrameStats frame_stats) {
+  MutexLock lock(&mutex_);
+  RTC_CHECK_EQ(state_, State::kActive)
+      << "Frames comparator has to be started before it will be used";
+  stream_stats_.at(stats_key).skipped_between_rendered.AddSample(
+      StatsSample(skipped_between_rendered, Now(),
+                  /*metadata=*/
+                  {{SampleMetadataKey::kFrameIdMetadataKey,
+                    std::to_string(frame_stats.frame_id)}}));
+  AddComparisonInternal(std::move(stats_key), std::move(captured),
+                        std::move(rendered), type, std::move(frame_stats));
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::AddComparisonInternal(
+    InternalStatsKey stats_key,
+    absl::optional<VideoFrame> captured,
+    absl::optional<VideoFrame> rendered,
+    FrameComparisonType type,
+    FrameStats frame_stats) {
+  cpu_measurer_.StartExcludingCpuThreadTime();
+  frames_comparator_stats_.comparisons_queue_size.AddSample(
+      StatsSample(comparisons_.size(), Now(), /*metadata=*/{}));
+  // If there too many computations waiting in the queue, we won't provide
+  // frames itself to make future computations lighter.
+  if (comparisons_.size() >= kMaxActiveComparisons) {
+    comparisons_.emplace_back(ValidateFrameComparison(
+        FrameComparison(std::move(stats_key), /*captured=*/absl::nullopt,
+                        /*rendered=*/absl::nullopt, type,
+                        std::move(frame_stats), OverloadReason::kCpu)));
+  } else {
+    OverloadReason overload_reason = OverloadReason::kNone;
+    if (!captured && type == FrameComparisonType::kRegular) {
+      overload_reason = OverloadReason::kMemory;
+    }
+    comparisons_.emplace_back(ValidateFrameComparison(FrameComparison(
+        std::move(stats_key), std::move(captured), std::move(rendered), type,
+        std::move(frame_stats), overload_reason)));
+  }
+  comparison_available_event_.Set();
+  cpu_measurer_.StopExcludingCpuThreadTime();
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::ProcessComparisons() {
+  while (true) {
+    // Try to pick next comparison to perform from the queue.
+    absl::optional<FrameComparison> comparison = absl::nullopt;
+    bool more_new_comparisons_expected;
+    {
+      MutexLock lock(&mutex_);
+      if (!comparisons_.empty()) {
+        comparison = comparisons_.front();
+        comparisons_.pop_front();
+        if (!comparisons_.empty()) {
+          comparison_available_event_.Set();
+        }
+      }
+      // If state is stopped => no new frame comparisons are expected.
+      more_new_comparisons_expected = state_ != State::kStopped;
+    }
+    if (!comparison) {
+      if (!more_new_comparisons_expected) {
+        comparison_available_event_.Set();
+        return;
+      }
+      comparison_available_event_.Wait(TimeDelta::Seconds(1));
+      continue;
+    }
+
+    cpu_measurer_.StartExcludingCpuThreadTime();
+    ProcessComparison(comparison.value());
+    cpu_measurer_.StopExcludingCpuThreadTime();
+  }
+}
+
+void DefaultVideoQualityAnalyzerFramesComparator::ProcessComparison(
+    const FrameComparison& comparison) {
+  // Comparison is checked to be valid before adding, so we can use this
+  // assumptions during computations.
+
+  // Perform expensive psnr and ssim calculations while not holding lock.
+  double psnr = -1.0;
+  double ssim = -1.0;
+  if ((options_.compute_psnr || options_.compute_ssim) &&
+      comparison.captured.has_value() && comparison.rendered.has_value()) {
+    rtc::scoped_refptr<I420BufferInterface> reference_buffer =
+        comparison.captured->video_frame_buffer()->ToI420();
+    rtc::scoped_refptr<I420BufferInterface> test_buffer =
+        comparison.rendered->video_frame_buffer()->ToI420();
+    if (options_.adjust_cropping_before_comparing_frames) {
+      test_buffer = ScaleVideoFrameBuffer(
+          *test_buffer, reference_buffer->width(), reference_buffer->height());
+      reference_buffer = test::AdjustCropping(reference_buffer, test_buffer);
+    }
+    if (options_.compute_psnr) {
+      psnr = options_.use_weighted_psnr
+                 ? I420WeightedPSNR(*reference_buffer, *test_buffer)
+                 : I420PSNR(*reference_buffer, *test_buffer);
+    }
+    if (options_.compute_ssim) {
+      ssim = I420SSIM(*reference_buffer, *test_buffer);
+    }
+  }
+
+  const FrameStats& frame_stats = comparison.frame_stats;
+
+  MutexLock lock(&mutex_);
+  auto stats_it = stream_stats_.find(comparison.stats_key);
+  RTC_CHECK(stats_it != stream_stats_.end()) << comparison.stats_key.ToString();
+  StreamStats* stats = &stats_it->second;
+
+  frames_comparator_stats_.comparisons_done++;
+  if (comparison.overload_reason == OverloadReason::kCpu) {
+    frames_comparator_stats_.cpu_overloaded_comparisons_done++;
+  } else if (comparison.overload_reason == OverloadReason::kMemory) {
+    frames_comparator_stats_.memory_overloaded_comparisons_done++;
+  }
+
+  std::map<std::string, std::string> metadata;
+  metadata.emplace(SampleMetadataKey::kFrameIdMetadataKey,
+                   std::to_string(frame_stats.frame_id));
+
+  if (psnr > 0) {
+    stats->psnr.AddSample(
+        StatsSample(psnr, frame_stats.rendered_time, metadata));
+  }
+  if (ssim > 0) {
+    stats->ssim.AddSample(
+        StatsSample(ssim, frame_stats.received_time, metadata));
+  }
+  stats->capture_frame_rate.AddEvent(frame_stats.captured_time);
+
+  // Compute dropped phase for dropped frame
+  if (comparison.type == FrameComparisonType::kDroppedFrame) {
+    FrameDropPhase dropped_phase;
+    if (frame_stats.decode_end_time.IsFinite()) {
+      dropped_phase = FrameDropPhase::kAfterDecoder;
+    } else if (frame_stats.decode_start_time.IsFinite()) {
+      dropped_phase = FrameDropPhase::kByDecoder;
+    } else if (frame_stats.encoded_time.IsFinite()) {
+      dropped_phase = FrameDropPhase::kTransport;
+    } else if (frame_stats.pre_encode_time.IsFinite()) {
+      dropped_phase = FrameDropPhase::kByEncoder;
+    } else {
+      dropped_phase = FrameDropPhase::kBeforeEncoder;
+    }
+    stats->dropped_by_phase[dropped_phase]++;
+  }
+
+  if (frame_stats.encoded_time.IsFinite()) {
+    stats->encode_time_ms.AddSample(
+        StatsSample(frame_stats.encoded_time - frame_stats.pre_encode_time,
+                    frame_stats.encoded_time, metadata));
+    stats->encode_frame_rate.AddEvent(frame_stats.encoded_time);
+    stats->total_encoded_images_payload +=
+        frame_stats.encoded_image_size.bytes();
+    stats->target_encode_bitrate.AddSample(StatsSample(
+        frame_stats.target_encode_bitrate, frame_stats.encoded_time, metadata));
+    for (const auto& [spatial_layer, qp_values] :
+         frame_stats.spatial_layers_qp) {
+      for (SamplesStatsCounter::StatsSample qp : qp_values.GetTimedSamples()) {
+        qp.metadata = metadata;
+        stats->spatial_layers_qp[spatial_layer].AddSample(std::move(qp));
+      }
+    }
+
+    // Stats sliced on encoded frame type.
+    if (frame_stats.encoded_frame_type == VideoFrameType::kVideoFrameKey) {
+      ++stats->num_send_key_frames;
+    }
+  }
+  // Next stats can be calculated only if frame was received on remote side.
+  if (comparison.type != FrameComparisonType::kDroppedFrame ||
+      comparison.frame_stats.decoder_failed) {
+    if (frame_stats.rendered_time.IsFinite()) {
+      stats->total_delay_incl_transport_ms.AddSample(
+          StatsSample(frame_stats.rendered_time - frame_stats.captured_time,
+                      frame_stats.received_time, metadata));
+      stats->receive_to_render_time_ms.AddSample(
+          StatsSample(frame_stats.rendered_time - frame_stats.received_time,
+                      frame_stats.rendered_time, metadata));
+    }
+    if (frame_stats.decode_start_time.IsFinite()) {
+      stats->transport_time_ms.AddSample(
+          StatsSample(frame_stats.decode_start_time - frame_stats.encoded_time,
+                      frame_stats.decode_start_time, metadata));
+
+      // Stats sliced on decoded frame type.
+      if (frame_stats.pre_decoded_frame_type ==
+          VideoFrameType::kVideoFrameKey) {
+        ++stats->num_recv_key_frames;
+        stats->recv_key_frame_size_bytes.AddSample(
+            StatsSample(frame_stats.pre_decoded_image_size.bytes(),
+                        frame_stats.decode_start_time, metadata));
+      } else if (frame_stats.pre_decoded_frame_type ==
+                 VideoFrameType::kVideoFrameDelta) {
+        stats->recv_delta_frame_size_bytes.AddSample(
+            StatsSample(frame_stats.pre_decoded_image_size.bytes(),
+                        frame_stats.decode_start_time, metadata));
+      }
+    }
+    if (frame_stats.decode_end_time.IsFinite()) {
+      stats->decode_time_ms.AddSample(StatsSample(
+          frame_stats.decode_end_time - frame_stats.decode_start_time,
+          frame_stats.decode_end_time, metadata));
+      stats->resolution_of_decoded_frame.AddSample(
+          StatsSample(*comparison.frame_stats.decoded_frame_width *
+                          *comparison.frame_stats.decoded_frame_height,
+                      frame_stats.decode_end_time, metadata));
+    }
+
+    if (frame_stats.prev_frame_rendered_time.IsFinite() &&
+        frame_stats.rendered_time.IsFinite()) {
+      TimeDelta time_between_rendered_frames =
+          frame_stats.rendered_time - frame_stats.prev_frame_rendered_time;
+      stats->time_between_rendered_frames_ms.AddSample(StatsSample(
+          time_between_rendered_frames, frame_stats.rendered_time, metadata));
+      TimeDelta average_time_between_rendered_frames = TimeDelta::Millis(
+          stats->time_between_rendered_frames_ms.GetAverage());
+      if (time_between_rendered_frames >
+          std::max(kFreezeThreshold + average_time_between_rendered_frames,
+                   3 * average_time_between_rendered_frames)) {
+        stats->freeze_time_ms.AddSample(StatsSample(
+            time_between_rendered_frames, frame_stats.rendered_time, metadata));
+        auto freeze_end_it =
+            stream_last_freeze_end_time_.find(comparison.stats_key);
+        RTC_DCHECK(freeze_end_it != stream_last_freeze_end_time_.end());
+        stats->time_between_freezes_ms.AddSample(StatsSample(
+            frame_stats.prev_frame_rendered_time - freeze_end_it->second,
+            frame_stats.rendered_time, metadata));
+        freeze_end_it->second = frame_stats.rendered_time;
+      }
+    }
+  }
+  // Compute stream codec info.
+  if (frame_stats.used_encoder.has_value()) {
+    if (stats->encoders.empty() || stats->encoders.back().codec_name !=
+                                       frame_stats.used_encoder->codec_name) {
+      stats->encoders.push_back(*frame_stats.used_encoder);
+    }
+    stats->encoders.back().last_frame_id =
+        frame_stats.used_encoder->last_frame_id;
+    stats->encoders.back().switched_from_at =
+        frame_stats.used_encoder->switched_from_at;
+  }
+
+  if (frame_stats.used_decoder.has_value()) {
+    if (stats->decoders.empty() || stats->decoders.back().codec_name !=
+                                       frame_stats.used_decoder->codec_name) {
+      stats->decoders.push_back(*frame_stats.used_decoder);
+    }
+    stats->decoders.back().last_frame_id =
+        frame_stats.used_decoder->last_frame_id;
+    stats->decoders.back().switched_from_at =
+        frame_stats.used_decoder->switched_from_at;
+  }
+}
+
+Timestamp DefaultVideoQualityAnalyzerFramesComparator::Now() {
+  return clock_->CurrentTime();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
new file mode 100644
index 0000000000..006c3eb9bf
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h
@@ -0,0 +1,157 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_FRAMES_COMPARATOR_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_FRAMES_COMPARATOR_H_
+
+#include <deque>
+#include <map>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/event.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "system_wrappers/include/clock.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+
+namespace webrtc {
+
+struct FramesComparatorStats {
+  // Size of analyzer internal comparisons queue, measured when new element
+  // id added to the queue.
+  SamplesStatsCounter comparisons_queue_size;
+  // Number of performed comparisons of 2 video frames from captured and
+  // rendered streams.
+  int64_t comparisons_done = 0;
+  // Number of cpu overloaded comparisons. Comparison is cpu overloaded if it is
+  // queued when there are too many not processed comparisons in the queue.
+  // Overloaded comparison doesn't include metrics like SSIM and PSNR that
+  // require heavy computations.
+  int64_t cpu_overloaded_comparisons_done = 0;
+  // Number of memory overloaded comparisons. Comparison is memory overloaded if
+  // it is queued when its captured frame was already removed due to high memory
+  // usage for that video stream.
+  int64_t memory_overloaded_comparisons_done = 0;
+};
+
+// Performs comparisons of added frames and tracks frames related statistics.
+// This class is thread safe.
+class DefaultVideoQualityAnalyzerFramesComparator {
+ public:
+  // Creates frames comparator.
+  // Frames comparator doesn't use `options.enable_receive_own_stream` for any
+  // purposes, because it's unrelated to its functionality.
+  DefaultVideoQualityAnalyzerFramesComparator(
+      webrtc::Clock* clock,
+      DefaultVideoQualityAnalyzerCpuMeasurer& cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions options = {})
+      : options_(options), clock_(clock), cpu_measurer_(cpu_measurer) {}
+  ~DefaultVideoQualityAnalyzerFramesComparator() { Stop({}); }
+
+  // Starts frames comparator. This method must be invoked before calling
+  // any other method on this object.
+  void Start(int max_threads_count);
+  // Stops frames comparator. This method will block until all added frame
+  // comparisons will be processed. After `Stop()` is invoked no more new
+  // comparisons can be added to this frames comparator.
+  //
+  // `last_rendered_frame_time` contains timestamps of last rendered frame for
+  //     each (stream, sender, receiver) tuple to properly update time between
+  //     freezes: it has include time from the last freeze until and of call.
+  void Stop(
+      const std::map<InternalStatsKey, Timestamp>& last_rendered_frame_times);
+
+  // Ensures that stream `stream_index` has stats objects created for all
+  // potential receivers. This method must be called before adding any
+  // frames comparison for that stream.
+  void EnsureStatsForStream(size_t stream_index,
+                            size_t sender_peer_index,
+                            size_t peers_count,
+                            Timestamp captured_time,
+                            Timestamp start_time);
+  // Ensures that newly added participant will have stream stats objects created
+  // for all streams which they can receive. This method must be called before
+  // any frames comparison will be added for the newly added participant.
+  //
+  // `stream_started_time` - start time of each stream for which stats object
+  //     has to be created.
+  // `start_time` - call start time.
+  void RegisterParticipantInCall(
+      rtc::ArrayView<std::pair<InternalStatsKey, Timestamp>>
+          stream_started_time,
+      Timestamp start_time);
+
+  // `captured` - video frame captured by sender to use for PSNR/SSIM
+  //     computation. If `type` is `FrameComparisonType::kRegular` and
+  //     `captured` is `absl::nullopt` comparison is assumed to be overloaded
+  //     due to memory constraints.
+  // `rendered` - video frame rendered by receiver to use for PSNR/SSIM
+  //     computation. Required only if `type` is
+  //     `FrameComparisonType::kRegular`, but can still be omitted if
+  //     `captured` is `absl::nullopt`.
+  void AddComparison(InternalStatsKey stats_key,
+                     absl::optional<VideoFrame> captured,
+                     absl::optional<VideoFrame> rendered,
+                     FrameComparisonType type,
+                     FrameStats frame_stats);
+  // `skipped_between_rendered` - amount of frames dropped on this stream before
+  //     last received frame and current frame.
+  void AddComparison(InternalStatsKey stats_key,
+                     int skipped_between_rendered,
+                     absl::optional<VideoFrame> captured,
+                     absl::optional<VideoFrame> rendered,
+                     FrameComparisonType type,
+                     FrameStats frame_stats);
+
+  std::map<InternalStatsKey, StreamStats> stream_stats() const {
+    MutexLock lock(&mutex_);
+    return stream_stats_;
+  }
+  FramesComparatorStats frames_comparator_stats() const {
+    MutexLock lock(&mutex_);
+    return frames_comparator_stats_;
+  }
+
+ private:
+  enum State { kNew, kActive, kStopped };
+
+  void AddComparisonInternal(InternalStatsKey stats_key,
+                             absl::optional<VideoFrame> captured,
+                             absl::optional<VideoFrame> rendered,
+                             FrameComparisonType type,
+                             FrameStats frame_stats)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  void ProcessComparisons();
+  void ProcessComparison(const FrameComparison& comparison);
+  Timestamp Now();
+
+  const DefaultVideoQualityAnalyzerOptions options_;
+  webrtc::Clock* const clock_;
+  DefaultVideoQualityAnalyzerCpuMeasurer& cpu_measurer_;
+
+  mutable Mutex mutex_;
+  State state_ RTC_GUARDED_BY(mutex_) = State::kNew;
+  std::map<InternalStatsKey, StreamStats> stream_stats_ RTC_GUARDED_BY(mutex_);
+  std::map<InternalStatsKey, Timestamp> stream_last_freeze_end_time_
+      RTC_GUARDED_BY(mutex_);
+  std::deque<FrameComparison> comparisons_ RTC_GUARDED_BY(mutex_);
+  FramesComparatorStats frames_comparator_stats_ RTC_GUARDED_BY(mutex_);
+
+  std::vector<rtc::PlatformThread> thread_pool_;
+  rtc::Event comparison_available_event_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_FRAMES_COMPARATOR_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc
new file mode 100644
index 0000000000..8d3cd47ed6
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc
@@ -0,0 +1,1648 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.h"
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "api/test/create_frame_generator.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_cpu_measurer.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Contains;
+using ::testing::DoubleEq;
+using ::testing::Each;
+using ::testing::Eq;
+using ::testing::IsEmpty;
+using ::testing::Pair;
+using ::testing::SizeIs;
+
+using StatsSample = ::webrtc::SamplesStatsCounter::StatsSample;
+
+constexpr int kMaxFramesInFlightPerStream = 10;
+
+DefaultVideoQualityAnalyzerOptions AnalyzerOptionsForTest() {
+  DefaultVideoQualityAnalyzerOptions options;
+  options.compute_psnr = false;
+  options.compute_ssim = false;
+  options.adjust_cropping_before_comparing_frames = false;
+  options.max_frames_in_flight_per_stream_count = kMaxFramesInFlightPerStream;
+  return options;
+}
+
+VideoFrame CreateFrame(uint16_t frame_id,
+                       int width,
+                       int height,
+                       Timestamp timestamp) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(width, height,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+  test::FrameGeneratorInterface::VideoFrameData frame_data =
+      frame_generator->NextFrame();
+  return VideoFrame::Builder()
+      .set_id(frame_id)
+      .set_video_frame_buffer(frame_data.buffer)
+      .set_update_rect(frame_data.update_rect)
+      .set_timestamp_us(timestamp.us())
+      .build();
+}
+
+StreamCodecInfo Vp8CodecForOneFrame(uint16_t frame_id, Timestamp time) {
+  StreamCodecInfo info;
+  info.codec_name = "VP8";
+  info.first_frame_id = frame_id;
+  info.last_frame_id = frame_id;
+  info.switched_on_at = time;
+  info.switched_from_at = time;
+  return info;
+}
+
+FrameStats FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
+    uint16_t frame_id,
+    Timestamp captured_time) {
+  FrameStats frame_stats(frame_id, captured_time);
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+  // Decode time is in microseconds.
+  frame_stats.decode_end_time = captured_time + TimeDelta::Micros(40010);
+  frame_stats.rendered_time = captured_time + TimeDelta::Millis(60);
+  frame_stats.used_encoder = Vp8CodecForOneFrame(1, frame_stats.encoded_time);
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(1, frame_stats.decode_end_time);
+  frame_stats.decoded_frame_width = 10;
+  frame_stats.decoded_frame_height = 10;
+  return frame_stats;
+}
+
+FrameStats ShiftStatsOn(const FrameStats& stats, TimeDelta delta) {
+  FrameStats frame_stats(stats.frame_id, stats.captured_time + delta);
+  frame_stats.pre_encode_time = stats.pre_encode_time + delta;
+  frame_stats.encoded_time = stats.encoded_time + delta;
+  frame_stats.received_time = stats.received_time + delta;
+  frame_stats.decode_start_time = stats.decode_start_time + delta;
+  frame_stats.decode_end_time = stats.decode_end_time + delta;
+  frame_stats.rendered_time = stats.rendered_time + delta;
+
+  frame_stats.used_encoder = stats.used_encoder;
+  frame_stats.used_decoder = stats.used_decoder;
+  frame_stats.decoded_frame_width = stats.decoded_frame_width;
+  frame_stats.decoded_frame_height = stats.decoded_frame_height;
+
+  return frame_stats;
+}
+
+SamplesStatsCounter StatsCounter(
+    const std::vector<std::pair<double, Timestamp>>& samples) {
+  SamplesStatsCounter counter;
+  for (const std::pair<double, Timestamp>& sample : samples) {
+    counter.AddSample(SamplesStatsCounter::StatsSample{.value = sample.first,
+                                                       .time = sample.second});
+  }
+  return counter;
+}
+
+double GetFirstOrDie(const SamplesStatsCounter& counter) {
+  EXPECT_FALSE(counter.IsEmpty()) << "Counter has to be not empty";
+  return counter.GetSamples()[0];
+}
+
+void AssertFirstMetadataHasField(const SamplesStatsCounter& counter,
+                                 const std::string& field_name,
+                                 const std::string& field_value) {
+  EXPECT_FALSE(counter.IsEmpty()) << "Coutner has to be not empty";
+  EXPECT_THAT(counter.GetTimedSamples()[0].metadata,
+              Contains(Pair(field_name, field_value)));
+}
+
+std::string ToString(const SamplesStatsCounter& counter) {
+  rtc::StringBuilder out;
+  for (const StatsSample& s : counter.GetTimedSamples()) {
+    out << "{ time_ms=" << s.time.ms() << "; value=" << s.value << "}, ";
+  }
+  return out.str();
+}
+
+void ExpectEmpty(const SamplesStatsCounter& counter) {
+  EXPECT_TRUE(counter.IsEmpty())
+      << "Expected empty SamplesStatsCounter, but got " << ToString(counter);
+}
+
+void ExpectEmpty(const SamplesRateCounter& counter) {
+  EXPECT_TRUE(counter.IsEmpty())
+      << "Expected empty SamplesRateCounter, but got "
+      << counter.GetEventsPerSecond();
+}
+
+void ExpectSizeAndAllElementsAre(const SamplesStatsCounter& counter,
+                                 int size,
+                                 double value) {
+  EXPECT_EQ(counter.NumSamples(), size);
+  EXPECT_THAT(counter.GetSamples(), Each(DoubleEq(value)));
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     StatsPresentedAfterAddingOneComparison) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer, AnalyzerOptionsForTest());
+
+  Timestamp stream_start_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  size_t peers_count = 2;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  FrameStats frame_stats = FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
+      /*frame_id=*/1, stream_start_time);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, peers_count,
+                                  stream_start_time, stream_start_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kRegular, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  std::map<InternalStatsKey, StreamStats> stats = comparator.stream_stats();
+  ExpectSizeAndAllElementsAre(stats.at(stats_key).transport_time_ms, /*size=*/1,
+                              /*value=*/20.0);
+  ExpectSizeAndAllElementsAre(stats.at(stats_key).total_delay_incl_transport_ms,
+                              /*size=*/1, /*value=*/60.0);
+  ExpectSizeAndAllElementsAre(stats.at(stats_key).encode_time_ms, /*size=*/1,
+                              /*value=*/10.0);
+  ExpectSizeAndAllElementsAre(stats.at(stats_key).decode_time_ms, /*size=*/1,
+                              /*value=*/0.01);
+  ExpectSizeAndAllElementsAre(stats.at(stats_key).receive_to_render_time_ms,
+                              /*size=*/1, /*value=*/30.0);
+  ExpectSizeAndAllElementsAre(stats.at(stats_key).resolution_of_decoded_frame,
+                              /*size=*/1, /*value=*/100.0);
+}
+
+TEST(
+    DefaultVideoQualityAnalyzerFramesComparatorTest,
+    MultiFrameStatsPresentedWithMetadataAfterAddingTwoComparisonWith10msDelay) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer, AnalyzerOptionsForTest());
+
+  Timestamp stream_start_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  size_t peers_count = 2;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  FrameStats frame_stats1 = FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
+      /*frame_id=*/1, stream_start_time);
+  FrameStats frame_stats2 = FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
+      /*frame_id=*/2, stream_start_time + TimeDelta::Millis(15));
+  frame_stats2.prev_frame_rendered_time = frame_stats1.rendered_time;
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, peers_count,
+                                  stream_start_time, stream_start_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kRegular, frame_stats1);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kRegular, frame_stats2);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  std::map<InternalStatsKey, StreamStats> stats = comparator.stream_stats();
+  ExpectSizeAndAllElementsAre(
+      stats.at(stats_key).time_between_rendered_frames_ms, /*size=*/1,
+      /*value=*/15.0);
+  AssertFirstMetadataHasField(
+      stats.at(stats_key).time_between_rendered_frames_ms, "frame_id", "2");
+  EXPECT_DOUBLE_EQ(stats.at(stats_key).encode_frame_rate.GetEventsPerSecond(),
+                   2.0 / 15 * 1000)
+      << "There should be 2 events with interval of 15 ms";
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     FrameInFlightStatsAreHandledCorrectly) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer, AnalyzerOptionsForTest());
+
+  Timestamp stream_start_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  size_t peers_count = 2;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // There are 7 different timings inside frame stats: captured, pre_encode,
+  // encoded, received, decode_start, decode_end, rendered. captured is always
+  // set and received is set together with decode_start. So we create 6
+  // different frame stats with interval of 15 ms, where for each stat next
+  // timings will be set
+  //   * 1st - captured
+  //   * 2nd - captured, pre_encode
+  //   * 3rd - captured, pre_encode, encoded
+  //   * 4th - captured, pre_encode, encoded, received, decode_start
+  //   * 5th - captured, pre_encode, encoded, received, decode_start, decode_end
+  //   * 6th - all of them set
+  std::vector<FrameStats> stats;
+  // 1st stat
+  FrameStats frame_stats(/*frame_id=*/1, stream_start_time);
+  stats.push_back(frame_stats);
+  // 2nd stat
+  frame_stats = ShiftStatsOn(frame_stats, TimeDelta::Millis(15));
+  frame_stats.frame_id = 2;
+  frame_stats.pre_encode_time =
+      frame_stats.captured_time + TimeDelta::Millis(10);
+  stats.push_back(frame_stats);
+  // 3rd stat
+  frame_stats = ShiftStatsOn(frame_stats, TimeDelta::Millis(15));
+  frame_stats.frame_id = 3;
+  frame_stats.encoded_time = frame_stats.captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder = Vp8CodecForOneFrame(1, frame_stats.encoded_time);
+  stats.push_back(frame_stats);
+  // 4th stat
+  frame_stats = ShiftStatsOn(frame_stats, TimeDelta::Millis(15));
+  frame_stats.frame_id = 4;
+  frame_stats.received_time = frame_stats.captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time =
+      frame_stats.captured_time + TimeDelta::Millis(40);
+  stats.push_back(frame_stats);
+  // 5th stat
+  frame_stats = ShiftStatsOn(frame_stats, TimeDelta::Millis(15));
+  frame_stats.frame_id = 5;
+  frame_stats.decode_end_time =
+      frame_stats.captured_time + TimeDelta::Millis(50);
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(1, frame_stats.decode_end_time);
+  frame_stats.decoded_frame_width = 10;
+  frame_stats.decoded_frame_height = 10;
+  stats.push_back(frame_stats);
+  // 6th stat
+  frame_stats = ShiftStatsOn(frame_stats, TimeDelta::Millis(15));
+  frame_stats.frame_id = 6;
+  frame_stats.rendered_time = frame_stats.captured_time + TimeDelta::Millis(60);
+  stats.push_back(frame_stats);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, peers_count,
+                                  stream_start_time, stream_start_time);
+  for (size_t i = 0; i < stats.size() - 1; ++i) {
+    comparator.AddComparison(stats_key,
+                             /*captured=*/absl::nullopt,
+                             /*rendered=*/absl::nullopt,
+                             FrameComparisonType::kFrameInFlight, stats[i]);
+  }
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kRegular,
+                           stats[stats.size() - 1]);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats result_stats = comparator.stream_stats().at(stats_key);
+
+  EXPECT_DOUBLE_EQ(result_stats.transport_time_ms.GetAverage(), 20.0)
+      << ToString(result_stats.transport_time_ms);
+  EXPECT_EQ(result_stats.transport_time_ms.NumSamples(), 3);
+
+  EXPECT_DOUBLE_EQ(result_stats.total_delay_incl_transport_ms.GetAverage(),
+                   60.0)
+      << ToString(result_stats.total_delay_incl_transport_ms);
+  EXPECT_EQ(result_stats.total_delay_incl_transport_ms.NumSamples(), 1);
+
+  EXPECT_DOUBLE_EQ(result_stats.encode_time_ms.GetAverage(), 10)
+      << ToString(result_stats.encode_time_ms);
+  EXPECT_EQ(result_stats.encode_time_ms.NumSamples(), 4);
+
+  EXPECT_DOUBLE_EQ(result_stats.decode_time_ms.GetAverage(), 10)
+      << ToString(result_stats.decode_time_ms);
+  EXPECT_EQ(result_stats.decode_time_ms.NumSamples(), 2);
+
+  EXPECT_DOUBLE_EQ(result_stats.receive_to_render_time_ms.GetAverage(), 30)
+      << ToString(result_stats.receive_to_render_time_ms);
+  EXPECT_EQ(result_stats.receive_to_render_time_ms.NumSamples(), 1);
+
+  EXPECT_DOUBLE_EQ(result_stats.resolution_of_decoded_frame.GetAverage(), 100)
+      << ToString(result_stats.resolution_of_decoded_frame);
+  EXPECT_EQ(result_stats.resolution_of_decoded_frame.NumSamples(), 2);
+
+  EXPECT_DOUBLE_EQ(result_stats.encode_frame_rate.GetEventsPerSecond(),
+                   4.0 / 45 * 1000)
+      << "There should be 4 events with interval of 15 ms";
+}
+
+// Tests to validate that stats for each possible input frame are computed
+// correctly.
+// Frame in flight start
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     CapturedOnlyInFlightFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kFrameInFlight, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectEmpty(stats.encode_time_ms);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectEmpty(stats.target_encode_bitrate);
+  EXPECT_THAT(stats.spatial_layers_qp, IsEmpty());
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 0);
+  EXPECT_EQ(stats.num_send_key_frames, 0);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_THAT(stats.encoders, IsEmpty());
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     PreEncodedInFlightFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kFrameInFlight, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectEmpty(stats.encode_time_ms);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectEmpty(stats.target_encode_bitrate);
+  EXPECT_THAT(stats.spatial_layers_qp, IsEmpty());
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 0);
+  EXPECT_EQ(stats.num_send_key_frames, 0);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_THAT(stats.encoders, IsEmpty());
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     EncodedInFlightKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kFrameInFlight, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     EncodedInFlightDeltaFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameDelta;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kFrameInFlight, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 0);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     PreDecodedInFlightKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kFrameInFlight, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectSizeAndAllElementsAre(stats.transport_time_ms, /*size=*/1,
+                              /*value=*/20.0);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectSizeAndAllElementsAre(stats.recv_key_frame_size_bytes, /*size=*/1,
+                              /*value=*/500.0);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 1);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     DecodedInFlightKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+  // Frame decoded
+  frame_stats.decode_end_time = captured_time + TimeDelta::Millis(50);
+  frame_stats.decoded_frame_width = 200;
+  frame_stats.decoded_frame_height = 100;
+
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.decode_end_time);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kFrameInFlight, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectSizeAndAllElementsAre(stats.transport_time_ms, /*size=*/1,
+                              /*value=*/20.0);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectSizeAndAllElementsAre(stats.decode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  EXPECT_GE(GetFirstOrDie(stats.resolution_of_decoded_frame), 200 * 100.0);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectSizeAndAllElementsAre(stats.recv_key_frame_size_bytes, /*size=*/1,
+                              /*value=*/500.0);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 1);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_EQ(stats.decoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_decoder});
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     DecoderFailureOnInFlightKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+  // Frame decoded
+  frame_stats.decoder_failed = true;
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.decode_end_time);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kFrameInFlight, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectSizeAndAllElementsAre(stats.transport_time_ms, /*size=*/1,
+                              /*value=*/20.0);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectSizeAndAllElementsAre(stats.recv_key_frame_size_bytes, /*size=*/1,
+                              /*value=*/500.0);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 1);
+  // All frame in flight are not considered as dropped.
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_EQ(stats.decoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_decoder});
+}
+// Frame in flight end
+
+// Dropped frame start
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     CapturedOnlyDroppedFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kDroppedFrame, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectEmpty(stats.encode_time_ms);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectEmpty(stats.target_encode_bitrate);
+  EXPECT_THAT(stats.spatial_layers_qp, IsEmpty());
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 0);
+  EXPECT_EQ(stats.num_send_key_frames, 0);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 1},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_THAT(stats.encoders, IsEmpty());
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     PreEncodedDroppedFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kDroppedFrame, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectEmpty(stats.encode_time_ms);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectEmpty(stats.target_encode_bitrate);
+  EXPECT_THAT(stats.spatial_layers_qp, IsEmpty());
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 0);
+  EXPECT_EQ(stats.num_send_key_frames, 0);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 1},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_THAT(stats.encoders, IsEmpty());
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     EncodedDroppedKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kDroppedFrame, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 1},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     EncodedDroppedDeltaFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameDelta;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kDroppedFrame, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 0);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 1},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     PreDecodedDroppedKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kDroppedFrame, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 1},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_THAT(stats.decoders, IsEmpty());
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     DecodedDroppedKeyFrameAccountedInStats) {
+  // We don't really drop frames after decoder, so it's a bit unclear what is
+  // correct way to account such frames in stats, so this test just fixes some
+  // current way.
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+  // Frame decoded
+  frame_stats.decode_end_time = captured_time + TimeDelta::Millis(50);
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.decode_end_time);
+  frame_stats.decoded_frame_width = 200;
+  frame_stats.decoded_frame_height = 100;
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kDroppedFrame, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectEmpty(stats.transport_time_ms);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectEmpty(stats.recv_key_frame_size_bytes);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 0);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 1}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_EQ(stats.decoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_decoder});
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     DecoderFailedDroppedKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+  // Frame decoded
+  frame_stats.decoder_failed = true;
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.decode_end_time);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kDroppedFrame, frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  ExpectEmpty(stats.psnr);
+  ExpectEmpty(stats.ssim);
+  ExpectSizeAndAllElementsAre(stats.transport_time_ms, /*size=*/1,
+                              /*value=*/20.0);
+  ExpectEmpty(stats.total_delay_incl_transport_ms);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  ExpectEmpty(stats.decode_time_ms);
+  ExpectEmpty(stats.receive_to_render_time_ms);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  ExpectEmpty(stats.resolution_of_decoded_frame);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectSizeAndAllElementsAre(stats.recv_key_frame_size_bytes, /*size=*/1,
+                              /*value=*/500.0);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 1);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 1},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_EQ(stats.decoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_decoder});
+}
+// Dropped frame end
+
+// Regular frame start
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     RenderedKeyFrameAccountedInStats) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  VideoFrame frame =
+      CreateFrame(frame_id, /*width=*/320, /*height=*/180, captured_time);
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+  // Frame decoded
+  frame_stats.decode_end_time = captured_time + TimeDelta::Millis(50);
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.decode_end_time);
+  frame_stats.decoded_frame_width = 200;
+  frame_stats.decoded_frame_height = 100;
+  // Frame rendered
+  frame_stats.rendered_time = captured_time + TimeDelta::Millis(60);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/frame,
+                           /*rendered=*/frame, FrameComparisonType::kRegular,
+                           frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  EXPECT_EQ(stats.stream_started_time, captured_time);
+  EXPECT_GE(GetFirstOrDie(stats.psnr), 20);
+  EXPECT_GE(GetFirstOrDie(stats.ssim), 0.5);
+  ExpectSizeAndAllElementsAre(stats.transport_time_ms, /*size=*/1,
+                              /*value=*/20.0);
+  EXPECT_GE(GetFirstOrDie(stats.total_delay_incl_transport_ms), 60.0);
+  ExpectEmpty(stats.time_between_rendered_frames_ms);
+  ExpectEmpty(stats.encode_frame_rate);
+  ExpectSizeAndAllElementsAre(stats.encode_time_ms, /*size=*/1, /*value=*/10.0);
+  EXPECT_GE(GetFirstOrDie(stats.decode_time_ms), 10.0);
+  EXPECT_GE(GetFirstOrDie(stats.receive_to_render_time_ms), 30.0);
+  ExpectEmpty(stats.skipped_between_rendered);
+  ExpectSizeAndAllElementsAre(stats.freeze_time_ms, /*size=*/1, /*value=*/0);
+  ExpectEmpty(stats.time_between_freezes_ms);
+  EXPECT_GE(GetFirstOrDie(stats.resolution_of_decoded_frame), 200 * 100.0);
+  ExpectSizeAndAllElementsAre(stats.target_encode_bitrate, /*size=*/1,
+                              /*value=*/2000.0);
+  EXPECT_THAT(stats.spatial_layers_qp, SizeIs(1));
+  ExpectSizeAndAllElementsAre(stats.spatial_layers_qp[0], /*size=*/2,
+                              /*value=*/5.0);
+  ExpectSizeAndAllElementsAre(stats.recv_key_frame_size_bytes, /*size=*/1,
+                              /*value=*/500.0);
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+  EXPECT_EQ(stats.total_encoded_images_payload, 1000);
+  EXPECT_EQ(stats.num_send_key_frames, 1);
+  EXPECT_EQ(stats.num_recv_key_frames, 1);
+  EXPECT_THAT(stats.dropped_by_phase, Eq(std::map<FrameDropPhase, int64_t>{
+                                          {FrameDropPhase::kBeforeEncoder, 0},
+                                          {FrameDropPhase::kByEncoder, 0},
+                                          {FrameDropPhase::kTransport, 0},
+                                          {FrameDropPhase::kByDecoder, 0},
+                                          {FrameDropPhase::kAfterDecoder, 0}}));
+  EXPECT_EQ(stats.encoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_encoder});
+  EXPECT_EQ(stats.decoders,
+            std::vector<StreamCodecInfo>{*frame_stats.used_decoder});
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest, AllStatsHaveMetadataSet) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer,
+      DefaultVideoQualityAnalyzerOptions());
+
+  Timestamp captured_time = Clock::GetRealTimeClock()->CurrentTime();
+  uint16_t frame_id = 1;
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  // Frame captured
+  VideoFrame frame =
+      CreateFrame(frame_id, /*width=*/320, /*height=*/180, captured_time);
+  FrameStats frame_stats(/*frame_id=*/1, captured_time);
+  // Frame pre encoded
+  frame_stats.pre_encode_time = captured_time + TimeDelta::Millis(10);
+  // Frame encoded
+  frame_stats.encoded_time = captured_time + TimeDelta::Millis(20);
+  frame_stats.used_encoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.encoded_time);
+  frame_stats.encoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.encoded_image_size = DataSize::Bytes(1000);
+  frame_stats.target_encode_bitrate = 2000;
+  frame_stats.spatial_layers_qp = {
+      {0, StatsCounter(
+              /*samples=*/{{5, Timestamp::Seconds(1)},
+                           {5, Timestamp::Seconds(2)}})}};
+  // Frame pre decoded
+  frame_stats.pre_decoded_frame_type = VideoFrameType::kVideoFrameKey;
+  frame_stats.pre_decoded_image_size = DataSize::Bytes(500);
+  frame_stats.received_time = captured_time + TimeDelta::Millis(30);
+  frame_stats.decode_start_time = captured_time + TimeDelta::Millis(40);
+  // Frame decoded
+  frame_stats.decode_end_time = captured_time + TimeDelta::Millis(50);
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(frame_id, frame_stats.decode_end_time);
+  // Frame rendered
+  frame_stats.rendered_time = captured_time + TimeDelta::Millis(60);
+  frame_stats.decoded_frame_width = 200;
+  frame_stats.decoded_frame_height = 100;
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, /*peers_count=*/2,
+                                  captured_time, captured_time);
+  comparator.AddComparison(stats_key,
+                           /*captured=*/frame,
+                           /*rendered=*/frame, FrameComparisonType::kRegular,
+                           frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  EXPECT_EQ(comparator.stream_stats().size(), 1lu);
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  AssertFirstMetadataHasField(stats.psnr, "frame_id", "1");
+  AssertFirstMetadataHasField(stats.ssim, "frame_id", "1");
+  AssertFirstMetadataHasField(stats.transport_time_ms, "frame_id", "1");
+  AssertFirstMetadataHasField(stats.total_delay_incl_transport_ms, "frame_id",
+                              "1");
+  AssertFirstMetadataHasField(stats.encode_time_ms, "frame_id", "1");
+  AssertFirstMetadataHasField(stats.decode_time_ms, "frame_id", "1");
+  AssertFirstMetadataHasField(stats.receive_to_render_time_ms, "frame_id", "1");
+  AssertFirstMetadataHasField(stats.resolution_of_decoded_frame, "frame_id",
+                              "1");
+  AssertFirstMetadataHasField(stats.target_encode_bitrate, "frame_id", "1");
+  AssertFirstMetadataHasField(stats.spatial_layers_qp[0], "frame_id", "1");
+  AssertFirstMetadataHasField(stats.recv_key_frame_size_bytes, "frame_id", "1");
+
+  ExpectEmpty(stats.recv_delta_frame_size_bytes);
+}
+// Regular frame end
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     FreezeStatsPresentedWithMetadataAfterAddFrameWithSkippedAndDelay) {
+  DefaultVideoQualityAnalyzerCpuMeasurer cpu_measurer;
+  DefaultVideoQualityAnalyzerFramesComparator comparator(
+      Clock::GetRealTimeClock(), cpu_measurer, AnalyzerOptionsForTest());
+
+  Timestamp stream_start_time = Clock::GetRealTimeClock()->CurrentTime();
+  size_t stream = 0;
+  size_t sender = 0;
+  size_t receiver = 1;
+  size_t peers_count = 2;
+  InternalStatsKey stats_key(stream, sender, receiver);
+
+  comparator.Start(/*max_threads_count=*/1);
+  comparator.EnsureStatsForStream(stream, sender, peers_count,
+                                  stream_start_time, stream_start_time);
+
+  // Add 5 frames which were rendered with 30 fps (~30ms between frames)
+  // Frame ids are in [1..5] and last frame is with 120ms offset from first.
+  Timestamp prev_frame_rendered_time = Timestamp::MinusInfinity();
+  for (int i = 0; i < 5; ++i) {
+    FrameStats frame_stats = FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
+        /*frame_id=*/i + 1, stream_start_time + TimeDelta::Millis(30 * i));
+    frame_stats.prev_frame_rendered_time = prev_frame_rendered_time;
+    prev_frame_rendered_time = frame_stats.rendered_time;
+
+    comparator.AddComparison(stats_key,
+                             /*captured=*/absl::nullopt,
+                             /*rendered=*/absl::nullopt,
+                             FrameComparisonType::kRegular, frame_stats);
+  }
+
+  // Next frame was rendered with 4 frames skipped and delay 300ms after last
+  // frame.
+  FrameStats freeze_frame_stats =
+      FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
+          /*frame_id=*/10, stream_start_time + TimeDelta::Millis(120 + 300));
+  freeze_frame_stats.prev_frame_rendered_time = prev_frame_rendered_time;
+
+  comparator.AddComparison(stats_key,
+                           /*skipped_between_rendered=*/4,
+                           /*captured=*/absl::nullopt,
+                           /*rendered=*/absl::nullopt,
+                           FrameComparisonType::kRegular, freeze_frame_stats);
+  comparator.Stop(/*last_rendered_frame_times=*/{});
+
+  StreamStats stats = comparator.stream_stats().at(stats_key);
+  ASSERT_THAT(GetFirstOrDie(stats.skipped_between_rendered), Eq(4));
+  AssertFirstMetadataHasField(stats.skipped_between_rendered, "frame_id", "10");
+  ASSERT_THAT(GetFirstOrDie(stats.freeze_time_ms), Eq(300));
+  AssertFirstMetadataHasField(stats.freeze_time_ms, "frame_id", "10");
+  // 180ms is time from the stream start to the rendered time of the last frame
+  // among first 5 frames which were received before freeze.
+  ASSERT_THAT(GetFirstOrDie(stats.time_between_freezes_ms), Eq(180));
+  AssertFirstMetadataHasField(stats.time_between_freezes_ms, "frame_id", "10");
+}
+// Stats validation tests end.
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc
new file mode 100644
index 0000000000..16f49ef154
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h"
+
+#include "api/video/video_frame.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+std::string InternalStatsKey::ToString() const {
+  rtc::StringBuilder out;
+  out << "stream=" << stream << "_sender=" << sender
+      << "_receiver=" << receiver;
+  return out.str();
+}
+
+bool operator<(const InternalStatsKey& a, const InternalStatsKey& b) {
+  if (a.stream != b.stream) {
+    return a.stream < b.stream;
+  }
+  if (a.sender != b.sender) {
+    return a.sender < b.sender;
+  }
+  return a.receiver < b.receiver;
+}
+
+bool operator==(const InternalStatsKey& a, const InternalStatsKey& b) {
+  return a.stream == b.stream && a.sender == b.sender &&
+         a.receiver == b.receiver;
+}
+
+FrameComparison::FrameComparison(InternalStatsKey stats_key,
+                                 absl::optional<VideoFrame> captured,
+                                 absl::optional<VideoFrame> rendered,
+                                 FrameComparisonType type,
+                                 FrameStats frame_stats,
+                                 OverloadReason overload_reason)
+    : stats_key(std::move(stats_key)),
+      captured(std::move(captured)),
+      rendered(std::move(rendered)),
+      type(type),
+      frame_stats(std::move(frame_stats)),
+      overload_reason(overload_reason) {}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h
new file mode 100644
index 0000000000..10f1314f46
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h
@@ -0,0 +1,132 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_INTERNAL_SHARED_OBJECTS_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_INTERNAL_SHARED_OBJECTS_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_type.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+
+namespace webrtc {
+
+struct InternalStatsKey {
+  InternalStatsKey(size_t stream, size_t sender, size_t receiver)
+      : stream(stream), sender(sender), receiver(receiver) {}
+
+  std::string ToString() const;
+
+  size_t stream;
+  size_t sender;
+  size_t receiver;
+};
+
+// Required to use InternalStatsKey as std::map key.
+bool operator<(const InternalStatsKey& a, const InternalStatsKey& b);
+bool operator==(const InternalStatsKey& a, const InternalStatsKey& b);
+
+// Final stats computed for frame after it went through the whole video
+// pipeline from capturing to rendering or dropping.
+struct FrameStats {
+  FrameStats(uint16_t frame_id, Timestamp captured_time)
+      : frame_id(frame_id), captured_time(captured_time) {}
+
+  uint16_t frame_id;
+  // Frame events timestamp.
+  Timestamp captured_time;
+  Timestamp pre_encode_time = Timestamp::MinusInfinity();
+  Timestamp encoded_time = Timestamp::MinusInfinity();
+  // Time when last packet of a frame was received.
+  Timestamp received_time = Timestamp::MinusInfinity();
+  Timestamp decode_start_time = Timestamp::MinusInfinity();
+  Timestamp decode_end_time = Timestamp::MinusInfinity();
+  Timestamp rendered_time = Timestamp::MinusInfinity();
+  Timestamp prev_frame_rendered_time = Timestamp::MinusInfinity();
+
+  VideoFrameType encoded_frame_type = VideoFrameType::kEmptyFrame;
+  DataSize encoded_image_size = DataSize::Bytes(0);
+  VideoFrameType pre_decoded_frame_type = VideoFrameType::kEmptyFrame;
+  DataSize pre_decoded_image_size = DataSize::Bytes(0);
+  uint32_t target_encode_bitrate = 0;
+  // Sender side qp values per spatial layer. In case when spatial layer is not
+  // set for `webrtc::EncodedImage`, 0 is used as default.
+  std::map<int, SamplesStatsCounter> spatial_layers_qp;
+
+  absl::optional<int> decoded_frame_width = absl::nullopt;
+  absl::optional<int> decoded_frame_height = absl::nullopt;
+
+  // Can be not set if frame was dropped by encoder.
+  absl::optional<StreamCodecInfo> used_encoder = absl::nullopt;
+  // Can be not set if frame was dropped in the network.
+  absl::optional<StreamCodecInfo> used_decoder = absl::nullopt;
+
+  bool decoder_failed = false;
+};
+
+// Describes why comparison was done in overloaded mode (without calculating
+// PSNR and SSIM).
+enum class OverloadReason {
+  kNone,
+  // Not enough CPU to process all incoming comparisons.
+  kCpu,
+  // Not enough memory to store captured frames for all comparisons.
+  kMemory
+};
+
+enum class FrameComparisonType {
+  // Comparison for captured and rendered frame.
+  kRegular,
+  // Comparison for captured frame that is known to be dropped somewhere in
+  // video pipeline.
+  kDroppedFrame,
+  // Comparison for captured frame that was still in the video pipeline when
+  // test was stopped. It's unknown is this frame dropped or would it be
+  // delivered if test continue.
+  kFrameInFlight
+};
+
+// Represents comparison between two VideoFrames. Contains video frames itself
+// and stats. Can be one of two types:
+//   1. Normal - in this case `captured` is presented and either `rendered` is
+//      presented and `dropped` is false, either `rendered` is omitted and
+//      `dropped` is true.
+//   2. Overloaded - in this case both `captured` and `rendered` are omitted
+//      because there were too many comparisons in the queue. `dropped` can be
+//      true or false showing was frame dropped or not.
+struct FrameComparison {
+  FrameComparison(InternalStatsKey stats_key,
+                  absl::optional<VideoFrame> captured,
+                  absl::optional<VideoFrame> rendered,
+                  FrameComparisonType type,
+                  FrameStats frame_stats,
+                  OverloadReason overload_reason);
+
+  InternalStatsKey stats_key;
+  // Frames can be omitted if there too many computations waiting in the
+  // queue.
+  absl::optional<VideoFrame> captured;
+  absl::optional<VideoFrame> rendered;
+  FrameComparisonType type;
+  FrameStats frame_stats;
+  OverloadReason overload_reason;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_INTERNAL_SHARED_OBJECTS_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc
new file mode 100644
index 0000000000..f5029ac956
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_metric_names_test.cc
@@ -0,0 +1,682 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/rtp_packet_info.h"
+#include "api/rtp_packet_infos.h"
+#include "api/test/create_frame_generator.h"
+#include "api/test/metrics/metric.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/metrics/stdout_metrics_exporter.h"
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_tools/frame_analyzer/video_geometry_aligner.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Contains;
+using ::testing::SizeIs;
+using ::testing::UnorderedElementsAre;
+
+using ::webrtc::test::DefaultMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Metric;
+using ::webrtc::test::MetricsExporter;
+using ::webrtc::test::StdoutMetricsExporter;
+using ::webrtc::test::Unit;
+
+constexpr int kAnalyzerMaxThreadsCount = 1;
+constexpr int kMaxFramesInFlightPerStream = 10;
+constexpr int kFrameWidth = 320;
+constexpr int kFrameHeight = 240;
+
+DefaultVideoQualityAnalyzerOptions AnalyzerOptionsForTest() {
+  DefaultVideoQualityAnalyzerOptions options;
+  options.compute_psnr = true;
+  options.compute_ssim = true;
+  options.adjust_cropping_before_comparing_frames = false;
+  options.max_frames_in_flight_per_stream_count = kMaxFramesInFlightPerStream;
+  options.report_detailed_frame_stats = true;
+  return options;
+}
+
+VideoFrame NextFrame(test::FrameGeneratorInterface* frame_generator,
+                     int64_t timestamp_us) {
+  test::FrameGeneratorInterface::VideoFrameData frame_data =
+      frame_generator->NextFrame();
+  return VideoFrame::Builder()
+      .set_video_frame_buffer(frame_data.buffer)
+      .set_update_rect(frame_data.update_rect)
+      .set_timestamp_us(timestamp_us)
+      .build();
+}
+
+EncodedImage FakeEncode(const VideoFrame& frame) {
+  EncodedImage image;
+  std::vector<RtpPacketInfo> packet_infos;
+  packet_infos.push_back(RtpPacketInfo(
+      /*ssrc=*/1,
+      /*csrcs=*/{},
+      /*rtp_timestamp=*/frame.timestamp(),
+      /*receive_time=*/Timestamp::Micros(frame.timestamp_us() + 10000)));
+  image.SetPacketInfos(RtpPacketInfos(packet_infos));
+  return image;
+}
+
+VideoFrame DeepCopy(const VideoFrame& frame) {
+  VideoFrame copy = frame;
+  copy.set_video_frame_buffer(
+      I420Buffer::Copy(*frame.video_frame_buffer()->ToI420()));
+  return copy;
+}
+
+void PassFramesThroughAnalyzer(DefaultVideoQualityAnalyzer& analyzer,
+                               absl::string_view sender,
+                               absl::string_view stream_label,
+                               std::vector<absl::string_view> receivers,
+                               int frames_count,
+                               test::FrameGeneratorInterface& frame_generator,
+                               int interframe_delay_ms = 0) {
+  for (int i = 0; i < frames_count; ++i) {
+    VideoFrame frame = NextFrame(&frame_generator, /*timestamp_us=*/1);
+    uint16_t frame_id =
+        analyzer.OnFrameCaptured(sender, std::string(stream_label), frame);
+    frame.set_id(frame_id);
+    analyzer.OnFramePreEncode(sender, frame);
+    analyzer.OnFrameEncoded(sender, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+    for (absl::string_view receiver : receivers) {
+      VideoFrame received_frame = DeepCopy(frame);
+      analyzer.OnFramePreDecode(receiver, received_frame.id(),
+                                FakeEncode(received_frame));
+      analyzer.OnFrameDecoded(receiver, received_frame,
+                              VideoQualityAnalyzerInterface::DecoderStats());
+      analyzer.OnFrameRendered(receiver, received_frame);
+    }
+    if (i < frames_count - 1 && interframe_delay_ms > 0) {
+      SleepMs(interframe_delay_ms);
+    }
+  }
+}
+
+// Metric fields to assert on
+struct MetricValidationInfo {
+  std::string test_case;
+  std::string name;
+  Unit unit;
+  ImprovementDirection improvement_direction;
+};
+
+bool operator==(const MetricValidationInfo& a, const MetricValidationInfo& b) {
+  return a.name == b.name && a.test_case == b.test_case && a.unit == b.unit &&
+         a.improvement_direction == b.improvement_direction;
+}
+
+std::ostream& operator<<(std::ostream& os, const MetricValidationInfo& m) {
+  os << "{ test_case=" << m.test_case << "; name=" << m.name
+     << "; unit=" << test::ToString(m.unit)
+     << "; improvement_direction=" << test::ToString(m.improvement_direction)
+     << " }";
+  return os;
+}
+
+std::vector<MetricValidationInfo> ToValidationInfo(
+    const std::vector<Metric>& metrics) {
+  std::vector<MetricValidationInfo> out;
+  for (const Metric& m : metrics) {
+    out.push_back(
+        MetricValidationInfo{.test_case = m.test_case,
+                             .name = m.name,
+                             .unit = m.unit,
+                             .improvement_direction = m.improvement_direction});
+  }
+  return out;
+}
+
+std::vector<std::string> ToTestCases(const std::vector<Metric>& metrics) {
+  std::vector<std::string> out;
+  for (const Metric& m : metrics) {
+    out.push_back(m.test_case);
+  }
+  return out;
+}
+
+TEST(DefaultVideoQualityAnalyzerMetricNamesTest, MetricNamesForP2PAreCorrect) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultMetricsLogger metrics_logger(Clock::GetRealTimeClock());
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       &metrics_logger, options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {"bob"},
+                            /*frames_count=*/5, *frame_generator,
+                            /*interframe_delay_ms=*/50);
+  analyzer.Stop();
+
+  std::vector<MetricValidationInfo> metrics =
+      ToValidationInfo(metrics_logger.GetCollectedMetrics());
+  EXPECT_THAT(
+      metrics,
+      UnorderedElementsAre(
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "ssim",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "transport_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "total_delay_incl_transport",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "time_between_rendered_frames",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "harmonic_framerate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "encode_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "encode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "time_between_freezes",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "freeze_time_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "pixels_per_frame",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "min_psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "decode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "receive_to_render_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "dropped_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "frames_in_flight",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "rendered_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "max_skipped",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "target_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "qp_sl0",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "actual_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "capture_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "num_encoded_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "num_decoded_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "num_send_key_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "num_recv_key_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "recv_key_frame_size_bytes",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "recv_delta_frame_size_bytes",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{.test_case = "test_case",
+                               .name = "cpu_usage_%",
+                               .unit = Unit::kUnitless,
+                               .improvement_direction =
+                                   ImprovementDirection::kSmallerIsBetter}));
+}
+
+TEST(DefaultVideoQualityAnalyzerMetricNamesTest,
+     MetricNamesFor3PeersAreCorrect) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultMetricsLogger metrics_logger(Clock::GetRealTimeClock());
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       &metrics_logger, options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{"alice", "bob", "charlie"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video",
+                            {"bob", "charlie"},
+                            /*frames_count=*/5, *frame_generator,
+                            /*interframe_delay_ms=*/50);
+  analyzer.Stop();
+
+  std::vector<MetricValidationInfo> metrics =
+      ToValidationInfo(metrics_logger.GetCollectedMetrics());
+  EXPECT_THAT(
+      metrics,
+      UnorderedElementsAre(
+          // Bob
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "ssim",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "transport_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "total_delay_incl_transport",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "time_between_rendered_frames",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "harmonic_framerate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "encode_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "encode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "time_between_freezes",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "freeze_time_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "pixels_per_frame",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "min_psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "decode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "receive_to_render_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "dropped_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "frames_in_flight",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "rendered_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "max_skipped",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "target_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "qp_sl0",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "actual_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "capture_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "num_encoded_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "num_decoded_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "num_send_key_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "num_recv_key_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "recv_key_frame_size_bytes",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_bob",
+              .name = "recv_delta_frame_size_bytes",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+
+          // Charlie
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "ssim",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "transport_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "total_delay_incl_transport",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "time_between_rendered_frames",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "harmonic_framerate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "encode_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "encode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "time_between_freezes",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "freeze_time_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "pixels_per_frame",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "min_psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "decode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "receive_to_render_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "dropped_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "frames_in_flight",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "rendered_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "max_skipped",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "target_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "qp_sl0",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "actual_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "capture_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "num_encoded_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "num_decoded_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "num_send_key_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "num_recv_key_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "recv_key_frame_size_bytes",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video_alice_charlie",
+              .name = "recv_delta_frame_size_bytes",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter},
+          MetricValidationInfo{.test_case = "test_case",
+                               .name = "cpu_usage_%",
+                               .unit = Unit::kUnitless,
+                               .improvement_direction =
+                                   ImprovementDirection::kSmallerIsBetter}));
+}
+
+TEST(DefaultVideoQualityAnalyzerMetricNamesTest,
+     TestCaseFor3PeerIsTheSameAfterAllPeersLeft) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultMetricsLogger metrics_logger(Clock::GetRealTimeClock());
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       &metrics_logger, options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{"alice", "bob", "charlie"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video",
+                            {"bob", "charlie"},
+                            /*frames_count=*/5, *frame_generator,
+                            /*interframe_delay_ms=*/50);
+  analyzer.UnregisterParticipantInCall("alice");
+  analyzer.UnregisterParticipantInCall("bob");
+  analyzer.UnregisterParticipantInCall("charlie");
+  analyzer.Stop();
+
+  std::vector<std::string> metrics =
+      ToTestCases(metrics_logger.GetCollectedMetrics());
+  EXPECT_THAT(metrics, SizeIs(57));
+  EXPECT_THAT(metrics, Contains("test_case/alice_video_alice_bob").Times(28));
+  EXPECT_THAT(metrics,
+              Contains("test_case/alice_video_alice_charlie").Times(28));
+  EXPECT_THAT(metrics, Contains("test_case").Times(1));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc
new file mode 100644
index 0000000000..79b9286e2d
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc
@@ -0,0 +1,172 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+
+#include <algorithm>
+#include <iterator>
+#include <ostream>
+#include <string>
+
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kMicrosPerSecond = 1000000;
+
+}  // namespace
+
+std::string StreamCodecInfo::ToString() const {
+  rtc::StringBuilder out;
+  out << "{codec_name=" << codec_name << "; first_frame_id=" << first_frame_id
+      << "; last_frame_id=" << last_frame_id
+      << "; switched_on_at=" << webrtc::ToString(switched_on_at)
+      << "; switched_from_at=" << webrtc::ToString(switched_from_at) << " }";
+  return out.str();
+}
+
+std::ostream& operator<<(std::ostream& os, const StreamCodecInfo& state) {
+  return os << state.ToString();
+}
+
+rtc::StringBuilder& operator<<(rtc::StringBuilder& sb,
+                               const StreamCodecInfo& state) {
+  return sb << state.ToString();
+}
+
+bool operator==(const StreamCodecInfo& a, const StreamCodecInfo& b) {
+  return a.codec_name == b.codec_name && a.first_frame_id == b.first_frame_id &&
+         a.last_frame_id == b.last_frame_id &&
+         a.switched_on_at == b.switched_on_at &&
+         a.switched_from_at == b.switched_from_at;
+}
+
+std::string ToString(FrameDropPhase phase) {
+  switch (phase) {
+    case FrameDropPhase::kBeforeEncoder:
+      return "kBeforeEncoder";
+    case FrameDropPhase::kByEncoder:
+      return "kByEncoder";
+    case FrameDropPhase::kTransport:
+      return "kTransport";
+    case FrameDropPhase::kByDecoder:
+      return "kByDecoder";
+    case FrameDropPhase::kAfterDecoder:
+      return "kAfterDecoder";
+    case FrameDropPhase::kLastValue:
+      return "kLastValue";
+  }
+}
+
+std::ostream& operator<<(std::ostream& os, FrameDropPhase phase) {
+  return os << ToString(phase);
+}
+rtc::StringBuilder& operator<<(rtc::StringBuilder& sb, FrameDropPhase phase) {
+  return sb << ToString(phase);
+}
+
+void SamplesRateCounter::AddEvent(Timestamp event_time) {
+  if (event_first_time_.IsMinusInfinity()) {
+    event_first_time_ = event_time;
+  }
+  event_last_time_ = event_time;
+  events_count_++;
+}
+
+double SamplesRateCounter::GetEventsPerSecond() const {
+  RTC_DCHECK(!IsEmpty());
+  // Divide on us and multiply on kMicrosPerSecond to correctly process cases
+  // where there were too small amount of events, so difference is less then 1
+  // sec. We can use us here, because Timestamp has us resolution.
+  return static_cast<double>(events_count_) /
+         (event_last_time_ - event_first_time_).us() * kMicrosPerSecond;
+}
+
+StreamStats::StreamStats(Timestamp stream_started_time)
+    : stream_started_time(stream_started_time) {
+  for (int i = static_cast<int>(FrameDropPhase::kBeforeEncoder);
+       i < static_cast<int>(FrameDropPhase::kLastValue); ++i) {
+    dropped_by_phase.emplace(static_cast<FrameDropPhase>(i), 0);
+  }
+}
+
+std::string StatsKey::ToString() const {
+  rtc::StringBuilder out;
+  out << stream_label << "_" << receiver;
+  return out.str();
+}
+
+bool operator<(const StatsKey& a, const StatsKey& b) {
+  if (a.stream_label != b.stream_label) {
+    return a.stream_label < b.stream_label;
+  }
+  return a.receiver < b.receiver;
+}
+
+bool operator==(const StatsKey& a, const StatsKey& b) {
+  return a.stream_label == b.stream_label && a.receiver == b.receiver;
+}
+
+VideoStreamsInfo::VideoStreamsInfo(
+    std::map<std::string, std::string> stream_to_sender,
+    std::map<std::string, std::set<std::string>> sender_to_streams,
+    std::map<std::string, std::set<std::string>> stream_to_receivers)
+    : stream_to_sender_(std::move(stream_to_sender)),
+      sender_to_streams_(std::move(sender_to_streams)),
+      stream_to_receivers_(std::move(stream_to_receivers)) {}
+
+std::set<StatsKey> VideoStreamsInfo::GetStatsKeys() const {
+  std::set<StatsKey> out;
+  for (const std::string& stream_label : GetStreams()) {
+    for (const std::string& receiver : GetReceivers(stream_label)) {
+      out.insert(StatsKey(stream_label, receiver));
+    }
+  }
+  return out;
+}
+
+std::set<std::string> VideoStreamsInfo::GetStreams() const {
+  std::set<std::string> out;
+  std::transform(stream_to_sender_.begin(), stream_to_sender_.end(),
+                 std::inserter(out, out.end()),
+                 [](auto map_entry) { return map_entry.first; });
+  return out;
+}
+
+std::set<std::string> VideoStreamsInfo::GetStreams(
+    absl::string_view sender_name) const {
+  auto it = sender_to_streams_.find(std::string(sender_name));
+  if (it == sender_to_streams_.end()) {
+    return {};
+  }
+  return it->second;
+}
+
+absl::optional<std::string> VideoStreamsInfo::GetSender(
+    absl::string_view stream_label) const {
+  auto it = stream_to_sender_.find(std::string(stream_label));
+  if (it == stream_to_sender_.end()) {
+    return absl::nullopt;
+  }
+  return it->second;
+}
+
+std::set<std::string> VideoStreamsInfo::GetReceivers(
+    absl::string_view stream_label) const {
+  auto it = stream_to_receivers_.find(std::string(stream_label));
+  if (it == stream_to_receivers_.end()) {
+    return {};
+  }
+  return it->second;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h
new file mode 100644
index 0000000000..175f777b68
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h
@@ -0,0 +1,284 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_SHARED_OBJECTS_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_SHARED_OBJECTS_H_
+
+#include <cstdint>
+#include <map>
+#include <memory>
+#include <ostream>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+
+// WebRTC will request a key frame after 3 seconds if no frames were received.
+// We assume max frame rate ~60 fps, so 270 frames will cover max freeze without
+// key frame request.
+constexpr size_t kDefaultMaxFramesInFlightPerStream = 270;
+
+class SamplesRateCounter {
+ public:
+  void AddEvent(Timestamp event_time);
+
+  bool IsEmpty() const { return event_first_time_ == event_last_time_; }
+
+  double GetEventsPerSecond() const;
+
+ private:
+  Timestamp event_first_time_ = Timestamp::MinusInfinity();
+  Timestamp event_last_time_ = Timestamp::MinusInfinity();
+  int64_t events_count_ = 0;
+};
+
+struct FrameCounters {
+  // Count of frames, that were passed into WebRTC pipeline by video stream
+  // source.
+  int64_t captured = 0;
+  // Count of frames that reached video encoder.
+  int64_t pre_encoded = 0;
+  // Count of encoded images that were produced by encoder for all requested
+  // spatial layers and simulcast streams.
+  int64_t encoded = 0;
+  // Count of encoded images received in decoder for all requested spatial
+  // layers and simulcast streams.
+  int64_t received = 0;
+  // Count of frames that were produced by decoder.
+  int64_t decoded = 0;
+  // Count of frames that went out from WebRTC pipeline to video sink.
+  int64_t rendered = 0;
+  // Count of frames that were dropped in any point between capturing and
+  // rendering.
+  int64_t dropped = 0;
+  // Count of frames for which decoder returned error when they were sent for
+  // decoding.
+  int64_t failed_to_decode = 0;
+};
+
+// Contains information about the codec that was used for encoding or decoding
+// the stream.
+struct StreamCodecInfo {
+  // Codec implementation name.
+  std::string codec_name;
+  // Id of the first frame for which this codec was used.
+  uint16_t first_frame_id;
+  // Id of the last frame for which this codec was used.
+  uint16_t last_frame_id;
+  // Timestamp when the first frame was handled by the encode/decoder.
+  Timestamp switched_on_at = Timestamp::PlusInfinity();
+  // Timestamp when this codec was used last time.
+  Timestamp switched_from_at = Timestamp::PlusInfinity();
+
+  std::string ToString() const;
+};
+
+std::ostream& operator<<(std::ostream& os, const StreamCodecInfo& state);
+rtc::StringBuilder& operator<<(rtc::StringBuilder& sb,
+                               const StreamCodecInfo& state);
+bool operator==(const StreamCodecInfo& a, const StreamCodecInfo& b);
+
+// Represents phases where video frame can be dropped and such drop will be
+// detected by analyzer.
+enum class FrameDropPhase : int {
+  kBeforeEncoder,
+  kByEncoder,
+  kTransport,
+  kByDecoder,
+  kAfterDecoder,
+  // kLastValue must be the last value in this enumeration.
+  kLastValue
+};
+
+std::string ToString(FrameDropPhase phase);
+std::ostream& operator<<(std::ostream& os, FrameDropPhase phase);
+rtc::StringBuilder& operator<<(rtc::StringBuilder& sb, FrameDropPhase phase);
+
+struct StreamStats {
+  explicit StreamStats(Timestamp stream_started_time);
+
+  // The time when the first frame of this stream was captured.
+  Timestamp stream_started_time;
+
+  // Spatial quality metrics.
+  SamplesStatsCounter psnr;
+  SamplesStatsCounter ssim;
+
+  // Time from frame encoded (time point on exit from encoder) to the
+  // encoded image received in decoder (time point on entrance to decoder).
+  SamplesStatsCounter transport_time_ms;
+  // Time from frame was captured on device to time frame was displayed on
+  // device.
+  SamplesStatsCounter total_delay_incl_transport_ms;
+  // Time between frames out from renderer.
+  SamplesStatsCounter time_between_rendered_frames_ms;
+  SamplesRateCounter capture_frame_rate;
+  SamplesRateCounter encode_frame_rate;
+  SamplesStatsCounter encode_time_ms;
+  SamplesStatsCounter decode_time_ms;
+  // Time from last packet of frame is received until it's sent to the renderer.
+  SamplesStatsCounter receive_to_render_time_ms;
+  // Max frames skipped between two nearest.
+  SamplesStatsCounter skipped_between_rendered;
+  // In the next 2 metrics freeze is a pause that is longer, than maximum:
+  //  1. 150ms
+  //  2. 3 * average time between two sequential frames.
+  // Item 1 will cover high fps video and is a duration, that is noticeable by
+  // human eye. Item 2 will cover low fps video like screen sharing.
+  // Freeze duration.
+  SamplesStatsCounter freeze_time_ms;
+  // Mean time between one freeze end and next freeze start.
+  SamplesStatsCounter time_between_freezes_ms;
+  SamplesStatsCounter resolution_of_decoded_frame;
+  SamplesStatsCounter target_encode_bitrate;
+  // Sender side qp values per spatial layer. In case when spatial layer is not
+  // set for `webrtc::EncodedImage`, 0 is used as default.
+  std::map<int, SamplesStatsCounter> spatial_layers_qp;
+
+  int64_t total_encoded_images_payload = 0;
+  // Counters on which phase how many frames were dropped.
+  std::map<FrameDropPhase, int64_t> dropped_by_phase;
+
+  // Frame count metrics.
+  int64_t num_send_key_frames = 0;
+  int64_t num_recv_key_frames = 0;
+
+  // Encoded frame size (in bytes) metrics.
+  SamplesStatsCounter recv_key_frame_size_bytes;
+  SamplesStatsCounter recv_delta_frame_size_bytes;
+
+  // Vector of encoders used for this stream by sending client.
+  std::vector<StreamCodecInfo> encoders;
+  // Vectors of decoders used for this stream by receiving client.
+  std::vector<StreamCodecInfo> decoders;
+};
+
+struct AnalyzerStats {
+  // Size of analyzer internal comparisons queue, measured when new element
+  // id added to the queue.
+  SamplesStatsCounter comparisons_queue_size;
+  // Number of performed comparisons of 2 video frames from captured and
+  // rendered streams.
+  int64_t comparisons_done = 0;
+  // Number of cpu overloaded comparisons. Comparison is cpu overloaded if it is
+  // queued when there are too many not processed comparisons in the queue.
+  // Overloaded comparison doesn't include metrics like SSIM and PSNR that
+  // require heavy computations.
+  int64_t cpu_overloaded_comparisons_done = 0;
+  // Number of memory overloaded comparisons. Comparison is memory overloaded if
+  // it is queued when its captured frame was already removed due to high memory
+  // usage for that video stream.
+  int64_t memory_overloaded_comparisons_done = 0;
+  // Count of frames in flight in analyzer measured when new comparison is added
+  // and after analyzer was stopped.
+  SamplesStatsCounter frames_in_flight_left_count;
+
+  // Next metrics are collected and reported iff
+  // `DefaultVideoQualityAnalyzerOptions::report_infra_metrics` is true.
+  SamplesStatsCounter on_frame_captured_processing_time_ms;
+  SamplesStatsCounter on_frame_pre_encode_processing_time_ms;
+  SamplesStatsCounter on_frame_encoded_processing_time_ms;
+  SamplesStatsCounter on_frame_pre_decode_processing_time_ms;
+  SamplesStatsCounter on_frame_decoded_processing_time_ms;
+  SamplesStatsCounter on_frame_rendered_processing_time_ms;
+  SamplesStatsCounter on_decoder_error_processing_time_ms;
+};
+
+struct StatsKey {
+  StatsKey(std::string stream_label, std::string receiver)
+      : stream_label(std::move(stream_label)), receiver(std::move(receiver)) {}
+
+  std::string ToString() const;
+
+  // Label of video stream to which stats belongs to.
+  std::string stream_label;
+  // Name of the peer on which stream was received.
+  std::string receiver;
+};
+
+// Required to use StatsKey as std::map key.
+bool operator<(const StatsKey& a, const StatsKey& b);
+bool operator==(const StatsKey& a, const StatsKey& b);
+
+// Contains all metadata related to the video streams that were seen by the
+// video analyzer.
+class VideoStreamsInfo {
+ public:
+  std::set<StatsKey> GetStatsKeys() const;
+
+  // Returns all stream labels that are known to the video analyzer.
+  std::set<std::string> GetStreams() const;
+
+  // Returns set of the stream for specified `sender_name`. If sender didn't
+  // send any streams or `sender_name` isn't known to the video analyzer
+  // empty set will be returned.
+  std::set<std::string> GetStreams(absl::string_view sender_name) const;
+
+  // Returns sender name for specified `stream_label`. Returns `absl::nullopt`
+  // if provided `stream_label` isn't known to the video analyzer.
+  absl::optional<std::string> GetSender(absl::string_view stream_label) const;
+
+  // Returns set of the receivers for specified `stream_label`. If stream wasn't
+  // received by any peer or `stream_label` isn't known to the video analyzer
+  // empty set will be returned.
+  std::set<std::string> GetReceivers(absl::string_view stream_label) const;
+
+ protected:
+  friend class DefaultVideoQualityAnalyzer;
+  VideoStreamsInfo(
+      std::map<std::string, std::string> stream_to_sender,
+      std::map<std::string, std::set<std::string>> sender_to_streams,
+      std::map<std::string, std::set<std::string>> stream_to_receivers);
+
+ private:
+  std::map<std::string, std::string> stream_to_sender_;
+  std::map<std::string, std::set<std::string>> sender_to_streams_;
+  std::map<std::string, std::set<std::string>> stream_to_receivers_;
+};
+
+struct DefaultVideoQualityAnalyzerOptions {
+  // Tells DefaultVideoQualityAnalyzer if heavy metrics have to be computed.
+  bool compute_psnr = true;
+  bool compute_ssim = true;
+  // If true, weights the luma plane more than the chroma planes in the PSNR.
+  bool use_weighted_psnr = false;
+  // Tells DefaultVideoQualityAnalyzer if detailed frame stats should be
+  // reported.
+  bool report_detailed_frame_stats = false;
+  // Tells DefaultVideoQualityAnalyzer if infra metrics related to the
+  // performance and stability of the analyzer itself should be reported.
+  bool report_infra_metrics = false;
+  // If true DefaultVideoQualityAnalyzer will try to adjust frames before
+  // computing PSNR and SSIM for them. In some cases picture may be shifted by
+  // a few pixels after the encode/decode step. Those difference is invisible
+  // for a human eye, but it affects the metrics. So the adjustment is used to
+  // get metrics that are closer to how human perceive the video. This feature
+  // significantly slows down the comparison, so turn it on only when it is
+  // needed.
+  bool adjust_cropping_before_comparing_frames = false;
+  // Amount of frames that are queued in the DefaultVideoQualityAnalyzer from
+  // the point they were captured to the point they were rendered on all
+  // receivers per stream.
+  size_t max_frames_in_flight_per_stream_count =
+      kDefaultMaxFramesInFlightPerStream;
+  // If true, the analyzer will expect peers to receive their own video streams.
+  bool enable_receive_own_stream = false;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_SHARED_OBJECTS_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc
new file mode 100644
index 0000000000..d59ef12c63
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.cc
@@ -0,0 +1,121 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h"
+
+#include <map>
+#include <set>
+
+#include "absl/types/optional.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+template <typename T>
+absl::optional<T> MaybeGetValue(const std::map<size_t, T>& map, size_t key) {
+  auto it = map.find(key);
+  if (it == map.end()) {
+    return absl::nullopt;
+  }
+  return it->second;
+}
+
+}  // namespace
+
+StreamState::StreamState(size_t sender,
+                         std::set<size_t> receivers,
+                         Timestamp stream_started_time)
+    : sender_(sender),
+      stream_started_time_(stream_started_time),
+      receivers_(receivers),
+      frame_ids_(std::move(receivers)) {
+  frame_ids_.AddReader(kAliveFramesQueueIndex);
+  RTC_CHECK_NE(sender_, kAliveFramesQueueIndex);
+  for (size_t receiver : receivers_) {
+    RTC_CHECK_NE(receiver, kAliveFramesQueueIndex);
+  }
+}
+
+uint16_t StreamState::PopFront(size_t peer) {
+  RTC_CHECK_NE(peer, kAliveFramesQueueIndex);
+  absl::optional<uint16_t> frame_id = frame_ids_.PopFront(peer);
+  RTC_DCHECK(frame_id.has_value());
+
+  // If alive's frame queue is longer than all others, than also pop frame from
+  // it, because that frame is received by all receivers.
+  size_t alive_size = frame_ids_.size(kAliveFramesQueueIndex);
+  size_t other_size = GetLongestReceiverQueue();
+  // Pops frame from alive queue if alive's queue is the longest one.
+  if (alive_size > other_size) {
+    absl::optional<uint16_t> alive_frame_id =
+        frame_ids_.PopFront(kAliveFramesQueueIndex);
+    RTC_DCHECK(alive_frame_id.has_value());
+    RTC_DCHECK_EQ(frame_id.value(), alive_frame_id.value());
+  }
+
+  return frame_id.value();
+}
+
+void StreamState::AddPeer(size_t peer) {
+  RTC_CHECK_NE(peer, kAliveFramesQueueIndex);
+  frame_ids_.AddReader(peer, kAliveFramesQueueIndex);
+  receivers_.insert(peer);
+}
+
+void StreamState::RemovePeer(size_t peer) {
+  RTC_CHECK_NE(peer, kAliveFramesQueueIndex);
+  frame_ids_.RemoveReader(peer);
+  receivers_.erase(peer);
+
+  // If we removed the last receiver for the alive frames, we need to pop them
+  // from the queue, because now they received by all receivers.
+  size_t alive_size = frame_ids_.size(kAliveFramesQueueIndex);
+  size_t other_size = GetLongestReceiverQueue();
+  while (alive_size > other_size) {
+    frame_ids_.PopFront(kAliveFramesQueueIndex);
+    alive_size--;
+  }
+}
+
+uint16_t StreamState::MarkNextAliveFrameAsDead() {
+  absl::optional<uint16_t> frame_id =
+      frame_ids_.PopFront(kAliveFramesQueueIndex);
+  RTC_DCHECK(frame_id.has_value());
+  return frame_id.value();
+}
+
+void StreamState::SetLastRenderedFrameTime(size_t peer, Timestamp time) {
+  auto it = last_rendered_frame_time_.find(peer);
+  if (it == last_rendered_frame_time_.end()) {
+    last_rendered_frame_time_.insert({peer, time});
+  } else {
+    it->second = time;
+  }
+}
+
+absl::optional<Timestamp> StreamState::last_rendered_frame_time(
+    size_t peer) const {
+  return MaybeGetValue(last_rendered_frame_time_, peer);
+}
+
+size_t StreamState::GetLongestReceiverQueue() const {
+  size_t max = 0;
+  for (size_t receiver : receivers_) {
+    size_t cur_size = frame_ids_.size(receiver);
+    if (cur_size > max) {
+      max = cur_size;
+    }
+  }
+  return max;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h
new file mode 100644
index 0000000000..829a79c7bf
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h
@@ -0,0 +1,100 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_STREAM_STATE_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_STREAM_STATE_H_
+
+#include <limits>
+#include <map>
+#include <set>
+
+#include "absl/types/optional.h"
+#include "api/units/timestamp.h"
+#include "test/pc/e2e/analyzer/video/multi_reader_queue.h"
+
+namespace webrtc {
+
+// Represents a current state of video stream inside
+// DefaultVideoQualityAnalyzer.
+//
+// Maintains the sequence of frames for each video stream and keeps track about
+// which frames were seen by each of the possible stream receiver.
+//
+// Keeps information about which frames are alive and which are dead. Frame is
+// alive if it contains VideoFrame payload for corresponding FrameInFlight
+// object inside DefaultVideoQualityAnalyzer, otherwise frame is considered
+// dead.
+//
+// Supports peer indexes from 0 to max(size_t) - 1.
+class StreamState {
+ public:
+  StreamState(size_t sender,
+              std::set<size_t> receivers,
+              Timestamp stream_started_time);
+
+  size_t sender() const { return sender_; }
+  Timestamp stream_started_time() const { return stream_started_time_; }
+
+  void PushBack(uint16_t frame_id) { frame_ids_.PushBack(frame_id); }
+  // Crash if state is empty.
+  uint16_t PopFront(size_t peer);
+  bool IsEmpty(size_t peer) const { return frame_ids_.IsEmpty(peer); }
+  // Crash if state is empty.
+  uint16_t Front(size_t peer) const { return frame_ids_.Front(peer).value(); }
+
+  // Adds a new peer to the state. All currently alive frames will be expected
+  // to be received by the newly added peer.
+  void AddPeer(size_t peer);
+
+  // Removes peer from the state. Frames that were expected to be received by
+  // this peer will be removed from it. On the other hand last rendered frame
+  // time for the removed peer will be preserved, because
+  // DefaultVideoQualityAnalyzer still may request it for stats processing.
+  void RemovePeer(size_t peer);
+
+  size_t GetAliveFramesCount() const {
+    return frame_ids_.size(kAliveFramesQueueIndex);
+  }
+  uint16_t MarkNextAliveFrameAsDead();
+
+  void SetLastRenderedFrameTime(size_t peer, Timestamp time);
+  absl::optional<Timestamp> last_rendered_frame_time(size_t peer) const;
+
+ private:
+  // Index of the `frame_ids_` queue which is used to track alive frames for
+  // this stream.
+  static constexpr size_t kAliveFramesQueueIndex =
+      std::numeric_limits<size_t>::max();
+
+  size_t GetLongestReceiverQueue() const;
+
+  // Index of the owner. Owner's queue in `frame_ids_` will keep alive frames.
+  const size_t sender_;
+  const Timestamp stream_started_time_;
+  std::set<size_t> receivers_;
+  // To correctly determine dropped frames we have to know sequence of frames
+  // in each stream so we will keep a list of frame ids inside the stream.
+  // This list is represented by multi head queue of frame ids with separate
+  // head for each receiver. When the frame is rendered, we will pop ids from
+  // the corresponding head until id will match with rendered one. All ids
+  // before matched one can be considered as dropped:
+  //
+  // | frame_id1 |->| frame_id2 |->| frame_id3 |->| frame_id4 |
+  //
+  // If we received frame with id frame_id3, then we will pop frame_id1 and
+  // frame_id2 and consider those frames as dropped and then compare received
+  // frame with the one from `FrameInFlight` with id frame_id3.
+  MultiReaderQueue<uint16_t> frame_ids_;
+  std::map<size_t, Timestamp> last_rendered_frame_time_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_DEFAULT_VIDEO_QUALITY_ANALYZER_STREAM_STATE_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state_test.cc
new file mode 100644
index 0000000000..01a6aab28a
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state_test.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_stream_state.h"
+
+#include <set>
+
+#include "api/units/timestamp.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+TEST(StreamStateTest, PopFrontAndFrontIndependentForEachPeer) {
+  StreamState state(/*sender=*/0,
+                    /*receivers=*/std::set<size_t>{1, 2},
+                    Timestamp::Seconds(1));
+  state.PushBack(/*frame_id=*/1);
+  state.PushBack(/*frame_id=*/2);
+
+  EXPECT_EQ(state.Front(/*peer=*/1), 1);
+  EXPECT_EQ(state.PopFront(/*peer=*/1), 1);
+  EXPECT_EQ(state.Front(/*peer=*/1), 2);
+  EXPECT_EQ(state.PopFront(/*peer=*/1), 2);
+  EXPECT_EQ(state.Front(/*peer=*/2), 1);
+  EXPECT_EQ(state.PopFront(/*peer=*/2), 1);
+  EXPECT_EQ(state.Front(/*peer=*/2), 2);
+  EXPECT_EQ(state.PopFront(/*peer=*/2), 2);
+}
+
+TEST(StreamStateTest, IsEmpty) {
+  StreamState state(/*sender=*/0,
+                    /*receivers=*/std::set<size_t>{1, 2},
+                    Timestamp::Seconds(1));
+  state.PushBack(/*frame_id=*/1);
+
+  EXPECT_FALSE(state.IsEmpty(/*peer=*/1));
+
+  state.PopFront(/*peer=*/1);
+
+  EXPECT_TRUE(state.IsEmpty(/*peer=*/1));
+}
+
+TEST(StreamStateTest, PopFrontForOnlyOnePeerDontChangeAliveFramesCount) {
+  StreamState state(/*sender=*/0,
+                    /*receivers=*/std::set<size_t>{1, 2},
+                    Timestamp::Seconds(1));
+  state.PushBack(/*frame_id=*/1);
+  state.PushBack(/*frame_id=*/2);
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 2lu);
+
+  state.PopFront(/*peer=*/1);
+  state.PopFront(/*peer=*/1);
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 2lu);
+}
+
+TEST(StreamStateTest, PopFrontForAllPeersReducesAliveFramesCount) {
+  StreamState state(/*sender=*/0,
+                    /*receivers=*/std::set<size_t>{1, 2},
+                    Timestamp::Seconds(1));
+  state.PushBack(/*frame_id=*/1);
+  state.PushBack(/*frame_id=*/2);
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 2lu);
+
+  state.PopFront(/*peer=*/1);
+  state.PopFront(/*peer=*/2);
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 1lu);
+}
+
+TEST(StreamStateTest, RemovePeerForLastExpectedReceiverUpdatesAliveFrames) {
+  StreamState state(/*sender=*/0,
+                    /*receivers=*/std::set<size_t>{1, 2},
+                    Timestamp::Seconds(1));
+  state.PushBack(/*frame_id=*/1);
+  state.PushBack(/*frame_id=*/2);
+
+  state.PopFront(/*peer=*/1);
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 2lu);
+
+  state.RemovePeer(/*peer=*/2);
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 1lu);
+}
+
+TEST(StreamStateTest, MarkNextAliveFrameAsDeadDecreseAliveFramesCount) {
+  StreamState state(/*sender=*/0,
+                    /*receivers=*/std::set<size_t>{1, 2},
+                    Timestamp::Seconds(1));
+  state.PushBack(/*frame_id=*/1);
+  state.PushBack(/*frame_id=*/2);
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 2lu);
+
+  state.MarkNextAliveFrameAsDead();
+
+  EXPECT_EQ(state.GetAliveFramesCount(), 1lu);
+}
+
+TEST(StreamStateTest, MarkNextAliveFrameAsDeadDoesntAffectFrontFrameForPeer) {
+  StreamState state(/*sender=*/0,
+                    /*receivers=*/std::set<size_t>{1, 2},
+                    Timestamp::Seconds(1));
+  state.PushBack(/*frame_id=*/1);
+  state.PushBack(/*frame_id=*/2);
+
+  EXPECT_EQ(state.Front(/*peer=*/1), 1);
+
+  state.MarkNextAliveFrameAsDead();
+
+  EXPECT_EQ(state.Front(/*peer=*/1), 1);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
new file mode 100644
index 0000000000..fc970e1ea2
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
@@ -0,0 +1,2204 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
+
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "api/rtp_packet_info.h"
+#include "api/rtp_packet_infos.h"
+#include "api/test/create_frame_generator.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_tools/frame_analyzer/video_geometry_aligner.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::TestWithParam;
+using ::testing::ValuesIn;
+
+using StatsSample = ::webrtc::SamplesStatsCounter::StatsSample;
+
+constexpr int kAnalyzerMaxThreadsCount = 1;
+constexpr int kMaxFramesInFlightPerStream = 10;
+constexpr int kFrameWidth = 320;
+constexpr int kFrameHeight = 240;
+constexpr double kMaxSsim = 1;
+constexpr char kStreamLabel[] = "video-stream";
+constexpr char kSenderPeerName[] = "alice";
+constexpr char kReceiverPeerName[] = "bob";
+
+DefaultVideoQualityAnalyzerOptions AnalyzerOptionsForTest() {
+  DefaultVideoQualityAnalyzerOptions options;
+  options.compute_psnr = false;
+  options.compute_ssim = false;
+  options.adjust_cropping_before_comparing_frames = false;
+  options.max_frames_in_flight_per_stream_count = kMaxFramesInFlightPerStream;
+  return options;
+}
+
+VideoFrame NextFrame(test::FrameGeneratorInterface* frame_generator,
+                     int64_t timestamp_us) {
+  test::FrameGeneratorInterface::VideoFrameData frame_data =
+      frame_generator->NextFrame();
+  return VideoFrame::Builder()
+      .set_video_frame_buffer(frame_data.buffer)
+      .set_update_rect(frame_data.update_rect)
+      .set_timestamp_us(timestamp_us)
+      .build();
+}
+
+EncodedImage FakeEncode(const VideoFrame& frame) {
+  EncodedImage image;
+  std::vector<RtpPacketInfo> packet_infos;
+  packet_infos.push_back(RtpPacketInfo(
+      /*ssrc=*/1,
+      /*csrcs=*/{},
+      /*rtp_timestamp=*/frame.timestamp(),
+      /*receive_time=*/Timestamp::Micros(frame.timestamp_us() + 10000)));
+  image.SetPacketInfos(RtpPacketInfos(packet_infos));
+  return image;
+}
+
+VideoFrame DeepCopy(const VideoFrame& frame) {
+  VideoFrame copy = frame;
+  copy.set_video_frame_buffer(
+      I420Buffer::Copy(*frame.video_frame_buffer()->ToI420()));
+  return copy;
+}
+
+std::vector<StatsSample> GetSortedSamples(const SamplesStatsCounter& counter) {
+  rtc::ArrayView<const StatsSample> view = counter.GetTimedSamples();
+  std::vector<StatsSample> out(view.begin(), view.end());
+  std::sort(out.begin(), out.end(),
+            [](const StatsSample& a, const StatsSample& b) {
+              return a.time < b.time;
+            });
+  return out;
+}
+
+std::string ToString(const std::vector<StatsSample>& values) {
+  rtc::StringBuilder out;
+  for (const auto& v : values) {
+    out << "{ time_ms=" << v.time.ms() << "; value=" << v.value << "}, ";
+  }
+  return out.str();
+}
+
+void FakeCPULoad() {
+  std::vector<int> temp(1000000);
+  for (size_t i = 0; i < temp.size(); ++i) {
+    temp[i] = rand();
+  }
+  std::sort(temp.begin(), temp.end());
+  ASSERT_TRUE(std::is_sorted(temp.begin(), temp.end()));
+}
+
+void PassFramesThroughAnalyzer(DefaultVideoQualityAnalyzer& analyzer,
+                               absl::string_view sender,
+                               absl::string_view stream_label,
+                               std::vector<absl::string_view> receivers,
+                               int frames_count,
+                               test::FrameGeneratorInterface& frame_generator,
+                               int interframe_delay_ms = 0) {
+  for (int i = 0; i < frames_count; ++i) {
+    VideoFrame frame = NextFrame(&frame_generator, /*timestamp_us=*/1);
+    uint16_t frame_id =
+        analyzer.OnFrameCaptured(sender, std::string(stream_label), frame);
+    frame.set_id(frame_id);
+    analyzer.OnFramePreEncode(sender, frame);
+    analyzer.OnFrameEncoded(sender, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+    for (absl::string_view receiver : receivers) {
+      VideoFrame received_frame = DeepCopy(frame);
+      analyzer.OnFramePreDecode(receiver, received_frame.id(),
+                                FakeEncode(received_frame));
+      analyzer.OnFrameDecoded(receiver, received_frame,
+                              VideoQualityAnalyzerInterface::DecoderStats());
+      analyzer.OnFrameRendered(receiver, received_frame);
+    }
+    if (i < frames_count - 1 && interframe_delay_ms > 0) {
+      SleepMs(interframe_delay_ms);
+    }
+  }
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     MemoryOverloadedAndThenAllFramesReceived) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  std::map<uint16_t, VideoFrame> captured_frames;
+  std::vector<uint16_t> frames_order;
+  for (int i = 0; i < kMaxFramesInFlightPerStream * 2; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  for (const uint16_t& frame_id : frames_order) {
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done,
+            kMaxFramesInFlightPerStream);
+  EXPECT_EQ(stats.comparisons_done, kMaxFramesInFlightPerStream * 2);
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, kMaxFramesInFlightPerStream * 2);
+  EXPECT_EQ(frame_counters.rendered, kMaxFramesInFlightPerStream * 2);
+  EXPECT_EQ(frame_counters.dropped, 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     FillMaxMemoryReceiveAllMemoryOverloadedAndThenAllFramesReceived) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  std::map<uint16_t, VideoFrame> captured_frames;
+  std::vector<uint16_t> frames_order;
+  // Feel analyzer's memory up to limit
+  for (int i = 0; i < kMaxFramesInFlightPerStream; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  // Receive all frames.
+  for (const uint16_t& frame_id : frames_order) {
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+  frames_order.clear();
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+
+  // Overload analyzer's memory up to limit
+  for (int i = 0; i < 2 * kMaxFramesInFlightPerStream; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  // Receive all frames.
+  for (const uint16_t& frame_id : frames_order) {
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done,
+            kMaxFramesInFlightPerStream);
+  EXPECT_EQ(stats.comparisons_done, kMaxFramesInFlightPerStream * 3);
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, kMaxFramesInFlightPerStream * 3);
+  EXPECT_EQ(frame_counters.rendered, kMaxFramesInFlightPerStream * 3);
+  EXPECT_EQ(frame_counters.dropped, 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     MemoryOverloadedHalfDroppedAndThenHalfFramesReceived) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  std::map<uint16_t, VideoFrame> captured_frames;
+  std::vector<uint16_t> frames_order;
+  for (int i = 0; i < kMaxFramesInFlightPerStream * 2; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  for (size_t i = kMaxFramesInFlightPerStream; i < frames_order.size(); ++i) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done, 0);
+  EXPECT_EQ(stats.comparisons_done, kMaxFramesInFlightPerStream * 2);
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, kMaxFramesInFlightPerStream * 2);
+  EXPECT_EQ(frame_counters.rendered, kMaxFramesInFlightPerStream);
+  EXPECT_EQ(frame_counters.dropped, kMaxFramesInFlightPerStream);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, NormalScenario) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  std::map<uint16_t, VideoFrame> captured_frames;
+  std::vector<uint16_t> frames_order;
+  for (int i = 0; i < kMaxFramesInFlightPerStream; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  for (size_t i = 1; i < frames_order.size(); i += 2) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done, 0);
+  EXPECT_EQ(stats.comparisons_done, kMaxFramesInFlightPerStream);
+
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 0)
+      << ToString(frames_in_flight_sizes);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, kMaxFramesInFlightPerStream);
+  EXPECT_EQ(frame_counters.received, kMaxFramesInFlightPerStream / 2);
+  EXPECT_EQ(frame_counters.decoded, kMaxFramesInFlightPerStream / 2);
+  EXPECT_EQ(frame_counters.rendered, kMaxFramesInFlightPerStream / 2);
+  EXPECT_EQ(frame_counters.dropped, kMaxFramesInFlightPerStream / 2);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, OneFrameReceivedTwice) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame captured_frame = NextFrame(frame_generator.get(), 0);
+  captured_frame.set_id(
+      analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, captured_frame));
+  analyzer.OnFramePreEncode(kSenderPeerName, captured_frame);
+  analyzer.OnFrameEncoded(kSenderPeerName, captured_frame.id(),
+                          FakeEncode(captured_frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  VideoFrame received_frame = DeepCopy(captured_frame);
+  analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                            FakeEncode(received_frame));
+  analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+
+  received_frame = DeepCopy(captured_frame);
+  analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                            FakeEncode(received_frame));
+  analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done, 0);
+  EXPECT_EQ(stats.comparisons_done, 1);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, 1);
+  EXPECT_EQ(frame_counters.received, 1);
+  EXPECT_EQ(frame_counters.decoded, 1);
+  EXPECT_EQ(frame_counters.rendered, 1);
+  EXPECT_EQ(frame_counters.dropped, 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, NormalScenario2Receivers) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  constexpr char kAlice[] = "alice";
+  constexpr char kBob[] = "bob";
+  constexpr char kCharlie[] = "charlie";
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case", std::vector<std::string>{kAlice, kBob, kCharlie},
+                 kAnalyzerMaxThreadsCount);
+
+  std::map<uint16_t, VideoFrame> captured_frames;
+  std::vector<uint16_t> frames_order;
+  for (int i = 0; i < kMaxFramesInFlightPerStream; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(analyzer.OnFrameCaptured(kAlice, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kAlice, frame);
+    SleepMs(20);
+    analyzer.OnFrameEncoded(kAlice, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  SleepMs(50);
+
+  for (size_t i = 1; i < frames_order.size(); i += 2) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kBob, received_frame.id(),
+                              FakeEncode(received_frame));
+    SleepMs(30);
+    analyzer.OnFrameDecoded(kBob, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    SleepMs(10);
+    analyzer.OnFrameRendered(kBob, received_frame);
+  }
+
+  for (size_t i = 1; i < frames_order.size(); i += 2) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kCharlie, received_frame.id(),
+                              FakeEncode(received_frame));
+    SleepMs(40);
+    analyzer.OnFrameDecoded(kCharlie, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    SleepMs(5);
+    analyzer.OnFrameRendered(kCharlie, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(analyzer_stats.memory_overloaded_comparisons_done, 0);
+  EXPECT_EQ(analyzer_stats.comparisons_done, kMaxFramesInFlightPerStream * 2);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, kMaxFramesInFlightPerStream);
+  EXPECT_EQ(frame_counters.received, kMaxFramesInFlightPerStream);
+  EXPECT_EQ(frame_counters.decoded, kMaxFramesInFlightPerStream);
+  EXPECT_EQ(frame_counters.rendered, kMaxFramesInFlightPerStream);
+  EXPECT_EQ(frame_counters.dropped, kMaxFramesInFlightPerStream);
+
+  VideoStreamsInfo streams_info = analyzer.GetKnownStreams();
+  EXPECT_EQ(streams_info.GetStreams(), std::set<std::string>{kStreamLabel});
+  EXPECT_EQ(streams_info.GetStreams(kAlice),
+            std::set<std::string>{kStreamLabel});
+  EXPECT_EQ(streams_info.GetSender(kStreamLabel), kAlice);
+  EXPECT_EQ(streams_info.GetReceivers(kStreamLabel),
+            (std::set<std::string>{kBob, kCharlie}));
+
+  EXPECT_EQ(streams_info.GetStatsKeys().size(), 2lu);
+  for (auto stream_key : streams_info.GetStatsKeys()) {
+    FrameCounters stream_conters =
+        analyzer.GetPerStreamCounters().at(stream_key);
+    // On some devices the pipeline can be too slow, so we actually can't
+    // force real constraints here. Lets just check, that at least 1
+    // frame passed whole pipeline.
+    EXPECT_GE(stream_conters.captured, 10);
+    EXPECT_GE(stream_conters.pre_encoded, 10);
+    EXPECT_GE(stream_conters.encoded, 10);
+    EXPECT_GE(stream_conters.received, 5);
+    EXPECT_GE(stream_conters.decoded, 5);
+    EXPECT_GE(stream_conters.rendered, 5);
+    EXPECT_GE(stream_conters.dropped, 5);
+  }
+
+  std::map<StatsKey, StreamStats> stats = analyzer.GetStats();
+  const StatsKey kAliceBobStats(kStreamLabel, kBob);
+  const StatsKey kAliceCharlieStats(kStreamLabel, kCharlie);
+  EXPECT_EQ(stats.size(), 2lu);
+  {
+    auto it = stats.find(kAliceBobStats);
+    EXPECT_FALSE(it == stats.end());
+    ASSERT_FALSE(it->second.encode_time_ms.IsEmpty());
+    EXPECT_GE(it->second.encode_time_ms.GetMin(), 20);
+    ASSERT_FALSE(it->second.decode_time_ms.IsEmpty());
+    EXPECT_GE(it->second.decode_time_ms.GetMin(), 30);
+    ASSERT_FALSE(it->second.resolution_of_decoded_frame.IsEmpty());
+    EXPECT_GE(it->second.resolution_of_decoded_frame.GetMin(),
+              kFrameWidth * kFrameHeight - 1);
+    EXPECT_LE(it->second.resolution_of_decoded_frame.GetMax(),
+              kFrameWidth * kFrameHeight + 1);
+  }
+  {
+    auto it = stats.find(kAliceCharlieStats);
+    EXPECT_FALSE(it == stats.end());
+    ASSERT_FALSE(it->second.encode_time_ms.IsEmpty());
+    EXPECT_GE(it->second.encode_time_ms.GetMin(), 20);
+    ASSERT_FALSE(it->second.decode_time_ms.IsEmpty());
+    EXPECT_GE(it->second.decode_time_ms.GetMin(), 30);
+    ASSERT_FALSE(it->second.resolution_of_decoded_frame.IsEmpty());
+    EXPECT_GE(it->second.resolution_of_decoded_frame.GetMin(),
+              kFrameWidth * kFrameHeight - 1);
+    EXPECT_LE(it->second.resolution_of_decoded_frame.GetMax(),
+              kFrameWidth * kFrameHeight + 1);
+  }
+}
+
+// Test the case which can happen when SFU is switching from one layer to
+// another, so the same frame can be received twice by the same peer.
+TEST(DefaultVideoQualityAnalyzerTest,
+     OneFrameReceivedTwiceBySamePeerWith2Receivers) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  constexpr char kAlice[] = "alice";
+  constexpr char kBob[] = "bob";
+  constexpr char kCharlie[] = "charlie";
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case", std::vector<std::string>{kAlice, kBob, kCharlie},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame captured_frame = NextFrame(frame_generator.get(), 0);
+  captured_frame.set_id(
+      analyzer.OnFrameCaptured(kAlice, kStreamLabel, captured_frame));
+  analyzer.OnFramePreEncode(kAlice, captured_frame);
+  analyzer.OnFrameEncoded(kAlice, captured_frame.id(),
+                          FakeEncode(captured_frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  VideoFrame received_frame = DeepCopy(captured_frame);
+  analyzer.OnFramePreDecode(kBob, received_frame.id(),
+                            FakeEncode(received_frame));
+  analyzer.OnFrameDecoded(kBob, received_frame,
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered(kBob, received_frame);
+
+  received_frame = DeepCopy(captured_frame);
+  analyzer.OnFramePreDecode(kBob, received_frame.id(),
+                            FakeEncode(received_frame));
+  analyzer.OnFrameDecoded(kBob, received_frame,
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered(kBob, received_frame);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done, 0);
+  // We have 2 comparisons here because 1 for the frame received by Bob and
+  // 1 for the frame in flight from Alice to Charlie.
+  EXPECT_EQ(stats.comparisons_done, 2);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, 1);
+  EXPECT_EQ(frame_counters.received, 1);
+  EXPECT_EQ(frame_counters.decoded, 1);
+  EXPECT_EQ(frame_counters.rendered, 1);
+  EXPECT_EQ(frame_counters.dropped, 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, HeavyQualityMetricsFromEqualFrames) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions analyzer_options;
+  analyzer_options.compute_psnr = true;
+  analyzer_options.compute_ssim = true;
+  analyzer_options.adjust_cropping_before_comparing_frames = false;
+  analyzer_options.max_frames_in_flight_per_stream_count =
+      kMaxFramesInFlightPerStream;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       analyzer_options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  for (int i = 0; i < kMaxFramesInFlightPerStream; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+
+    VideoFrame received_frame = DeepCopy(frame);
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. Heavy metrics
+  // computation is turned on, so giving some extra time to be sure that
+  // computatio have ended.
+  SleepMs(500);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done, 0);
+  EXPECT_EQ(stats.comparisons_done, kMaxFramesInFlightPerStream);
+
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 0)
+      << ToString(frames_in_flight_sizes);
+
+  std::map<StatsKey, StreamStats> stream_stats = analyzer.GetStats();
+  const StatsKey kAliceBobStats(kStreamLabel, kReceiverPeerName);
+  EXPECT_EQ(stream_stats.size(), 1lu);
+
+  auto it = stream_stats.find(kAliceBobStats);
+  EXPECT_GE(it->second.psnr.GetMin(), kPerfectPSNR);
+  EXPECT_GE(it->second.ssim.GetMin(), kMaxSsim);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     HeavyQualityMetricsFromShiftedFramesWithAdjustment) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions analyzer_options;
+  analyzer_options.compute_psnr = true;
+  analyzer_options.compute_ssim = true;
+  analyzer_options.adjust_cropping_before_comparing_frames = true;
+  analyzer_options.max_frames_in_flight_per_stream_count =
+      kMaxFramesInFlightPerStream;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       analyzer_options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  for (int i = 0; i < kMaxFramesInFlightPerStream; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+
+    VideoFrame received_frame = frame;
+    // Shift frame by a few pixels.
+    test::CropRegion crop_region{0, 1, 3, 0};
+    rtc::scoped_refptr<VideoFrameBuffer> cropped_buffer =
+        CropAndZoom(crop_region, received_frame.video_frame_buffer()->ToI420());
+    received_frame.set_video_frame_buffer(cropped_buffer);
+
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. Heavy metrics
+  // computation is turned on, so giving some extra time to be sure that
+  // computatio have ended.
+  SleepMs(500);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done, 0);
+  EXPECT_EQ(stats.comparisons_done, kMaxFramesInFlightPerStream);
+
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 0)
+      << ToString(frames_in_flight_sizes);
+
+  std::map<StatsKey, StreamStats> stream_stats = analyzer.GetStats();
+  const StatsKey kAliceBobStats(kStreamLabel, kReceiverPeerName);
+  EXPECT_EQ(stream_stats.size(), 1lu);
+
+  auto it = stream_stats.find(kAliceBobStats);
+  EXPECT_GE(it->second.psnr.GetMin(), kPerfectPSNR);
+  EXPECT_GE(it->second.ssim.GetMin(), kMaxSsim);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, CpuUsage) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  std::map<uint16_t, VideoFrame> captured_frames;
+  std::vector<uint16_t> frames_order;
+  for (int i = 0; i < kMaxFramesInFlightPerStream; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  // Windows CPU clock has low accuracy. We need to fake some additional load to
+  // be sure that the clock ticks (https://crbug.com/webrtc/12249).
+  FakeCPULoad();
+
+  for (size_t i = 1; i < frames_order.size(); i += 2) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  double cpu_usage = analyzer.GetCpuUsagePercent();
+  ASSERT_GT(cpu_usage, 0);
+
+  SleepMs(100);
+  analyzer.Stop();
+
+  EXPECT_EQ(analyzer.GetCpuUsagePercent(), cpu_usage);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, RuntimeParticipantsAdding) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  constexpr char kAlice[] = "alice";
+  constexpr char kBob[] = "bob";
+  constexpr char kCharlie[] = "charlie";
+  constexpr char kKatie[] = "katie";
+
+  constexpr int kFramesCount = 9;
+  constexpr int kOneThirdFrames = kFramesCount / 3;
+  constexpr int kTwoThirdFrames = 2 * kOneThirdFrames;
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case", {}, kAnalyzerMaxThreadsCount);
+
+  std::map<uint16_t, VideoFrame> captured_frames;
+  std::vector<uint16_t> frames_order;
+  analyzer.RegisterParticipantInCall(kAlice);
+  analyzer.RegisterParticipantInCall(kBob);
+
+  // Alice is sending frames.
+  for (int i = 0; i < kFramesCount; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), i);
+    frame.set_id(analyzer.OnFrameCaptured(kAlice, kStreamLabel, frame));
+    frames_order.push_back(frame.id());
+    captured_frames.insert({frame.id(), frame});
+    analyzer.OnFramePreEncode(kAlice, frame);
+    analyzer.OnFrameEncoded(kAlice, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  // Bob receives one third of the sent frames.
+  for (int i = 0; i < kOneThirdFrames; ++i) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kBob, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kBob, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kBob, received_frame);
+  }
+
+  analyzer.RegisterParticipantInCall(kCharlie);
+  analyzer.RegisterParticipantInCall(kKatie);
+
+  // New participants were dynamically added. Bob and Charlie receive second
+  // third of the sent frames. Katie drops the frames.
+  for (int i = kOneThirdFrames; i < kTwoThirdFrames; ++i) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame bob_received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kBob, bob_received_frame.id(),
+                              FakeEncode(bob_received_frame));
+    analyzer.OnFrameDecoded(kBob, bob_received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kBob, bob_received_frame);
+
+    VideoFrame charlie_received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kCharlie, charlie_received_frame.id(),
+                              FakeEncode(charlie_received_frame));
+    analyzer.OnFrameDecoded(kCharlie, charlie_received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kCharlie, charlie_received_frame);
+  }
+
+  // Bob, Charlie and Katie receive the rest of the sent frames.
+  for (int i = kTwoThirdFrames; i < kFramesCount; ++i) {
+    uint16_t frame_id = frames_order.at(i);
+    VideoFrame bob_received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kBob, bob_received_frame.id(),
+                              FakeEncode(bob_received_frame));
+    analyzer.OnFrameDecoded(kBob, bob_received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kBob, bob_received_frame);
+
+    VideoFrame charlie_received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kCharlie, charlie_received_frame.id(),
+                              FakeEncode(charlie_received_frame));
+    analyzer.OnFrameDecoded(kCharlie, charlie_received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kCharlie, charlie_received_frame);
+
+    VideoFrame katie_received_frame = DeepCopy(captured_frames.at(frame_id));
+    analyzer.OnFramePreDecode(kKatie, katie_received_frame.id(),
+                              FakeEncode(katie_received_frame));
+    analyzer.OnFrameDecoded(kKatie, katie_received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kKatie, katie_received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.memory_overloaded_comparisons_done, 0);
+  EXPECT_EQ(stats.comparisons_done, kFramesCount + 2 * kTwoThirdFrames);
+
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 0)
+      << ToString(frames_in_flight_sizes);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, kFramesCount);
+  EXPECT_EQ(frame_counters.received, 2 * kFramesCount);
+  EXPECT_EQ(frame_counters.decoded, 2 * kFramesCount);
+  EXPECT_EQ(frame_counters.rendered, 2 * kFramesCount);
+  EXPECT_EQ(frame_counters.dropped, kOneThirdFrames);
+
+  const StatsKey kAliceBobStats(kStreamLabel, kBob);
+  const StatsKey kAliceCharlieStats(kStreamLabel, kCharlie);
+  const StatsKey kAliceKatieStats(kStreamLabel, kKatie);
+  EXPECT_EQ(analyzer.GetKnownStreams().GetStatsKeys(),
+            (std::set<StatsKey>{kAliceBobStats, kAliceCharlieStats,
+                                kAliceKatieStats}));
+  {
+    FrameCounters stream_conters =
+        analyzer.GetPerStreamCounters().at(kAliceBobStats);
+    EXPECT_EQ(stream_conters.captured, kFramesCount);
+    EXPECT_EQ(stream_conters.pre_encoded, kFramesCount);
+    EXPECT_EQ(stream_conters.encoded, kFramesCount);
+    EXPECT_EQ(stream_conters.received, kFramesCount);
+    EXPECT_EQ(stream_conters.decoded, kFramesCount);
+    EXPECT_EQ(stream_conters.rendered, kFramesCount);
+  }
+  {
+    FrameCounters stream_conters =
+        analyzer.GetPerStreamCounters().at(kAliceCharlieStats);
+    EXPECT_EQ(stream_conters.captured, kFramesCount);
+    EXPECT_EQ(stream_conters.pre_encoded, kFramesCount);
+    EXPECT_EQ(stream_conters.encoded, kFramesCount);
+    EXPECT_EQ(stream_conters.received, kTwoThirdFrames);
+    EXPECT_EQ(stream_conters.decoded, kTwoThirdFrames);
+    EXPECT_EQ(stream_conters.rendered, kTwoThirdFrames);
+  }
+  {
+    FrameCounters stream_conters =
+        analyzer.GetPerStreamCounters().at(kAliceKatieStats);
+    EXPECT_EQ(stream_conters.captured, kFramesCount);
+    EXPECT_EQ(stream_conters.pre_encoded, kFramesCount);
+    EXPECT_EQ(stream_conters.encoded, kFramesCount);
+    EXPECT_EQ(stream_conters.received, kOneThirdFrames);
+    EXPECT_EQ(stream_conters.decoded, kOneThirdFrames);
+    EXPECT_EQ(stream_conters.rendered, kOneThirdFrames);
+  }
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     SimulcastFrameWasFullyReceivedByAllPeersBeforeEncodeFinish) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  constexpr char kAlice[] = "alice";
+  constexpr char kBob[] = "bob";
+  constexpr char kCharlie[] = "charlie";
+  analyzer.Start("test_case", std::vector<std::string>{kAlice, kBob, kCharlie},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), 1);
+
+  frame.set_id(analyzer.OnFrameCaptured(kAlice, kStreamLabel, frame));
+  analyzer.OnFramePreEncode(kAlice, frame);
+  // Encode 1st simulcast layer
+  analyzer.OnFrameEncoded(kAlice, frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  // Receive by Bob
+  VideoFrame received_frame = DeepCopy(frame);
+  analyzer.OnFramePreDecode(kBob, received_frame.id(),
+                            FakeEncode(received_frame));
+  analyzer.OnFrameDecoded(kBob, received_frame,
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered(kBob, received_frame);
+  // Receive by Charlie
+  received_frame = DeepCopy(frame);
+  analyzer.OnFramePreDecode(kCharlie, received_frame.id(),
+                            FakeEncode(received_frame));
+  analyzer.OnFrameDecoded(kCharlie, received_frame,
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered(kCharlie, received_frame);
+
+  // Encode 2nd simulcast layer
+  analyzer.OnFrameEncoded(kAlice, frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.comparisons_done, 2);
+
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 0)
+      << ToString(frames_in_flight_sizes);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, 1);
+  EXPECT_EQ(frame_counters.rendered, 2);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     FrameCanBeReceivedBySenderAfterItWasReceivedByReceiver) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.enable_receive_own_stream = true;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  std::vector<VideoFrame> frames;
+  for (int i = 0; i < 3; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), 1);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames.push_back(frame);
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  // Receive by 2nd peer.
+  for (VideoFrame& frame : frames) {
+    VideoFrame received_frame = DeepCopy(frame);
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Check that we still have that frame in flight.
+  AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats();
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(analyzer_stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 3)
+      << "Expected that frame is still in flight, "
+      << "because it wasn't received by sender"
+      << ToString(frames_in_flight_sizes);
+
+  // Receive by sender
+  for (VideoFrame& frame : frames) {
+    VideoFrame received_frame = DeepCopy(frame);
+    analyzer.OnFramePreDecode(kSenderPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kSenderPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kSenderPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  analyzer_stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(analyzer_stats.comparisons_done, 6);
+
+  frames_in_flight_sizes =
+      GetSortedSamples(analyzer_stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 0)
+      << ToString(frames_in_flight_sizes);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, 3);
+  EXPECT_EQ(frame_counters.rendered, 6);
+
+  EXPECT_EQ(analyzer.GetStats().size(), 2lu);
+  {
+    FrameCounters stream_conters = analyzer.GetPerStreamCounters().at(
+        StatsKey(kStreamLabel, kReceiverPeerName));
+    EXPECT_EQ(stream_conters.captured, 3);
+    EXPECT_EQ(stream_conters.pre_encoded, 3);
+    EXPECT_EQ(stream_conters.encoded, 3);
+    EXPECT_EQ(stream_conters.received, 3);
+    EXPECT_EQ(stream_conters.decoded, 3);
+    EXPECT_EQ(stream_conters.rendered, 3);
+  }
+  {
+    FrameCounters stream_conters = analyzer.GetPerStreamCounters().at(
+        StatsKey(kStreamLabel, kSenderPeerName));
+    EXPECT_EQ(stream_conters.captured, 3);
+    EXPECT_EQ(stream_conters.pre_encoded, 3);
+    EXPECT_EQ(stream_conters.encoded, 3);
+    EXPECT_EQ(stream_conters.received, 3);
+    EXPECT_EQ(stream_conters.decoded, 3);
+    EXPECT_EQ(stream_conters.rendered, 3);
+  }
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     FrameCanBeReceivedByReceiverAfterItWasReceivedBySender) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.enable_receive_own_stream = true;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  std::vector<VideoFrame> frames;
+  for (int i = 0; i < 3; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), 1);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames.push_back(frame);
+    analyzer.OnFramePreEncode(kSenderPeerName, frame);
+    analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+
+  // Receive by sender
+  for (VideoFrame& frame : frames) {
+    VideoFrame received_frame = DeepCopy(frame);
+    analyzer.OnFramePreDecode(kSenderPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kSenderPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kSenderPeerName, received_frame);
+  }
+
+  // Check that we still have that frame in flight.
+  AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats();
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(analyzer_stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 3)
+      << "Expected that frame is still in flight, "
+      << "because it wasn't received by sender"
+      << ToString(frames_in_flight_sizes);
+
+  // Receive by 2nd peer.
+  for (VideoFrame& frame : frames) {
+    VideoFrame received_frame = DeepCopy(frame);
+    analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                              FakeEncode(received_frame));
+    analyzer.OnFrameDecoded(kReceiverPeerName, received_frame,
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  analyzer_stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(analyzer_stats.comparisons_done, 6);
+
+  frames_in_flight_sizes =
+      GetSortedSamples(analyzer_stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 0)
+      << ToString(frames_in_flight_sizes);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, 3);
+  EXPECT_EQ(frame_counters.rendered, 6);
+
+  EXPECT_EQ(analyzer.GetStats().size(), 2lu);
+  {
+    FrameCounters stream_conters = analyzer.GetPerStreamCounters().at(
+        StatsKey(kStreamLabel, kReceiverPeerName));
+    EXPECT_EQ(stream_conters.captured, 3);
+    EXPECT_EQ(stream_conters.pre_encoded, 3);
+    EXPECT_EQ(stream_conters.encoded, 3);
+    EXPECT_EQ(stream_conters.received, 3);
+    EXPECT_EQ(stream_conters.decoded, 3);
+    EXPECT_EQ(stream_conters.rendered, 3);
+  }
+  {
+    FrameCounters stream_conters = analyzer.GetPerStreamCounters().at(
+        StatsKey(kStreamLabel, kSenderPeerName));
+    EXPECT_EQ(stream_conters.captured, 3);
+    EXPECT_EQ(stream_conters.pre_encoded, 3);
+    EXPECT_EQ(stream_conters.encoded, 3);
+    EXPECT_EQ(stream_conters.received, 3);
+    EXPECT_EQ(stream_conters.decoded, 3);
+    EXPECT_EQ(stream_conters.rendered, 3);
+  }
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, CodecTrackedCorrectly) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(),
+                                       AnalyzerOptionsForTest());
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoQualityAnalyzerInterface::EncoderStats encoder_stats;
+  std::vector<std::string> codec_names = {"codec_1", "codec_2"};
+  std::vector<VideoFrame> frames;
+  // Send 3 frame for each codec.
+  for (size_t i = 0; i < codec_names.size(); ++i) {
+    for (size_t j = 0; j < 3; ++j) {
+      VideoFrame frame = NextFrame(frame_generator.get(), 3 * i + j);
+      frame.set_id(
+          analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+      analyzer.OnFramePreEncode(kSenderPeerName, frame);
+      encoder_stats.encoder_name = codec_names[i];
+      analyzer.OnFrameEncoded(kSenderPeerName, frame.id(), FakeEncode(frame),
+                              encoder_stats, false);
+      frames.push_back(std::move(frame));
+    }
+  }
+
+  // Receive 3 frame for each codec.
+  VideoQualityAnalyzerInterface::DecoderStats decoder_stats;
+  for (size_t i = 0; i < codec_names.size(); ++i) {
+    for (size_t j = 0; j < 3; ++j) {
+      VideoFrame received_frame = DeepCopy(frames[3 * i + j]);
+      analyzer.OnFramePreDecode(kReceiverPeerName, received_frame.id(),
+                                FakeEncode(received_frame));
+      decoder_stats.decoder_name = codec_names[i];
+      analyzer.OnFrameDecoded(kReceiverPeerName, received_frame, decoder_stats);
+      analyzer.OnFrameRendered(kReceiverPeerName, received_frame);
+    }
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  std::map<StatsKey, StreamStats> stats = analyzer.GetStats();
+  ASSERT_EQ(stats.size(), 1lu);
+  const StreamStats& stream_stats =
+      stats.at(StatsKey(kStreamLabel, kReceiverPeerName));
+  ASSERT_EQ(stream_stats.encoders.size(), 2lu);
+  EXPECT_EQ(stream_stats.encoders[0].codec_name, codec_names[0]);
+  EXPECT_EQ(stream_stats.encoders[0].first_frame_id, frames[0].id());
+  EXPECT_EQ(stream_stats.encoders[0].last_frame_id, frames[2].id());
+  EXPECT_EQ(stream_stats.encoders[1].codec_name, codec_names[1]);
+  EXPECT_EQ(stream_stats.encoders[1].first_frame_id, frames[3].id());
+  EXPECT_EQ(stream_stats.encoders[1].last_frame_id, frames[5].id());
+
+  ASSERT_EQ(stream_stats.decoders.size(), 2lu);
+  EXPECT_EQ(stream_stats.decoders[0].codec_name, codec_names[0]);
+  EXPECT_EQ(stream_stats.decoders[0].first_frame_id, frames[0].id());
+  EXPECT_EQ(stream_stats.decoders[0].last_frame_id, frames[2].id());
+  EXPECT_EQ(stream_stats.decoders[1].codec_name, codec_names[1]);
+  EXPECT_EQ(stream_stats.decoders[1].first_frame_id, frames[3].id());
+  EXPECT_EQ(stream_stats.decoders[1].last_frame_id, frames[5].id());
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     FramesInFlightAreCorrectlySentToTheComparatorAfterStop) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  // There are 7 different timings inside frame stats: captured, pre_encode,
+  // encoded, received, decode_start, decode_end, rendered. captured is always
+  // set and received is set together with decode_start. So we create 6
+  // different frames, where for each frame next timings will be set
+  //   * 1st - all of them set
+  //   * 2nd - captured, pre_encode, encoded, received, decode_start, decode_end
+  //   * 3rd - captured, pre_encode, encoded, received, decode_start
+  //   * 4th - captured, pre_encode, encoded
+  //   * 5th - captured, pre_encode
+  //   * 6th - captured
+  std::vector<VideoFrame> frames;
+  // Sender side actions
+  for (int i = 0; i < 6; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), 1);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames.push_back(frame);
+  }
+  for (int i = 0; i < 5; ++i) {
+    analyzer.OnFramePreEncode(kSenderPeerName, frames[i]);
+  }
+  for (int i = 0; i < 4; ++i) {
+    analyzer.OnFrameEncoded(
+        kSenderPeerName, frames[i].id(), FakeEncode(frames[i]),
+        VideoQualityAnalyzerInterface::EncoderStats(), false);
+  }
+
+  // Receiver side actions
+  for (int i = 0; i < 3; ++i) {
+    analyzer.OnFramePreDecode(kReceiverPeerName, frames[i].id(),
+                              FakeEncode(frames[i]));
+  }
+  for (int i = 0; i < 2; ++i) {
+    analyzer.OnFrameDecoded(kReceiverPeerName, DeepCopy(frames[i]),
+                            VideoQualityAnalyzerInterface::DecoderStats());
+  }
+  for (int i = 0; i < 1; ++i) {
+    analyzer.OnFrameRendered(kReceiverPeerName, DeepCopy(frames[i]));
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(analyzer_stats.comparisons_done, 6);
+
+  // The last frames in flight size has to reflect the amount of frame in flight
+  // before all of them were sent to the comparison when Stop() was invoked.
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(analyzer_stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 5)
+      << ToString(frames_in_flight_sizes);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, 6);
+  EXPECT_EQ(frame_counters.pre_encoded, 5);
+  EXPECT_EQ(frame_counters.encoded, 4);
+  EXPECT_EQ(frame_counters.received, 3);
+  EXPECT_EQ(frame_counters.decoded, 2);
+  EXPECT_EQ(frame_counters.rendered, 1);
+
+  EXPECT_EQ(analyzer.GetStats().size(), 1lu);
+  {
+    FrameCounters stream_conters = analyzer.GetPerStreamCounters().at(
+        StatsKey(kStreamLabel, kReceiverPeerName));
+    EXPECT_EQ(stream_conters.captured, 6);
+    EXPECT_EQ(stream_conters.pre_encoded, 5);
+    EXPECT_EQ(stream_conters.encoded, 4);
+    EXPECT_EQ(stream_conters.received, 3);
+    EXPECT_EQ(stream_conters.decoded, 2);
+    EXPECT_EQ(stream_conters.rendered, 1);
+  }
+}
+
+TEST(
+    DefaultVideoQualityAnalyzerTest,
+    FramesInFlightAreCorrectlySentToTheComparatorAfterStopForSenderAndReceiver) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.enable_receive_own_stream = true;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{kSenderPeerName, kReceiverPeerName},
+                 kAnalyzerMaxThreadsCount);
+
+  // There are 7 different timings inside frame stats: captured, pre_encode,
+  // encoded, received, decode_start, decode_end, rendered. captured is always
+  // set and received is set together with decode_start. So we create 6
+  // different frames, where for each frame next timings will be set
+  //   * 1st - all of them set
+  //   * 2nd - captured, pre_encode, encoded, received, decode_start, decode_end
+  //   * 3rd - captured, pre_encode, encoded, received, decode_start
+  //   * 4th - captured, pre_encode, encoded
+  //   * 5th - captured, pre_encode
+  //   * 6th - captured
+  std::vector<VideoFrame> frames;
+  // Sender side actions
+  for (int i = 0; i < 6; ++i) {
+    VideoFrame frame = NextFrame(frame_generator.get(), 1);
+    frame.set_id(
+        analyzer.OnFrameCaptured(kSenderPeerName, kStreamLabel, frame));
+    frames.push_back(frame);
+  }
+  for (int i = 0; i < 5; ++i) {
+    analyzer.OnFramePreEncode(kSenderPeerName, frames[i]);
+  }
+  for (int i = 0; i < 4; ++i) {
+    analyzer.OnFrameEncoded(
+        kSenderPeerName, frames[i].id(), FakeEncode(frames[i]),
+        VideoQualityAnalyzerInterface::EncoderStats(), false);
+  }
+
+  // Receiver side actions
+  for (int i = 0; i < 3; ++i) {
+    analyzer.OnFramePreDecode(kSenderPeerName, frames[i].id(),
+                              FakeEncode(frames[i]));
+    analyzer.OnFramePreDecode(kReceiverPeerName, frames[i].id(),
+                              FakeEncode(frames[i]));
+  }
+  for (int i = 0; i < 2; ++i) {
+    analyzer.OnFrameDecoded(kSenderPeerName, DeepCopy(frames[i]),
+                            VideoQualityAnalyzerInterface::DecoderStats());
+    analyzer.OnFrameDecoded(kReceiverPeerName, DeepCopy(frames[i]),
+                            VideoQualityAnalyzerInterface::DecoderStats());
+  }
+  for (int i = 0; i < 1; ++i) {
+    analyzer.OnFrameRendered(kSenderPeerName, DeepCopy(frames[i]));
+    analyzer.OnFrameRendered(kReceiverPeerName, DeepCopy(frames[i]));
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats analyzer_stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(analyzer_stats.comparisons_done, 12);
+
+  // The last frames in flight size has to reflect the amount of frame in flight
+  // before all of them were sent to the comparison when Stop() was invoked.
+  std::vector<StatsSample> frames_in_flight_sizes =
+      GetSortedSamples(analyzer_stats.frames_in_flight_left_count);
+  EXPECT_EQ(frames_in_flight_sizes.back().value, 5)
+      << ToString(frames_in_flight_sizes);
+
+  FrameCounters frame_counters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(frame_counters.captured, 6);
+  EXPECT_EQ(frame_counters.pre_encoded, 5);
+  EXPECT_EQ(frame_counters.encoded, 4);
+  EXPECT_EQ(frame_counters.received, 6);
+  EXPECT_EQ(frame_counters.decoded, 4);
+  EXPECT_EQ(frame_counters.rendered, 2);
+
+  EXPECT_EQ(analyzer.GetStats().size(), 2lu);
+  {
+    FrameCounters stream_conters = analyzer.GetPerStreamCounters().at(
+        StatsKey(kStreamLabel, kReceiverPeerName));
+    EXPECT_EQ(stream_conters.captured, 6);
+    EXPECT_EQ(stream_conters.pre_encoded, 5);
+    EXPECT_EQ(stream_conters.encoded, 4);
+    EXPECT_EQ(stream_conters.received, 3);
+    EXPECT_EQ(stream_conters.decoded, 2);
+    EXPECT_EQ(stream_conters.rendered, 1);
+  }
+  {
+    FrameCounters stream_conters = analyzer.GetPerStreamCounters().at(
+        StatsKey(kStreamLabel, kSenderPeerName));
+    EXPECT_EQ(stream_conters.captured, 6);
+    EXPECT_EQ(stream_conters.pre_encoded, 5);
+    EXPECT_EQ(stream_conters.encoded, 4);
+    EXPECT_EQ(stream_conters.received, 3);
+    EXPECT_EQ(stream_conters.decoded, 2);
+    EXPECT_EQ(stream_conters.rendered, 1);
+  }
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, GetStreamFrames) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  // The order in which peers captured frames and passed them to analyzer.
+  std::vector<std::string> frame_capturers_sequence{
+      "alice", "alice", "bob",   "bob",   "bob",
+      "bob",   "bob",   "alice", "alice", "alice",
+  };
+
+  std::map<std::string, std::vector<uint16_t>> stream_to_frame_ids;
+  stream_to_frame_ids.emplace("alice_video", std::vector<uint16_t>{});
+  stream_to_frame_ids.emplace("bob_video", std::vector<uint16_t>{});
+
+  std::vector<VideoFrame> frames;
+  for (const std::string& sender : frame_capturers_sequence) {
+    VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+    uint16_t frame_id =
+        analyzer.OnFrameCaptured(sender, sender + "_video", frame);
+    frame.set_id(frame_id);
+    stream_to_frame_ids.find(sender + "_video")->second.push_back(frame_id);
+    frames.push_back(frame);
+    analyzer.OnFramePreEncode(sender, frame);
+    analyzer.OnFrameEncoded(sender, frame.id(), FakeEncode(frame),
+                            VideoQualityAnalyzerInterface::EncoderStats(),
+                            false);
+  }
+  // We don't need to receive frames for stats to be gathered correctly.
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  EXPECT_EQ(analyzer.GetStreamFrames(), stream_to_frame_ids);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, ReceiverReceivedFramesWhenSenderRemoved) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", frame);
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  analyzer.UnregisterParticipantInCall("alice");
+
+  analyzer.OnFramePreDecode("bob", frame.id(), FakeEncode(frame));
+  analyzer.OnFrameDecoded("bob", DeepCopy(frame),
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered("bob", DeepCopy(frame));
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(stream_conters.captured, 1);
+  EXPECT_EQ(stream_conters.pre_encoded, 1);
+  EXPECT_EQ(stream_conters.encoded, 1);
+  EXPECT_EQ(stream_conters.received, 1);
+  EXPECT_EQ(stream_conters.decoded, 1);
+  EXPECT_EQ(stream_conters.rendered, 1);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     ReceiverReceivedFramesWhenSenderRemovedWithSelfview) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.enable_receive_own_stream = true;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", frame);
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  analyzer.UnregisterParticipantInCall("alice");
+
+  analyzer.OnFramePreDecode("bob", frame.id(), FakeEncode(frame));
+  analyzer.OnFrameDecoded("bob", DeepCopy(frame),
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered("bob", DeepCopy(frame));
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(stream_conters.captured, 1);
+  EXPECT_EQ(stream_conters.pre_encoded, 1);
+  EXPECT_EQ(stream_conters.encoded, 1);
+  EXPECT_EQ(stream_conters.received, 1);
+  EXPECT_EQ(stream_conters.decoded, 1);
+  EXPECT_EQ(stream_conters.rendered, 1);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     SenderReceivedFramesWhenReceiverRemovedWithSelfview) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.enable_receive_own_stream = true;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", frame);
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  analyzer.UnregisterParticipantInCall("bob");
+
+  analyzer.OnFramePreDecode("alice", frame.id(), FakeEncode(frame));
+  analyzer.OnFrameDecoded("alice", DeepCopy(frame),
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered("alice", DeepCopy(frame));
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "alice"));
+  EXPECT_EQ(stream_conters.captured, 1);
+  EXPECT_EQ(stream_conters.pre_encoded, 1);
+  EXPECT_EQ(stream_conters.encoded, 1);
+  EXPECT_EQ(stream_conters.received, 1);
+  EXPECT_EQ(stream_conters.decoded, 1);
+  EXPECT_EQ(stream_conters.rendered, 1);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     SenderAndReceiverReceivedFramesWhenReceiverRemovedWithSelfview) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.enable_receive_own_stream = true;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", frame);
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  analyzer.OnFramePreDecode("bob", frame.id(), FakeEncode(frame));
+  analyzer.OnFrameDecoded("bob", DeepCopy(frame),
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered("bob", DeepCopy(frame));
+
+  analyzer.UnregisterParticipantInCall("bob");
+
+  analyzer.OnFramePreDecode("alice", frame.id(), FakeEncode(frame));
+  analyzer.OnFrameDecoded("alice", DeepCopy(frame),
+                          VideoQualityAnalyzerInterface::DecoderStats());
+  analyzer.OnFrameRendered("alice", DeepCopy(frame));
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters alice_alice_stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "alice"));
+  EXPECT_EQ(alice_alice_stream_conters.captured, 1);
+  EXPECT_EQ(alice_alice_stream_conters.pre_encoded, 1);
+  EXPECT_EQ(alice_alice_stream_conters.encoded, 1);
+  EXPECT_EQ(alice_alice_stream_conters.received, 1);
+  EXPECT_EQ(alice_alice_stream_conters.decoded, 1);
+  EXPECT_EQ(alice_alice_stream_conters.rendered, 1);
+
+  FrameCounters alice_bob_stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(alice_bob_stream_conters.captured, 1);
+  EXPECT_EQ(alice_bob_stream_conters.pre_encoded, 1);
+  EXPECT_EQ(alice_bob_stream_conters.encoded, 1);
+  EXPECT_EQ(alice_bob_stream_conters.received, 1);
+  EXPECT_EQ(alice_bob_stream_conters.decoded, 1);
+  EXPECT_EQ(alice_bob_stream_conters.rendered, 1);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforeCapturing2ndFrame) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {"bob"},
+                            /*frames_count=*/1, *frame_generator);
+  analyzer.UnregisterParticipantInCall("bob");
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {},
+                            /*frames_count=*/1, *frame_generator);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters global_stream_conters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(global_stream_conters.captured, 2);
+  EXPECT_EQ(global_stream_conters.pre_encoded, 2);
+  EXPECT_EQ(global_stream_conters.encoded, 2);
+  EXPECT_EQ(global_stream_conters.received, 1);
+  EXPECT_EQ(global_stream_conters.decoded, 1);
+  EXPECT_EQ(global_stream_conters.rendered, 1);
+  FrameCounters stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(stream_conters.captured, 2);
+  EXPECT_EQ(stream_conters.pre_encoded, 2);
+  EXPECT_EQ(stream_conters.encoded, 2);
+  EXPECT_EQ(stream_conters.received, 1);
+  EXPECT_EQ(stream_conters.decoded, 1);
+  EXPECT_EQ(stream_conters.rendered, 1);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforePreEncoded) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.UnregisterParticipantInCall("bob");
+  analyzer.OnFramePreEncode("alice", frame);
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters global_stream_conters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(global_stream_conters.captured, 1);
+  EXPECT_EQ(global_stream_conters.pre_encoded, 1);
+  EXPECT_EQ(global_stream_conters.encoded, 1);
+  EXPECT_EQ(global_stream_conters.received, 0);
+  EXPECT_EQ(global_stream_conters.decoded, 0);
+  EXPECT_EQ(global_stream_conters.rendered, 0);
+  FrameCounters stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(stream_conters.captured, 1);
+  EXPECT_EQ(stream_conters.pre_encoded, 1);
+  EXPECT_EQ(stream_conters.encoded, 1);
+  EXPECT_EQ(stream_conters.received, 0);
+  EXPECT_EQ(stream_conters.decoded, 0);
+  EXPECT_EQ(stream_conters.rendered, 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, ReceiverRemovedBeforeEncoded) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", frame);
+  analyzer.UnregisterParticipantInCall("bob");
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters global_stream_conters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(global_stream_conters.captured, 1);
+  EXPECT_EQ(global_stream_conters.pre_encoded, 1);
+  EXPECT_EQ(global_stream_conters.encoded, 1);
+  EXPECT_EQ(global_stream_conters.received, 0);
+  EXPECT_EQ(global_stream_conters.decoded, 0);
+  EXPECT_EQ(global_stream_conters.rendered, 0);
+  FrameCounters stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(stream_conters.captured, 1);
+  EXPECT_EQ(stream_conters.pre_encoded, 1);
+  EXPECT_EQ(stream_conters.encoded, 1);
+  EXPECT_EQ(stream_conters.received, 0);
+  EXPECT_EQ(stream_conters.decoded, 0);
+  EXPECT_EQ(stream_conters.rendered, 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     ReceiverRemovedBetweenSimulcastLayersEncoded) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", frame);
+  // 1st simulcast layer encoded
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+  analyzer.UnregisterParticipantInCall("bob");
+  // 2nd simulcast layer encoded
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters global_stream_conters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(global_stream_conters.captured, 1);
+  EXPECT_EQ(global_stream_conters.pre_encoded, 1);
+  EXPECT_EQ(global_stream_conters.encoded, 1);
+  EXPECT_EQ(global_stream_conters.received, 0);
+  EXPECT_EQ(global_stream_conters.decoded, 0);
+  EXPECT_EQ(global_stream_conters.rendered, 0);
+  FrameCounters stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(stream_conters.captured, 1);
+  EXPECT_EQ(stream_conters.pre_encoded, 1);
+  EXPECT_EQ(stream_conters.encoded, 1);
+  EXPECT_EQ(stream_conters.received, 0);
+  EXPECT_EQ(stream_conters.decoded, 0);
+  EXPECT_EQ(stream_conters.rendered, 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, UnregisterOneAndRegisterAnother) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{"alice", "bob", "charlie"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video",
+                            {"bob", "charlie"},
+                            /*frames_count=*/2, *frame_generator);
+  analyzer.UnregisterParticipantInCall("bob");
+  analyzer.RegisterParticipantInCall("david");
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video",
+                            {"charlie", "david"},
+                            /*frames_count=*/4, *frame_generator);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters global_stream_conters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(global_stream_conters.captured, 6);
+  EXPECT_EQ(global_stream_conters.pre_encoded, 6);
+  EXPECT_EQ(global_stream_conters.encoded, 6);
+  EXPECT_EQ(global_stream_conters.received, 12);
+  EXPECT_EQ(global_stream_conters.decoded, 12);
+  EXPECT_EQ(global_stream_conters.rendered, 12);
+  FrameCounters alice_bob_stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(alice_bob_stream_conters.captured, 6);
+  EXPECT_EQ(alice_bob_stream_conters.pre_encoded, 6);
+  EXPECT_EQ(alice_bob_stream_conters.encoded, 6);
+  EXPECT_EQ(alice_bob_stream_conters.received, 2);
+  EXPECT_EQ(alice_bob_stream_conters.decoded, 2);
+  EXPECT_EQ(alice_bob_stream_conters.rendered, 2);
+  FrameCounters alice_charlie_stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "charlie"));
+  EXPECT_EQ(alice_charlie_stream_conters.captured, 6);
+  EXPECT_EQ(alice_charlie_stream_conters.pre_encoded, 6);
+  EXPECT_EQ(alice_charlie_stream_conters.encoded, 6);
+  EXPECT_EQ(alice_charlie_stream_conters.received, 6);
+  EXPECT_EQ(alice_charlie_stream_conters.decoded, 6);
+  EXPECT_EQ(alice_charlie_stream_conters.rendered, 6);
+  FrameCounters alice_david_stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "david"));
+  EXPECT_EQ(alice_david_stream_conters.captured, 6);
+  EXPECT_EQ(alice_david_stream_conters.pre_encoded, 6);
+  EXPECT_EQ(alice_david_stream_conters.encoded, 6);
+  EXPECT_EQ(alice_david_stream_conters.received, 4);
+  EXPECT_EQ(alice_david_stream_conters.decoded, 4);
+  EXPECT_EQ(alice_david_stream_conters.rendered, 4);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     UnregisterOneAndRegisterAnotherRegisterBack) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case",
+                 std::vector<std::string>{"alice", "bob", "charlie"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video",
+                            {"bob", "charlie"},
+                            /*frames_count=*/2, *frame_generator);
+  analyzer.UnregisterParticipantInCall("bob");
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {"charlie"},
+                            /*frames_count=*/4, *frame_generator);
+  analyzer.RegisterParticipantInCall("bob");
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video",
+                            {"bob", "charlie"},
+                            /*frames_count=*/6, *frame_generator);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  FrameCounters global_stream_conters = analyzer.GetGlobalCounters();
+  EXPECT_EQ(global_stream_conters.captured, 12);
+  EXPECT_EQ(global_stream_conters.pre_encoded, 12);
+  EXPECT_EQ(global_stream_conters.encoded, 12);
+  EXPECT_EQ(global_stream_conters.received, 20);
+  EXPECT_EQ(global_stream_conters.decoded, 20);
+  EXPECT_EQ(global_stream_conters.rendered, 20);
+  FrameCounters alice_bob_stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "bob"));
+  EXPECT_EQ(alice_bob_stream_conters.captured, 12);
+  EXPECT_EQ(alice_bob_stream_conters.pre_encoded, 12);
+  EXPECT_EQ(alice_bob_stream_conters.encoded, 12);
+  EXPECT_EQ(alice_bob_stream_conters.received, 8);
+  EXPECT_EQ(alice_bob_stream_conters.decoded, 8);
+  EXPECT_EQ(alice_bob_stream_conters.rendered, 8);
+  FrameCounters alice_charlie_stream_conters =
+      analyzer.GetPerStreamCounters().at(StatsKey("alice_video", "charlie"));
+  EXPECT_EQ(alice_charlie_stream_conters.captured, 12);
+  EXPECT_EQ(alice_charlie_stream_conters.pre_encoded, 12);
+  EXPECT_EQ(alice_charlie_stream_conters.encoded, 12);
+  EXPECT_EQ(alice_charlie_stream_conters.received, 12);
+  EXPECT_EQ(alice_charlie_stream_conters.decoded, 12);
+  EXPECT_EQ(alice_charlie_stream_conters.rendered, 12);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     FramesInFlightAreAccountedForUnregisterPeers) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  // Add one frame in flight which has encode time >= 10ms.
+  VideoFrame frame = NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id = analyzer.OnFrameCaptured("alice", "alice_video", frame);
+  frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", frame);
+  SleepMs(10);
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+
+  analyzer.UnregisterParticipantInCall("bob");
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  StreamStats stats = analyzer.GetStats().at(StatsKey("alice_video", "bob"));
+  ASSERT_EQ(stats.encode_time_ms.NumSamples(), 1);
+  EXPECT_GE(stats.encode_time_ms.GetAverage(), 10);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, InfraMetricsAreReportedWhenRequested) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.report_infra_metrics = true;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {"bob"},
+                            /*frames_count=*/1, *frame_generator);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.on_frame_captured_processing_time_ms.NumSamples(), 1);
+  EXPECT_EQ(stats.on_frame_pre_encode_processing_time_ms.NumSamples(), 1);
+  EXPECT_EQ(stats.on_frame_encoded_processing_time_ms.NumSamples(), 1);
+  EXPECT_EQ(stats.on_frame_pre_decode_processing_time_ms.NumSamples(), 1);
+  EXPECT_EQ(stats.on_frame_decoded_processing_time_ms.NumSamples(), 1);
+  EXPECT_EQ(stats.on_frame_rendered_processing_time_ms.NumSamples(), 1);
+  EXPECT_EQ(stats.on_decoder_error_processing_time_ms.NumSamples(), 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest, InfraMetricsNotCollectedByDefault) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.report_infra_metrics = false;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {"bob"},
+                            /*frames_count=*/1, *frame_generator);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  AnalyzerStats stats = analyzer.GetAnalyzerStats();
+  EXPECT_EQ(stats.on_frame_captured_processing_time_ms.NumSamples(), 0);
+  EXPECT_EQ(stats.on_frame_pre_encode_processing_time_ms.NumSamples(), 0);
+  EXPECT_EQ(stats.on_frame_encoded_processing_time_ms.NumSamples(), 0);
+  EXPECT_EQ(stats.on_frame_pre_decode_processing_time_ms.NumSamples(), 0);
+  EXPECT_EQ(stats.on_frame_decoded_processing_time_ms.NumSamples(), 0);
+  EXPECT_EQ(stats.on_frame_rendered_processing_time_ms.NumSamples(), 0);
+  EXPECT_EQ(stats.on_decoder_error_processing_time_ms.NumSamples(), 0);
+}
+
+TEST(DefaultVideoQualityAnalyzerTest,
+     FrameDroppedByDecoderIsAccountedCorrectly) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  options.report_infra_metrics = false;
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  VideoFrame to_be_dropped_frame =
+      NextFrame(frame_generator.get(), /*timestamp_us=*/1);
+  uint16_t frame_id =
+      analyzer.OnFrameCaptured("alice", "alice_video", to_be_dropped_frame);
+  to_be_dropped_frame.set_id(frame_id);
+  analyzer.OnFramePreEncode("alice", to_be_dropped_frame);
+  analyzer.OnFrameEncoded("alice", to_be_dropped_frame.id(),
+                          FakeEncode(to_be_dropped_frame),
+                          VideoQualityAnalyzerInterface::EncoderStats(), false);
+  VideoFrame received_to_be_dropped_frame = DeepCopy(to_be_dropped_frame);
+  analyzer.OnFramePreDecode("bob", received_to_be_dropped_frame.id(),
+                            FakeEncode(received_to_be_dropped_frame));
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {"bob"},
+                            /*frames_count=*/1, *frame_generator);
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(100);
+  analyzer.Stop();
+
+  StreamStats stats = analyzer.GetStats().at(StatsKey("alice_video", "bob"));
+  ASSERT_EQ(stats.dropped_by_phase[FrameDropPhase::kByDecoder], 1);
+}
+
+class DefaultVideoQualityAnalyzerTimeBetweenFreezesTest
+    : public TestWithParam<bool> {};
+
+TEST_P(DefaultVideoQualityAnalyzerTimeBetweenFreezesTest,
+       TimeBetweenFreezesIsEqualToStreamDurationWhenThereAreNoFeeezes) {
+  std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+      test::CreateSquareFrameGenerator(kFrameWidth, kFrameHeight,
+                                       /*type=*/absl::nullopt,
+                                       /*num_squares=*/absl::nullopt);
+
+  DefaultVideoQualityAnalyzerOptions options = AnalyzerOptionsForTest();
+  DefaultVideoQualityAnalyzer analyzer(Clock::GetRealTimeClock(),
+                                       test::GetGlobalMetricsLogger(), options);
+  analyzer.Start("test_case", std::vector<std::string>{"alice", "bob"},
+                 kAnalyzerMaxThreadsCount);
+
+  PassFramesThroughAnalyzer(analyzer, "alice", "alice_video", {"bob"},
+                            /*frames_count=*/5, *frame_generator,
+                            /*interframe_delay_ms=*/50);
+  if (GetParam()) {
+    analyzer.UnregisterParticipantInCall("bob");
+  }
+
+  // Give analyzer some time to process frames on async thread. The computations
+  // have to be fast (heavy metrics are disabled!), so if doesn't fit 100ms it
+  // means we have an issue!
+  SleepMs(50);
+  analyzer.Stop();
+
+  StreamStats stats = analyzer.GetStats().at(StatsKey("alice_video", "bob"));
+  ASSERT_EQ(stats.time_between_freezes_ms.NumSamples(), 1);
+  EXPECT_GE(stats.time_between_freezes_ms.GetAverage(), 200);
+}
+
+INSTANTIATE_TEST_SUITE_P(WithRegisteredAndUnregisteredPeerAtTheEndOfTheCall,
+                         DefaultVideoQualityAnalyzerTimeBetweenFreezesTest,
+                         ValuesIn({true, false}));
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/encoded_image_data_injector.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/encoded_image_data_injector.h
new file mode 100644
index 0000000000..384e901462
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/encoded_image_data_injector.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_ENCODED_IMAGE_DATA_INJECTOR_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_ENCODED_IMAGE_DATA_INJECTOR_H_
+
+#include <cstdint>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/video/encoded_image.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Injects frame id into EncodedImage on encoder side
+class EncodedImageDataInjector {
+ public:
+  virtual ~EncodedImageDataInjector() = default;
+
+  // Return encoded image with specified `id` and `discard` flag injected into
+  // its payload. `discard` flag mean does analyzing decoder should discard this
+  // encoded image because it belongs to unnecessary simulcast stream or spatial
+  // layer.
+  virtual EncodedImage InjectData(uint16_t id,
+                                  bool discard,
+                                  const EncodedImage& source) = 0;
+};
+
+struct EncodedImageExtractionResult {
+  absl::optional<uint16_t> id;
+  EncodedImage image;
+  // Is true if encoded image should be discarded. It is used to filter out
+  // unnecessary spatial layers and simulcast streams.
+  bool discard;
+};
+
+// Extracts frame id from EncodedImage on decoder side.
+class EncodedImageDataExtractor {
+ public:
+  virtual ~EncodedImageDataExtractor() = default;
+
+  // Invoked by framework before any image will come to the extractor.
+  // `expected_receivers_count` is the expected amount of receivers for each
+  // encoded image.
+  virtual void Start(int expected_receivers_count) = 0;
+
+  // Invoked by framework when it is required to add one more receiver for
+  // frames. Will be invoked before that receiver will start receive data.
+  virtual void AddParticipantInCall() = 0;
+
+  // Invoked by framework when it is required to remove receiver for frames.
+  // Will be invoked after that receiver will stop receiving data.
+  virtual void RemoveParticipantInCall() = 0;
+
+  // Returns encoded image id, extracted from payload and also encoded image
+  // with its original payload. For concatenated spatial layers it should be the
+  // same id.
+  virtual EncodedImageExtractionResult ExtractData(
+      const EncodedImage& source) = 0;
+};
+
+class EncodedImageDataPropagator : public EncodedImageDataInjector,
+                                   public EncodedImageDataExtractor {
+ public:
+  ~EncodedImageDataPropagator() override = default;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_ENCODED_IMAGE_DATA_INJECTOR_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
new file mode 100644
index 0000000000..da9c53beb9
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/example_video_quality_analyzer.h"
+
+#include "api/array_view.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+ExampleVideoQualityAnalyzer::ExampleVideoQualityAnalyzer() = default;
+ExampleVideoQualityAnalyzer::~ExampleVideoQualityAnalyzer() = default;
+
+void ExampleVideoQualityAnalyzer::Start(
+    std::string test_case_name,
+    rtc::ArrayView<const std::string> peer_names,
+    int max_threads_count) {}
+
+uint16_t ExampleVideoQualityAnalyzer::OnFrameCaptured(
+    absl::string_view peer_name,
+    const std::string& stream_label,
+    const webrtc::VideoFrame& frame) {
+  MutexLock lock(&lock_);
+  uint16_t frame_id = next_frame_id_++;
+  if (frame_id == VideoFrame::kNotSetId) {
+    frame_id = next_frame_id_++;
+  }
+  auto it = frames_in_flight_.find(frame_id);
+  if (it == frames_in_flight_.end()) {
+    frames_in_flight_.insert(frame_id);
+    frames_to_stream_label_.insert({frame_id, stream_label});
+  } else {
+    RTC_LOG(LS_WARNING) << "Meet new frame with the same id: " << frame_id
+                        << ". Assumes old one as dropped";
+    // We needn't insert frame to frames_in_flight_, because it is already
+    // there.
+    ++frames_dropped_;
+    auto stream_it = frames_to_stream_label_.find(frame_id);
+    RTC_CHECK(stream_it != frames_to_stream_label_.end());
+    stream_it->second = stream_label;
+  }
+  ++frames_captured_;
+  return frame_id;
+}
+
+void ExampleVideoQualityAnalyzer::OnFramePreEncode(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame) {
+  MutexLock lock(&lock_);
+  ++frames_pre_encoded_;
+}
+
+void ExampleVideoQualityAnalyzer::OnFrameEncoded(
+    absl::string_view peer_name,
+    uint16_t frame_id,
+    const webrtc::EncodedImage& encoded_image,
+    const EncoderStats& stats,
+    bool discarded) {
+  MutexLock lock(&lock_);
+  ++frames_encoded_;
+}
+
+void ExampleVideoQualityAnalyzer::OnFrameDropped(
+    absl::string_view peer_name,
+    webrtc::EncodedImageCallback::DropReason reason) {
+  RTC_LOG(LS_INFO) << "Frame dropped by encoder";
+  MutexLock lock(&lock_);
+  ++frames_dropped_;
+}
+
+void ExampleVideoQualityAnalyzer::OnFramePreDecode(
+    absl::string_view peer_name,
+    uint16_t frame_id,
+    const webrtc::EncodedImage& encoded_image) {
+  MutexLock lock(&lock_);
+  ++frames_received_;
+}
+
+void ExampleVideoQualityAnalyzer::OnFrameDecoded(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame,
+    const DecoderStats& stats) {
+  MutexLock lock(&lock_);
+  ++frames_decoded_;
+}
+
+void ExampleVideoQualityAnalyzer::OnFrameRendered(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame) {
+  MutexLock lock(&lock_);
+  frames_in_flight_.erase(frame.id());
+  ++frames_rendered_;
+}
+
+void ExampleVideoQualityAnalyzer::OnEncoderError(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame,
+    int32_t error_code) {
+  RTC_LOG(LS_ERROR) << "Failed to encode frame " << frame.id()
+                    << ". Code: " << error_code;
+}
+
+void ExampleVideoQualityAnalyzer::OnDecoderError(absl::string_view peer_name,
+                                                 uint16_t frame_id,
+                                                 int32_t error_code,
+                                                 const DecoderStats& stats) {
+  RTC_LOG(LS_ERROR) << "Failed to decode frame " << frame_id
+                    << ". Code: " << error_code;
+}
+
+void ExampleVideoQualityAnalyzer::Stop() {
+  MutexLock lock(&lock_);
+  RTC_LOG(LS_INFO) << "There are " << frames_in_flight_.size()
+                   << " frames in flight, assuming all of them are dropped";
+  frames_dropped_ += frames_in_flight_.size();
+}
+
+std::string ExampleVideoQualityAnalyzer::GetStreamLabel(uint16_t frame_id) {
+  MutexLock lock(&lock_);
+  auto it = frames_to_stream_label_.find(frame_id);
+  RTC_DCHECK(it != frames_to_stream_label_.end())
+      << "Unknown frame_id=" << frame_id;
+  return it->second;
+}
+
+uint64_t ExampleVideoQualityAnalyzer::frames_captured() const {
+  MutexLock lock(&lock_);
+  return frames_captured_;
+}
+
+uint64_t ExampleVideoQualityAnalyzer::frames_pre_encoded() const {
+  MutexLock lock(&lock_);
+  return frames_pre_encoded_;
+}
+
+uint64_t ExampleVideoQualityAnalyzer::frames_encoded() const {
+  MutexLock lock(&lock_);
+  return frames_encoded_;
+}
+
+uint64_t ExampleVideoQualityAnalyzer::frames_received() const {
+  MutexLock lock(&lock_);
+  return frames_received_;
+}
+
+uint64_t ExampleVideoQualityAnalyzer::frames_decoded() const {
+  MutexLock lock(&lock_);
+  return frames_decoded_;
+}
+
+uint64_t ExampleVideoQualityAnalyzer::frames_rendered() const {
+  MutexLock lock(&lock_);
+  return frames_rendered_;
+}
+
+uint64_t ExampleVideoQualityAnalyzer::frames_dropped() const {
+  MutexLock lock(&lock_);
+  return frames_dropped_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
new file mode 100644
index 0000000000..af4868a961
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_EXAMPLE_VIDEO_QUALITY_ANALYZER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_EXAMPLE_VIDEO_QUALITY_ANALYZER_H_
+
+#include <atomic>
+#include <map>
+#include <set>
+#include <string>
+
+#include "api/array_view.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// This class is an example implementation of
+// webrtc::VideoQualityAnalyzerInterface and calculates simple metrics
+// just to demonstration purposes. Assumed to be used in the single process
+// test cases, where both peers are in the same process.
+class ExampleVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
+ public:
+  ExampleVideoQualityAnalyzer();
+  ~ExampleVideoQualityAnalyzer() override;
+
+  void Start(std::string test_case_name,
+             rtc::ArrayView<const std::string> peer_names,
+             int max_threads_count) override;
+  uint16_t OnFrameCaptured(absl::string_view peer_name,
+                           const std::string& stream_label,
+                           const VideoFrame& frame) override;
+  void OnFramePreEncode(absl::string_view peer_name,
+                        const VideoFrame& frame) override;
+  void OnFrameEncoded(absl::string_view peer_name,
+                      uint16_t frame_id,
+                      const EncodedImage& encoded_image,
+                      const EncoderStats& stats,
+                      bool discarded) override;
+  void OnFrameDropped(absl::string_view peer_name,
+                      EncodedImageCallback::DropReason reason) override;
+  void OnFramePreDecode(absl::string_view peer_name,
+                        uint16_t frame_id,
+                        const EncodedImage& encoded_image) override;
+  void OnFrameDecoded(absl::string_view peer_name,
+                      const VideoFrame& frame,
+                      const DecoderStats& stats) override;
+  void OnFrameRendered(absl::string_view peer_name,
+                       const VideoFrame& frame) override;
+  void OnEncoderError(absl::string_view peer_name,
+                      const VideoFrame& frame,
+                      int32_t error_code) override;
+  void OnDecoderError(absl::string_view peer_name,
+                      uint16_t frame_id,
+                      int32_t error_code,
+                      const DecoderStats& stats) override;
+  void Stop() override;
+  std::string GetStreamLabel(uint16_t frame_id) override;
+
+  uint64_t frames_captured() const;
+  uint64_t frames_pre_encoded() const;
+  uint64_t frames_encoded() const;
+  uint64_t frames_received() const;
+  uint64_t frames_decoded() const;
+  uint64_t frames_rendered() const;
+  uint64_t frames_dropped() const;
+
+ private:
+  // When peer A captured the frame it will come into analyzer's OnFrameCaptured
+  // and will be stored in frames_in_flight_. It will be removed from there
+  // when it will be received in peer B, so we need to guard it with lock.
+  // Also because analyzer will serve for all video streams it can be called
+  // from different threads inside one peer.
+  mutable Mutex lock_;
+  // Stores frame ids, that are currently going from one peer to another. We
+  // need to keep them to correctly determine dropped frames and also correctly
+  // process frame id overlap.
+  std::set<uint16_t> frames_in_flight_ RTC_GUARDED_BY(lock_);
+  std::map<uint16_t, std::string> frames_to_stream_label_ RTC_GUARDED_BY(lock_);
+  uint16_t next_frame_id_ RTC_GUARDED_BY(lock_) = 1;
+  uint64_t frames_captured_ RTC_GUARDED_BY(lock_) = 0;
+  uint64_t frames_pre_encoded_ RTC_GUARDED_BY(lock_) = 0;
+  uint64_t frames_encoded_ RTC_GUARDED_BY(lock_) = 0;
+  uint64_t frames_received_ RTC_GUARDED_BY(lock_) = 0;
+  uint64_t frames_decoded_ RTC_GUARDED_BY(lock_) = 0;
+  uint64_t frames_rendered_ RTC_GUARDED_BY(lock_) = 0;
+  uint64_t frames_dropped_ RTC_GUARDED_BY(lock_) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_EXAMPLE_VIDEO_QUALITY_ANALYZER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/multi_reader_queue.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/multi_reader_queue.h
new file mode 100644
index 0000000000..39d26b42bc
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/multi_reader_queue.h
@@ -0,0 +1,168 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_MULTI_READER_QUEUE_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_MULTI_READER_QUEUE_H_
+
+#include <deque>
+#include <memory>
+#include <set>
+#include <unordered_map>
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Represents the queue which can be read by multiple readers. Each reader reads
+// from its own queue head. When an element is added it will become visible for
+// all readers. When an element will be removed by all the readers, the element
+// will be removed from the queue.
+template <typename T>
+class MultiReaderQueue {
+ public:
+  // Creates queue with exactly `readers_count` readers named from 0 to
+  // `readers_count - 1`.
+  explicit MultiReaderQueue(size_t readers_count) {
+    for (size_t i = 0; i < readers_count; ++i) {
+      heads_[i] = 0;
+    }
+  }
+  // Creates queue with specified readers.
+  explicit MultiReaderQueue(std::set<size_t> readers) {
+    for (size_t reader : readers) {
+      heads_[reader] = 0;
+    }
+  }
+
+  // Adds a new `reader`, initializing its reading position (the reader's head)
+  // equal to the one of `reader_to_copy`.
+  // Complexity O(MultiReaderQueue::size(reader_to_copy)).
+  void AddReader(size_t reader, size_t reader_to_copy) {
+    size_t pos = GetHeadPositionOrDie(reader_to_copy);
+
+    auto it = heads_.find(reader);
+    RTC_CHECK(it == heads_.end())
+        << "Reader " << reader << " is already in the queue";
+    heads_[reader] = heads_[reader_to_copy];
+    for (size_t i = pos; i < queue_.size(); ++i) {
+      in_queues_[i]++;
+    }
+  }
+
+  // Adds a new `reader`, initializing its reading position equal to first
+  // element in the queue.
+  // Complexity O(MultiReaderQueue::size()).
+  void AddReader(size_t reader) {
+    auto it = heads_.find(reader);
+    RTC_CHECK(it == heads_.end())
+        << "Reader " << reader << " is already in the queue";
+    heads_[reader] = removed_elements_count_;
+    for (size_t i = 0; i < queue_.size(); ++i) {
+      in_queues_[i]++;
+    }
+  }
+
+  // Removes specified `reader` from the queue.
+  // Complexity O(MultiReaderQueue::size(reader)).
+  void RemoveReader(size_t reader) {
+    size_t pos = GetHeadPositionOrDie(reader);
+    for (size_t i = pos; i < queue_.size(); ++i) {
+      in_queues_[i]--;
+    }
+    while (!in_queues_.empty() && in_queues_[0] == 0) {
+      PopFront();
+    }
+    heads_.erase(reader);
+  }
+
+  // Add value to the end of the queue. Complexity O(1).
+  void PushBack(T value) {
+    queue_.push_back(value);
+    in_queues_.push_back(heads_.size());
+  }
+
+  // Extract element from specified head. Complexity O(1).
+  absl::optional<T> PopFront(size_t reader) {
+    size_t pos = GetHeadPositionOrDie(reader);
+    if (pos >= queue_.size()) {
+      return absl::nullopt;
+    }
+
+    T out = queue_[pos];
+
+    in_queues_[pos]--;
+    heads_[reader]++;
+
+    if (in_queues_[pos] == 0) {
+      RTC_DCHECK_EQ(pos, 0);
+      PopFront();
+    }
+    return out;
+  }
+
+  // Returns element at specified head. Complexity O(1).
+  absl::optional<T> Front(size_t reader) const {
+    size_t pos = GetHeadPositionOrDie(reader);
+    if (pos >= queue_.size()) {
+      return absl::nullopt;
+    }
+    return queue_[pos];
+  }
+
+  // Returns true if for specified head there are no more elements in the queue
+  // or false otherwise. Complexity O(1).
+  bool IsEmpty(size_t reader) const {
+    size_t pos = GetHeadPositionOrDie(reader);
+    return pos >= queue_.size();
+  }
+
+  // Returns size of the longest queue between all readers.
+  // Complexity O(1).
+  size_t size() const { return queue_.size(); }
+
+  // Returns size of the specified queue. Complexity O(1).
+  size_t size(size_t reader) const {
+    size_t pos = GetHeadPositionOrDie(reader);
+    return queue_.size() - pos;
+  }
+
+  // Complexity O(1).
+  size_t readers_count() const { return heads_.size(); }
+
+ private:
+  size_t GetHeadPositionOrDie(size_t reader) const {
+    auto it = heads_.find(reader);
+    RTC_CHECK(it != heads_.end()) << "No queue for reader " << reader;
+    return it->second - removed_elements_count_;
+  }
+
+  void PopFront() {
+    RTC_DCHECK(!queue_.empty());
+    RTC_DCHECK_EQ(in_queues_[0], 0);
+    queue_.pop_front();
+    in_queues_.pop_front();
+    removed_elements_count_++;
+  }
+
+  // Number of the elements that were removed from the queue. It is used to
+  // subtract from each head to compute the right index inside `queue_`;
+  size_t removed_elements_count_ = 0;
+  std::deque<T> queue_;
+  // In how may queues the element at index `i` is. An element can be removed
+  // from the front if and only if it is in 0 queues.
+  std::deque<size_t> in_queues_;
+  // Map from the reader to the head position in the queue.
+  std::unordered_map<size_t, size_t> heads_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_MULTI_READER_QUEUE_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/multi_reader_queue_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/multi_reader_queue_test.cc
new file mode 100644
index 0000000000..ea6aa0a416
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/multi_reader_queue_test.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/multi_reader_queue.h"
+
+#include "absl/types/optional.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+TEST(MultiReaderQueueTest, EmptyQueueEmptyForAllHeads) {
+  MultiReaderQueue<int> queue = MultiReaderQueue<int>(/*readers_count=*/10);
+  EXPECT_EQ(queue.size(), 0lu);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_TRUE(queue.IsEmpty(/*reader=*/i));
+    EXPECT_EQ(queue.size(/*reader=*/i), 0lu);
+    EXPECT_FALSE(queue.PopFront(/*reader=*/i).has_value());
+    EXPECT_FALSE(queue.Front(/*reader=*/i).has_value());
+  }
+}
+
+TEST(MultiReaderQueueTest, SizeIsEqualForAllHeadsAfterAddOnly) {
+  MultiReaderQueue<int> queue = MultiReaderQueue<int>(/*readers_count=*/10);
+  queue.PushBack(1);
+  queue.PushBack(2);
+  queue.PushBack(3);
+  EXPECT_EQ(queue.size(), 3lu);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_FALSE(queue.IsEmpty(/*reader=*/i));
+    EXPECT_EQ(queue.size(/*reader=*/i), 3lu);
+  }
+}
+
+TEST(MultiReaderQueueTest, SizeIsCorrectAfterRemoveFromOnlyOneHead) {
+  MultiReaderQueue<int> queue = MultiReaderQueue<int>(/*readers_count=*/10);
+  for (int i = 0; i < 5; ++i) {
+    queue.PushBack(i);
+  }
+  EXPECT_EQ(queue.size(), 5lu);
+  // Removing elements from queue #0
+  for (int i = 0; i < 5; ++i) {
+    EXPECT_EQ(queue.size(/*reader=*/0), static_cast<size_t>(5 - i));
+    EXPECT_EQ(queue.PopFront(/*reader=*/0), absl::optional<int>(i));
+    for (int j = 1; j < 10; ++j) {
+      EXPECT_EQ(queue.size(/*reader=*/j), 5lu);
+    }
+  }
+  EXPECT_EQ(queue.size(/*reader=*/0), 0lu);
+  EXPECT_TRUE(queue.IsEmpty(/*reader=*/0));
+}
+
+TEST(MultiReaderQueueTest, SingleHeadOneAddOneRemove) {
+  MultiReaderQueue<int> queue = MultiReaderQueue<int>(/*readers_count=*/1);
+  queue.PushBack(1);
+  EXPECT_EQ(queue.size(), 1lu);
+  EXPECT_TRUE(queue.Front(/*reader=*/0).has_value());
+  EXPECT_EQ(queue.Front(/*reader=*/0).value(), 1);
+  absl::optional<int> value = queue.PopFront(/*reader=*/0);
+  EXPECT_TRUE(value.has_value());
+  EXPECT_EQ(value.value(), 1);
+  EXPECT_EQ(queue.size(), 0lu);
+  EXPECT_TRUE(queue.IsEmpty(/*reader=*/0));
+}
+
+TEST(MultiReaderQueueTest, SingleHead) {
+  MultiReaderQueue<size_t> queue =
+      MultiReaderQueue<size_t>(/*readers_count=*/1);
+  for (size_t i = 0; i < 10; ++i) {
+    queue.PushBack(i);
+    EXPECT_EQ(queue.size(), i + 1);
+  }
+  for (size_t i = 0; i < 10; ++i) {
+    EXPECT_EQ(queue.Front(/*reader=*/0), absl::optional<size_t>(i));
+    EXPECT_EQ(queue.PopFront(/*reader=*/0), absl::optional<size_t>(i));
+    EXPECT_EQ(queue.size(), 10 - i - 1);
+  }
+}
+
+TEST(MultiReaderQueueTest, ThreeHeadsAddAllRemoveAllPerHead) {
+  MultiReaderQueue<size_t> queue =
+      MultiReaderQueue<size_t>(/*readers_count=*/3);
+  for (size_t i = 0; i < 10; ++i) {
+    queue.PushBack(i);
+    EXPECT_EQ(queue.size(), i + 1);
+  }
+  for (size_t i = 0; i < 10; ++i) {
+    absl::optional<size_t> value = queue.PopFront(/*reader=*/0);
+    EXPECT_EQ(queue.size(), 10lu);
+    ASSERT_TRUE(value.has_value());
+    EXPECT_EQ(value.value(), i);
+  }
+  for (size_t i = 0; i < 10; ++i) {
+    absl::optional<size_t> value = queue.PopFront(/*reader=*/1);
+    EXPECT_EQ(queue.size(), 10lu);
+    ASSERT_TRUE(value.has_value());
+    EXPECT_EQ(value.value(), i);
+  }
+  for (size_t i = 0; i < 10; ++i) {
+    absl::optional<size_t> value = queue.PopFront(/*reader=*/2);
+    EXPECT_EQ(queue.size(), 10 - i - 1);
+    ASSERT_TRUE(value.has_value());
+    EXPECT_EQ(value.value(), i);
+  }
+}
+
+TEST(MultiReaderQueueTest, ThreeHeadsAddAllRemoveAll) {
+  MultiReaderQueue<size_t> queue =
+      MultiReaderQueue<size_t>(/*readers_count=*/3);
+  for (size_t i = 0; i < 10; ++i) {
+    queue.PushBack(i);
+    EXPECT_EQ(queue.size(), i + 1);
+  }
+  for (size_t i = 0; i < 10; ++i) {
+    absl::optional<size_t> value1 = queue.PopFront(/*reader=*/0);
+    absl::optional<size_t> value2 = queue.PopFront(/*reader=*/1);
+    absl::optional<size_t> value3 = queue.PopFront(/*reader=*/2);
+    EXPECT_EQ(queue.size(), 10 - i - 1);
+    ASSERT_TRUE(value1.has_value());
+    ASSERT_TRUE(value2.has_value());
+    ASSERT_TRUE(value3.has_value());
+    EXPECT_EQ(value1.value(), i);
+    EXPECT_EQ(value2.value(), i);
+    EXPECT_EQ(value3.value(), i);
+  }
+}
+
+TEST(MultiReaderQueueTest, AddReaderSeeElementsOnlyFromReaderToCopy) {
+  MultiReaderQueue<size_t> queue =
+      MultiReaderQueue<size_t>(/*readers_count=*/2);
+  for (size_t i = 0; i < 10; ++i) {
+    queue.PushBack(i);
+  }
+  for (size_t i = 0; i < 5; ++i) {
+    queue.PopFront(0);
+  }
+
+  queue.AddReader(/*reader=*/2, /*reader_to_copy=*/0);
+
+  EXPECT_EQ(queue.readers_count(), 3lu);
+  for (size_t i = 5; i < 10; ++i) {
+    absl::optional<size_t> value = queue.PopFront(/*reader=*/2);
+    EXPECT_EQ(queue.size(/*reader=*/2), 10 - i - 1);
+    ASSERT_TRUE(value.has_value());
+    EXPECT_EQ(value.value(), i);
+  }
+}
+
+TEST(MultiReaderQueueTest, AddReaderWithoutReaderToCopySeeFullQueue) {
+  MultiReaderQueue<size_t> queue =
+      MultiReaderQueue<size_t>(/*readers_count=*/2);
+  for (size_t i = 0; i < 10; ++i) {
+    queue.PushBack(i);
+  }
+  for (size_t i = 0; i < 5; ++i) {
+    queue.PopFront(/*reader=*/0);
+  }
+
+  queue.AddReader(/*reader=*/2);
+
+  EXPECT_EQ(queue.readers_count(), 3lu);
+  for (size_t i = 0; i < 10; ++i) {
+    absl::optional<size_t> value = queue.PopFront(/*reader=*/2);
+    EXPECT_EQ(queue.size(/*reader=*/2), 10 - i - 1);
+    ASSERT_TRUE(value.has_value());
+    EXPECT_EQ(value.value(), i);
+  }
+}
+
+TEST(MultiReaderQueueTest, RemoveReaderWontChangeOthers) {
+  MultiReaderQueue<size_t> queue =
+      MultiReaderQueue<size_t>(/*readers_count=*/2);
+  for (size_t i = 0; i < 10; ++i) {
+    queue.PushBack(i);
+  }
+  EXPECT_EQ(queue.size(/*reader=*/1), 10lu);
+
+  queue.RemoveReader(0);
+
+  EXPECT_EQ(queue.readers_count(), 1lu);
+  EXPECT_EQ(queue.size(/*reader=*/1), 10lu);
+}
+
+TEST(MultiReaderQueueTest, RemoveLastReaderMakesQueueEmpty) {
+  MultiReaderQueue<size_t> queue =
+      MultiReaderQueue<size_t>(/*readers_count=*/1);
+  for (size_t i = 0; i < 10; ++i) {
+    queue.PushBack(i);
+  }
+  EXPECT_EQ(queue.size(), 10lu);
+
+  queue.RemoveReader(0);
+
+  EXPECT_EQ(queue.size(), 0lu);
+  EXPECT_EQ(queue.readers_count(), 0lu);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection.cc
new file mode 100644
index 0000000000..3ccab620f8
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection.cc
@@ -0,0 +1,101 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/names_collection.h"
+
+#include <set>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+
+namespace webrtc {
+
+NamesCollection::NamesCollection(rtc::ArrayView<const std::string> names) {
+  names_ = std::vector<std::string>(names.begin(), names.end());
+  for (size_t i = 0; i < names_.size(); ++i) {
+    index_.emplace(names_[i], i);
+    removed_.emplace_back(false);
+  }
+  size_ = names_.size();
+}
+
+bool NamesCollection::HasName(absl::string_view name) const {
+  auto it = index_.find(name);
+  if (it == index_.end()) {
+    return false;
+  }
+  return !removed_[it->second];
+}
+
+size_t NamesCollection::AddIfAbsent(absl::string_view name) {
+  auto it = index_.find(name);
+  if (it != index_.end()) {
+    // Name was registered in the collection before: we need to restore it.
+    size_t index = it->second;
+    if (removed_[index]) {
+      removed_[index] = false;
+      size_++;
+    }
+    return index;
+  }
+  size_t out = names_.size();
+  size_t old_capacity = names_.capacity();
+  names_.emplace_back(name);
+  removed_.emplace_back(false);
+  size_++;
+  size_t new_capacity = names_.capacity();
+
+  if (old_capacity == new_capacity) {
+    index_.emplace(names_[out], out);
+  } else {
+    // Reallocation happened in the vector, so we need to rebuild `index_` to
+    // fix absl::string_view internal references.
+    index_.clear();
+    for (size_t i = 0; i < names_.size(); ++i) {
+      index_.emplace(names_[i], i);
+    }
+  }
+  return out;
+}
+
+absl::optional<size_t> NamesCollection::RemoveIfPresent(
+    absl::string_view name) {
+  auto it = index_.find(name);
+  if (it == index_.end()) {
+    return absl::nullopt;
+  }
+  size_t index = it->second;
+  if (removed_[index]) {
+    return absl::nullopt;
+  }
+  removed_[index] = true;
+  size_--;
+  return index;
+}
+
+std::set<size_t> NamesCollection::GetPresentIndexes() const {
+  std::set<size_t> out;
+  for (size_t i = 0; i < removed_.size(); ++i) {
+    if (!removed_[i]) {
+      out.insert(i);
+    }
+  }
+  return out;
+}
+
+std::set<size_t> NamesCollection::GetAllIndexes() const {
+  std::set<size_t> out;
+  for (size_t i = 0; i < names_.size(); ++i) {
+    out.insert(i);
+  }
+  return out;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection.h
new file mode 100644
index 0000000000..f9a13a2a11
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection.h
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_NAMES_COLLECTION_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_NAMES_COLLECTION_H_
+
+#include <map>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Contains mapping between string names and unique size_t values (indexes).
+// Once the name is added to the collection it is guaranteed:
+//   1. Name will have the same index until collection will be destructed
+//   2. Adding, removing and re-adding name won't change its index
+//
+// The name is considered in the collection if it was added and wasn't removed.
+// Adding the name when it is in the collection won't change the collection, the
+// same as removing the name when it is removed.
+//
+// Collection will return name's index and name for the index independently from
+// was name removed or not. Once the name was added to the collection the index
+// will be allocated for it. To check if name is in collection right now user
+// has to explicitly call to `HasName` function.
+class NamesCollection {
+ public:
+  NamesCollection() = default;
+
+  explicit NamesCollection(rtc::ArrayView<const std::string> names);
+
+  // Returns amount of currently presented names in the collection.
+  size_t size() const { return size_; }
+
+  // Returns amount of all names known to this collection.
+  size_t GetKnownSize() const { return names_.size(); }
+
+  // Returns index of the `name` which was known to the collection. Crashes
+  // if `name` was never registered in the collection.
+  size_t index(absl::string_view name) const { return index_.at(name); }
+
+  // Returns name which was known to the collection for the specified `index`.
+  // Crashes if there was no any name registered in the collection for such
+  // `index`.
+  const std::string& name(size_t index) const { return names_.at(index); }
+
+  // Returns if `name` is currently presented in this collection.
+  bool HasName(absl::string_view name) const;
+
+  // Adds specified `name` to the collection if it isn't presented.
+  // Returns index which corresponds to specified `name`.
+  size_t AddIfAbsent(absl::string_view name);
+
+  // Removes specified `name` from the collection if it is presented.
+  //
+  // After name was removed, collection size will be decreased, but `name` index
+  // will be preserved. Collection will return false for `HasName(name)`, but
+  // will continue to return previously known index for `index(name)` and return
+  // `name` for `name(index(name))`.
+  //
+  // Returns the index of the removed value or absl::nullopt if no such `name`
+  // registered in the collection.
+  absl::optional<size_t> RemoveIfPresent(absl::string_view name);
+
+  // Returns a set of indexes for all currently present names in the
+  // collection.
+  std::set<size_t> GetPresentIndexes() const;
+
+  // Returns a set of all indexes known to the collection including indexes for
+  // names that were removed.
+  std::set<size_t> GetAllIndexes() const;
+
+ private:
+  std::vector<std::string> names_;
+  std::vector<bool> removed_;
+  std::map<absl::string_view, size_t> index_;
+  size_t size_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_NAMES_COLLECTION_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection_test.cc
new file mode 100644
index 0000000000..6c52f96975
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection_test.cc
@@ -0,0 +1,152 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/names_collection.h"
+
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::Eq;
+using ::testing::Ne;
+
+TEST(NamesCollectionTest, NamesFromCtorHasUniqueIndexes) {
+  NamesCollection collection(std::vector<std::string>{"alice", "bob"});
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(2)));
+  EXPECT_TRUE(collection.HasName("alice"));
+  EXPECT_THAT(collection.name(collection.index("alice")), Eq("alice"));
+
+  EXPECT_TRUE(collection.HasName("bob"));
+  EXPECT_THAT(collection.name(collection.index("bob")), Eq("bob"));
+
+  EXPECT_THAT(collection.index("bob"), Ne(collection.index("alice")));
+}
+
+TEST(NamesCollectionTest, AddedNamesHasIndexes) {
+  NamesCollection collection(std::vector<std::string>{});
+  collection.AddIfAbsent("alice");
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+  EXPECT_TRUE(collection.HasName("alice"));
+  EXPECT_THAT(collection.name(collection.index("alice")), Eq("alice"));
+}
+
+TEST(NamesCollectionTest, AddBobDoesNotChangeAliceIndex) {
+  NamesCollection collection(std::vector<std::string>{"alice"});
+
+  size_t alice_index = collection.index("alice");
+
+  collection.AddIfAbsent("bob");
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(2)));
+  EXPECT_THAT(collection.index("alice"), Eq(alice_index));
+  EXPECT_THAT(collection.index("bob"), Ne(alice_index));
+}
+
+TEST(NamesCollectionTest, AddAliceSecondTimeDoesNotChangeIndex) {
+  NamesCollection collection(std::vector<std::string>{"alice"});
+
+  size_t alice_index = collection.index("alice");
+
+  EXPECT_THAT(collection.AddIfAbsent("alice"), Eq(alice_index));
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+  EXPECT_THAT(collection.index("alice"), Eq(alice_index));
+}
+
+TEST(NamesCollectionTest, RemoveRemovesFromCollectionButNotIndex) {
+  NamesCollection collection(std::vector<std::string>{"alice", "bob"});
+
+  size_t bob_index = collection.index("bob");
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(2)));
+
+  EXPECT_THAT(collection.RemoveIfPresent("bob"),
+              Eq(absl::optional<size_t>(bob_index)));
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+  EXPECT_FALSE(collection.HasName("bob"));
+
+  EXPECT_THAT(collection.index("bob"), Eq(bob_index));
+  EXPECT_THAT(collection.name(bob_index), Eq("bob"));
+}
+
+TEST(NamesCollectionTest, RemoveOfAliceDoesNotChangeBobIndex) {
+  NamesCollection collection(std::vector<std::string>{"alice", "bob"});
+
+  size_t alice_index = collection.index("alice");
+  size_t bob_index = collection.index("bob");
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(2)));
+
+  EXPECT_THAT(collection.RemoveIfPresent("alice"),
+              Eq(absl::optional<size_t>(alice_index)));
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+  EXPECT_THAT(collection.index("bob"), Eq(bob_index));
+  EXPECT_THAT(collection.name(bob_index), Eq("bob"));
+}
+
+TEST(NamesCollectionTest, RemoveSecondTimeHasNoEffect) {
+  NamesCollection collection(std::vector<std::string>{"bob"});
+
+  size_t bob_index = collection.index("bob");
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+  EXPECT_THAT(collection.RemoveIfPresent("bob"),
+              Eq(absl::optional<size_t>(bob_index)));
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(0)));
+  EXPECT_THAT(collection.RemoveIfPresent("bob"), Eq(absl::nullopt));
+}
+
+TEST(NamesCollectionTest, RemoveOfNotExistingHasNoEffect) {
+  NamesCollection collection(std::vector<std::string>{"bob"});
+
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+  EXPECT_THAT(collection.RemoveIfPresent("alice"), Eq(absl::nullopt));
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+}
+
+TEST(NamesCollectionTest, AddRemoveAddPreserveTheIndex) {
+  NamesCollection collection(std::vector<std::string>{});
+
+  size_t alice_index = collection.AddIfAbsent("alice");
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+
+  EXPECT_THAT(collection.RemoveIfPresent("alice"),
+              Eq(absl::optional<size_t>(alice_index)));
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(0)));
+
+  EXPECT_THAT(collection.AddIfAbsent("alice"), Eq(alice_index));
+  EXPECT_THAT(collection.index("alice"), Eq(alice_index));
+  EXPECT_THAT(collection.size(), Eq(static_cast<size_t>(1)));
+}
+
+TEST(NamesCollectionTest, GetKnownSizeReturnsForRemovedNames) {
+  NamesCollection collection(std::vector<std::string>{});
+
+  size_t alice_index = collection.AddIfAbsent("alice");
+  EXPECT_THAT(collection.GetKnownSize(), Eq(static_cast<size_t>(1)));
+
+  EXPECT_THAT(collection.RemoveIfPresent("alice"),
+              Eq(absl::optional<size_t>(alice_index)));
+  EXPECT_THAT(collection.GetKnownSize(), Eq(static_cast<size_t>(1)));
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
new file mode 100644
index 0000000000..b958f4d027
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
@@ -0,0 +1,272 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h"
+
+#include <cstdint>
+#include <cstring>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/logging.h"
+#include "test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+QualityAnalyzingVideoDecoder::QualityAnalyzingVideoDecoder(
+    absl::string_view peer_name,
+    std::unique_ptr<VideoDecoder> delegate,
+    EncodedImageDataExtractor* extractor,
+    VideoQualityAnalyzerInterface* analyzer)
+    : peer_name_(peer_name),
+      implementation_name_("AnalyzingDecoder-" +
+                           std::string(delegate->ImplementationName())),
+      delegate_(std::move(delegate)),
+      extractor_(extractor),
+      analyzer_(analyzer) {
+  analyzing_callback_ = std::make_unique<DecoderCallback>(this);
+}
+QualityAnalyzingVideoDecoder::~QualityAnalyzingVideoDecoder() = default;
+
+bool QualityAnalyzingVideoDecoder::Configure(const Settings& settings) {
+  {
+    MutexLock lock(&mutex_);
+    codec_name_ = std::string(CodecTypeToPayloadString(settings.codec_type())) +
+                  "_" + delegate_->GetDecoderInfo().implementation_name;
+  }
+  return delegate_->Configure(settings);
+}
+
+int32_t QualityAnalyzingVideoDecoder::Decode(const EncodedImage& input_image,
+                                             bool missing_frames,
+                                             int64_t render_time_ms) {
+  // Image  extractor extracts id from provided EncodedImage and also returns
+  // the image with the original buffer. Buffer can be modified in place, so
+  // owner of original buffer will be responsible for deleting it, or extractor
+  // can create a new buffer. In such case extractor will be responsible for
+  // deleting it.
+  EncodedImageExtractionResult out = extractor_->ExtractData(input_image);
+
+  if (out.discard) {
+    // To partly emulate behavior of Selective Forwarding Unit (SFU) in the
+    // test, on receiver side we will "discard" frames from irrelevant streams.
+    // When all encoded images were marked to discarded, black frame have to be
+    // returned. Because simulcast streams will be received by receiver as 3
+    // different independent streams we don't want that irrelevant streams
+    // affect video quality metrics and also we don't want to use CPU time to
+    // decode them to prevent regressions on relevant streams. Also we can't
+    // just drop frame, because in such case, receiving part will be confused
+    // with all frames missing and will request a key frame, which will result
+    // into extra load on network and sender side. Because of it, discarded
+    // image will be always decoded as black frame and will be passed to
+    // callback directly without reaching decoder and video quality analyzer.
+    //
+    // For more details see QualityAnalyzingVideoEncoder.
+    return analyzing_callback_->IrrelevantSimulcastStreamDecoded(
+        out.id.value_or(VideoFrame::kNotSetId), input_image.Timestamp());
+  }
+
+  EncodedImage* origin_image;
+  {
+    MutexLock lock(&mutex_);
+    // Store id to be able to retrieve it in analyzing callback.
+    timestamp_to_frame_id_.insert({input_image.Timestamp(), out.id});
+    // Store encoded image to prevent its destruction while it is used in
+    // decoder.
+    origin_image = &(
+        decoding_images_.insert({input_image.Timestamp(), std::move(out.image)})
+            .first->second);
+  }
+  // We can safely dereference `origin_image`, because it can be removed from
+  // the map only after `delegate_` Decode method will be invoked. Image will
+  // be removed inside DecodedImageCallback, which can be done on separate
+  // thread.
+  analyzer_->OnFramePreDecode(
+      peer_name_, out.id.value_or(VideoFrame::kNotSetId), *origin_image);
+  int32_t result =
+      delegate_->Decode(*origin_image, missing_frames, render_time_ms);
+  if (result != WEBRTC_VIDEO_CODEC_OK) {
+    // If delegate decoder failed, then cleanup data for this image.
+    VideoQualityAnalyzerInterface::DecoderStats stats;
+    {
+      MutexLock lock(&mutex_);
+      timestamp_to_frame_id_.erase(input_image.Timestamp());
+      decoding_images_.erase(input_image.Timestamp());
+      stats.decoder_name = codec_name_;
+    }
+    analyzer_->OnDecoderError(
+        peer_name_, out.id.value_or(VideoFrame::kNotSetId), result, stats);
+  }
+  return result;
+}
+
+int32_t QualityAnalyzingVideoDecoder::RegisterDecodeCompleteCallback(
+    DecodedImageCallback* callback) {
+  analyzing_callback_->SetDelegateCallback(callback);
+  return delegate_->RegisterDecodeCompleteCallback(analyzing_callback_.get());
+}
+
+int32_t QualityAnalyzingVideoDecoder::Release() {
+  // Release decoder first. During release process it can still decode some
+  // frames, so we don't take a lock to prevent deadlock.
+  int32_t result = delegate_->Release();
+
+  MutexLock lock(&mutex_);
+  analyzing_callback_->SetDelegateCallback(nullptr);
+  timestamp_to_frame_id_.clear();
+  decoding_images_.clear();
+  return result;
+}
+
+VideoDecoder::DecoderInfo QualityAnalyzingVideoDecoder::GetDecoderInfo() const {
+  DecoderInfo info = delegate_->GetDecoderInfo();
+  info.implementation_name = implementation_name_;
+  return info;
+}
+
+const char* QualityAnalyzingVideoDecoder::ImplementationName() const {
+  return implementation_name_.c_str();
+}
+
+QualityAnalyzingVideoDecoder::DecoderCallback::DecoderCallback(
+    QualityAnalyzingVideoDecoder* decoder)
+    : decoder_(decoder), delegate_callback_(nullptr) {}
+QualityAnalyzingVideoDecoder::DecoderCallback::~DecoderCallback() = default;
+
+void QualityAnalyzingVideoDecoder::DecoderCallback::SetDelegateCallback(
+    DecodedImageCallback* delegate) {
+  MutexLock lock(&callback_mutex_);
+  delegate_callback_ = delegate;
+}
+
+// We have to implement all next 3 methods because we don't know which one
+// exactly is implemented in `delegate_callback_`, so we need to call the same
+// method on `delegate_callback_`, as was called on `this` callback.
+int32_t QualityAnalyzingVideoDecoder::DecoderCallback::Decoded(
+    VideoFrame& decodedImage) {
+  decoder_->OnFrameDecoded(&decodedImage, /*decode_time_ms=*/absl::nullopt,
+                           /*qp=*/absl::nullopt);
+
+  MutexLock lock(&callback_mutex_);
+  RTC_DCHECK(delegate_callback_);
+  return delegate_callback_->Decoded(decodedImage);
+}
+
+int32_t QualityAnalyzingVideoDecoder::DecoderCallback::Decoded(
+    VideoFrame& decodedImage,
+    int64_t decode_time_ms) {
+  decoder_->OnFrameDecoded(&decodedImage, decode_time_ms, /*qp=*/absl::nullopt);
+
+  MutexLock lock(&callback_mutex_);
+  RTC_DCHECK(delegate_callback_);
+  return delegate_callback_->Decoded(decodedImage, decode_time_ms);
+}
+
+void QualityAnalyzingVideoDecoder::DecoderCallback::Decoded(
+    VideoFrame& decodedImage,
+    absl::optional<int32_t> decode_time_ms,
+    absl::optional<uint8_t> qp) {
+  decoder_->OnFrameDecoded(&decodedImage, decode_time_ms, qp);
+
+  MutexLock lock(&callback_mutex_);
+  RTC_DCHECK(delegate_callback_);
+  delegate_callback_->Decoded(decodedImage, decode_time_ms, qp);
+}
+
+int32_t
+QualityAnalyzingVideoDecoder::DecoderCallback::IrrelevantSimulcastStreamDecoded(
+    uint16_t frame_id,
+    uint32_t timestamp_ms) {
+  webrtc::VideoFrame dummy_frame =
+      webrtc::VideoFrame::Builder()
+          .set_video_frame_buffer(GetDummyFrameBuffer())
+          .set_timestamp_rtp(timestamp_ms)
+          .set_id(frame_id)
+          .build();
+  MutexLock lock(&callback_mutex_);
+  RTC_DCHECK(delegate_callback_);
+  delegate_callback_->Decoded(dummy_frame, absl::nullopt, absl::nullopt);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+rtc::scoped_refptr<webrtc::VideoFrameBuffer>
+QualityAnalyzingVideoDecoder::DecoderCallback::GetDummyFrameBuffer() {
+  if (!dummy_frame_buffer_) {
+    dummy_frame_buffer_ = CreateDummyFrameBuffer();
+  }
+
+  return dummy_frame_buffer_;
+}
+
+void QualityAnalyzingVideoDecoder::OnFrameDecoded(
+    VideoFrame* frame,
+    absl::optional<int32_t> decode_time_ms,
+    absl::optional<uint8_t> qp) {
+  absl::optional<uint16_t> frame_id;
+  std::string codec_name;
+  {
+    MutexLock lock(&mutex_);
+    auto it = timestamp_to_frame_id_.find(frame->timestamp());
+    if (it == timestamp_to_frame_id_.end()) {
+      // Ensure, that we have info about this frame. It can happen that for some
+      // reasons decoder response, that it failed to decode, when we were
+      // posting frame to it, but then call the callback for this frame.
+      RTC_LOG(LS_ERROR) << "QualityAnalyzingVideoDecoder::OnFrameDecoded: No "
+                           "frame id for frame for frame->timestamp()="
+                        << frame->timestamp();
+      return;
+    }
+    frame_id = it->second;
+    timestamp_to_frame_id_.erase(it);
+    decoding_images_.erase(frame->timestamp());
+    codec_name = codec_name_;
+  }
+  // Set frame id to the value, that was extracted from corresponding encoded
+  // image.
+  frame->set_id(frame_id.value_or(VideoFrame::kNotSetId));
+  VideoQualityAnalyzerInterface::DecoderStats stats;
+  stats.decoder_name = codec_name;
+  stats.decode_time_ms = decode_time_ms;
+  analyzer_->OnFrameDecoded(peer_name_, *frame, stats);
+}
+
+QualityAnalyzingVideoDecoderFactory::QualityAnalyzingVideoDecoderFactory(
+    absl::string_view peer_name,
+    std::unique_ptr<VideoDecoderFactory> delegate,
+    EncodedImageDataExtractor* extractor,
+    VideoQualityAnalyzerInterface* analyzer)
+    : peer_name_(peer_name),
+      delegate_(std::move(delegate)),
+      extractor_(extractor),
+      analyzer_(analyzer) {}
+QualityAnalyzingVideoDecoderFactory::~QualityAnalyzingVideoDecoderFactory() =
+    default;
+
+std::vector<SdpVideoFormat>
+QualityAnalyzingVideoDecoderFactory::GetSupportedFormats() const {
+  return delegate_->GetSupportedFormats();
+}
+
+std::unique_ptr<VideoDecoder>
+QualityAnalyzingVideoDecoderFactory::CreateVideoDecoder(
+    const SdpVideoFormat& format) {
+  std::unique_ptr<VideoDecoder> decoder = delegate_->CreateVideoDecoder(format);
+  return std::make_unique<QualityAnalyzingVideoDecoder>(
+      peer_name_, std::move(decoder), extractor_, analyzer_);
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
new file mode 100644
index 0000000000..a86f4196b0
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h
@@ -0,0 +1,153 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_QUALITY_ANALYZING_VIDEO_DECODER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_QUALITY_ANALYZING_VIDEO_DECODER_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/pc/e2e/analyzer/video/encoded_image_data_injector.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// QualityAnalyzingVideoDecoder is used to wrap origin video decoder and inject
+// VideoQualityAnalyzerInterface before and after decoder.
+//
+// QualityAnalyzingVideoDecoder propagates all calls to the origin decoder.
+// It registers its own DecodedImageCallback in the origin decoder and will
+// store user specified callback inside itself.
+//
+// When Decode(...) will be invoked, quality decoder first will extract frame id
+// from passed EncodedImage with EncodedImageIdExtracor that was specified in
+// constructor, then will call video quality analyzer, with correct
+// EncodedImage and only then will pass image to origin decoder.
+//
+// When origin decoder decodes the image it will call quality decoder's special
+// callback, where video analyzer will be called again and then decoded frame
+// will be passed to origin callback, provided by user.
+//
+// Quality decoder registers its own callback in origin decoder, at the same
+// time the user registers their callback in quality decoder.
+class QualityAnalyzingVideoDecoder : public VideoDecoder {
+ public:
+  QualityAnalyzingVideoDecoder(absl::string_view peer_name,
+                               std::unique_ptr<VideoDecoder> delegate,
+                               EncodedImageDataExtractor* extractor,
+                               VideoQualityAnalyzerInterface* analyzer);
+  ~QualityAnalyzingVideoDecoder() override;
+
+  // Methods of VideoDecoder interface.
+  bool Configure(const Settings& settings) override;
+  int32_t Decode(const EncodedImage& input_image,
+                 bool missing_frames,
+                 int64_t render_time_ms) override;
+  int32_t RegisterDecodeCompleteCallback(
+      DecodedImageCallback* callback) override;
+  int32_t Release() override;
+  DecoderInfo GetDecoderInfo() const override;
+  const char* ImplementationName() const override;
+
+ private:
+  class DecoderCallback : public DecodedImageCallback {
+   public:
+    explicit DecoderCallback(QualityAnalyzingVideoDecoder* decoder);
+    ~DecoderCallback() override;
+
+    void SetDelegateCallback(DecodedImageCallback* delegate);
+
+    // Methods of DecodedImageCallback interface.
+    int32_t Decoded(VideoFrame& decodedImage) override;
+    int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) override;
+    void Decoded(VideoFrame& decodedImage,
+                 absl::optional<int32_t> decode_time_ms,
+                 absl::optional<uint8_t> qp) override;
+
+    int32_t IrrelevantSimulcastStreamDecoded(uint16_t frame_id,
+                                             uint32_t timestamp_ms);
+
+   private:
+    rtc::scoped_refptr<webrtc::VideoFrameBuffer> GetDummyFrameBuffer();
+
+    QualityAnalyzingVideoDecoder* const decoder_;
+
+    rtc::scoped_refptr<webrtc::VideoFrameBuffer> dummy_frame_buffer_;
+
+    Mutex callback_mutex_;
+    DecodedImageCallback* delegate_callback_ RTC_GUARDED_BY(callback_mutex_);
+  };
+
+  void OnFrameDecoded(VideoFrame* frame,
+                      absl::optional<int32_t> decode_time_ms,
+                      absl::optional<uint8_t> qp);
+
+  const std::string peer_name_;
+  const std::string implementation_name_;
+  std::unique_ptr<VideoDecoder> delegate_;
+  EncodedImageDataExtractor* const extractor_;
+  VideoQualityAnalyzerInterface* const analyzer_;
+  std::unique_ptr<DecoderCallback> analyzing_callback_;
+
+  // VideoDecoder interface assumes async delivery of decoded video frames.
+  // This lock is used to protect shared state, that have to be propagated
+  // from received EncodedImage to resulted VideoFrame.
+  Mutex mutex_;
+
+  // Name of the video codec type used. Ex: VP8, VP9, H264 etc.
+  std::string codec_name_ RTC_GUARDED_BY(mutex_);
+  std::map<uint32_t, absl::optional<uint16_t>> timestamp_to_frame_id_
+      RTC_GUARDED_BY(mutex_);
+  // Stores currently being decoded images by timestamp. Because
+  // EncodedImageDataExtractor can create new copy on EncodedImage we need to
+  // ensure, that this image won't be deleted during async decoding. To do it
+  // all images are putted into this map and removed from here inside callback.
+  std::map<uint32_t, EncodedImage> decoding_images_ RTC_GUARDED_BY(mutex_);
+};
+
+// Produces QualityAnalyzingVideoDecoder, which hold decoders, produced by
+// specified factory as delegates. Forwards all other calls to specified
+// factory.
+class QualityAnalyzingVideoDecoderFactory : public VideoDecoderFactory {
+ public:
+  QualityAnalyzingVideoDecoderFactory(
+      absl::string_view peer_name,
+      std::unique_ptr<VideoDecoderFactory> delegate,
+      EncodedImageDataExtractor* extractor,
+      VideoQualityAnalyzerInterface* analyzer);
+  ~QualityAnalyzingVideoDecoderFactory() override;
+
+  // Methods of VideoDecoderFactory interface.
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
+      const SdpVideoFormat& format) override;
+
+ private:
+  const std::string peer_name_;
+  std::unique_ptr<VideoDecoderFactory> delegate_;
+  EncodedImageDataExtractor* const extractor_;
+  VideoQualityAnalyzerInterface* const analyzer_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_QUALITY_ANALYZING_VIDEO_DECODER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
new file mode 100644
index 0000000000..e814ba88b7
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
@@ -0,0 +1,403 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h"
+
+#include <cmath>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/video/video_codec_type.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using EmulatedSFUConfigMap =
+    ::webrtc::webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap;
+
+constexpr size_t kMaxFrameInPipelineCount = 1000;
+constexpr double kNoMultiplier = 1.0;
+constexpr double kEps = 1e-6;
+
+std::pair<uint32_t, uint32_t> GetMinMaxBitratesBps(const VideoCodec& codec,
+                                                   size_t spatial_idx) {
+  uint32_t min_bitrate = codec.minBitrate;
+  uint32_t max_bitrate = codec.maxBitrate;
+  if (spatial_idx < codec.numberOfSimulcastStreams &&
+      codec.codecType != VideoCodecType::kVideoCodecVP9) {
+    min_bitrate =
+        std::max(min_bitrate, codec.simulcastStream[spatial_idx].minBitrate);
+    max_bitrate =
+        std::min(max_bitrate, codec.simulcastStream[spatial_idx].maxBitrate);
+  }
+  if (codec.codecType == VideoCodecType::kVideoCodecVP9 &&
+      spatial_idx < codec.VP9().numberOfSpatialLayers) {
+    min_bitrate =
+        std::max(min_bitrate, codec.spatialLayers[spatial_idx].minBitrate);
+    max_bitrate =
+        std::min(max_bitrate, codec.spatialLayers[spatial_idx].maxBitrate);
+  }
+  RTC_DCHECK_GT(max_bitrate, min_bitrate);
+  return {min_bitrate * 1000, max_bitrate * 1000};
+}
+
+}  // namespace
+
+QualityAnalyzingVideoEncoder::QualityAnalyzingVideoEncoder(
+    absl::string_view peer_name,
+    std::unique_ptr<VideoEncoder> delegate,
+    double bitrate_multiplier,
+    EmulatedSFUConfigMap stream_to_sfu_config,
+    EncodedImageDataInjector* injector,
+    VideoQualityAnalyzerInterface* analyzer)
+    : peer_name_(peer_name),
+      delegate_(std::move(delegate)),
+      bitrate_multiplier_(bitrate_multiplier),
+      stream_to_sfu_config_(std::move(stream_to_sfu_config)),
+      injector_(injector),
+      analyzer_(analyzer),
+      mode_(SimulcastMode::kNormal),
+      delegate_callback_(nullptr) {}
+QualityAnalyzingVideoEncoder::~QualityAnalyzingVideoEncoder() = default;
+
+void QualityAnalyzingVideoEncoder::SetFecControllerOverride(
+    FecControllerOverride* fec_controller_override) {
+  // Ignored.
+}
+
+int32_t QualityAnalyzingVideoEncoder::InitEncode(
+    const VideoCodec* codec_settings,
+    const Settings& settings) {
+  MutexLock lock(&mutex_);
+  codec_settings_ = *codec_settings;
+  mode_ = SimulcastMode::kNormal;
+  absl::optional<InterLayerPredMode> inter_layer_pred_mode;
+  if (codec_settings->GetScalabilityMode().has_value()) {
+    inter_layer_pred_mode = ScalabilityModeToInterLayerPredMode(
+        *codec_settings->GetScalabilityMode());
+  } else if (codec_settings->codecType == kVideoCodecVP9) {
+    if (codec_settings->VP9().numberOfSpatialLayers > 1) {
+      inter_layer_pred_mode = codec_settings->VP9().interLayerPred;
+    }
+  }
+  if (inter_layer_pred_mode.has_value()) {
+    switch (*inter_layer_pred_mode) {
+      case InterLayerPredMode::kOn:
+        mode_ = SimulcastMode::kSVC;
+        break;
+      case InterLayerPredMode::kOnKeyPic:
+        mode_ = SimulcastMode::kKSVC;
+        break;
+      case InterLayerPredMode::kOff:
+        mode_ = SimulcastMode::kSimulcast;
+        break;
+      default:
+        RTC_DCHECK_NOTREACHED()
+            << "Unknown InterLayerPredMode value " << *inter_layer_pred_mode;
+        break;
+    }
+  }
+  if (codec_settings->numberOfSimulcastStreams > 1) {
+    mode_ = SimulcastMode::kSimulcast;
+  }
+  return delegate_->InitEncode(codec_settings, settings);
+}
+
+int32_t QualityAnalyzingVideoEncoder::RegisterEncodeCompleteCallback(
+    EncodedImageCallback* callback) {
+  // We need to get a lock here because delegate_callback can be hypothetically
+  // accessed from different thread (encoder one) concurrently.
+  MutexLock lock(&mutex_);
+  delegate_callback_ = callback;
+  return delegate_->RegisterEncodeCompleteCallback(this);
+}
+
+int32_t QualityAnalyzingVideoEncoder::Release() {
+  // Release encoder first. During release process it can still encode some
+  // frames, so we don't take a lock to prevent deadlock.
+  int32_t result = delegate_->Release();
+
+  MutexLock lock(&mutex_);
+  delegate_callback_ = nullptr;
+  return result;
+}
+
+int32_t QualityAnalyzingVideoEncoder::Encode(
+    const VideoFrame& frame,
+    const std::vector<VideoFrameType>* frame_types) {
+  {
+    MutexLock lock(&mutex_);
+    // Store id to be able to retrieve it in analyzing callback.
+    timestamp_to_frame_id_list_.push_back({frame.timestamp(), frame.id()});
+    // If this list is growing, it means that we are not receiving new encoded
+    // images from encoder. So it should be a bug in setup on in the encoder.
+    RTC_DCHECK_LT(timestamp_to_frame_id_list_.size(), kMaxFrameInPipelineCount);
+  }
+  analyzer_->OnFramePreEncode(peer_name_, frame);
+  int32_t result = delegate_->Encode(frame, frame_types);
+  if (result != WEBRTC_VIDEO_CODEC_OK) {
+    // If origin encoder failed, then cleanup data for this frame.
+    {
+      MutexLock lock(&mutex_);
+      // The timestamp-frame_id pair can be not the last one, so we need to
+      // find it first and then remove. We will search from the end, because
+      // usually it will be the last or close to the last one.
+      auto it = timestamp_to_frame_id_list_.end();
+      while (it != timestamp_to_frame_id_list_.begin()) {
+        --it;
+        if (it->first == frame.timestamp()) {
+          timestamp_to_frame_id_list_.erase(it);
+          break;
+        }
+      }
+    }
+    analyzer_->OnEncoderError(peer_name_, frame, result);
+  }
+  return result;
+}
+
+void QualityAnalyzingVideoEncoder::SetRates(
+    const VideoEncoder::RateControlParameters& parameters) {
+  RTC_DCHECK_GT(bitrate_multiplier_, 0.0);
+  if (fabs(bitrate_multiplier_ - kNoMultiplier) < kEps) {
+    {
+      MutexLock lock(&mutex_);
+      bitrate_allocation_ = parameters.bitrate;
+    }
+    return delegate_->SetRates(parameters);
+  }
+
+  RateControlParameters adjusted_params = parameters;
+  {
+    MutexLock lock(&mutex_);
+    // Simulating encoder overshooting target bitrate, by configuring actual
+    // encoder too high. Take care not to adjust past limits of config,
+    // otherwise encoders may crash on DCHECK.
+    VideoBitrateAllocation multiplied_allocation;
+    for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
+      const uint32_t spatial_layer_bitrate_bps =
+          parameters.bitrate.GetSpatialLayerSum(si);
+      if (spatial_layer_bitrate_bps == 0) {
+        continue;
+      }
+
+      uint32_t min_bitrate_bps;
+      uint32_t max_bitrate_bps;
+      std::tie(min_bitrate_bps, max_bitrate_bps) =
+          GetMinMaxBitratesBps(codec_settings_, si);
+      double bitrate_multiplier = bitrate_multiplier_;
+      const uint32_t corrected_bitrate = rtc::checked_cast<uint32_t>(
+          bitrate_multiplier * spatial_layer_bitrate_bps);
+      if (corrected_bitrate < min_bitrate_bps) {
+        bitrate_multiplier = min_bitrate_bps / spatial_layer_bitrate_bps;
+      } else if (corrected_bitrate > max_bitrate_bps) {
+        bitrate_multiplier = max_bitrate_bps / spatial_layer_bitrate_bps;
+      }
+
+      for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
+        if (parameters.bitrate.HasBitrate(si, ti)) {
+          multiplied_allocation.SetBitrate(
+              si, ti,
+              rtc::checked_cast<uint32_t>(
+                  bitrate_multiplier * parameters.bitrate.GetBitrate(si, ti)));
+        }
+      }
+    }
+
+    adjusted_params.bitrate = multiplied_allocation;
+    bitrate_allocation_ = adjusted_params.bitrate;
+  }
+  return delegate_->SetRates(adjusted_params);
+}
+
+VideoEncoder::EncoderInfo QualityAnalyzingVideoEncoder::GetEncoderInfo() const {
+  return delegate_->GetEncoderInfo();
+}
+
+// It is assumed, that encoded callback will be always invoked with encoded
+// images that correspond to the frames in the same sequence, that frames
+// arrived. In other words, assume we have frames F1, F2 and F3 and they have
+// corresponding encoded images I1, I2 and I3. In such case if we will call
+// encode first with F1, then with F2 and then with F3, then encoder callback
+// will be called first with all spatial layers for F1 (I1), then F2 (I2) and
+// then F3 (I3).
+//
+// Basing on it we will use a list of timestamp-frame_id pairs like this:
+//  1. If current encoded image timestamp is equals to timestamp in the front
+//     pair - pick frame id from that pair
+//  2. If current encoded image timestamp isn't equals to timestamp in the front
+//     pair - remove the front pair and got to the step 1.
+EncodedImageCallback::Result QualityAnalyzingVideoEncoder::OnEncodedImage(
+    const EncodedImage& encoded_image,
+    const CodecSpecificInfo* codec_specific_info) {
+  uint16_t frame_id;
+  bool discard = false;
+  uint32_t target_encode_bitrate = 0;
+  std::string codec_name;
+  {
+    MutexLock lock(&mutex_);
+    std::pair<uint32_t, uint16_t> timestamp_frame_id;
+    while (!timestamp_to_frame_id_list_.empty()) {
+      timestamp_frame_id = timestamp_to_frame_id_list_.front();
+      if (timestamp_frame_id.first == encoded_image.Timestamp()) {
+        break;
+      }
+      timestamp_to_frame_id_list_.pop_front();
+    }
+
+    // After the loop the first element should point to current `encoded_image`
+    // frame id. We don't remove it from the list, because there may be
+    // multiple spatial layers for this frame, so encoder can produce more
+    // encoded images with this timestamp. The first element will be removed
+    // when the next frame would be encoded and EncodedImageCallback would be
+    // called with the next timestamp.
+
+    if (timestamp_to_frame_id_list_.empty()) {
+      // Ensure, that we have info about this frame. It can happen that for some
+      // reasons encoder response, that he failed to decode, when we were
+      // posting frame to it, but then call the callback for this frame.
+      RTC_LOG(LS_ERROR) << "QualityAnalyzingVideoEncoder::OnEncodedImage: No "
+                           "frame id for encoded_image.Timestamp()="
+                        << encoded_image.Timestamp();
+      return EncodedImageCallback::Result(
+          EncodedImageCallback::Result::Error::OK);
+    }
+    frame_id = timestamp_frame_id.second;
+
+    discard = ShouldDiscard(frame_id, encoded_image);
+    if (!discard) {
+      target_encode_bitrate = bitrate_allocation_.GetSpatialLayerSum(
+          encoded_image.SpatialIndex().value_or(0));
+    }
+    codec_name =
+        std::string(CodecTypeToPayloadString(codec_settings_.codecType)) + "_" +
+        delegate_->GetEncoderInfo().implementation_name;
+  }
+
+  VideoQualityAnalyzerInterface::EncoderStats stats;
+  stats.encoder_name = codec_name;
+  stats.target_encode_bitrate = target_encode_bitrate;
+  stats.qp = encoded_image.qp_;
+  analyzer_->OnFrameEncoded(peer_name_, frame_id, encoded_image, stats,
+                            discard);
+
+  // Image data injector injects frame id and discard flag into provided
+  // EncodedImage and returns the image with a) modified original buffer (in
+  // such case the current owner of the buffer will be responsible for deleting
+  // it) or b) a new buffer (in such case injector will be responsible for
+  // deleting it).
+  const EncodedImage& image =
+      injector_->InjectData(frame_id, discard, encoded_image);
+  {
+    MutexLock lock(&mutex_);
+    RTC_DCHECK(delegate_callback_);
+    return delegate_callback_->OnEncodedImage(image, codec_specific_info);
+  }
+}
+
+void QualityAnalyzingVideoEncoder::OnDroppedFrame(
+    EncodedImageCallback::DropReason reason) {
+  MutexLock lock(&mutex_);
+  analyzer_->OnFrameDropped(peer_name_, reason);
+  RTC_DCHECK(delegate_callback_);
+  delegate_callback_->OnDroppedFrame(reason);
+}
+
+bool QualityAnalyzingVideoEncoder::ShouldDiscard(
+    uint16_t frame_id,
+    const EncodedImage& encoded_image) {
+  std::string stream_label = analyzer_->GetStreamLabel(frame_id);
+  EmulatedSFUConfigMap::mapped_type emulated_sfu_config =
+      stream_to_sfu_config_[stream_label];
+
+  if (!emulated_sfu_config)
+    return false;
+
+  int cur_spatial_index = encoded_image.SpatialIndex().value_or(0);
+  int cur_temporal_index = encoded_image.TemporalIndex().value_or(0);
+
+  if (emulated_sfu_config->target_temporal_index &&
+      cur_temporal_index > *emulated_sfu_config->target_temporal_index)
+    return true;
+
+  if (emulated_sfu_config->target_layer_index) {
+    switch (mode_) {
+      case SimulcastMode::kSimulcast:
+        // In simulcast mode only encoded images with required spatial index are
+        // interested, so all others have to be discarded.
+        return cur_spatial_index != *emulated_sfu_config->target_layer_index;
+      case SimulcastMode::kSVC:
+        // In SVC mode encoded images with spatial indexes that are equal or
+        // less than required one are interesting, so all above have to be
+        // discarded.
+        return cur_spatial_index > *emulated_sfu_config->target_layer_index;
+      case SimulcastMode::kKSVC:
+        // In KSVC mode for key frame encoded images with spatial indexes that
+        // are equal or less than required one are interesting, so all above
+        // have to be discarded. For other frames only required spatial index
+        // is interesting, so all others except the ones depending on the
+        // keyframes can be discarded. There's no good test for that, so we keep
+        // all of temporal layer 0 for now.
+        if (encoded_image._frameType == VideoFrameType::kVideoFrameKey ||
+            cur_temporal_index == 0)
+          return cur_spatial_index > *emulated_sfu_config->target_layer_index;
+        return cur_spatial_index != *emulated_sfu_config->target_layer_index;
+      case SimulcastMode::kNormal:
+        RTC_DCHECK_NOTREACHED() << "Analyzing encoder is in kNormal mode, but "
+                                   "target_layer_index is set";
+    }
+  }
+  return false;
+}
+
+QualityAnalyzingVideoEncoderFactory::QualityAnalyzingVideoEncoderFactory(
+    absl::string_view peer_name,
+    std::unique_ptr<VideoEncoderFactory> delegate,
+    double bitrate_multiplier,
+    EmulatedSFUConfigMap stream_to_sfu_config,
+    EncodedImageDataInjector* injector,
+    VideoQualityAnalyzerInterface* analyzer)
+    : peer_name_(peer_name),
+      delegate_(std::move(delegate)),
+      bitrate_multiplier_(bitrate_multiplier),
+      stream_to_sfu_config_(std::move(stream_to_sfu_config)),
+      injector_(injector),
+      analyzer_(analyzer) {}
+QualityAnalyzingVideoEncoderFactory::~QualityAnalyzingVideoEncoderFactory() =
+    default;
+
+std::vector<SdpVideoFormat>
+QualityAnalyzingVideoEncoderFactory::GetSupportedFormats() const {
+  return delegate_->GetSupportedFormats();
+}
+
+VideoEncoderFactory::CodecSupport
+QualityAnalyzingVideoEncoderFactory::QueryCodecSupport(
+    const SdpVideoFormat& format,
+    absl::optional<std::string> scalability_mode) const {
+  return delegate_->QueryCodecSupport(format, scalability_mode);
+}
+
+std::unique_ptr<VideoEncoder>
+QualityAnalyzingVideoEncoderFactory::CreateVideoEncoder(
+    const SdpVideoFormat& format) {
+  return std::make_unique<QualityAnalyzingVideoEncoder>(
+      peer_name_, delegate_->CreateVideoEncoder(format), bitrate_multiplier_,
+      stream_to_sfu_config_, injector_, analyzer_);
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
new file mode 100644
index 0000000000..4adeacc0cd
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_QUALITY_ANALYZING_VIDEO_ENCODER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_QUALITY_ANALYZING_VIDEO_ENCODER_H_
+
+#include <list>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/pc/e2e/analyzer/video/encoded_image_data_injector.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// QualityAnalyzingVideoEncoder is used to wrap origin video encoder and inject
+// VideoQualityAnalyzerInterface before and after encoder.
+//
+// QualityAnalyzingVideoEncoder propagates all calls to the origin encoder.
+// It registers its own EncodedImageCallback in the origin encoder and will
+// store user specified callback inside itself.
+//
+// When Encode(...) will be invoked, quality encoder first calls video quality
+// analyzer with original frame, then encodes frame with original encoder.
+//
+// When origin encoder encodes the image it will call quality encoder's special
+// callback, where video analyzer will be called again and then frame id will be
+// injected into EncodedImage with passed EncodedImageDataInjector. Then new
+// EncodedImage will be passed to origin callback, provided by user.
+//
+// Quality encoder registers its own callback in origin encoder, at the same
+// time the user registers their callback in quality encoder.
+class QualityAnalyzingVideoEncoder : public VideoEncoder,
+                                     public EncodedImageCallback {
+ public:
+  using EmulatedSFUConfigMap =
+      std::map<std::string, absl::optional<EmulatedSFUConfig>>;
+
+  QualityAnalyzingVideoEncoder(absl::string_view peer_name,
+                               std::unique_ptr<VideoEncoder> delegate,
+                               double bitrate_multiplier,
+                               EmulatedSFUConfigMap stream_to_sfu_config,
+                               EncodedImageDataInjector* injector,
+                               VideoQualityAnalyzerInterface* analyzer);
+  ~QualityAnalyzingVideoEncoder() override;
+
+  // Methods of VideoEncoder interface.
+  void SetFecControllerOverride(
+      FecControllerOverride* fec_controller_override) override;
+  int32_t InitEncode(const VideoCodec* codec_settings,
+                     const Settings& settings) override;
+  int32_t RegisterEncodeCompleteCallback(
+      EncodedImageCallback* callback) override;
+  int32_t Release() override;
+  int32_t Encode(const VideoFrame& frame,
+                 const std::vector<VideoFrameType>* frame_types) override;
+  void SetRates(const VideoEncoder::RateControlParameters& parameters) override;
+  EncoderInfo GetEncoderInfo() const override;
+
+  // Methods of EncodedImageCallback interface.
+  EncodedImageCallback::Result OnEncodedImage(
+      const EncodedImage& encoded_image,
+      const CodecSpecificInfo* codec_specific_info) override;
+  void OnDroppedFrame(DropReason reason) override;
+
+ private:
+  enum SimulcastMode {
+    // In this mode encoder assumes not more than 1 encoded image per video
+    // frame
+    kNormal,
+
+    // Next modes are to test video conference behavior. For conference sender
+    // will send multiple spatial layers/simulcast streams for single video
+    // track and there is some Selective Forwarding Unit (SFU), that forwards
+    // only best one, that will pass through downlink to the receiver.
+    //
+    // Here this behavior will be partly emulated. Sender will send all spatial
+    // layers/simulcast streams and then some of them will be filtered out on
+    // the receiver side. During test setup user can specify which spatial
+    // layer/simulcast stream is required, what will simulated which spatial
+    // layer/simulcast stream will be chosen by SFU in the real world. Then
+    // sender will mark encoded images for all spatial layers above required or
+    // all simulcast streams except required as to be discarded and on receiver
+    // side they will be discarded in quality analyzing decoder and won't be
+    // passed into delegate decoder.
+    //
+    // If the sender for some reasons won't send specified spatial layer, then
+    // receiver still will fall back on lower spatial layers. But for simulcast
+    // streams if required one won't be sent, receiver will assume all frames
+    // in that period as dropped and will experience video freeze.
+    //
+    // Test based on this simulation will be used to evaluate video quality
+    // of concrete spatial layers/simulcast streams and also check distribution
+    // of bandwidth between spatial layers/simulcast streams by BWE.
+
+    // In this mode encoder assumes that for each frame simulcast encoded
+    // images will be produced. So all simulcast streams except required will
+    // be marked as to be discarded in decoder and won't reach video quality
+    // analyzer.
+    kSimulcast,
+    // In this mode encoder assumes that for each frame encoded images for
+    // different spatial layers will be produced. So all spatial layers above
+    // required will be marked to be discarded in decoder and won't reach
+    // video quality analyzer.
+    kSVC,
+    // In this mode encoder assumes that for each frame encoded images for
+    // different spatial layers will be produced. Compared to kSVC mode
+    // spatial layers that are above required will be marked to be discarded
+    // only for key frames and for regular frames all except required spatial
+    // layer will be marked as to be discarded in decoder and won't reach video
+    // quality analyzer.
+    kKSVC
+  };
+
+  bool ShouldDiscard(uint16_t frame_id, const EncodedImage& encoded_image)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  const std::string peer_name_;
+  std::unique_ptr<VideoEncoder> delegate_;
+  const double bitrate_multiplier_;
+  // Contains mapping from stream label to optional spatial index.
+  // If we have stream label "Foo" and mapping contains
+  // 1. `absl::nullopt` means all streams are required
+  // 2. Concrete value means that particular simulcast/SVC stream have to be
+  //    analyzed.
+  EmulatedSFUConfigMap stream_to_sfu_config_;
+  EncodedImageDataInjector* const injector_;
+  VideoQualityAnalyzerInterface* const analyzer_;
+
+  // VideoEncoder interface assumes async delivery of encoded images.
+  // This lock is used to protect shared state, that have to be propagated
+  // from received VideoFrame to resulted EncodedImage.
+  Mutex mutex_;
+
+  VideoCodec codec_settings_ RTC_GUARDED_BY(mutex_);
+  SimulcastMode mode_ RTC_GUARDED_BY(mutex_);
+  EncodedImageCallback* delegate_callback_ RTC_GUARDED_BY(mutex_);
+  std::list<std::pair<uint32_t, uint16_t>> timestamp_to_frame_id_list_
+      RTC_GUARDED_BY(mutex_);
+  VideoBitrateAllocation bitrate_allocation_ RTC_GUARDED_BY(mutex_);
+};
+
+// Produces QualityAnalyzingVideoEncoder, which hold decoders, produced by
+// specified factory as delegates. Forwards all other calls to specified
+// factory.
+class QualityAnalyzingVideoEncoderFactory : public VideoEncoderFactory {
+ public:
+  QualityAnalyzingVideoEncoderFactory(
+      absl::string_view peer_name,
+      std::unique_ptr<VideoEncoderFactory> delegate,
+      double bitrate_multiplier,
+      QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap stream_to_sfu_config,
+      EncodedImageDataInjector* injector,
+      VideoQualityAnalyzerInterface* analyzer);
+  ~QualityAnalyzingVideoEncoderFactory() override;
+
+  // Methods of VideoEncoderFactory interface.
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  VideoEncoderFactory::CodecSupport QueryCodecSupport(
+      const SdpVideoFormat& format,
+      absl::optional<std::string> scalability_mode) const override;
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override;
+
+ private:
+  const std::string peer_name_;
+  std::unique_ptr<VideoEncoderFactory> delegate_;
+  const double bitrate_multiplier_;
+  QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap stream_to_sfu_config_;
+  EncodedImageDataInjector* const injector_;
+  VideoQualityAnalyzerInterface* const analyzer_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_QUALITY_ANALYZING_VIDEO_ENCODER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc
new file mode 100644
index 0000000000..7a73b9f4f1
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h"
+
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+constexpr char kIrrelatedSimulcastStreamFrameData[] = "Dummy!";
+
+}  // namespace
+
+rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateDummyFrameBuffer() {
+  // Use i420 buffer here as default one and supported by all codecs.
+  rtc::scoped_refptr<webrtc::I420Buffer> buffer =
+      webrtc::I420Buffer::Create(2, 2);
+  memcpy(buffer->MutableDataY(), kIrrelatedSimulcastStreamFrameData, 2);
+  memcpy(buffer->MutableDataY() + buffer->StrideY(),
+         kIrrelatedSimulcastStreamFrameData + 2, 2);
+  memcpy(buffer->MutableDataU(), kIrrelatedSimulcastStreamFrameData + 4, 1);
+  memcpy(buffer->MutableDataV(), kIrrelatedSimulcastStreamFrameData + 5, 1);
+  return buffer;
+}
+
+bool IsDummyFrame(const webrtc::VideoFrame& video_frame) {
+  if (video_frame.width() != 2 || video_frame.height() != 2) {
+    return false;
+  }
+  rtc::scoped_refptr<webrtc::I420BufferInterface> buffer =
+      video_frame.video_frame_buffer()->ToI420();
+  if (memcmp(buffer->DataY(), kIrrelatedSimulcastStreamFrameData, 2) != 0) {
+    return false;
+  }
+  if (memcmp(buffer->DataY() + buffer->StrideY(),
+             kIrrelatedSimulcastStreamFrameData + 2, 2) != 0) {
+    return false;
+  }
+  if (memcmp(buffer->DataU(), kIrrelatedSimulcastStreamFrameData + 4, 1) != 0) {
+    return false;
+  }
+  if (memcmp(buffer->DataV(), kIrrelatedSimulcastStreamFrameData + 5, 1) != 0) {
+    return false;
+  }
+  return true;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h
new file mode 100644
index 0000000000..8ecfae7385
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_SIMULCAST_DUMMY_BUFFER_HELPER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_SIMULCAST_DUMMY_BUFFER_HELPER_H_
+
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Creates a special video frame buffer that should be used to create frames
+// during Selective Forwarding Unit (SFU) emulation. Such frames are used when
+// original was discarded and some frame is required to be passed upstream
+// to make WebRTC pipeline happy and not request key frame on the received
+// stream due to lack of incoming frames.
+rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateDummyFrameBuffer();
+
+// Tests if provided frame contains a buffer created by
+// `CreateDummyFrameBuffer`.
+bool IsDummyFrame(const webrtc::VideoFrame& video_frame);
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_SIMULCAST_DUMMY_BUFFER_HELPER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper_test.cc
new file mode 100644
index 0000000000..db1030232d
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper_test.cc
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h"
+
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "rtc_base/random.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+uint8_t RandByte(Random& random) {
+  return random.Rand(255);
+}
+
+VideoFrame CreateRandom2x2VideoFrame(uint16_t id, Random& random) {
+  rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(2, 2);
+
+  uint8_t data[6] = {RandByte(random), RandByte(random), RandByte(random),
+                     RandByte(random), RandByte(random), RandByte(random)};
+
+  memcpy(buffer->MutableDataY(), data, 2);
+  memcpy(buffer->MutableDataY() + buffer->StrideY(), data + 2, 2);
+  memcpy(buffer->MutableDataU(), data + 4, 1);
+  memcpy(buffer->MutableDataV(), data + 5, 1);
+
+  return VideoFrame::Builder()
+      .set_id(id)
+      .set_video_frame_buffer(buffer)
+      .set_timestamp_us(1)
+      .build();
+}
+
+TEST(CreateDummyFrameBufferTest, CreatedBufferIsDummy) {
+  VideoFrame dummy_frame = VideoFrame::Builder()
+                               .set_video_frame_buffer(CreateDummyFrameBuffer())
+                               .build();
+
+  EXPECT_TRUE(IsDummyFrame(dummy_frame));
+}
+
+TEST(IsDummyFrameTest, NotEveryFrameIsDummy) {
+  Random random(/*seed=*/100);
+  VideoFrame frame = CreateRandom2x2VideoFrame(1, random);
+  EXPECT_FALSE(IsDummyFrame(frame));
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
new file mode 100644
index 0000000000..ccd2f03537
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
@@ -0,0 +1,187 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h"
+
+#include <algorithm>
+#include <cstddef>
+
+#include "absl/memory/memory.h"
+#include "api/video/encoded_image.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+SingleProcessEncodedImageDataInjector::SingleProcessEncodedImageDataInjector() =
+    default;
+SingleProcessEncodedImageDataInjector::
+    ~SingleProcessEncodedImageDataInjector() = default;
+
+EncodedImage SingleProcessEncodedImageDataInjector::InjectData(
+    uint16_t id,
+    bool discard,
+    const EncodedImage& source) {
+  RTC_CHECK(source.size() >= ExtractionInfo::kUsedBufferSize);
+
+  ExtractionInfo info;
+  info.discard = discard;
+  size_t insertion_pos = source.size() - ExtractionInfo::kUsedBufferSize;
+  memcpy(info.origin_data, &source.data()[insertion_pos],
+         ExtractionInfo::kUsedBufferSize);
+  {
+    MutexLock lock(&lock_);
+    // Will create new one if missed.
+    ExtractionInfoVector& ev = extraction_cache_[id];
+    info.sub_id = ev.next_sub_id++;
+    ev.infos[info.sub_id] = info;
+  }
+
+  auto buffer = EncodedImageBuffer::Create(source.data(), source.size());
+  buffer->data()[insertion_pos] = id & 0x00ff;
+  buffer->data()[insertion_pos + 1] = (id & 0xff00) >> 8;
+  buffer->data()[insertion_pos + 2] = info.sub_id;
+
+  EncodedImage out = source;
+  out.SetEncodedData(buffer);
+  return out;
+}
+
+void SingleProcessEncodedImageDataInjector::AddParticipantInCall() {
+  MutexLock crit(&lock_);
+  expected_receivers_count_++;
+}
+
+void SingleProcessEncodedImageDataInjector::RemoveParticipantInCall() {
+  MutexLock crit(&lock_);
+  expected_receivers_count_--;
+  // Now we need go over `extraction_cache_` and removed frames which have been
+  // received by `expected_receivers_count_`.
+  for (auto& [frame_id, extraction_infos] : extraction_cache_) {
+    for (auto it = extraction_infos.infos.begin();
+         it != extraction_infos.infos.end();) {
+      // Frame is received if `received_count` equals to
+      // `expected_receivers_count_`.
+      if (it->second.received_count == expected_receivers_count_) {
+        it = extraction_infos.infos.erase(it);
+      } else {
+        ++it;
+      }
+    }
+  }
+}
+
+EncodedImageExtractionResult SingleProcessEncodedImageDataInjector::ExtractData(
+    const EncodedImage& source) {
+  size_t size = source.size();
+  auto buffer = EncodedImageBuffer::Create(source.data(), source.size());
+  EncodedImage out = source;
+  out.SetEncodedData(buffer);
+
+  std::vector<size_t> frame_sizes;
+  std::vector<size_t> frame_sl_index;
+  size_t max_spatial_index = out.SpatialIndex().value_or(0);
+  for (size_t i = 0; i <= max_spatial_index; ++i) {
+    auto frame_size = source.SpatialLayerFrameSize(i);
+    if (frame_size.value_or(0)) {
+      frame_sl_index.push_back(i);
+      frame_sizes.push_back(frame_size.value());
+    }
+  }
+  if (frame_sizes.empty()) {
+    frame_sizes.push_back(size);
+  }
+
+  size_t prev_frames_size = 0;
+  absl::optional<uint16_t> id = absl::nullopt;
+  bool discard = true;
+  std::vector<ExtractionInfo> extraction_infos;
+  for (size_t frame_size : frame_sizes) {
+    size_t insertion_pos =
+        prev_frames_size + frame_size - ExtractionInfo::kUsedBufferSize;
+    // Extract frame id from first 2 bytes starting from insertion pos.
+    uint16_t next_id = buffer->data()[insertion_pos] +
+                       (buffer->data()[insertion_pos + 1] << 8);
+    // Extract frame sub id from second 3 byte starting from insertion pos.
+    uint8_t sub_id = buffer->data()[insertion_pos + 2];
+    RTC_CHECK(!id || *id == next_id)
+        << "Different frames encoded into single encoded image: " << *id
+        << " vs " << next_id;
+    id = next_id;
+    ExtractionInfo info;
+    {
+      MutexLock lock(&lock_);
+      auto ext_vector_it = extraction_cache_.find(next_id);
+      RTC_CHECK(ext_vector_it != extraction_cache_.end())
+          << "Unknown frame_id=" << next_id;
+
+      auto info_it = ext_vector_it->second.infos.find(sub_id);
+      RTC_CHECK(info_it != ext_vector_it->second.infos.end())
+          << "Unknown sub_id=" << sub_id << " for frame_id=" << next_id;
+      info_it->second.received_count++;
+      info = info_it->second;
+      if (info.received_count == expected_receivers_count_) {
+        ext_vector_it->second.infos.erase(info_it);
+      }
+    }
+    // We need to discard encoded image only if all concatenated encoded images
+    // have to be discarded.
+    discard = discard && info.discard;
+
+    extraction_infos.push_back(info);
+    prev_frames_size += frame_size;
+  }
+  RTC_CHECK(id);
+
+  if (discard) {
+    out.set_size(0);
+    for (size_t i = 0; i <= max_spatial_index; ++i) {
+      out.SetSpatialLayerFrameSize(i, 0);
+    }
+    return EncodedImageExtractionResult{*id, out, true};
+  }
+
+  // Make a pass from begin to end to restore origin payload and erase discarded
+  // encoded images.
+  size_t pos = 0;
+  for (size_t frame_index = 0; frame_index < frame_sizes.size();
+       ++frame_index) {
+    RTC_CHECK(pos < size);
+    const size_t frame_size = frame_sizes[frame_index];
+    const ExtractionInfo& info = extraction_infos[frame_index];
+    if (info.discard) {
+      // If this encoded image is marked to be discarded - erase it's payload
+      // from the buffer.
+      memmove(&buffer->data()[pos], &buffer->data()[pos + frame_size],
+              size - pos - frame_size);
+      RTC_CHECK_LT(frame_index, frame_sl_index.size())
+          << "codec doesn't support discard option or the image, that was "
+             "supposed to be discarded, is lost";
+      out.SetSpatialLayerFrameSize(frame_sl_index[frame_index], 0);
+      size -= frame_size;
+    } else {
+      memcpy(
+          &buffer->data()[pos + frame_size - ExtractionInfo::kUsedBufferSize],
+          info.origin_data, ExtractionInfo::kUsedBufferSize);
+      pos += frame_size;
+    }
+  }
+  out.set_size(pos);
+
+  return EncodedImageExtractionResult{*id, out, discard};
+}
+
+SingleProcessEncodedImageDataInjector::ExtractionInfoVector::
+    ExtractionInfoVector() = default;
+SingleProcessEncodedImageDataInjector::ExtractionInfoVector::
+    ~ExtractionInfoVector() = default;
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h
new file mode 100644
index 0000000000..1082440e2f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_SINGLE_PROCESS_ENCODED_IMAGE_DATA_INJECTOR_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_SINGLE_PROCESS_ENCODED_IMAGE_DATA_INJECTOR_H_
+
+#include <cstdint>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/video/encoded_image.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/pc/e2e/analyzer/video/encoded_image_data_injector.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Based on assumption that all call participants are in the same OS process
+// and uses same QualityAnalyzingVideoContext to obtain
+// EncodedImageDataInjector.
+//
+// To inject frame id and discard flag into EncodedImage injector uses last 3rd
+// and 2nd bytes of EncodedImage payload. Then it uses last byte for frame
+// sub id, that is required to distinguish different spatial layers. The origin
+// data from these 3 bytes will be stored inside injector's internal storage and
+// then will be restored during extraction phase.
+//
+// This injector won't add any extra overhead into EncodedImage payload and
+// support frames with any size of payload. Also assumes that every EncodedImage
+// payload size is greater or equals to 3 bytes
+//
+// This injector doesn't support video frames/encoded images without frame ID.
+class SingleProcessEncodedImageDataInjector
+    : public EncodedImageDataPropagator {
+ public:
+  SingleProcessEncodedImageDataInjector();
+  ~SingleProcessEncodedImageDataInjector() override;
+
+  // Id and discard flag will be injected into EncodedImage buffer directly.
+  // This buffer won't be fully copied, so `source` image buffer will be also
+  // changed.
+  EncodedImage InjectData(uint16_t id,
+                          bool discard,
+                          const EncodedImage& source) override;
+
+  void Start(int expected_receivers_count) override {
+    MutexLock crit(&lock_);
+    expected_receivers_count_ = expected_receivers_count;
+  }
+  void AddParticipantInCall() override;
+  void RemoveParticipantInCall() override;
+  EncodedImageExtractionResult ExtractData(const EncodedImage& source) override;
+
+ private:
+  // Contains data required to extract frame id from EncodedImage and restore
+  // original buffer.
+  struct ExtractionInfo {
+    // Number of bytes from the beginning of the EncodedImage buffer that will
+    // be used to store frame id and sub id.
+    const static size_t kUsedBufferSize = 3;
+    // Frame sub id to distinguish encoded images for different spatial layers.
+    uint8_t sub_id;
+    // Flag to show is this encoded images should be discarded by analyzing
+    // decoder because of not required spatial layer/simulcast stream.
+    bool discard;
+    // Data from first 3 bytes of origin encoded image's payload.
+    uint8_t origin_data[ExtractionInfo::kUsedBufferSize];
+    // Count of how many times this frame was received.
+    int received_count = 0;
+  };
+
+  struct ExtractionInfoVector {
+    ExtractionInfoVector();
+    ~ExtractionInfoVector();
+
+    // Next sub id, that have to be used for this frame id.
+    uint8_t next_sub_id = 0;
+    std::map<uint8_t, ExtractionInfo> infos;
+  };
+
+  Mutex lock_;
+  int expected_receivers_count_ RTC_GUARDED_BY(lock_);
+  // Stores a mapping from frame id to extraction info for spatial layers
+  // for this frame id. There can be a lot of them, because if frame was
+  // dropped we can't clean it up, because we won't receive a signal on
+  // decoder side about that frame. In such case it will be replaced
+  // when sub id will overlap.
+  std::map<uint16_t, ExtractionInfoVector> extraction_cache_
+      RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_SINGLE_PROCESS_ENCODED_IMAGE_DATA_INJECTOR_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector_unittest.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector_unittest.cc
new file mode 100644
index 0000000000..f6fa40455a
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector_unittest.cc
@@ -0,0 +1,445 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h"
+
+#include <utility>
+
+#include "api/video/encoded_image.h"
+#include "rtc_base/buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+rtc::scoped_refptr<EncodedImageBuffer>
+CreateEncodedImageBufferOfSizeNFilledWithValuesFromX(size_t n, uint8_t x) {
+  auto buffer = EncodedImageBuffer::Create(n);
+  for (size_t i = 0; i < n; ++i) {
+    buffer->data()[i] = static_cast<uint8_t>(x + i);
+  }
+  return buffer;
+}
+
+EncodedImage CreateEncodedImageOfSizeNFilledWithValuesFromX(size_t n,
+                                                            uint8_t x) {
+  EncodedImage image;
+  image.SetEncodedData(
+      CreateEncodedImageBufferOfSizeNFilledWithValuesFromX(n, x));
+  return image;
+}
+
+EncodedImage DeepCopyEncodedImage(const EncodedImage& source) {
+  EncodedImage copy = source;
+  copy.SetEncodedData(EncodedImageBuffer::Create(source.data(), source.size()));
+  return copy;
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest, InjectExtractDiscardFalse) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/1);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+
+  EncodedImageExtractionResult out =
+      injector.ExtractData(injector.InjectData(512, false, source));
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest, InjectExtractDiscardTrue) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/1);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+
+  EncodedImageExtractionResult out =
+      injector.ExtractData(injector.InjectData(512, true, source));
+  EXPECT_EQ(out.id, 512);
+  EXPECT_TRUE(out.discard);
+  EXPECT_EQ(out.image.size(), 0ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest,
+     InjectWithUnsetSpatialLayerSizes) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/1);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+
+  EncodedImage intermediate = injector.InjectData(512, false, source);
+  intermediate.SetSpatialIndex(2);
+
+  EncodedImageExtractionResult out = injector.ExtractData(intermediate);
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+  EXPECT_EQ(out.image.SpatialIndex().value_or(0), 2);
+  for (int i = 0; i < 3; ++i) {
+    EXPECT_EQ(out.image.SpatialLayerFrameSize(i).value_or(0), 0ul);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest,
+     InjectWithZeroSpatialLayerSizes) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/1);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+
+  EncodedImage intermediate = injector.InjectData(512, false, source);
+  intermediate.SetSpatialIndex(2);
+  intermediate.SetSpatialLayerFrameSize(0, 0);
+  intermediate.SetSpatialLayerFrameSize(1, 0);
+  intermediate.SetSpatialLayerFrameSize(2, 0);
+
+  EncodedImageExtractionResult out = injector.ExtractData(intermediate);
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+  EXPECT_EQ(out.image.SpatialIndex().value_or(0), 2);
+  for (int i = 0; i < 3; ++i) {
+    EXPECT_EQ(out.image.SpatialLayerFrameSize(i).value_or(0), 0ul);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest, Inject3Extract3) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/1);
+
+  // 1st frame
+  EncodedImage source1 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source1.SetTimestamp(123456710);
+  // 2nd frame 1st spatial layer
+  EncodedImage source2 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/11);
+  source2.SetTimestamp(123456720);
+  // 2nd frame 2nd spatial layer
+  EncodedImage source3 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/21);
+  source3.SetTimestamp(123456720);
+
+  EncodedImage intermediate1 = injector.InjectData(510, false, source1);
+  EncodedImage intermediate2 = injector.InjectData(520, true, source2);
+  EncodedImage intermediate3 = injector.InjectData(520, false, source3);
+
+  // Extract ids in different order.
+  EncodedImageExtractionResult out3 = injector.ExtractData(intermediate3);
+  EncodedImageExtractionResult out1 = injector.ExtractData(intermediate1);
+  EncodedImageExtractionResult out2 = injector.ExtractData(intermediate2);
+
+  EXPECT_EQ(out1.id, 510);
+  EXPECT_FALSE(out1.discard);
+  EXPECT_EQ(out1.image.size(), 10ul);
+  EXPECT_EQ(out1.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out1.image.data()[i], i + 1);
+  }
+  EXPECT_EQ(out2.id, 520);
+  EXPECT_TRUE(out2.discard);
+  EXPECT_EQ(out2.image.size(), 0ul);
+  EXPECT_EQ(out2.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  EXPECT_EQ(out3.id, 520);
+  EXPECT_FALSE(out3.discard);
+  EXPECT_EQ(out3.image.size(), 10ul);
+  EXPECT_EQ(out3.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out3.image.data()[i], i + 21);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest, InjectExtractFromConcatenated) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/1);
+
+  EncodedImage source1 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source1.SetTimestamp(123456710);
+  EncodedImage source2 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/11);
+  source2.SetTimestamp(123456710);
+  EncodedImage source3 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/21);
+  source3.SetTimestamp(123456710);
+
+  // Inject id into 3 images with same frame id.
+  EncodedImage intermediate1 = injector.InjectData(512, false, source1);
+  EncodedImage intermediate2 = injector.InjectData(512, true, source2);
+  EncodedImage intermediate3 = injector.InjectData(512, false, source3);
+
+  // Concatenate them into single encoded image, like it can be done in jitter
+  // buffer.
+  size_t concatenated_length =
+      intermediate1.size() + intermediate2.size() + intermediate3.size();
+  rtc::Buffer concatenated_buffer;
+  concatenated_buffer.AppendData(intermediate1.data(), intermediate1.size());
+  concatenated_buffer.AppendData(intermediate2.data(), intermediate2.size());
+  concatenated_buffer.AppendData(intermediate3.data(), intermediate3.size());
+  EncodedImage concatenated;
+  concatenated.SetEncodedData(EncodedImageBuffer::Create(
+      concatenated_buffer.data(), concatenated_length));
+  concatenated.SetSpatialIndex(2);
+  concatenated.SetSpatialLayerFrameSize(0, intermediate1.size());
+  concatenated.SetSpatialLayerFrameSize(1, intermediate2.size());
+  concatenated.SetSpatialLayerFrameSize(2, intermediate3.size());
+
+  // Extract frame id from concatenated image
+  EncodedImageExtractionResult out = injector.ExtractData(concatenated);
+
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 2 * 10ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+    EXPECT_EQ(out.image.data()[i + 10], i + 21);
+  }
+  EXPECT_EQ(out.image.SpatialIndex().value_or(0), 2);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 10ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(1).value_or(0), 0ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(2).value_or(0), 10ul);
+}
+
+TEST(SingleProcessEncodedImageDataInjector,
+     InjectExtractFromConcatenatedAllDiscarded) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/1);
+
+  EncodedImage source1 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source1.SetTimestamp(123456710);
+  EncodedImage source2 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/11);
+  source2.SetTimestamp(123456710);
+  EncodedImage source3 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/21);
+  source3.SetTimestamp(123456710);
+
+  // Inject id into 3 images with same frame id.
+  EncodedImage intermediate1 = injector.InjectData(512, true, source1);
+  EncodedImage intermediate2 = injector.InjectData(512, true, source2);
+  EncodedImage intermediate3 = injector.InjectData(512, true, source3);
+
+  // Concatenate them into single encoded image, like it can be done in jitter
+  // buffer.
+  size_t concatenated_length =
+      intermediate1.size() + intermediate2.size() + intermediate3.size();
+  rtc::Buffer concatenated_buffer;
+  concatenated_buffer.AppendData(intermediate1.data(), intermediate1.size());
+  concatenated_buffer.AppendData(intermediate2.data(), intermediate2.size());
+  concatenated_buffer.AppendData(intermediate3.data(), intermediate3.size());
+  EncodedImage concatenated;
+  concatenated.SetEncodedData(EncodedImageBuffer::Create(
+      concatenated_buffer.data(), concatenated_length));
+  concatenated.SetSpatialIndex(2);
+  concatenated.SetSpatialLayerFrameSize(0, intermediate1.size());
+  concatenated.SetSpatialLayerFrameSize(1, intermediate2.size());
+  concatenated.SetSpatialLayerFrameSize(2, intermediate3.size());
+
+  // Extract frame id from concatenated image
+  EncodedImageExtractionResult out = injector.ExtractData(concatenated);
+
+  EXPECT_EQ(out.id, 512);
+  EXPECT_TRUE(out.discard);
+  EXPECT_EQ(out.image.size(), 0ul);
+  EXPECT_EQ(out.image.SpatialIndex().value_or(0), 2);
+  for (int i = 0; i < 3; ++i) {
+    EXPECT_EQ(out.image.SpatialLayerFrameSize(i).value_or(0), 0ul);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest, InjectOnceExtractTwice) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/2);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+
+  EncodedImageExtractionResult out = injector.ExtractData(
+      injector.InjectData(/*id=*/512, /*discard=*/false, source));
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+  out = injector.ExtractData(
+      injector.InjectData(/*id=*/512, /*discard=*/false, source));
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest, Add1stReceiverAfterStart) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/0);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+  EncodedImage modified_image = injector.InjectData(
+      /*id=*/512, /*discard=*/false, source);
+
+  injector.AddParticipantInCall();
+  EncodedImageExtractionResult out = injector.ExtractData(modified_image);
+
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest, Add3rdReceiverAfterStart) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/2);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+  EncodedImage modified_image = injector.InjectData(
+      /*id=*/512, /*discard=*/false, source);
+  injector.ExtractData(modified_image);
+
+  injector.AddParticipantInCall();
+  injector.ExtractData(modified_image);
+  EncodedImageExtractionResult out = injector.ExtractData(modified_image);
+
+  EXPECT_EQ(out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTest,
+     RemoveReceiverRemovesOnlyFullyReceivedFrames) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/3);
+
+  EncodedImage source1 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source1.SetTimestamp(10);
+  EncodedImage source2 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source2.SetTimestamp(20);
+
+  EncodedImage modified_image1 = injector.InjectData(
+      /*id=*/512, /*discard=*/false, source1);
+  EncodedImage modified_image2 = injector.InjectData(
+      /*id=*/513, /*discard=*/false, source2);
+
+  // Out of 3 receivers 1st image received by 2 and 2nd image by 1
+  injector.ExtractData(DeepCopyEncodedImage(modified_image1));
+  injector.ExtractData(DeepCopyEncodedImage(modified_image1));
+  injector.ExtractData(DeepCopyEncodedImage(modified_image2));
+
+  // When we removed one receiver, 2nd image should still be available for
+  // extraction.
+  injector.RemoveParticipantInCall();
+
+  EncodedImageExtractionResult out =
+      injector.ExtractData(DeepCopyEncodedImage(modified_image2));
+
+  EXPECT_EQ(out.id, 513);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  EXPECT_EQ(out.image.SpatialLayerFrameSize(0).value_or(0), 0ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(out.image.data()[i], i + 1);
+  }
+}
+
+// Death tests.
+// Disabled on Android because death tests misbehave on Android, see
+// base/test/gtest_util.h.
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST(SingleProcessEncodedImageDataInjectorTestDeathTest,
+     InjectOnceExtractMoreThenExpected) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/2);
+
+  EncodedImage source =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source.SetTimestamp(123456789);
+
+  EncodedImage modified =
+      injector.InjectData(/*id=*/512, /*discard=*/false, source);
+
+  injector.ExtractData(DeepCopyEncodedImage(modified));
+  injector.ExtractData(DeepCopyEncodedImage(modified));
+  EXPECT_DEATH(injector.ExtractData(DeepCopyEncodedImage(modified)),
+               "Unknown sub_id=0 for frame_id=512");
+}
+
+TEST(SingleProcessEncodedImageDataInjectorTestDeathTest,
+     RemoveReceiverRemovesOnlyFullyReceivedFramesVerifyFrameIsRemoved) {
+  SingleProcessEncodedImageDataInjector injector;
+  injector.Start(/*expected_receivers_count=*/3);
+
+  EncodedImage source1 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source1.SetTimestamp(10);
+  EncodedImage source2 =
+      CreateEncodedImageOfSizeNFilledWithValuesFromX(/*n=*/10, /*x=*/1);
+  source2.SetTimestamp(20);
+
+  EncodedImage modified_image1 = injector.InjectData(
+      /*id=*/512, /*discard=*/false, source1);
+  EncodedImage modified_image2 = injector.InjectData(
+      /*id=*/513, /*discard=*/false, source2);
+
+  // Out of 3 receivers 1st image received by 2 and 2nd image by 1
+  injector.ExtractData(DeepCopyEncodedImage(modified_image1));
+  injector.ExtractData(DeepCopyEncodedImage(modified_image1));
+  injector.ExtractData(DeepCopyEncodedImage(modified_image2));
+
+  // When we removed one receiver 1st image should be removed.
+  injector.RemoveParticipantInCall();
+
+  EXPECT_DEATH(injector.ExtractData(DeepCopyEncodedImage(modified_image1)),
+               "Unknown sub_id=0 for frame_id=512");
+}
+#endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping.cc
new file mode 100644
index 0000000000..4fec0a8f9e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/video_dumping.h"
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/test/video/video_frame_writer.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/video_frame_writer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+class VideoFrameIdsWriter final : public test::VideoFrameWriter {
+ public:
+  explicit VideoFrameIdsWriter(absl::string_view file_name)
+      : file_name_(file_name) {
+    output_file_ = fopen(file_name_.c_str(), "wb");
+    RTC_LOG(LS_INFO) << "Writing VideoFrame IDs into " << file_name_;
+    RTC_CHECK(output_file_ != nullptr)
+        << "Failed to open file to dump frame ids for writing: " << file_name_;
+  }
+  ~VideoFrameIdsWriter() override { Close(); }
+
+  bool WriteFrame(const VideoFrame& frame) override {
+    RTC_CHECK(output_file_ != nullptr) << "Writer is already closed";
+    int chars_written = fprintf(output_file_, "%d\n", frame.id());
+    if (chars_written < 2) {
+      RTC_LOG(LS_ERROR) << "Failed to write frame id to the output file: "
+                        << file_name_;
+      return false;
+    }
+    return true;
+  }
+
+  void Close() override {
+    if (output_file_ != nullptr) {
+      RTC_LOG(LS_INFO) << "Closing file for VideoFrame IDs: " << file_name_;
+      fclose(output_file_);
+      output_file_ = nullptr;
+    }
+  }
+
+ private:
+  const std::string file_name_;
+  FILE* output_file_;
+};
+
+// Broadcast received frame to multiple underlying frame writers.
+class BroadcastingFrameWriter final : public test::VideoFrameWriter {
+ public:
+  explicit BroadcastingFrameWriter(
+      std::vector<std::unique_ptr<test::VideoFrameWriter>> delegates)
+      : delegates_(std::move(delegates)) {}
+  ~BroadcastingFrameWriter() override { Close(); }
+
+  bool WriteFrame(const webrtc::VideoFrame& frame) override {
+    for (auto& delegate : delegates_) {
+      if (!delegate->WriteFrame(frame)) {
+        return false;
+      }
+    }
+    return true;
+  }
+
+  void Close() override {
+    for (auto& delegate : delegates_) {
+      delegate->Close();
+    }
+  }
+
+ private:
+  std::vector<std::unique_ptr<test::VideoFrameWriter>> delegates_;
+};
+
+}  // namespace
+
+VideoWriter::VideoWriter(test::VideoFrameWriter* video_writer,
+                         int sampling_modulo)
+    : video_writer_(video_writer), sampling_modulo_(sampling_modulo) {}
+
+void VideoWriter::OnFrame(const VideoFrame& frame) {
+  if (frames_counter_++ % sampling_modulo_ != 0) {
+    return;
+  }
+  bool result = video_writer_->WriteFrame(frame);
+  RTC_CHECK(result) << "Failed to write frame";
+}
+
+std::unique_ptr<test::VideoFrameWriter> CreateVideoFrameWithIdsWriter(
+    std::unique_ptr<test::VideoFrameWriter> video_writer_delegate,
+    absl::string_view frame_ids_dump_file_name) {
+  std::vector<std::unique_ptr<test::VideoFrameWriter>> requested_writers;
+  requested_writers.push_back(std::move(video_writer_delegate));
+  requested_writers.push_back(
+      std::make_unique<VideoFrameIdsWriter>(frame_ids_dump_file_name));
+  return std::make_unique<BroadcastingFrameWriter>(
+      std::move(requested_writers));
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping.h
new file mode 100644
index 0000000000..cad4e1bdbf
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_DUMPING_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_DUMPING_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/test/video/video_frame_writer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "test/testsupport/video_frame_writer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// `VideoSinkInterface` to dump incoming video frames into specified video
+// writer.
+class VideoWriter final : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  // Creates video writer. Caller keeps ownership of `video_writer` and is
+  // responsible for closing it after VideoWriter will be destroyed.
+  VideoWriter(test::VideoFrameWriter* video_writer, int sampling_modulo);
+  VideoWriter(const VideoWriter&) = delete;
+  VideoWriter& operator=(const VideoWriter&) = delete;
+  ~VideoWriter() override = default;
+
+  void OnFrame(const VideoFrame& frame) override;
+
+ private:
+  test::VideoFrameWriter* const video_writer_;
+  const int sampling_modulo_;
+
+  int64_t frames_counter_ = 0;
+};
+
+// Creates a `VideoFrameWriter` to dump video frames together with their ids.
+// It uses provided `video_writer_delegate` to write video itself. Frame ids
+// will be logged into the specified file.
+std::unique_ptr<test::VideoFrameWriter> CreateVideoFrameWithIdsWriter(
+    std::unique_ptr<test::VideoFrameWriter> video_writer_delegate,
+    absl::string_view frame_ids_dump_file_name);
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_DUMPING_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping_test.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping_test.cc
new file mode 100644
index 0000000000..5dd4021516
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_dumping_test.cc
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/analyzer/video/video_dumping.h"
+
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_frame_buffer.h"
+#include "rtc_base/random.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+#include "test/testsupport/video_frame_writer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::Eq;
+using ::testing::Test;
+
+uint8_t RandByte(Random& random) {
+  return random.Rand(255);
+}
+
+VideoFrame CreateRandom2x2VideoFrame(uint16_t id, Random& random) {
+  rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(2, 2);
+
+  uint8_t data[6] = {RandByte(random), RandByte(random), RandByte(random),
+                     RandByte(random), RandByte(random), RandByte(random)};
+
+  memcpy(buffer->MutableDataY(), data, 2);
+  memcpy(buffer->MutableDataY() + buffer->StrideY(), data + 2, 2);
+  memcpy(buffer->MutableDataU(), data + 4, 1);
+  memcpy(buffer->MutableDataV(), data + 5, 1);
+
+  return VideoFrame::Builder()
+      .set_id(id)
+      .set_video_frame_buffer(buffer)
+      .set_timestamp_us(1)
+      .build();
+}
+
+std::vector<uint8_t> AsVector(const uint8_t* data, size_t size) {
+  std::vector<uint8_t> out;
+  out.assign(data, data + size);
+  return out;
+}
+
+void AssertFramesEqual(rtc::scoped_refptr<webrtc::I420BufferInterface> actual,
+                       rtc::scoped_refptr<VideoFrameBuffer> expected) {
+  ASSERT_THAT(actual->width(), Eq(expected->width()));
+  ASSERT_THAT(actual->height(), Eq(expected->height()));
+  rtc::scoped_refptr<webrtc::I420BufferInterface> expected_i420 =
+      expected->ToI420();
+
+  int height = actual->height();
+
+  EXPECT_THAT(AsVector(actual->DataY(), actual->StrideY() * height),
+              ElementsAreArray(expected_i420->DataY(),
+                               expected_i420->StrideY() * height));
+  EXPECT_THAT(AsVector(actual->DataU(), actual->StrideU() * (height + 1) / 2),
+              ElementsAreArray(expected_i420->DataU(),
+                               expected_i420->StrideU() * (height + 1) / 2));
+  EXPECT_THAT(AsVector(actual->DataV(), actual->StrideV() * (height + 1) / 2),
+              ElementsAreArray(expected_i420->DataV(),
+                               expected_i420->StrideV() * (height + 1) / 2));
+}
+
+void AssertFrameIdsAre(const std::string& filename,
+                       std::vector<std::string> expected_ids) {
+  FILE* file = fopen(filename.c_str(), "r");
+  ASSERT_TRUE(file != nullptr);
+  std::vector<std::string> actual_ids;
+  char buffer[8];
+  while (fgets(buffer, sizeof buffer, file) != nullptr) {
+    std::string current_id(buffer);
+    ASSERT_GE(current_id.size(), 2lu);
+    // Trim "\n" at the end.
+    actual_ids.push_back(current_id.substr(0, current_id.size() - 1));
+  }
+  EXPECT_THAT(actual_ids, ElementsAreArray(expected_ids));
+}
+
+class VideoDumpingTest : public Test {
+ protected:
+  ~VideoDumpingTest() override = default;
+
+  void SetUp() override {
+    video_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                 "video_dumping_test");
+    ids_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                               "video_dumping_test");
+  }
+
+  void TearDown() override {
+    remove(video_filename_.c_str());
+    remove(ids_filename_.c_str());
+  }
+
+  std::string video_filename_;
+  std::string ids_filename_;
+};
+
+using CreateVideoFrameWithIdsWriterTest = VideoDumpingTest;
+
+TEST_F(CreateVideoFrameWithIdsWriterTest, VideoIsWritenWithFrameIds) {
+  Random random(/*seed=*/100);
+  VideoFrame frame1 = CreateRandom2x2VideoFrame(1, random);
+  VideoFrame frame2 = CreateRandom2x2VideoFrame(2, random);
+
+  std::unique_ptr<test::VideoFrameWriter> writer =
+      CreateVideoFrameWithIdsWriter(
+          std::make_unique<test::Y4mVideoFrameWriterImpl>(
+              std::string(video_filename_),
+              /*width=*/2, /*height=*/2, /*fps=*/2),
+          ids_filename_);
+
+  ASSERT_TRUE(writer->WriteFrame(frame1));
+  ASSERT_TRUE(writer->WriteFrame(frame2));
+  writer->Close();
+
+  auto frame_reader = test::CreateY4mFrameReader(video_filename_);
+  EXPECT_THAT(frame_reader->num_frames(), Eq(2));
+  AssertFramesEqual(frame_reader->PullFrame(), frame1.video_frame_buffer());
+  AssertFramesEqual(frame_reader->PullFrame(), frame2.video_frame_buffer());
+  AssertFrameIdsAre(ids_filename_, {"1", "2"});
+}
+
+using VideoWriterTest = VideoDumpingTest;
+
+TEST_F(VideoWriterTest, AllFramesAreWrittenWithSamplingModulo1) {
+  Random random(/*seed=*/100);
+  VideoFrame frame1 = CreateRandom2x2VideoFrame(1, random);
+  VideoFrame frame2 = CreateRandom2x2VideoFrame(2, random);
+
+  {
+    test::Y4mVideoFrameWriterImpl frame_writer(std::string(video_filename_),
+                                               /*width=*/2, /*height=*/2,
+                                               /*fps=*/2);
+    VideoWriter writer(&frame_writer, /*sampling_modulo=*/1);
+
+    writer.OnFrame(frame1);
+    writer.OnFrame(frame2);
+    frame_writer.Close();
+  }
+
+  auto frame_reader = test::CreateY4mFrameReader(video_filename_);
+  EXPECT_THAT(frame_reader->num_frames(), Eq(2));
+  AssertFramesEqual(frame_reader->PullFrame(), frame1.video_frame_buffer());
+  AssertFramesEqual(frame_reader->PullFrame(), frame2.video_frame_buffer());
+}
+
+TEST_F(VideoWriterTest, OnlyEvery2ndFramesIsWrittenWithSamplingModulo2) {
+  Random random(/*seed=*/100);
+  VideoFrame frame1 = CreateRandom2x2VideoFrame(1, random);
+  VideoFrame frame2 = CreateRandom2x2VideoFrame(2, random);
+  VideoFrame frame3 = CreateRandom2x2VideoFrame(3, random);
+
+  {
+    test::Y4mVideoFrameWriterImpl frame_writer(std::string(video_filename_),
+                                               /*width=*/2, /*height=*/2,
+                                               /*fps=*/2);
+    VideoWriter writer(&frame_writer, /*sampling_modulo=*/2);
+
+    writer.OnFrame(frame1);
+    writer.OnFrame(frame2);
+    writer.OnFrame(frame3);
+    frame_writer.Close();
+  }
+
+  auto frame_reader = test::CreateY4mFrameReader(video_filename_);
+  EXPECT_THAT(frame_reader->num_frames(), Eq(2));
+  AssertFramesEqual(frame_reader->PullFrame(), frame1.video_frame_buffer());
+  AssertFramesEqual(frame_reader->PullFrame(), frame3.video_frame_buffer());
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.cc
new file mode 100644
index 0000000000..5a74d60250
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.h"
+
+#include "absl/memory/memory.h"
+#include "api/video/encoded_image.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+EncodedImage VideoFrameTrackingIdInjector::InjectData(
+    uint16_t id,
+    bool unused_discard,
+    const EncodedImage& source) {
+  RTC_CHECK(!unused_discard);
+  EncodedImage out = source;
+  out.SetVideoFrameTrackingId(id);
+  return out;
+}
+
+EncodedImageExtractionResult VideoFrameTrackingIdInjector::ExtractData(
+    const EncodedImage& source) {
+  return EncodedImageExtractionResult{source.VideoFrameTrackingId(), source,
+                                      /*discard=*/false};
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.h
new file mode 100644
index 0000000000..ecc3cd3f51
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_FRAME_TRACKING_ID_INJECTOR_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_FRAME_TRACKING_ID_INJECTOR_H_
+
+#include <cstdint>
+
+#include "api/video/encoded_image.h"
+#include "test/pc/e2e/analyzer/video/encoded_image_data_injector.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// This injector sets and retrieves the provided id in the EncodedImage
+// video_frame_tracking_id field. This is only possible with the RTP header
+// extension VideoFrameTrackingIdExtension that will propagate the input
+// tracking id to the received EncodedImage. This RTP header extension is
+// enabled with the field trial WebRTC-VideoFrameTrackingIdAdvertised
+// (http://www.webrtc.org/experiments/rtp-hdrext/video-frame-tracking-id).
+//
+// Note that this injector doesn't allow to discard frames.
+class VideoFrameTrackingIdInjector : public EncodedImageDataPropagator {
+ public:
+  EncodedImage InjectData(uint16_t id,
+                          bool unused_discard,
+                          const EncodedImage& source) override;
+
+  EncodedImageExtractionResult ExtractData(const EncodedImage& source) override;
+
+  void Start(int) override {}
+  void AddParticipantInCall() override {}
+  void RemoveParticipantInCall() override {}
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_FRAME_TRACKING_ID_INJECTOR_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector_unittest.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector_unittest.cc
new file mode 100644
index 0000000000..c7d453c4bb
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_frame_tracking_id_injector_unittest.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.h"
+
+#include "api/video/encoded_image.h"
+#include "rtc_base/buffer.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+EncodedImage CreateEncodedImageOfSizeN(size_t n) {
+  EncodedImage image;
+  rtc::scoped_refptr<EncodedImageBuffer> buffer = EncodedImageBuffer::Create(n);
+  for (size_t i = 0; i < n; ++i) {
+    buffer->data()[i] = static_cast<uint8_t>(i);
+  }
+  image.SetEncodedData(buffer);
+  return image;
+}
+
+TEST(VideoFrameTrackingIdInjectorTest, InjectExtractDiscardFalse) {
+  VideoFrameTrackingIdInjector injector;
+  EncodedImage source = CreateEncodedImageOfSizeN(10);
+  EncodedImageExtractionResult out =
+      injector.ExtractData(injector.InjectData(512, false, source));
+
+  ASSERT_TRUE(out.id.has_value());
+  EXPECT_EQ(*out.id, 512);
+  EXPECT_FALSE(out.discard);
+  EXPECT_EQ(out.image.size(), 10ul);
+  for (int i = 0; i < 10; ++i) {
+    EXPECT_EQ(source.data()[i], out.image.data()[i]);
+  }
+}
+
+#if GTEST_HAS_DEATH_TEST
+TEST(VideoFrameTrackingIdInjectorTest, InjectExtractDiscardTrue) {
+  VideoFrameTrackingIdInjector injector;
+  EncodedImage source = CreateEncodedImageOfSizeN(10);
+
+  EXPECT_DEATH(injector.InjectData(512, true, source), "");
+}
+#endif  // GTEST_HAS_DEATH_TEST
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc
new file mode 100644
index 0000000000..87c11886cc
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc
@@ -0,0 +1,264 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h"
+
+#include <stdio.h>
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/video/i420_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/clock.h"
+#include "test/pc/e2e/analyzer/video/analyzing_video_sink.h"
+#include "test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h"
+#include "test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h"
+#include "test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h"
+#include "test/pc/e2e/analyzer/video/video_dumping.h"
+#include "test/testsupport/fixed_fps_video_frame_writer_adapter.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using webrtc::webrtc_pc_e2e::VideoConfig;
+using EmulatedSFUConfigMap =
+    ::webrtc::webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap;
+
+class AnalyzingFramePreprocessor
+    : public test::TestVideoCapturer::FramePreprocessor {
+ public:
+  AnalyzingFramePreprocessor(
+      absl::string_view peer_name,
+      absl::string_view stream_label,
+      VideoQualityAnalyzerInterface* analyzer,
+      std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks)
+      : peer_name_(peer_name),
+        stream_label_(stream_label),
+        analyzer_(analyzer),
+        sinks_(std::move(sinks)) {}
+  ~AnalyzingFramePreprocessor() override = default;
+
+  VideoFrame Preprocess(const VideoFrame& source_frame) override {
+    // Copy VideoFrame to be able to set id on it.
+    VideoFrame frame = source_frame;
+    uint16_t frame_id =
+        analyzer_->OnFrameCaptured(peer_name_, stream_label_, frame);
+    frame.set_id(frame_id);
+
+    for (auto& sink : sinks_) {
+      sink->OnFrame(frame);
+    }
+    return frame;
+  }
+
+ private:
+  const std::string peer_name_;
+  const std::string stream_label_;
+  VideoQualityAnalyzerInterface* const analyzer_;
+  const std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>>
+      sinks_;
+};
+
+}  // namespace
+
+VideoQualityAnalyzerInjectionHelper::VideoQualityAnalyzerInjectionHelper(
+    Clock* clock,
+    std::unique_ptr<VideoQualityAnalyzerInterface> analyzer,
+    EncodedImageDataInjector* injector,
+    EncodedImageDataExtractor* extractor)
+    : clock_(clock),
+      analyzer_(std::move(analyzer)),
+      injector_(injector),
+      extractor_(extractor) {
+  RTC_DCHECK(clock_);
+  RTC_DCHECK(injector_);
+  RTC_DCHECK(extractor_);
+}
+VideoQualityAnalyzerInjectionHelper::~VideoQualityAnalyzerInjectionHelper() =
+    default;
+
+std::unique_ptr<VideoEncoderFactory>
+VideoQualityAnalyzerInjectionHelper::WrapVideoEncoderFactory(
+    absl::string_view peer_name,
+    std::unique_ptr<VideoEncoderFactory> delegate,
+    double bitrate_multiplier,
+    EmulatedSFUConfigMap stream_to_sfu_config) const {
+  return std::make_unique<QualityAnalyzingVideoEncoderFactory>(
+      peer_name, std::move(delegate), bitrate_multiplier,
+      std::move(stream_to_sfu_config), injector_, analyzer_.get());
+}
+
+std::unique_ptr<VideoDecoderFactory>
+VideoQualityAnalyzerInjectionHelper::WrapVideoDecoderFactory(
+    absl::string_view peer_name,
+    std::unique_ptr<VideoDecoderFactory> delegate) const {
+  return std::make_unique<QualityAnalyzingVideoDecoderFactory>(
+      peer_name, std::move(delegate), extractor_, analyzer_.get());
+}
+
+std::unique_ptr<test::TestVideoCapturer::FramePreprocessor>
+VideoQualityAnalyzerInjectionHelper::CreateFramePreprocessor(
+    absl::string_view peer_name,
+    const VideoConfig& config) {
+  std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
+  if (config.input_dump_options.has_value()) {
+    std::unique_ptr<test::VideoFrameWriter> writer =
+        config.input_dump_options->CreateInputDumpVideoFrameWriter(
+            *config.stream_label, config.GetResolution());
+    sinks.push_back(std::make_unique<VideoWriter>(
+        writer.get(), config.input_dump_options->sampling_modulo()));
+    video_writers_.push_back(std::move(writer));
+  }
+  if (config.show_on_screen) {
+    sinks.push_back(absl::WrapUnique(
+        test::VideoRenderer::Create((*config.stream_label + "-capture").c_str(),
+                                    config.width, config.height)));
+  }
+  sinks_helper_.AddConfig(peer_name, config);
+  {
+    MutexLock lock(&mutex_);
+    known_video_configs_.insert({*config.stream_label, config});
+  }
+  return std::make_unique<AnalyzingFramePreprocessor>(
+      peer_name, std::move(*config.stream_label), analyzer_.get(),
+      std::move(sinks));
+}
+
+std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>
+VideoQualityAnalyzerInjectionHelper::CreateVideoSink(
+    absl::string_view peer_name) {
+  return std::make_unique<AnalyzingVideoSink2>(peer_name, this);
+}
+
+std::unique_ptr<AnalyzingVideoSink>
+VideoQualityAnalyzerInjectionHelper::CreateVideoSink(
+    absl::string_view peer_name,
+    const VideoSubscription& subscription,
+    bool report_infra_metrics) {
+  return std::make_unique<AnalyzingVideoSink>(peer_name, clock_, *analyzer_,
+                                              sinks_helper_, subscription,
+                                              report_infra_metrics);
+}
+
+void VideoQualityAnalyzerInjectionHelper::Start(
+    std::string test_case_name,
+    rtc::ArrayView<const std::string> peer_names,
+    int max_threads_count) {
+  analyzer_->Start(std::move(test_case_name), peer_names, max_threads_count);
+  extractor_->Start(peer_names.size());
+  MutexLock lock(&mutex_);
+  peers_count_ = peer_names.size();
+}
+
+void VideoQualityAnalyzerInjectionHelper::RegisterParticipantInCall(
+    absl::string_view peer_name) {
+  analyzer_->RegisterParticipantInCall(peer_name);
+  extractor_->AddParticipantInCall();
+  MutexLock lock(&mutex_);
+  peers_count_++;
+}
+
+void VideoQualityAnalyzerInjectionHelper::UnregisterParticipantInCall(
+    absl::string_view peer_name) {
+  analyzer_->UnregisterParticipantInCall(peer_name);
+  extractor_->RemoveParticipantInCall();
+  MutexLock lock(&mutex_);
+  peers_count_--;
+}
+
+void VideoQualityAnalyzerInjectionHelper::OnStatsReports(
+    absl::string_view pc_label,
+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
+  analyzer_->OnStatsReports(pc_label, report);
+}
+
+void VideoQualityAnalyzerInjectionHelper::Stop() {
+  analyzer_->Stop();
+  for (const auto& video_writer : video_writers_) {
+    video_writer->Close();
+  }
+  video_writers_.clear();
+  sinks_helper_.Clear();
+}
+
+void VideoQualityAnalyzerInjectionHelper::OnFrame(absl::string_view peer_name,
+                                                  const VideoFrame& frame) {
+  if (IsDummyFrame(frame)) {
+    // This is dummy frame, so we  don't need to process it further.
+    return;
+  }
+  // Copy entire video frame including video buffer to ensure that analyzer
+  // won't hold any WebRTC internal buffers.
+  VideoFrame frame_copy = frame;
+  frame_copy.set_video_frame_buffer(
+      I420Buffer::Copy(*frame.video_frame_buffer()->ToI420()));
+  analyzer_->OnFrameRendered(peer_name, frame_copy);
+
+  if (frame.id() != VideoFrame::kNotSetId) {
+    std::string stream_label = analyzer_->GetStreamLabel(frame.id());
+    std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>>* sinks =
+        PopulateSinks(ReceiverStream(peer_name, stream_label));
+    if (sinks == nullptr) {
+      return;
+    }
+    for (auto& sink : *sinks) {
+      sink->OnFrame(frame);
+    }
+  }
+}
+
+std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>>*
+VideoQualityAnalyzerInjectionHelper::PopulateSinks(
+    const ReceiverStream& receiver_stream) {
+  MutexLock lock(&mutex_);
+  auto sinks_it = sinks_.find(receiver_stream);
+  if (sinks_it != sinks_.end()) {
+    return &sinks_it->second;
+  }
+  auto it = known_video_configs_.find(receiver_stream.stream_label);
+  RTC_DCHECK(it != known_video_configs_.end())
+      << "No video config for stream " << receiver_stream.stream_label;
+  const VideoConfig& config = it->second;
+
+  std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>> sinks;
+  if (config.output_dump_options.has_value()) {
+    std::unique_ptr<test::VideoFrameWriter> writer =
+        config.output_dump_options->CreateOutputDumpVideoFrameWriter(
+            receiver_stream.stream_label, receiver_stream.peer_name,
+            config.GetResolution());
+    if (config.output_dump_use_fixed_framerate) {
+      writer = std::make_unique<test::FixedFpsVideoFrameWriterAdapter>(
+          config.fps, clock_, std::move(writer));
+    }
+    sinks.push_back(std::make_unique<VideoWriter>(
+        writer.get(), config.output_dump_options->sampling_modulo()));
+    video_writers_.push_back(std::move(writer));
+  }
+  if (config.show_on_screen) {
+    sinks.push_back(absl::WrapUnique(
+        test::VideoRenderer::Create((*config.stream_label + "-render").c_str(),
+                                    config.width, config.height)));
+  }
+  sinks_.insert({receiver_stream, std::move(sinks)});
+  return &(sinks_.find(receiver_stream)->second);
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h
new file mode 100644
index 0000000000..7421c8e4a7
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_QUALITY_ANALYZER_INJECTION_HELPER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_QUALITY_ANALYZER_INJECTION_HELPER_H_
+
+#include <stdio.h>
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/stats_observer_interface.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "system_wrappers/include/clock.h"
+#include "test/pc/e2e/analyzer/video/analyzing_video_sink.h"
+#include "test/pc/e2e/analyzer/video/analyzing_video_sinks_helper.h"
+#include "test/pc/e2e/analyzer/video/encoded_image_data_injector.h"
+#include "test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h"
+#include "test/test_video_capturer.h"
+#include "test/testsupport/video_frame_writer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Provides factory methods for components, that will be used to inject
+// VideoQualityAnalyzerInterface into PeerConnection pipeline.
+class VideoQualityAnalyzerInjectionHelper : public StatsObserverInterface {
+ public:
+  VideoQualityAnalyzerInjectionHelper(
+      Clock* clock,
+      std::unique_ptr<VideoQualityAnalyzerInterface> analyzer,
+      EncodedImageDataInjector* injector,
+      EncodedImageDataExtractor* extractor);
+  ~VideoQualityAnalyzerInjectionHelper() override;
+
+  // Wraps video encoder factory to give video quality analyzer access to frames
+  // before encoding and encoded images after.
+  std::unique_ptr<VideoEncoderFactory> WrapVideoEncoderFactory(
+      absl::string_view peer_name,
+      std::unique_ptr<VideoEncoderFactory> delegate,
+      double bitrate_multiplier,
+      QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap stream_to_sfu_config)
+      const;
+  // Wraps video decoder factory to give video quality analyzer access to
+  // received encoded images and frames, that were decoded from them.
+  std::unique_ptr<VideoDecoderFactory> WrapVideoDecoderFactory(
+      absl::string_view peer_name,
+      std::unique_ptr<VideoDecoderFactory> delegate) const;
+
+  // Creates VideoFrame preprocessor, that will allow video quality analyzer to
+  // get access to the captured frames. If provided config also specifies
+  // `input_dump_file_name`, video will be written into that file.
+  std::unique_ptr<test::TestVideoCapturer::FramePreprocessor>
+  CreateFramePreprocessor(absl::string_view peer_name,
+                          const webrtc::webrtc_pc_e2e::VideoConfig& config);
+  // Creates sink, that will allow video quality analyzer to get access to
+  // the rendered frames. If corresponding video track has
+  // `output_dump_file_name` in its VideoConfig, which was used for
+  // CreateFramePreprocessor(...), then video also will be written
+  // into that file.
+  // TODO(titovartem): Remove method with `peer_name` only parameter.
+  std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> CreateVideoSink(
+      absl::string_view peer_name);
+  std::unique_ptr<AnalyzingVideoSink> CreateVideoSink(
+      absl::string_view peer_name,
+      const VideoSubscription& subscription,
+      bool report_infra_metrics);
+
+  void Start(std::string test_case_name,
+             rtc::ArrayView<const std::string> peer_names,
+             int max_threads_count = 1);
+
+  // Registers new call participant to the underlying video quality analyzer.
+  // The method should be called before the participant is actually added.
+  void RegisterParticipantInCall(absl::string_view peer_name);
+
+  // Will be called after test removed existing participant in the middle of the
+  // call.
+  void UnregisterParticipantInCall(absl::string_view peer_name);
+
+  // Forwards `stats_reports` for Peer Connection `pc_label` to
+  // `analyzer_`.
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override;
+
+  // Stops VideoQualityAnalyzerInterface to populate final data and metrics.
+  // Should be invoked after analyzed video tracks are disposed.
+  void Stop();
+
+ private:
+  // Deprecated, to be removed when old API isn't used anymore.
+  class AnalyzingVideoSink2 final : public rtc::VideoSinkInterface<VideoFrame> {
+   public:
+    explicit AnalyzingVideoSink2(absl::string_view peer_name,
+                                 VideoQualityAnalyzerInjectionHelper* helper)
+        : peer_name_(peer_name), helper_(helper) {}
+    ~AnalyzingVideoSink2() override = default;
+
+    void OnFrame(const VideoFrame& frame) override {
+      helper_->OnFrame(peer_name_, frame);
+    }
+
+   private:
+    const std::string peer_name_;
+    VideoQualityAnalyzerInjectionHelper* const helper_;
+  };
+
+  struct ReceiverStream {
+    ReceiverStream(absl::string_view peer_name, absl::string_view stream_label)
+        : peer_name(peer_name), stream_label(stream_label) {}
+
+    std::string peer_name;
+    std::string stream_label;
+
+    // Define operators required to use ReceiverStream as std::map key.
+    bool operator==(const ReceiverStream& o) const {
+      return peer_name == o.peer_name && stream_label == o.stream_label;
+    }
+    bool operator<(const ReceiverStream& o) const {
+      return (peer_name == o.peer_name) ? stream_label < o.stream_label
+                                        : peer_name < o.peer_name;
+    }
+  };
+
+  // Creates a deep copy of the frame and passes it to the video analyzer, while
+  // passing real frame to the sinks
+  void OnFrame(absl::string_view peer_name, const VideoFrame& frame);
+  std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>>*
+  PopulateSinks(const ReceiverStream& receiver_stream);
+
+  Clock* const clock_;
+  std::unique_ptr<VideoQualityAnalyzerInterface> analyzer_;
+  EncodedImageDataInjector* injector_;
+  EncodedImageDataExtractor* extractor_;
+
+  std::vector<std::unique_ptr<test::VideoFrameWriter>> video_writers_;
+
+  AnalyzingVideoSinksHelper sinks_helper_;
+  Mutex mutex_;
+  int peers_count_ RTC_GUARDED_BY(mutex_);
+  // Map from stream label to the video config.
+  std::map<std::string, webrtc::webrtc_pc_e2e::VideoConfig> known_video_configs_
+      RTC_GUARDED_BY(mutex_);
+  std::map<ReceiverStream,
+           std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>>>
+      sinks_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_QUALITY_ANALYZER_INJECTION_HELPER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc
new file mode 100644
index 0000000000..8049af308e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc
@@ -0,0 +1,162 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h"
+
+#include <map>
+#include <string>
+
+#include "api/stats/rtc_stats.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/test/metrics/metric.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+using ::webrtc::webrtc_pc_e2e::MetricMetadataKey;
+
+SamplesStatsCounter BytesPerSecondToKbps(const SamplesStatsCounter& counter) {
+  return counter * 0.008;
+}
+
+}  // namespace
+
+VideoQualityMetricsReporter::VideoQualityMetricsReporter(
+    Clock* const clock,
+    test::MetricsLogger* const metrics_logger)
+    : clock_(clock), metrics_logger_(metrics_logger) {
+  RTC_CHECK(metrics_logger_);
+}
+
+void VideoQualityMetricsReporter::Start(
+    absl::string_view test_case_name,
+    const TrackIdStreamInfoMap* /*reporter_helper*/) {
+  test_case_name_ = std::string(test_case_name);
+  start_time_ = Now();
+}
+
+void VideoQualityMetricsReporter::OnStatsReports(
+    absl::string_view pc_label,
+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
+  RTC_CHECK(start_time_)
+      << "Please invoke Start(...) method before calling OnStatsReports(...)";
+
+  auto transport_stats = report->GetStatsOfType<RTCTransportStats>();
+  if (transport_stats.size() == 0u ||
+      !transport_stats[0]->selected_candidate_pair_id.is_defined()) {
+    return;
+  }
+  RTC_DCHECK_EQ(transport_stats.size(), 1);
+  std::string selected_ice_id =
+      transport_stats[0]->selected_candidate_pair_id.ValueToString();
+  // Use the selected ICE candidate pair ID to get the appropriate ICE stats.
+  const RTCIceCandidatePairStats ice_candidate_pair_stats =
+      report->Get(selected_ice_id)->cast_to<const RTCIceCandidatePairStats>();
+
+  auto outbound_rtp_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
+  StatsSample sample;
+  for (auto& s : outbound_rtp_stats) {
+    if (!s->kind.is_defined()) {
+      continue;
+    }
+    if (!(*s->kind == RTCMediaStreamTrackKind::kVideo)) {
+      continue;
+    }
+    if (s->timestamp() > sample.sample_time) {
+      sample.sample_time = s->timestamp();
+    }
+    sample.retransmitted_bytes_sent +=
+        DataSize::Bytes(s->retransmitted_bytes_sent.ValueOrDefault(0ul));
+    sample.bytes_sent += DataSize::Bytes(s->bytes_sent.ValueOrDefault(0ul));
+    sample.header_bytes_sent +=
+        DataSize::Bytes(s->header_bytes_sent.ValueOrDefault(0ul));
+  }
+
+  MutexLock lock(&video_bwe_stats_lock_);
+  VideoBweStats& video_bwe_stats = video_bwe_stats_[std::string(pc_label)];
+  if (ice_candidate_pair_stats.available_outgoing_bitrate.is_defined()) {
+    video_bwe_stats.available_send_bandwidth.AddSample(
+        DataRate::BitsPerSec(
+            *ice_candidate_pair_stats.available_outgoing_bitrate)
+            .bytes_per_sec());
+  }
+
+  StatsSample prev_sample = last_stats_sample_[std::string(pc_label)];
+  if (prev_sample.sample_time.IsZero()) {
+    prev_sample.sample_time = start_time_.value();
+  }
+  last_stats_sample_[std::string(pc_label)] = sample;
+
+  TimeDelta time_between_samples = sample.sample_time - prev_sample.sample_time;
+  if (time_between_samples.IsZero()) {
+    return;
+  }
+
+  DataRate retransmission_bitrate =
+      (sample.retransmitted_bytes_sent - prev_sample.retransmitted_bytes_sent) /
+      time_between_samples;
+  video_bwe_stats.retransmission_bitrate.AddSample(
+      retransmission_bitrate.bytes_per_sec());
+  DataRate transmission_bitrate =
+      (sample.bytes_sent + sample.header_bytes_sent - prev_sample.bytes_sent -
+       prev_sample.header_bytes_sent) /
+      time_between_samples;
+  video_bwe_stats.transmission_bitrate.AddSample(
+      transmission_bitrate.bytes_per_sec());
+}
+
+void VideoQualityMetricsReporter::StopAndReportResults() {
+  MutexLock video_bwemutex_(&video_bwe_stats_lock_);
+  for (const auto& item : video_bwe_stats_) {
+    ReportVideoBweResults(item.first, item.second);
+  }
+}
+
+std::string VideoQualityMetricsReporter::GetTestCaseName(
+    const std::string& peer_name) const {
+  return test_case_name_ + "/" + peer_name;
+}
+
+void VideoQualityMetricsReporter::ReportVideoBweResults(
+    const std::string& peer_name,
+    const VideoBweStats& video_bwe_stats) {
+  std::string test_case_name = GetTestCaseName(peer_name);
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  std::map<std::string, std::string> metric_metadata{
+      {MetricMetadataKey::kPeerMetadataKey, peer_name},
+      {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}};
+
+  metrics_logger_->LogMetric(
+      "available_send_bandwidth", test_case_name,
+      BytesPerSecondToKbps(video_bwe_stats.available_send_bandwidth),
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter,
+      metric_metadata);
+  metrics_logger_->LogMetric(
+      "transmission_bitrate", test_case_name,
+      BytesPerSecondToKbps(video_bwe_stats.transmission_bitrate),
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter,
+      metric_metadata);
+  metrics_logger_->LogMetric(
+      "retransmission_bitrate", test_case_name,
+      BytesPerSecondToKbps(video_bwe_stats.retransmission_bitrate),
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter,
+      metric_metadata);
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h
new file mode 100644
index 0000000000..d3d976343b
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_QUALITY_METRICS_REPORTER_H_
+#define TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_QUALITY_METRICS_REPORTER_H_
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/test/track_id_stream_info_map.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+struct VideoBweStats {
+  SamplesStatsCounter available_send_bandwidth;
+  SamplesStatsCounter transmission_bitrate;
+  SamplesStatsCounter retransmission_bitrate;
+};
+
+class VideoQualityMetricsReporter
+    : public PeerConnectionE2EQualityTestFixture::QualityMetricsReporter {
+ public:
+  VideoQualityMetricsReporter(Clock* const clock,
+                              test::MetricsLogger* const metrics_logger);
+  ~VideoQualityMetricsReporter() override = default;
+
+  void Start(absl::string_view test_case_name,
+             const TrackIdStreamInfoMap* reporter_helper) override;
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override;
+  void StopAndReportResults() override;
+
+ private:
+  struct StatsSample {
+    DataSize bytes_sent = DataSize::Zero();
+    DataSize header_bytes_sent = DataSize::Zero();
+    DataSize retransmitted_bytes_sent = DataSize::Zero();
+
+    Timestamp sample_time = Timestamp::Zero();
+  };
+
+  std::string GetTestCaseName(const std::string& peer_name) const;
+  void ReportVideoBweResults(const std::string& peer_name,
+                             const VideoBweStats& video_bwe_stats);
+  Timestamp Now() const { return clock_->CurrentTime(); }
+
+  Clock* const clock_;
+  test::MetricsLogger* const metrics_logger_;
+
+  std::string test_case_name_;
+  absl::optional<Timestamp> start_time_;
+
+  Mutex video_bwe_stats_lock_;
+  // Map between a peer connection label (provided by the framework) and
+  // its video BWE stats.
+  std::map<std::string, VideoBweStats> video_bwe_stats_
+      RTC_GUARDED_BY(video_bwe_stats_lock_);
+  std::map<std::string, StatsSample> last_stats_sample_
+      RTC_GUARDED_BY(video_bwe_stats_lock_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_VIDEO_VIDEO_QUALITY_METRICS_REPORTER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer_helper.cc b/third_party/libwebrtc/test/pc/e2e/analyzer_helper.cc
new file mode 100644
index 0000000000..76cd9a7c78
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer_helper.cc
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/analyzer_helper.h"
+
+#include <string>
+#include <utility>
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+AnalyzerHelper::AnalyzerHelper() {
+  signaling_sequence_checker_.Detach();
+}
+
+void AnalyzerHelper::AddTrackToStreamMapping(
+    absl::string_view track_id,
+    absl::string_view receiver_peer,
+    absl::string_view stream_label,
+    absl::optional<std::string> sync_group) {
+  RTC_DCHECK_RUN_ON(&signaling_sequence_checker_);
+  track_to_stream_map_.insert(
+      {std::string(track_id),
+       StreamInfo{.receiver_peer = std::string(receiver_peer),
+                  .stream_label = std::string(stream_label),
+                  .sync_group = sync_group.has_value()
+                                    ? *sync_group
+                                    : std::string(stream_label)}});
+}
+
+void AnalyzerHelper::AddTrackToStreamMapping(std::string track_id,
+                                             std::string stream_label) {
+  RTC_DCHECK_RUN_ON(&signaling_sequence_checker_);
+  track_to_stream_map_.insert(
+      {std::move(track_id), StreamInfo{stream_label, stream_label}});
+}
+
+void AnalyzerHelper::AddTrackToStreamMapping(std::string track_id,
+                                             std::string stream_label,
+                                             std::string sync_group) {
+  RTC_DCHECK_RUN_ON(&signaling_sequence_checker_);
+  track_to_stream_map_.insert(
+      {std::move(track_id),
+       StreamInfo{std::move(stream_label), std::move(sync_group)}});
+}
+
+AnalyzerHelper::StreamInfo AnalyzerHelper::GetStreamInfoFromTrackId(
+    absl::string_view track_id) const {
+  RTC_DCHECK_RUN_ON(&signaling_sequence_checker_);
+  auto track_to_stream_pair = track_to_stream_map_.find(std::string(track_id));
+  RTC_CHECK(track_to_stream_pair != track_to_stream_map_.end());
+  return track_to_stream_pair->second;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/analyzer_helper.h b/third_party/libwebrtc/test/pc/e2e/analyzer_helper.h
new file mode 100644
index 0000000000..d0b47c4fb9
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer_helper.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ANALYZER_HELPER_H_
+#define TEST_PC_E2E_ANALYZER_HELPER_H_
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/test/track_id_stream_info_map.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// This class is a utility that provides bookkeeping capabilities that
+// are useful to associate stats reports track_ids to the remote stream info.
+// The framework will populate an instance of this class and it will pass
+// it to the Start method of Media Quality Analyzers.
+// An instance of AnalyzerHelper must only be accessed from a single
+// thread and since stats collection happens on the signaling thread,
+// AddTrackToStreamMapping, GetStreamLabelFromTrackId and
+// GetSyncGroupLabelFromTrackId must be invoked from the signaling thread. Get
+// methods should be invoked only after all data is added. Mixing Get methods
+// with adding new data may lead to undefined behavior.
+class AnalyzerHelper : public TrackIdStreamInfoMap {
+ public:
+  AnalyzerHelper();
+
+  void AddTrackToStreamMapping(absl::string_view track_id,
+                               absl::string_view receiver_peer,
+                               absl::string_view stream_label,
+                               absl::optional<std::string> sync_group);
+  void AddTrackToStreamMapping(std::string track_id, std::string stream_label);
+  void AddTrackToStreamMapping(std::string track_id,
+                               std::string stream_label,
+                               std::string sync_group);
+
+  StreamInfo GetStreamInfoFromTrackId(
+      absl::string_view track_id) const override;
+
+ private:
+  SequenceChecker signaling_sequence_checker_;
+  std::map<std::string, StreamInfo> track_to_stream_map_
+      RTC_GUARDED_BY(signaling_sequence_checker_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ANALYZER_HELPER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.cc
new file mode 100644
index 0000000000..0d4fe7478d
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/cross_media_metrics_reporter.h"
+
+#include <utility>
+#include <vector>
+
+#include "api/stats/rtc_stats.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/test/metrics/metric.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+CrossMediaMetricsReporter::CrossMediaMetricsReporter(
+    test::MetricsLogger* metrics_logger)
+    : metrics_logger_(metrics_logger) {
+  RTC_CHECK(metrics_logger_);
+}
+
+void CrossMediaMetricsReporter::Start(
+    absl::string_view test_case_name,
+    const TrackIdStreamInfoMap* reporter_helper) {
+  test_case_name_ = std::string(test_case_name);
+  reporter_helper_ = reporter_helper;
+}
+
+void CrossMediaMetricsReporter::OnStatsReports(
+    absl::string_view pc_label,
+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
+  auto inbound_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
+  std::map<std::string, std::vector<const RTCInboundRTPStreamStats*>>
+      sync_group_stats;
+  for (const auto& stat : inbound_stats) {
+    if (stat->estimated_playout_timestamp.ValueOrDefault(0.) > 0 &&
+        stat->track_identifier.is_defined()) {
+      sync_group_stats[reporter_helper_
+                           ->GetStreamInfoFromTrackId(*stat->track_identifier)
+                           .sync_group]
+          .push_back(stat);
+    }
+  }
+
+  MutexLock lock(&mutex_);
+  for (const auto& pair : sync_group_stats) {
+    // If there is less than two streams, it is not a sync group.
+    if (pair.second.size() < 2) {
+      continue;
+    }
+    auto sync_group = std::string(pair.first);
+    const RTCInboundRTPStreamStats* audio_stat = pair.second[0];
+    const RTCInboundRTPStreamStats* video_stat = pair.second[1];
+
+    RTC_CHECK(pair.second.size() == 2 && audio_stat->kind.is_defined() &&
+              video_stat->kind.is_defined() &&
+              *audio_stat->kind != *video_stat->kind)
+        << "Sync group should consist of one audio and one video stream.";
+
+    if (*audio_stat->kind == RTCMediaStreamTrackKind::kVideo) {
+      std::swap(audio_stat, video_stat);
+    }
+    // Stream labels of a sync group are same for all polls, so we need it add
+    // it only once.
+    if (stats_info_.find(sync_group) == stats_info_.end()) {
+      RTC_CHECK(audio_stat->track_identifier.is_defined());
+      RTC_CHECK(video_stat->track_identifier.is_defined());
+      stats_info_[sync_group].audio_stream_info =
+          reporter_helper_->GetStreamInfoFromTrackId(
+              *audio_stat->track_identifier);
+      stats_info_[sync_group].video_stream_info =
+          reporter_helper_->GetStreamInfoFromTrackId(
+              *video_stat->track_identifier);
+    }
+
+    double audio_video_playout_diff = *audio_stat->estimated_playout_timestamp -
+                                      *video_stat->estimated_playout_timestamp;
+    if (audio_video_playout_diff > 0) {
+      stats_info_[sync_group].audio_ahead_ms.AddSample(
+          audio_video_playout_diff);
+      stats_info_[sync_group].video_ahead_ms.AddSample(0);
+    } else {
+      stats_info_[sync_group].audio_ahead_ms.AddSample(0);
+      stats_info_[sync_group].video_ahead_ms.AddSample(
+          std::abs(audio_video_playout_diff));
+    }
+  }
+}
+
+void CrossMediaMetricsReporter::StopAndReportResults() {
+  MutexLock lock(&mutex_);
+  for (const auto& pair : stats_info_) {
+    const std::string& sync_group = pair.first;
+    // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+    std::map<std::string, std::string> audio_metric_metadata{
+        {MetricMetadataKey::kPeerSyncGroupMetadataKey, sync_group},
+        {MetricMetadataKey::kAudioStreamMetadataKey,
+         pair.second.audio_stream_info.stream_label},
+        {MetricMetadataKey::kPeerMetadataKey,
+         pair.second.audio_stream_info.receiver_peer},
+        {MetricMetadataKey::kReceiverMetadataKey,
+         pair.second.audio_stream_info.receiver_peer},
+        {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}};
+    metrics_logger_->LogMetric(
+        "audio_ahead_ms",
+        GetTestCaseName(pair.second.audio_stream_info.stream_label, sync_group),
+        pair.second.audio_ahead_ms, Unit::kMilliseconds,
+        webrtc::test::ImprovementDirection::kSmallerIsBetter,
+        std::move(audio_metric_metadata));
+
+    // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+    std::map<std::string, std::string> video_metric_metadata{
+        {MetricMetadataKey::kPeerSyncGroupMetadataKey, sync_group},
+        {MetricMetadataKey::kAudioStreamMetadataKey,
+         pair.second.video_stream_info.stream_label},
+        {MetricMetadataKey::kPeerMetadataKey,
+         pair.second.video_stream_info.receiver_peer},
+        {MetricMetadataKey::kReceiverMetadataKey,
+         pair.second.video_stream_info.receiver_peer},
+        {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}};
+    metrics_logger_->LogMetric(
+        "video_ahead_ms",
+        GetTestCaseName(pair.second.video_stream_info.stream_label, sync_group),
+        pair.second.video_ahead_ms, Unit::kMilliseconds,
+        webrtc::test::ImprovementDirection::kSmallerIsBetter,
+        std::move(video_metric_metadata));
+  }
+}
+
+std::string CrossMediaMetricsReporter::GetTestCaseName(
+    const std::string& stream_label,
+    const std::string& sync_group) const {
+  return test_case_name_ + "/" + sync_group + "_" + stream_label;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.h
new file mode 100644
index 0000000000..2d51ebb20f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_CROSS_MEDIA_METRICS_REPORTER_H_
+#define TEST_PC_E2E_CROSS_MEDIA_METRICS_REPORTER_H_
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/test/track_id_stream_info_map.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class CrossMediaMetricsReporter
+    : public PeerConnectionE2EQualityTestFixture::QualityMetricsReporter {
+ public:
+  explicit CrossMediaMetricsReporter(test::MetricsLogger* metrics_logger);
+  ~CrossMediaMetricsReporter() override = default;
+
+  void Start(absl::string_view test_case_name,
+             const TrackIdStreamInfoMap* reporter_helper) override;
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override;
+  void StopAndReportResults() override;
+
+ private:
+  struct StatsInfo {
+    SamplesStatsCounter audio_ahead_ms;
+    SamplesStatsCounter video_ahead_ms;
+
+    TrackIdStreamInfoMap::StreamInfo audio_stream_info;
+    TrackIdStreamInfoMap::StreamInfo video_stream_info;
+    std::string audio_stream_label;
+    std::string video_stream_label;
+  };
+
+  std::string GetTestCaseName(const std::string& stream_label,
+                              const std::string& sync_group) const;
+
+  test::MetricsLogger* const metrics_logger_;
+
+  std::string test_case_name_;
+  const TrackIdStreamInfoMap* reporter_helper_;
+
+  Mutex mutex_;
+  std::map<std::string, StatsInfo> stats_info_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_CROSS_MEDIA_METRICS_REPORTER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.cc b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.cc
new file mode 100644
index 0000000000..8fdabeb16f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.cc
@@ -0,0 +1,117 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/echo/echo_emulation.h"
+
+#include <limits>
+#include <utility>
+
+#include "api/test/pclf/media_configuration.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+constexpr int kSingleBufferDurationMs = 10;
+
+}  // namespace
+
+EchoEmulatingCapturer::EchoEmulatingCapturer(
+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+    EchoEmulationConfig config)
+    : delegate_(std::move(capturer)),
+      config_(config),
+      renderer_queue_(2 * config_.echo_delay.ms() / kSingleBufferDurationMs),
+      queue_input_(TestAudioDeviceModule::SamplesPerFrame(
+                       delegate_->SamplingFrequency()) *
+                   delegate_->NumChannels()),
+      queue_output_(TestAudioDeviceModule::SamplesPerFrame(
+                        delegate_->SamplingFrequency()) *
+                    delegate_->NumChannels()) {
+  renderer_thread_.Detach();
+  capturer_thread_.Detach();
+}
+
+void EchoEmulatingCapturer::OnAudioRendered(
+    rtc::ArrayView<const int16_t> data) {
+  RTC_DCHECK_RUN_ON(&renderer_thread_);
+  if (!recording_started_) {
+    // Because rendering can start before capturing in the beginning we can have
+    // a set of empty audio data frames. So we will skip them and will start
+    // fill the queue only after 1st non-empty audio data frame will arrive.
+    bool is_empty = true;
+    for (auto d : data) {
+      if (d != 0) {
+        is_empty = false;
+        break;
+      }
+    }
+    if (is_empty) {
+      return;
+    }
+    recording_started_ = true;
+  }
+  queue_input_.assign(data.begin(), data.end());
+  if (!renderer_queue_.Insert(&queue_input_)) {
+    RTC_LOG(LS_WARNING) << "Echo queue is full";
+  }
+}
+
+bool EchoEmulatingCapturer::Capture(rtc::BufferT<int16_t>* buffer) {
+  RTC_DCHECK_RUN_ON(&capturer_thread_);
+  bool result = delegate_->Capture(buffer);
+  // Now we have to reduce input signal to avoid saturation when mixing in the
+  // fake echo.
+  for (size_t i = 0; i < buffer->size(); ++i) {
+    (*buffer)[i] /= 2;
+  }
+
+  // When we accumulated enough delay in the echo buffer we will pop from
+  // that buffer on each ::Capture(...) call. If the buffer become empty it
+  // will mean some bug, so we will crash during removing item from the queue.
+  if (!delay_accumulated_) {
+    delay_accumulated_ =
+        renderer_queue_.SizeAtLeast() >=
+        static_cast<size_t>(config_.echo_delay.ms() / kSingleBufferDurationMs);
+  }
+
+  if (delay_accumulated_) {
+    RTC_CHECK(renderer_queue_.Remove(&queue_output_));
+    for (size_t i = 0; i < buffer->size() && i < queue_output_.size(); ++i) {
+      int32_t res = (*buffer)[i] + queue_output_[i];
+      if (res < std::numeric_limits<int16_t>::min()) {
+        res = std::numeric_limits<int16_t>::min();
+      }
+      if (res > std::numeric_limits<int16_t>::max()) {
+        res = std::numeric_limits<int16_t>::max();
+      }
+      (*buffer)[i] = static_cast<int16_t>(res);
+    }
+  }
+
+  return result;
+}
+
+EchoEmulatingRenderer::EchoEmulatingRenderer(
+    std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+    EchoEmulatingCapturer* echo_emulating_capturer)
+    : delegate_(std::move(renderer)),
+      echo_emulating_capturer_(echo_emulating_capturer) {
+  RTC_DCHECK(echo_emulating_capturer_);
+}
+
+bool EchoEmulatingRenderer::Render(rtc::ArrayView<const int16_t> data) {
+  if (data.size() > 0) {
+    echo_emulating_capturer_->OnAudioRendered(data);
+  }
+  return delegate_->Render(data);
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.h b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.h
new file mode 100644
index 0000000000..359a481e46
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_ECHO_ECHO_EMULATION_H_
+#define TEST_PC_E2E_ECHO_ECHO_EMULATION_H_
+
+#include <atomic>
+#include <deque>
+#include <memory>
+#include <vector>
+
+#include "api/test/pclf/media_configuration.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "rtc_base/swap_queue.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Reduces audio input strength from provided capturer twice and adds input
+// provided into EchoEmulatingCapturer::OnAudioRendered(...).
+class EchoEmulatingCapturer : public TestAudioDeviceModule::Capturer {
+ public:
+  EchoEmulatingCapturer(
+      std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+      EchoEmulationConfig config);
+
+  void OnAudioRendered(rtc::ArrayView<const int16_t> data);
+
+  int SamplingFrequency() const override {
+    return delegate_->SamplingFrequency();
+  }
+  int NumChannels() const override { return delegate_->NumChannels(); }
+  bool Capture(rtc::BufferT<int16_t>* buffer) override;
+
+ private:
+  std::unique_ptr<TestAudioDeviceModule::Capturer> delegate_;
+  const EchoEmulationConfig config_;
+
+  SwapQueue<std::vector<int16_t>> renderer_queue_;
+
+  SequenceChecker renderer_thread_;
+  std::vector<int16_t> queue_input_ RTC_GUARDED_BY(renderer_thread_);
+  bool recording_started_ RTC_GUARDED_BY(renderer_thread_) = false;
+
+  SequenceChecker capturer_thread_;
+  std::vector<int16_t> queue_output_ RTC_GUARDED_BY(capturer_thread_);
+  bool delay_accumulated_ RTC_GUARDED_BY(capturer_thread_) = false;
+};
+
+// Renders output into provided renderer and also copy output into provided
+// EchoEmulationCapturer.
+class EchoEmulatingRenderer : public TestAudioDeviceModule::Renderer {
+ public:
+  EchoEmulatingRenderer(
+      std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
+      EchoEmulatingCapturer* echo_emulating_capturer);
+
+  int SamplingFrequency() const override {
+    return delegate_->SamplingFrequency();
+  }
+  int NumChannels() const override { return delegate_->NumChannels(); }
+  bool Render(rtc::ArrayView<const int16_t> data) override;
+
+ private:
+  std::unique_ptr<TestAudioDeviceModule::Renderer> delegate_;
+  EchoEmulatingCapturer* echo_emulating_capturer_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_ECHO_ECHO_EMULATION_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/g3doc/architecture.md b/third_party/libwebrtc/test/pc/e2e/g3doc/architecture.md
new file mode 100644
index 0000000000..1b68c6db2c
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/architecture.md
@@ -0,0 +1,209 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'titovartem' reviewed: '2021-04-12'} *-->
+
+# PeerConnection level framework fixture architecture
+
+## Overview
+
+The main implementation of
+[`webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture`][1] is
+[`webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTest`][2]. Internally it owns
+the next main pieces:
+
+*   [`MediaHelper`][3] - responsible for adding audio and video tracks to the
+    peers.
+*   [`VideoQualityAnalyzerInjectionHelper`][4] and
+    [`SingleProcessEncodedImageDataInjector`][5] - used to inject video quality
+    analysis and properly match captured and rendered video frames. You can read
+    more about it in
+    [DefaultVideoQualityAnalyzer](default_video_quality_analyzer.md) section.
+*   [`AudioQualityAnalyzerInterface`][6] - used to measure audio quality metrics
+*   [`TestActivitiesExecutor`][7] - used to support [`ExecuteAt(...)`][8] and
+    [`ExecuteEvery(...)`][9] API of `PeerConnectionE2EQualityTestFixture` to run
+    any arbitrary action during test execution timely synchronized with a test
+    call.
+*   A vector of [`QualityMetricsReporter`][10] added by the
+    `PeerConnectionE2EQualityTestFixture` user.
+*   Two peers: Alice and Bob represented by instances of [`TestPeer`][11]
+    object.
+
+Also it keeps a reference to [`webrtc::TimeController`][12], which is used to
+create all required threads, task queues, task queue factories and time related
+objects.
+
+## TestPeer
+
+Call participants are represented by instances of `TestPeer` object.
+[`TestPeerFactory`][13] is used to create them. `TestPeer` owns all instances
+related to the `webrtc::PeerConnection`, including required listeners and
+callbacks. Also it provides an API to do offer/answer exchange and ICE candidate
+exchange. For this purposes internally it uses an instance of
+[`webrtc::PeerConnectionWrapper`][14].
+
+The `TestPeer` also owns the `PeerConnection` worker thread. The signaling
+thread for all `PeerConnection`'s is owned by
+`PeerConnectionE2EQualityTestFixture` and shared between all participants in the
+call. The network thread is owned by the network layer (it maybe either emulated
+network provided by [Network Emulation Framework][24] or network thread and
+`rtc::NetworkManager` provided by user) and provided when peer is added to the
+fixture via [`AddPeer(...)`][15] API.
+
+## GetStats API based metrics reporters
+
+`PeerConnectionE2EQualityTestFixture` gives the user ability to provide
+different `QualityMetricsReporter`s which will listen for `PeerConnection`
+[`GetStats`][16] API. Then such reporters will be able to report various metrics
+that user wants to measure.
+
+`PeerConnectionE2EQualityTestFixture` itself also uses this mechanism to
+measure:
+
+*   Audio quality metrics
+*   Audio/Video sync metrics (with help of [`CrossMediaMetricsReporter`][17])
+
+Also framework provides a [`StatsBasedNetworkQualityMetricsReporter`][18] to
+measure network related WebRTC metrics and print debug raw emulated network
+statistic. This reporter should be added by user via
+[`AddQualityMetricsReporter(...)`][19] API if requried.
+
+Internally stats gathering is done by [`StatsPoller`][20]. Stats are requested
+once per second for each `PeerConnection` and then resulted object is provided
+into each stats listener.
+
+## Offer/Answer exchange
+
+`PeerConnectionE2EQualityTest` provides ability to test Simulcast and SVC for
+video. These features aren't supported by P2P call and in general requires a
+Selective Forwarding Unit (SFU). So special logic is applied to mimic SFU
+behavior in P2P call. This logic is located inside [`SignalingInterceptor`][21],
+[`QualityAnalyzingVideoEncoder`][22] and [`QualityAnalyzingVideoDecoder`][23]
+and consist of SDP modification during offer/answer exchange and special
+handling of video frames from unrelated Simulcast/SVC streams during decoding.
+
+### Simulcast
+
+In case of Simulcast we have a video track, which internally contains multiple
+video streams, for example low resolution, medium resolution and high
+resolution. WebRTC client doesn't support receiving an offer with multiple
+streams in it, because usually SFU will keep only single stream for the client.
+To bypass it framework will modify offer by converting a single track with three
+video streams into three independent video tracks. Then sender will think that
+it send simulcast, but receiver will think that it receives 3 independent
+tracks.
+
+To achieve such behavior some extra tweaks are required:
+
+*   MID RTP header extension from original offer have to be removed
+*   RID RTP header extension from original offer is replaced with MID RTP header
+    extension, so the ID that sender uses for RID on receiver will be parsed as
+    MID.
+*   Answer have to be modified in the opposite way.
+
+Described modifications are illustrated on the picture below.
+
+![VP8 Simulcast offer modification](vp8_simulcast_offer_modification.png "VP8 Simulcast offer modification")
+
+The exchange will look like this:
+
+1.  Alice creates an offer
+2.  Alice sets offer as local description
+3.  Do described offer modification
+4.  Alice sends modified offer to Bob
+5.  Bob sets modified offer as remote description
+6.  Bob creates answer
+7.  Bob sets answer as local description
+8.  Do reverse modifications on answer
+9.  Bob sends modified answer to Alice
+10. Alice sets modified answer as remote description
+
+Such mechanism put a constraint that RTX streams are not supported, because they
+don't have RID RTP header extension in their packets.
+
+### SVC
+
+In case of SVC the framework will update the sender's offer before even setting
+it as local description on the sender side. Then no changes to answer will be
+required.
+
+`ssrc` is a 32 bit random value that is generated in RTP to denote a specific
+source used to send media in an RTP connection. In original offer video track
+section will look like this:
+
+```
+m=video 9 UDP/TLS/RTP/SAVPF 98 100 99 101
+...
+a=ssrc-group:FID <primary ssrc> <retransmission ssrc>
+a=ssrc:<primary ssrc> cname:...
+....
+a=ssrc:<retransmission ssrc> cname:...
+....
+```
+
+To enable SVC for such video track framework will add extra `ssrc`s for each SVC
+stream that is required like this:
+
+```
+a=ssrc-group:FID <Low resolution primary ssrc> <Low resolution retransmission ssrc>
+a=ssrc:<Low resolution primary ssrc> cname:...
+....
+a=ssrc:<Low resolution retransmission ssrc> cname:....
+...
+a=ssrc-group:FID <Medium resolution primary ssrc> <Medium resolution retransmission ssrc>
+a=ssrc:<Medium resolution primary ssrc> cname:...
+....
+a=ssrc:<Medium resolution retransmission ssrc> cname:....
+...
+a=ssrc-group:FID <High resolution primary ssrc> <High resolution retransmission ssrc>
+a=ssrc:<High resolution primary ssrc> cname:...
+....
+a=ssrc:<High resolution retransmission ssrc> cname:....
+...
+```
+
+The next line will also be added to the video track section of the offer:
+
+```
+a=ssrc-group:SIM <Low resolution primary ssrc> <Medium resolution primary ssrc> <High resolution primary ssrc>
+```
+
+It will tell PeerConnection that this track should be configured as SVC. It
+utilize WebRTC Plan B offer structure to achieve SVC behavior, also it modifies
+offer before setting it as local description which violates WebRTC standard.
+Also it adds limitations that on lossy networks only top resolution streams can
+be analyzed, because WebRTC won't try to restore low resolution streams in case
+of loss, because it still receives higher stream.
+
+### Handling in encoder/decoder
+
+In the encoder, the framework for each encoded video frame will propagate
+information requried for the fake SFU to know if it belongs to an interesting
+simulcast stream/spatial layer of if it should be "discarded".
+
+On the decoder side frames that should be "discarded" by fake SFU will be auto
+decoded into single pixel images and only the interesting simulcast
+stream/spatial layer will go into real decoder and then will be analyzed.
+
+[1]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;l=55;drc=484acf27231d931dbc99aedce85bc27e06486b96
+[2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/peer_connection_quality_test.h;l=44;drc=6cc893ad778a0965e2b7a8e614f3c98aa81bee5b
+[3]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/media/media_helper.h;l=27;drc=d46db9f1523ae45909b4a6fdc90a140443068bc6
+[4]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h;l=38;drc=79020414fd5c71f9ec1f25445ea5f1c8001e1a49
+[5]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h;l=40;drc=79020414fd5c71f9ec1f25445ea5f1c8001e1a49
+[6]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/audio_quality_analyzer_interface.h;l=23;drc=20f45823e37fd7272aa841831c029c21f29742c2
+[7]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/test_activities_executor.h;l=28;drc=6cc893ad778a0965e2b7a8e614f3c98aa81bee5b
+[8]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;l=439;drc=484acf27231d931dbc99aedce85bc27e06486b96
+[9]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;l=445;drc=484acf27231d931dbc99aedce85bc27e06486b96
+[10]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;l=413;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[11]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/test_activities_executor.h;l=28;drc=6cc893ad778a0965e2b7a8e614f3c98aa81bee5b
+[12]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/test_activities_executor.h;l=28;drc=6cc893ad778a0965e2b7a8e614f3c98aa81bee5b
+[13]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/test_peer_factory.h;l=46;drc=0ef4a2488a466a24ab97b31fdddde55440d451f9
+[14]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/pc/peer_connection_wrapper.h;l=47;drc=5ab79e62f691875a237ea28ca3975ea1f0ed62ec
+[15]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;l=459;drc=484acf27231d931dbc99aedce85bc27e06486b96
+[16]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/peer_connection_interface.h;l=886;drc=9438fb3fff97c803d1ead34c0e4f223db168526f
+[17]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/cross_media_metrics_reporter.h;l=29;drc=9d777620236ec76754cfce19f6e82dd18e52d22c
+[18]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/cross_media_metrics_reporter.h;l=29;drc=9d777620236ec76754cfce19f6e82dd18e52d22c
+[19]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;l=450;drc=484acf27231d931dbc99aedce85bc27e06486b96
+[20]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/stats_poller.h;l=52;drc=9b526180c9e9722d3fc7f8689da6ec094fc7fc0a
+[21]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/sdp/sdp_changer.h;l=79;drc=ee558dcca89fd8b105114ededf9e74d948da85e8
+[22]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h;l=54;drc=79020414fd5c71f9ec1f25445ea5f1c8001e1a49
+[23]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.h;l=50;drc=79020414fd5c71f9ec1f25445ea5f1c8001e1a49
+[24]: /test/network/g3doc/index.md
diff --git a/third_party/libwebrtc/test/pc/e2e/g3doc/default_video_quality_analyzer.md b/third_party/libwebrtc/test/pc/e2e/g3doc/default_video_quality_analyzer.md
new file mode 100644
index 0000000000..67596777f2
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/default_video_quality_analyzer.md
@@ -0,0 +1,197 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'titovartem' reviewed: '2021-02-21'} *-->
+
+# DefaultVideoQualityAnalyzer
+
+## Audience
+
+This document is for users of
+[`webrtc::webrtc_pc_e2e::DefaultVideoQualityAnalyzer`][1].
+
+## Overview
+
+`DefaultVideoQualityAnalyzer` implements
+[`webrtc::VideoQualityAnalyzerInterface`][2] and is a main
+implementation of video quality analyzer for WebRTC. To operate correctly it
+requires to receive video frame on each step:
+
+1.  On frame captured - analyzer will generate a unique ID for the frame, that
+    caller should attach to the it.
+2.  Immediately before frame enter the encoder.
+3.  Immediately after the frame was encoded.
+4.  After the frame was received and immediately before it entered the decoder.
+5.  Immediately after the frame was decoded.
+6.  When the frame was rendered.
+
+![VideoQualityAnalyzerInterface pipeline](video_quality_analyzer_pipeline.png "VideoQualityAnalyzerInterface pipeline")
+
+The analyzer updates its internal metrics per frame when it was rendered and
+reports all of them after it was stopped through
+[WebRTC perf results reporting system][10].
+
+To properly inject `DefaultVideoQualityAnalyzer` into pipeline the following helpers can be used:
+
+### VideoQualityAnalyzerInjectionHelper
+
+[`webrtc::webrtc_pc_e2e::VideoQualityAnalyzerInjectionHelper`][3] provides
+factory methods for components, that will be used to inject
+`VideoQualityAnalyzerInterface` into the `PeerConnection` pipeline:
+
+*   Wrappers for [`webrtc::VideoEncoderFactory`][4] and
+    [`webrtc::VideoDecodeFactory`][5] which will properly pass
+    [`webrtc::VideoFrame`][6] and [`webrtc::EncodedImage`][7] into analyzer
+    before and after real video encode and decoder.
+*   [`webrtc::test::TestVideoCapturer::FramePreprocessor`][8] which is used to
+    pass generated frames into analyzer on capturing and then set the returned
+    frame ID. It also configures dumping of captured frames if requried.
+*   [`rtc::VideoSinkInterface<VideoFrame>`][9] which is used to pass frames to
+    the analyzer before they will be rendered to compute per frame metrics. It
+    also configures dumping of rendered video if requried.
+
+Besides factories `VideoQualityAnalyzerInjectionHelper` has method to
+orchestrate `VideoQualityAnalyzerInterface` workflow:
+
+*   `Start` - to start video analyzer, so it will be able to receive and analyze
+    video frames.
+*   `RegisterParticipantInCall` - to add new participants after analyzer was
+    started.
+*   `Stop` - to stop analyzer, compute all metrics for frames that were recevied
+    before and report them.
+
+Also `VideoQualityAnalyzerInjectionHelper` implements
+[`webrtc::webrtc_pc_e2e::StatsObserverInterface`][11] to propagate WebRTC stats
+to `VideoQualityAnalyzerInterface`.
+
+### EncodedImageDataInjector and EncodedImageDataExtractor
+
+[`webrtc::webrtc_pc_e2e::EncodedImageDataInjector`][14] and
+[`webrtc::webrtc_pc_e2e::EncodedImageDataInjector`][15] are used to inject and
+extract data into `webrtc::EncodedImage` to propagate frame ID and other
+required information through the network.
+
+By default [`webrtc::webrtc_pc_e2e::SingleProcessEncodedImageDataInjector`][16]
+is used. It assumes `webrtc::EncodedImage` payload as black box which is
+remaining unchanged from encoder to decoder and stores the information required
+for its work in the last 3 bytes of the payload, replacing the original data
+during injection and restoring it back during extraction. Also
+`SingleProcessEncodedImageDataInjector` requires that sender and receiver were
+inside single process.
+
+![SingleProcessEncodedImageDataInjector](single_process_encoded_image_data_injector.png "SingleProcessEncodedImageDataInjector")
+
+## Exported metrics
+
+Exported metrics are reported to WebRTC perf results reporting system.
+
+### General
+
+*   *`cpu_usage`* - CPU usage excluding video analyzer
+
+### Video
+
+*   *`psnr`* - peak signal-to-noise ratio:
+    [wikipedia](https://en.wikipedia.org/wiki/Peak_signal-to-noise_ratio)
+*   *`ssim`* - structural similarity:
+    [wikipedia](https://en.wikipedia.org/wiki/Structural_similarity).
+*   *`min_psnr`* - minimum value of psnr across all frames of video stream.
+*   *`encode_time`* - time to encode a single frame.
+*   *`decode_time`* - time to decode a single frame.
+*   *`transport_time`* - time from frame encoded to frame received for decoding.
+*   *`receive_to_render_time`* - time from frame received for decoding to frame
+    rendered.
+*   *`total_delay_incl_transport`* - time from frame was captured on device to
+    time when frame was displayed on device.
+*   *`encode_frame_rate`* - frame rate after encoder.
+*   *`harmonic_framerate`* - video duration divided on squared sum of interframe
+    delays. Reflects render frame rate penalized by freezes.
+*   *`time_between_rendered_frames`* - time between frames out to renderer.
+*   *`dropped_frames`* - amount of frames that were sent, but weren't rendered
+    and are known not to be “on the way� from sender to receiver.
+
+Freeze is a pause when no new frames from decoder arrived for 150ms + avg time
+between frames or 3 * avg time between frames.
+
+*   *`time_between_freezes`* - mean time from previous freeze end to new freeze
+    start.
+*   *`freeze_time_ms`* - total freeze time in ms.
+*   *`max_skipped`* - frames skipped between two nearest rendered.
+*   *`pixels_per_frame`* - amount of pixels on frame (width * height).
+*   *`target_encode_bitrate`* - target encode bitrate provided by BWE to
+    encoder.
+*   *`actual_encode_bitrate -`* - actual encode bitrate produced by encoder.
+*   *`available_send_bandwidth -`* - available send bandwidth estimated by BWE.
+*   *`transmission_bitrate`* - bitrate of media in the emulated network, not
+    counting retransmissions FEC, and RTCP messages
+*   *`retransmission_bitrate`* - bitrate of retransmission streams only.
+
+### Framework stability
+
+*   *`frames_in_flight`* - amount of frames that were captured but wasn't seen
+    on receiver.
+
+## Debug metrics
+
+Debug metrics are not reported to WebRTC perf results reporting system, but are
+available through `DefaultVideoQualityAnalyzer` API.
+
+### [FrameCounters][12]
+
+Frame counters consist of next counters:
+
+*   *`captured`* - count of frames, that were passed into WebRTC pipeline by
+    video stream source
+*   *`pre_encoded`* - count of frames that reached video encoder.
+*   *`encoded`* - count of encoded images that were produced by encoder for all
+    requested spatial layers and simulcast streams.
+*   *`received`* - count of encoded images received in decoder for all requested
+    spatial layers and simulcast streams.
+*   *`decoded`* - count of frames that were produced by decoder.
+*   *`rendered`* - count of frames that went out from WebRTC pipeline to video
+    sink.
+*   *`dropped`* - count of frames that were dropped in any point between
+    capturing and rendering.
+
+`DefaultVideoQualityAnalyzer` exports these frame counters:
+
+*   *`GlobalCounters`* - frame counters for frames met on each stage of analysis
+    for all media streams.
+*   *`PerStreamCounters`* - frame counters for frames met on each stage of
+    analysis separated per individual video track (single media section in the
+    SDP offer).
+
+### [AnalyzerStats][13]
+
+Contains metrics about internal state of video analyzer during its work
+
+*   *`comparisons_queue_size`* - size of analyzer internal queue used to perform
+    captured and rendered frames comparisons measured when new element is added
+    to the queue.
+*   *`comparisons_done`* - number of performed comparisons of 2 video frames
+    from captured and rendered streams.
+*   *`cpu_overloaded_comparisons_done`* - number of cpu overloaded comparisons.
+    Comparison is cpu overloaded if it is queued when there are too many not
+    processed comparisons in the queue. Overloaded comparison doesn't include
+    metrics like SSIM and PSNR that require heavy computations.
+*   *`memory_overloaded_comparisons_done`* - number of memory overloaded
+    comparisons. Comparison is memory overloaded if it is queued when its
+    captured frame was already removed due to high memory usage for that video
+    stream.
+*   *`frames_in_flight_left_count`* - count of frames in flight in analyzer
+    measured when new comparison is added and after analyzer was stopped.
+
+[1]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h;l=188;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[2]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/test/video_quality_analyzer_interface.h;l=56;drc=d7808f1c464a07c8f1e2f97ec7ee92fda998d590
+[3]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h;l=39;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[4]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/video_codecs/video_encoder_factory.h;l=27;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[5]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/video_codecs/video_decoder_factory.h;l=27;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[6]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/video/video_frame.h;l=30;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[7]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/video/encoded_image.h;l=71;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[8]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/test_video_capturer.h;l=28;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[9]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/video/video_sink_interface.h;l=19;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[10]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/testsupport/perf_test.h;drc=0710b401b1e5b500b8e84946fb657656ba1b58b7
+[11]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/test/stats_observer_interface.h;l=21;drc=9b526180c9e9722d3fc7f8689da6ec094fc7fc0a
+[12]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h;l=57;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[13]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h;l=113;drc=08f46909a8735cf181b99ef2f7e1791c5a7531d2
+[14]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/pc/e2e/analyzer/video/encoded_image_data_injector.h;l=23;drc=c57089a97a3df454f4356d882cc8df173e8b3ead
+[15]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/pc/e2e/analyzer/video/encoded_image_data_injector.h;l=46;drc=c57089a97a3df454f4356d882cc8df173e8b3ead
+[16]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h;l=40;drc=c57089a97a3df454f4356d882cc8df173e8b3ead
diff --git a/third_party/libwebrtc/test/pc/e2e/g3doc/in_test_psnr_plot.png b/third_party/libwebrtc/test/pc/e2e/g3doc/in_test_psnr_plot.png
new file mode 100644
index 0000000000..3f36725727
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/in_test_psnr_plot.png
diff --git a/third_party/libwebrtc/test/pc/e2e/g3doc/index.md b/third_party/libwebrtc/test/pc/e2e/g3doc/index.md
new file mode 100644
index 0000000000..678262bb2b
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/index.md
@@ -0,0 +1,224 @@
+<!-- go/cmark -->
+<!--* freshness: {owner: 'titovartem' reviewed: '2021-04-12'} *-->
+
+# PeerConnection Level Framework
+
+## API
+
+*   [Fixture][1]
+*   [Fixture factory function][2]
+
+## Documentation
+
+The PeerConnection level framework is designed for end-to-end media quality
+testing through the PeerConnection level public API. The framework uses the
+*Unified plan* API to generate offers/answers during the signaling phase. The
+framework also wraps the video encoder/decoder and inject it into
+*`webrtc::PeerConnection`* to measure video quality, performing 1:1 frames
+matching between captured and rendered frames without any extra requirements to
+input video. For audio quality evaluation the standard `GetStats()` API from
+PeerConnection is used.
+
+The framework API is located in the namespace *`webrtc::webrtc_pc_e2e`*.
+
+### Supported features
+
+*   Single or bidirectional media in the call
+*   RTC Event log dump per peer
+*   AEC dump per peer
+*   Compatible with *`webrtc::TimeController`* for both real and simulated time
+*   Media
+    *   AV sync
+*   Video
+    *   Any amount of video tracks both from caller and callee sides
+    *   Input video from
+        *   Video generator
+        *   Specified file
+        *   Any instance of *`webrtc::test::FrameGeneratorInterface`*
+    *   Dumping of captured/rendered video into file
+    *   Screen sharing
+    *   Vp8 simulcast from caller side
+    *   Vp9 SVC from caller side
+    *   Choosing of video codec (name and parameters), having multiple codecs
+        negotiated to support codec-switching testing.
+    *   FEC (ULP or Flex)
+    *   Forced codec overshooting (for encoder overshoot emulation on some
+        mobile devices, when hardware encoder can overshoot target bitrate)
+*   Audio
+    *   Up to 1 audio track both from caller and callee sides
+    *   Generated audio
+    *   Audio from specified file
+    *   Dumping of captured/rendered audio into file
+    *   Parameterizing of `cricket::AudioOptions`
+    *   Echo emulation
+*   Injection of various WebRTC components into underlying
+    *`webrtc::PeerConnection`* or *`webrtc::PeerConnectionFactory`*. You can see
+    the full list [here][11]
+*   Scheduling of events, that can happen during the test, for example:
+    *   Changes in network configuration
+    *   User statistics measurements
+    *   Custom defined actions
+*   User defined statistics reporting via
+    *`webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::QualityMetricsReporter`*
+    interface
+
+## Exported metrics
+
+### General
+
+*   *`<peer_name>_connected`* - peer successfully established connection to
+    remote side
+*   *`cpu_usage`* - CPU usage excluding video analyzer
+*   *`audio_ahead_ms`* - Used to estimate how much audio and video is out of
+    sync when the two tracks were from the same source. Stats are polled
+    periodically during a call. The metric represents how much earlier was audio
+    played out on average over the call. If, during a stats poll, video is
+    ahead, then audio_ahead_ms will be equal to 0 for this poll.
+*   *`video_ahead_ms`* - Used to estimate how much audio and video is out of
+    sync when the two tracks were from the same source. Stats are polled
+    periodically during a call. The metric represents how much earlier was video
+    played out on average over the call. If, during a stats poll, audio is
+    ahead, then video_ahead_ms will be equal to 0 for this poll.
+
+### Video
+
+See documentation for
+[*`DefaultVideoQualityAnalyzer`*](default_video_quality_analyzer.md#exported-metrics)
+
+### Audio
+
+*   *`accelerate_rate`* - when playout is sped up, this counter is increased by
+    the difference between the number of samples received and the number of
+    samples played out. If speedup is achieved by removing samples, this will be
+    the count of samples removed. Rate is calculated as difference between
+    nearby samples divided on sample interval.
+*   *`expand_rate`* - the total number of samples that are concealed samples
+    over time. A concealed sample is a sample that was replaced with synthesized
+    samples generated locally before being played out. Examples of samples that
+    have to be concealed are samples from lost packets or samples from packets
+    that arrive too late to be played out
+*   *`speech_expand_rate`* - the total number of samples that are concealed
+    samples minus the total number of concealed samples inserted that are
+    "silent" over time. Playing out silent samples results in silence or comfort
+    noise.
+*   *`preemptive_rate`* - when playout is slowed down, this counter is increased
+    by the difference between the number of samples received and the number of
+    samples played out. If playout is slowed down by inserting samples, this
+    will be the number of inserted samples. Rate is calculated as difference
+    between nearby samples divided on sample interval.
+*   *`average_jitter_buffer_delay_ms`* - average size of NetEQ jitter buffer.
+*   *`preferred_buffer_size_ms`* - preferred size of NetEQ jitter buffer.
+*   *`visqol_mos`* - proxy for audio quality itself.
+*   *`asdm_samples`* - measure of how much acceleration/deceleration was in the
+    signal.
+*   *`word_error_rate`* - measure of how intelligible the audio was (percent of
+    words that could not be recognized in output audio).
+
+### Network
+
+*   *`bytes_sent`* - represents the total number of payload bytes sent on this
+    PeerConnection, i.e., not including headers or padding
+*   *`packets_sent`* - represents the total number of packets sent over this
+    PeerConnection’s transports.
+*   *`average_send_rate`* - average send rate calculated on bytes_sent divided
+    by test duration.
+*   *`payload_bytes_sent`* - total number of bytes sent for all SSRC plus total
+    number of RTP header and padding bytes sent for all SSRC. This does not
+    include the size of transport layer headers such as IP or UDP.
+*   *`sent_packets_loss`* - packets_sent minus corresponding packets_received.
+*   *`bytes_received`* - represents the total number of bytes received on this
+    PeerConnection, i.e., not including headers or padding.
+*   *`packets_received`* - represents the total number of packets received on
+    this PeerConnection’s transports.
+*   *`average_receive_rate`* - average receive rate calculated on bytes_received
+    divided by test duration.
+*   *`payload_bytes_received`* - total number of bytes received for all SSRC
+    plus total number of RTP header and padding bytes received for all SSRC.
+    This does not include the size of transport layer headers such as IP or UDP.
+
+### Framework stability
+
+*   *`frames_in_flight`* - amount of frames that were captured but wasn't seen
+    on receiver in the way that also all frames after also weren't seen on
+    receiver.
+*   *`bytes_discarded_no_receiver`* - total number of bytes that were received
+    on network interfaces related to the peer, but destination port was closed.
+*   *`packets_discarded_no_receiver`* - total number of packets that were
+    received on network interfaces related to the peer, but destination port was
+    closed.
+
+## Examples
+
+Examples can be found in
+
+*   [peer_connection_e2e_smoke_test.cc][3]
+*   [pc_full_stack_tests.cc][4]
+
+## Stats plotting
+
+### Description
+
+Stats plotting provides ability to plot statistic collected during the test.
+Right now it is used in PeerConnection level framework and give ability to see
+how video quality metrics changed during test execution.
+
+### Usage
+
+To make any metrics plottable you need:
+
+1.  Collect metric data with [SamplesStatsCounter][5] which internally will
+    store all intermediate points and timestamps when these points were added.
+2.  Then you need to report collected data with
+    [`webrtc::test::PrintResult(...)`][6]. By using these method you will also
+    specify name of the plottable metric.
+
+After these steps it will be possible to export your metric for plotting. There
+are several options how you can do this:
+
+1.  Use [`webrtc::TestMain::Create()`][7] as `main` function implementation, for
+    example use [`test/test_main.cc`][8] as `main` function for your test.
+
+    In such case your binary will have flag `--plot`, where you can provide a
+    list of metrics, that you want to plot or specify `all` to plot all
+    available metrics.
+
+    If `--plot` is specified, the binary will output metrics data into `stdout`.
+    Then you need to pipe this `stdout` into python plotter script
+    [`rtc_tools/metrics_plotter.py`][9], which will plot data.
+
+    Examples:
+
+    ```shell
+    $ ./out/Default/test_support_unittests \
+      --gtest_filter=PeerConnectionE2EQualityTestSmokeTest.Svc \
+      --nologs \
+      --plot=all \
+    | python rtc_tools/metrics_plotter.py
+    ```
+
+    ```shell
+    $ ./out/Default/test_support_unittests \
+      --gtest_filter=PeerConnectionE2EQualityTestSmokeTest.Svc \
+      --nologs \
+      --plot=psnr,ssim \
+    | python rtc_tools/metrics_plotter.py
+    ```
+
+    Example chart: ![PSNR changes during the test](in_test_psnr_plot.png)
+
+2.  Use API from [`test/testsupport/perf_test.h`][10] directly by invoking
+    `webrtc::test::PrintPlottableResults(const std::vector<std::string>&
+    desired_graphs)` to print plottable metrics to stdout. Then as in previous
+    option you need to pipe result into plotter script.
+
+[1]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;drc=cbe6e8a2589a925d4c91a2ac2c69201f03de9c39
+[2]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/create_peerconnection_quality_test_fixture.h;drc=cbe6e8a2589a925d4c91a2ac2c69201f03de9c39
+[3]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/pc/e2e/peer_connection_e2e_smoke_test.cc;drc=cbe6e8a2589a925d4c91a2ac2c69201f03de9c39
+[4]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/pc_full_stack_tests.cc;drc=cbe6e8a2589a925d4c91a2ac2c69201f03de9c39
+[5]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/numerics/samples_stats_counter.h;drc=cbe6e8a2589a925d4c91a2ac2c69201f03de9c39
+[6]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/testsupport/perf_test.h;l=86;drc=0710b401b1e5b500b8e84946fb657656ba1b58b7
+[7]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/test_main_lib.h;l=23;drc=bcb42f1e4be136c390986a40d9d5cb3ad0de260b
+[8]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/test_main.cc;drc=bcb42f1e4be136c390986a40d9d5cb3ad0de260b
+[9]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/rtc_tools/metrics_plotter.py;drc=8cc6695652307929edfc877cd64b75cd9ec2d615
+[10]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/test/testsupport/perf_test.h;l=105;drc=0710b401b1e5b500b8e84946fb657656ba1b58b7
+[11]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/test/peerconnection_quality_test_fixture.h;l=272;drc=484acf27231d931dbc99aedce85bc27e06486b96
diff --git a/third_party/libwebrtc/test/pc/e2e/g3doc/single_process_encoded_image_data_injector.png b/third_party/libwebrtc/test/pc/e2e/g3doc/single_process_encoded_image_data_injector.png
new file mode 100644
index 0000000000..73480bafbe
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/single_process_encoded_image_data_injector.png
diff --git a/third_party/libwebrtc/test/pc/e2e/g3doc/video_quality_analyzer_pipeline.png b/third_party/libwebrtc/test/pc/e2e/g3doc/video_quality_analyzer_pipeline.png
new file mode 100644
index 0000000000..6cddb91110
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/video_quality_analyzer_pipeline.png
diff --git a/third_party/libwebrtc/test/pc/e2e/g3doc/vp8_simulcast_offer_modification.png b/third_party/libwebrtc/test/pc/e2e/g3doc/vp8_simulcast_offer_modification.png
new file mode 100644
index 0000000000..c7eaa04c0e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/vp8_simulcast_offer_modification.png
diff --git a/third_party/libwebrtc/test/pc/e2e/media/media_helper.cc b/third_party/libwebrtc/test/pc/e2e/media/media_helper.cc
new file mode 100644
index 0000000000..e945bd4dae
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/media/media_helper.cc
@@ -0,0 +1,128 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/media/media_helper.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/types/variant.h"
+#include "api/media_stream_interface.h"
+#include "api/test/create_frame_generator.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "test/frame_generator_capturer.h"
+#include "test/platform_video_capturer.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+void MediaHelper::MaybeAddAudio(TestPeer* peer) {
+  if (!peer->params().audio_config) {
+    return;
+  }
+  const AudioConfig& audio_config = peer->params().audio_config.value();
+  rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+      peer->pc_factory()->CreateAudioSource(audio_config.audio_options);
+  rtc::scoped_refptr<AudioTrackInterface> track =
+      peer->pc_factory()->CreateAudioTrack(*audio_config.stream_label,
+                                           source.get());
+  std::string sync_group = audio_config.sync_group
+                               ? audio_config.sync_group.value()
+                               : audio_config.stream_label.value() + "-sync";
+  peer->AddTrack(track, {sync_group, *audio_config.stream_label});
+}
+
+std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>>
+MediaHelper::MaybeAddVideo(TestPeer* peer) {
+  // Params here valid because of pre-run validation.
+  const Params& params = peer->params();
+  const ConfigurableParams& configurable_params = peer->configurable_params();
+  std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>> out;
+  for (size_t i = 0; i < configurable_params.video_configs.size(); ++i) {
+    const VideoConfig& video_config = configurable_params.video_configs[i];
+    // Setup input video source into peer connection.
+    std::unique_ptr<test::TestVideoCapturer> capturer = CreateVideoCapturer(
+        video_config, peer->ReleaseVideoSource(i),
+        video_quality_analyzer_injection_helper_->CreateFramePreprocessor(
+            params.name.value(), video_config));
+    bool is_screencast =
+        video_config.content_hint == VideoTrackInterface::ContentHint::kText ||
+        video_config.content_hint ==
+            VideoTrackInterface::ContentHint::kDetailed;
+    rtc::scoped_refptr<TestVideoCapturerVideoTrackSource> source =
+        rtc::make_ref_counted<TestVideoCapturerVideoTrackSource>(
+            std::move(capturer), is_screencast);
+    out.push_back(source);
+    RTC_LOG(LS_INFO) << "Adding video with video_config.stream_label="
+                     << video_config.stream_label.value();
+    rtc::scoped_refptr<VideoTrackInterface> track =
+        peer->pc_factory()->CreateVideoTrack(video_config.stream_label.value(),
+                                             source.get());
+    if (video_config.content_hint.has_value()) {
+      track->set_content_hint(video_config.content_hint.value());
+    }
+    std::string sync_group = video_config.sync_group
+                                 ? video_config.sync_group.value()
+                                 : video_config.stream_label.value() + "-sync";
+    RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> sender =
+        peer->AddTrack(track, {sync_group, *video_config.stream_label});
+    RTC_CHECK(sender.ok());
+    if (video_config.temporal_layers_count ||
+        video_config.degradation_preference) {
+      RtpParameters rtp_parameters = sender.value()->GetParameters();
+      if (video_config.temporal_layers_count) {
+        for (auto& encoding_parameters : rtp_parameters.encodings) {
+          encoding_parameters.num_temporal_layers =
+              video_config.temporal_layers_count;
+        }
+      }
+      if (video_config.degradation_preference) {
+        rtp_parameters.degradation_preference =
+            video_config.degradation_preference;
+      }
+      RTCError res = sender.value()->SetParameters(rtp_parameters);
+      RTC_CHECK(res.ok()) << "Failed to set RTP parameters";
+    }
+  }
+  return out;
+}
+
+std::unique_ptr<test::TestVideoCapturer> MediaHelper::CreateVideoCapturer(
+    const VideoConfig& video_config,
+    PeerConfigurer::VideoSource source,
+    std::unique_ptr<test::TestVideoCapturer::FramePreprocessor>
+        frame_preprocessor) {
+  CapturingDeviceIndex* capturing_device_index =
+      absl::get_if<CapturingDeviceIndex>(&source);
+  if (capturing_device_index != nullptr) {
+    std::unique_ptr<test::TestVideoCapturer> capturer =
+        test::CreateVideoCapturer(video_config.width, video_config.height,
+                                  video_config.fps,
+                                  static_cast<size_t>(*capturing_device_index));
+    RTC_CHECK(capturer)
+        << "Failed to obtain input stream from capturing device #"
+        << *capturing_device_index;
+    capturer->SetFramePreprocessor(std::move(frame_preprocessor));
+    return capturer;
+  }
+
+  auto capturer = std::make_unique<test::FrameGeneratorCapturer>(
+      clock_,
+      absl::get<std::unique_ptr<test::FrameGeneratorInterface>>(
+          std::move(source)),
+      video_config.fps, *task_queue_factory_);
+  capturer->SetFramePreprocessor(std::move(frame_preprocessor));
+  capturer->Init();
+  return capturer;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/media/media_helper.h b/third_party/libwebrtc/test/pc/e2e/media/media_helper.h
new file mode 100644
index 0000000000..2d163d009e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/media/media_helper.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_MEDIA_MEDIA_HELPER_H_
+#define TEST_PC_E2E_MEDIA_MEDIA_HELPER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/test/frame_generator_interface.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h"
+#include "test/pc/e2e/media/test_video_capturer_video_track_source.h"
+#include "test/pc/e2e/test_peer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class MediaHelper {
+ public:
+  MediaHelper(VideoQualityAnalyzerInjectionHelper*
+                  video_quality_analyzer_injection_helper,
+              TaskQueueFactory* task_queue_factory,
+              Clock* clock)
+      : clock_(clock),
+        task_queue_factory_(task_queue_factory),
+        video_quality_analyzer_injection_helper_(
+            video_quality_analyzer_injection_helper) {}
+
+  void MaybeAddAudio(TestPeer* peer);
+
+  std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>>
+  MaybeAddVideo(TestPeer* peer);
+
+ private:
+  std::unique_ptr<test::TestVideoCapturer> CreateVideoCapturer(
+      const VideoConfig& video_config,
+      PeerConfigurer::VideoSource source,
+      std::unique_ptr<test::TestVideoCapturer::FramePreprocessor>
+          frame_preprocessor);
+
+  Clock* const clock_;
+  TaskQueueFactory* const task_queue_factory_;
+  VideoQualityAnalyzerInjectionHelper* video_quality_analyzer_injection_helper_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_MEDIA_MEDIA_HELPER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/media/test_video_capturer_video_track_source.h b/third_party/libwebrtc/test/pc/e2e/media/test_video_capturer_video_track_source.h
new file mode 100644
index 0000000000..c883a2e8e9
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/media/test_video_capturer_video_track_source.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_MEDIA_TEST_VIDEO_CAPTURER_VIDEO_TRACK_SOURCE_H_
+#define TEST_PC_E2E_MEDIA_TEST_VIDEO_CAPTURER_VIDEO_TRACK_SOURCE_H_
+
+#include <memory>
+#include <utility>
+
+#include "api/video/video_frame.h"
+#include "api/video/video_source_interface.h"
+#include "pc/video_track_source.h"
+#include "test/test_video_capturer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class TestVideoCapturerVideoTrackSource : public VideoTrackSource {
+ public:
+  TestVideoCapturerVideoTrackSource(
+      std::unique_ptr<test::TestVideoCapturer> video_capturer,
+      bool is_screencast)
+      : VideoTrackSource(/*remote=*/false),
+        video_capturer_(std::move(video_capturer)),
+        is_screencast_(is_screencast) {}
+
+  ~TestVideoCapturerVideoTrackSource() = default;
+
+  void Start() { SetState(kLive); }
+
+  void Stop() { SetState(kMuted); }
+
+  bool is_screencast() const override { return is_screencast_; }
+
+ protected:
+  rtc::VideoSourceInterface<VideoFrame>* source() override {
+    return video_capturer_.get();
+  }
+
+ private:
+  std::unique_ptr<test::TestVideoCapturer> video_capturer_;
+  const bool is_screencast_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_MEDIA_TEST_VIDEO_CAPTURER_VIDEO_TRACK_SOURCE_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/metric_metadata_keys.h b/third_party/libwebrtc/test/pc/e2e/metric_metadata_keys.h
new file mode 100644
index 0000000000..fbcd3b90fe
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/metric_metadata_keys.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PC_E2E_METRIC_METADATA_KEYS_H_
+#define TEST_PC_E2E_METRIC_METADATA_KEYS_H_
+
+#include <string>
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// All metadata fields are present only if applicable for particular metric.
+class MetricMetadataKey {
+ public:
+  // Represents on peer with whom the metric is associated.
+  static constexpr char kPeerMetadataKey[] = "peer";
+  // Represents sender of the media stream.
+  static constexpr char kSenderMetadataKey[] = "sender";
+  // Represents receiver of the media stream.
+  static constexpr char kReceiverMetadataKey[] = "receiver";
+  // Represents name of the audio stream.
+  static constexpr char kAudioStreamMetadataKey[] = "audio_stream";
+  // Represents name of the video stream.
+  static constexpr char kVideoStreamMetadataKey[] = "video_stream";
+  // Represents name of the sync group to which stream belongs.
+  static constexpr char kPeerSyncGroupMetadataKey[] = "peer_sync_group";
+  // Represents the test name (without any peer and stream data appended to it
+  // as it currently happens with the webrtc.test_metrics.Metric.test_case
+  // field). This metadata is temporary and it will be removed once this
+  // information is moved to webrtc.test_metrics.Metric.test_case.
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  static constexpr char kExperimentalTestNameMetadataKey[] =
+      "experimental_test_name";
+  // Represents index of a video spatial layer to which metric belongs.
+  static constexpr char kSpatialLayerMetadataKey[] = "spatial_layer";
+
+ private:
+  MetricMetadataKey() = default;
+};
+
+// All metadata fields are presented only if applicable for particular metric.
+class SampleMetadataKey {
+ public:
+  // Represents a frame ID with which data point is associated.
+  static constexpr char kFrameIdMetadataKey[] = "frame_id";
+
+ private:
+  SampleMetadataKey() = default;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_METRIC_METADATA_KEYS_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
new file mode 100644
index 0000000000..0bb28f0847
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
@@ -0,0 +1,183 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/network_quality_metrics_reporter.h"
+
+#include <utility>
+
+#include "api/stats/rtc_stats.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/test/metrics/metric.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1);
+
+// Field trial which controls whether to report standard-compliant bytes
+// sent/received per stream.  If enabled, padding and headers are not included
+// in bytes sent or received.
+constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
+
+}  // namespace
+
+NetworkQualityMetricsReporter::NetworkQualityMetricsReporter(
+    EmulatedNetworkManagerInterface* alice_network,
+    EmulatedNetworkManagerInterface* bob_network,
+    test::MetricsLogger* metrics_logger)
+    : alice_network_(alice_network),
+      bob_network_(bob_network),
+      metrics_logger_(metrics_logger) {
+  RTC_CHECK(metrics_logger_);
+}
+
+void NetworkQualityMetricsReporter::Start(
+    absl::string_view test_case_name,
+    const TrackIdStreamInfoMap* /*reporter_helper*/) {
+  test_case_name_ = std::string(test_case_name);
+  // Check that network stats are clean before test execution.
+  EmulatedNetworkStats alice_stats = PopulateStats(alice_network_);
+  RTC_CHECK_EQ(alice_stats.overall_outgoing_stats.packets_sent, 0);
+  RTC_CHECK_EQ(alice_stats.overall_incoming_stats.packets_received, 0);
+  EmulatedNetworkStats bob_stats = PopulateStats(bob_network_);
+  RTC_CHECK_EQ(bob_stats.overall_outgoing_stats.packets_sent, 0);
+  RTC_CHECK_EQ(bob_stats.overall_incoming_stats.packets_received, 0);
+}
+
+void NetworkQualityMetricsReporter::OnStatsReports(
+    absl::string_view pc_label,
+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
+  DataSize payload_received = DataSize::Zero();
+  DataSize payload_sent = DataSize::Zero();
+
+  auto inbound_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
+  for (const auto& stat : inbound_stats) {
+    payload_received +=
+        DataSize::Bytes(stat->bytes_received.ValueOrDefault(0ul) +
+                        stat->header_bytes_received.ValueOrDefault(0ul));
+  }
+
+  auto outbound_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
+  for (const auto& stat : outbound_stats) {
+    payload_sent +=
+        DataSize::Bytes(stat->bytes_sent.ValueOrDefault(0ul) +
+                        stat->header_bytes_sent.ValueOrDefault(0ul));
+  }
+
+  MutexLock lock(&lock_);
+  PCStats& stats = pc_stats_[std::string(pc_label)];
+  stats.payload_received = payload_received;
+  stats.payload_sent = payload_sent;
+}
+
+void NetworkQualityMetricsReporter::StopAndReportResults() {
+  EmulatedNetworkStats alice_stats = PopulateStats(alice_network_);
+  EmulatedNetworkStats bob_stats = PopulateStats(bob_network_);
+  int64_t alice_packets_loss =
+      alice_stats.overall_outgoing_stats.packets_sent -
+      bob_stats.overall_incoming_stats.packets_received;
+  int64_t bob_packets_loss =
+      bob_stats.overall_outgoing_stats.packets_sent -
+      alice_stats.overall_incoming_stats.packets_received;
+  ReportStats("alice", alice_stats, alice_packets_loss);
+  ReportStats("bob", bob_stats, bob_packets_loss);
+
+  if (!webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
+    RTC_LOG(LS_ERROR)
+        << "Non-standard GetStats; \"payload\" counts include RTP headers";
+  }
+
+  MutexLock lock(&lock_);
+  for (const auto& pair : pc_stats_) {
+    ReportPCStats(pair.first, pair.second);
+  }
+}
+
+EmulatedNetworkStats NetworkQualityMetricsReporter::PopulateStats(
+    EmulatedNetworkManagerInterface* network) {
+  rtc::Event wait;
+  EmulatedNetworkStats stats;
+  network->GetStats([&](EmulatedNetworkStats s) {
+    stats = std::move(s);
+    wait.Set();
+  });
+  bool stats_received = wait.Wait(kStatsWaitTimeout);
+  RTC_CHECK(stats_received);
+  return stats;
+}
+
+void NetworkQualityMetricsReporter::ReportStats(
+    const std::string& network_label,
+    const EmulatedNetworkStats& stats,
+    int64_t packet_loss) {
+  metrics_logger_->LogSingleValueMetric(
+      "bytes_sent", GetTestCaseName(network_label),
+      stats.overall_outgoing_stats.bytes_sent.bytes(), Unit::kBytes,
+      ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "packets_sent", GetTestCaseName(network_label),
+      stats.overall_outgoing_stats.packets_sent, Unit::kUnitless,
+      ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "average_send_rate", GetTestCaseName(network_label),
+      stats.overall_outgoing_stats.packets_sent >= 2
+          ? stats.overall_outgoing_stats.AverageSendRate().kbps<double>()
+          : 0,
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "bytes_discarded_no_receiver", GetTestCaseName(network_label),
+      stats.overall_incoming_stats.bytes_discarded_no_receiver.bytes(),
+      Unit::kBytes, ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "packets_discarded_no_receiver", GetTestCaseName(network_label),
+      stats.overall_incoming_stats.packets_discarded_no_receiver,
+      Unit::kUnitless, ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "bytes_received", GetTestCaseName(network_label),
+      stats.overall_incoming_stats.bytes_received.bytes(), Unit::kBytes,
+      ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "packets_received", GetTestCaseName(network_label),
+      stats.overall_incoming_stats.packets_received, Unit::kUnitless,
+      ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "average_receive_rate", GetTestCaseName(network_label),
+      stats.overall_incoming_stats.packets_received >= 2
+          ? stats.overall_incoming_stats.AverageReceiveRate().kbps<double>()
+          : 0,
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "sent_packets_loss", GetTestCaseName(network_label), packet_loss,
+      Unit::kUnitless, ImprovementDirection::kNeitherIsBetter);
+}
+
+void NetworkQualityMetricsReporter::ReportPCStats(const std::string& pc_label,
+                                                  const PCStats& stats) {
+  metrics_logger_->LogSingleValueMetric(
+      "payload_bytes_received", pc_label, stats.payload_received.bytes(),
+      Unit::kBytes, ImprovementDirection::kNeitherIsBetter);
+  metrics_logger_->LogSingleValueMetric(
+      "payload_bytes_sent", pc_label, stats.payload_sent.bytes(), Unit::kBytes,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+std::string NetworkQualityMetricsReporter::GetTestCaseName(
+    const std::string& network_label) const {
+  return test_case_name_ + "/" + network_label;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
new file mode 100644
index 0000000000..ed894bcf54
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
@@ -0,0 +1,72 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_NETWORK_QUALITY_METRICS_REPORTER_H_
+#define TEST_PC_E2E_NETWORK_QUALITY_METRICS_REPORTER_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/test/track_id_stream_info_map.h"
+#include "api/units/data_size.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class NetworkQualityMetricsReporter
+    : public PeerConnectionE2EQualityTestFixture::QualityMetricsReporter {
+ public:
+  NetworkQualityMetricsReporter(EmulatedNetworkManagerInterface* alice_network,
+                                EmulatedNetworkManagerInterface* bob_network,
+                                test::MetricsLogger* metrics_logger);
+  ~NetworkQualityMetricsReporter() override = default;
+
+  // Network stats must be empty when this method will be invoked.
+  void Start(absl::string_view test_case_name,
+             const TrackIdStreamInfoMap* reporter_helper) override;
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override;
+  void StopAndReportResults() override;
+
+ private:
+  struct PCStats {
+    // TODO(nisse): Separate audio and video counters. Depends on standard stat
+    // counters, enabled by field trial "WebRTC-UseStandardBytesStats".
+    DataSize payload_received = DataSize::Zero();
+    DataSize payload_sent = DataSize::Zero();
+  };
+
+  static EmulatedNetworkStats PopulateStats(
+      EmulatedNetworkManagerInterface* network);
+  void ReportStats(const std::string& network_label,
+                   const EmulatedNetworkStats& stats,
+                   int64_t packet_loss);
+  void ReportPCStats(const std::string& pc_label, const PCStats& stats);
+  std::string GetTestCaseName(const std::string& network_label) const;
+
+  std::string test_case_name_;
+
+  EmulatedNetworkManagerInterface* const alice_network_;
+  EmulatedNetworkManagerInterface* const bob_network_;
+  test::MetricsLogger* const metrics_logger_;
+  Mutex lock_;
+  std::map<std::string, PCStats> pc_stats_ RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_NETWORK_QUALITY_METRICS_REPORTER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_e2e_smoke_test.cc b/third_party/libwebrtc/test/pc/e2e/peer_connection_e2e_smoke_test.cc
new file mode 100644
index 0000000000..0e7993e5be
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_e2e_smoke_test.cc
@@ -0,0 +1,536 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <cstdint>
+#include <memory>
+#include <string>
+
+#include "api/media_stream_interface.h"
+#include "api/test/create_network_emulation_manager.h"
+#include "api/test/create_peer_connection_quality_test_frame_generator.h"
+#include "api/test/create_peerconnection_quality_test_fixture.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "call/simulated_network.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h"
+#include "test/pc/e2e/stats_based_network_quality_metrics_reporter.h"
+#include "test/testsupport/file_utils.h"
+
+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"  // nogncheck
+#endif
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+class PeerConnectionE2EQualityTestSmokeTest : public ::testing::Test {
+ public:
+  void SetUp() override {
+    network_emulation_ = CreateNetworkEmulationManager();
+    auto video_quality_analyzer = std::make_unique<DefaultVideoQualityAnalyzer>(
+        network_emulation_->time_controller()->GetClock(),
+        test::GetGlobalMetricsLogger());
+    video_quality_analyzer_ = video_quality_analyzer.get();
+    fixture_ = CreatePeerConnectionE2EQualityTestFixture(
+        testing::UnitTest::GetInstance()->current_test_info()->name(),
+        *network_emulation_->time_controller(),
+        /*audio_quality_analyzer=*/nullptr, std::move(video_quality_analyzer));
+  }
+
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+  CreateNetwork() {
+    EmulatedNetworkNode* alice_node = network_emulation_->CreateEmulatedNode(
+        std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+    EmulatedNetworkNode* bob_node = network_emulation_->CreateEmulatedNode(
+        std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig()));
+
+    EmulatedEndpoint* alice_endpoint =
+        network_emulation_->CreateEndpoint(EmulatedEndpointConfig());
+    EmulatedEndpoint* bob_endpoint =
+        network_emulation_->CreateEndpoint(EmulatedEndpointConfig());
+
+    network_emulation_->CreateRoute(alice_endpoint, {alice_node}, bob_endpoint);
+    network_emulation_->CreateRoute(bob_endpoint, {bob_node}, alice_endpoint);
+
+    EmulatedNetworkManagerInterface* alice_network =
+        network_emulation_->CreateEmulatedNetworkManagerInterface(
+            {alice_endpoint});
+    EmulatedNetworkManagerInterface* bob_network =
+        network_emulation_->CreateEmulatedNetworkManagerInterface(
+            {bob_endpoint});
+
+    return std::make_pair(alice_network, bob_network);
+  }
+
+  void AddPeer(EmulatedNetworkManagerInterface* network,
+               rtc::FunctionView<void(PeerConfigurer*)> update_configurer) {
+    auto configurer =
+        std::make_unique<PeerConfigurer>(network->network_dependencies());
+    update_configurer(configurer.get());
+    fixture_->AddPeer(std::move(configurer));
+  }
+
+  void RunAndCheckEachVideoStreamReceivedFrames(const RunParams& run_params) {
+    fixture_->Run(run_params);
+
+    EXPECT_GE(fixture_->GetRealTestDuration(), run_params.run_duration);
+    VideoStreamsInfo known_streams = video_quality_analyzer_->GetKnownStreams();
+    for (const StatsKey& stream_key : known_streams.GetStatsKeys()) {
+      FrameCounters stream_conters =
+          video_quality_analyzer_->GetPerStreamCounters().at(stream_key);
+      // On some devices the pipeline can be too slow, so we actually can't
+      // force real constraints here. Lets just check, that at least 1
+      // frame passed whole pipeline.
+      int64_t expected_min_fps = run_params.run_duration.seconds() * 15;
+      EXPECT_GE(stream_conters.captured, expected_min_fps)
+          << stream_key.ToString();
+      EXPECT_GE(stream_conters.pre_encoded, 1) << stream_key.ToString();
+      EXPECT_GE(stream_conters.encoded, 1) << stream_key.ToString();
+      EXPECT_GE(stream_conters.received, 1) << stream_key.ToString();
+      EXPECT_GE(stream_conters.decoded, 1) << stream_key.ToString();
+      EXPECT_GE(stream_conters.rendered, 1) << stream_key.ToString();
+    }
+  }
+
+  NetworkEmulationManager* network_emulation() {
+    return network_emulation_.get();
+  }
+
+  PeerConnectionE2EQualityTestFixture* fixture() { return fixture_.get(); }
+
+ private:
+  std::unique_ptr<NetworkEmulationManager> network_emulation_;
+  DefaultVideoQualityAnalyzer* video_quality_analyzer_;
+  std::unique_ptr<PeerConnectionE2EQualityTestFixture> fixture_;
+};
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_Smoke DISABLED_Smoke
+#else
+#define MAYBE_Smoke Smoke
+#endif
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Smoke) {
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    VideoConfig video(160, 120, 15);
+    video.stream_label = "alice-video";
+    video.sync_group = "alice-media";
+    alice->AddVideoConfig(std::move(video));
+
+    AudioConfig audio;
+    audio.stream_label = "alice-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_alice_source", "wav");
+    audio.sampling_frequency_in_hz = 48000;
+    audio.sync_group = "alice-media";
+    alice->SetAudioConfig(std::move(audio));
+    alice->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+
+    alice->SetUseFlexFEC(true);
+    alice->SetUseUlpFEC(true);
+    alice->SetVideoEncoderBitrateMultiplier(1.1);
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* charlie) {
+    charlie->SetName("charlie");
+    VideoConfig video(160, 120, 15);
+    video.stream_label = "charlie-video";
+    video.temporal_layers_count = 2;
+    charlie->AddVideoConfig(std::move(video));
+
+    AudioConfig audio;
+    audio.stream_label = "charlie-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_bob_source", "wav");
+    charlie->SetAudioConfig(std::move(audio));
+    charlie->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+
+    charlie->SetUseFlexFEC(true);
+    charlie->SetUseUlpFEC(true);
+    charlie->SetVideoEncoderBitrateMultiplier(1.1);
+  });
+  fixture()->AddQualityMetricsReporter(
+      std::make_unique<StatsBasedNetworkQualityMetricsReporter>(
+          std::map<std::string, std::vector<EmulatedEndpoint*>>(
+              {{"alice", network_links.first->endpoints()},
+               {"charlie", network_links.second->endpoints()}}),
+          network_emulation(), test::GetGlobalMetricsLogger()));
+  RunParams run_params(TimeDelta::Seconds(2));
+  run_params.enable_flex_fec_support = true;
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_Smoke DISABLED_Smoke
+#else
+#define MAYBE_SendAndReceivePacketsOnOneThread \
+  SmokeSendAndReceivePacketsOnOneThread
+#endif
+// Only use the network thread for sending and receiving packets.
+// The one and only network thread is used as a worker thread in all
+// PeerConnections. Pacing when sending packets is done on the worker thread.
+// See bugs.webrtc.org/14502.
+TEST_F(PeerConnectionE2EQualityTestSmokeTest,
+       MAYBE_SendAndReceivePacketsOnOneThread) {
+  test::ScopedFieldTrials trials(
+      std::string(field_trial::GetFieldTrialString()) +
+      "WebRTC-SendPacketsOnWorkerThread/Enabled/");
+
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    // Peerconnection use the network thread as the worker thread.
+    alice->SetUseNetworkThreadAsWorkerThread();
+    VideoConfig video(160, 120, 15);
+    video.stream_label = "alice-video";
+    video.sync_group = "alice-media";
+    alice->AddVideoConfig(std::move(video));
+
+    AudioConfig audio;
+    audio.stream_label = "alice-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_alice_source", "wav");
+    audio.sampling_frequency_in_hz = 48000;
+    audio.sync_group = "alice-media";
+    alice->SetAudioConfig(std::move(audio));
+    alice->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* charlie) {
+    // Peerconnection use the network thread as the worker thread.
+    charlie->SetUseNetworkThreadAsWorkerThread();
+    charlie->SetName("charlie");
+    VideoConfig video(160, 120, 15);
+    video.stream_label = "charlie-video";
+    video.temporal_layers_count = 2;
+    charlie->AddVideoConfig(std::move(video));
+
+    AudioConfig audio;
+    audio.stream_label = "charlie-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_bob_source", "wav");
+    charlie->SetAudioConfig(std::move(audio));
+    charlie->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+    charlie->SetVideoEncoderBitrateMultiplier(1.1);
+  });
+  fixture()->AddQualityMetricsReporter(
+      std::make_unique<StatsBasedNetworkQualityMetricsReporter>(
+          std::map<std::string, std::vector<EmulatedEndpoint*>>(
+              {{"alice", network_links.first->endpoints()},
+               {"charlie", network_links.second->endpoints()}}),
+          network_emulation(), test::GetGlobalMetricsLogger()));
+  RunParams run_params(TimeDelta::Seconds(2));
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+
+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, SmokeH264) {
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    VideoConfig video(160, 120, 15);
+    video.stream_label = "alice-video";
+    video.sync_group = "alice-media";
+    alice->AddVideoConfig(std::move(video));
+
+    AudioConfig audio;
+    audio.stream_label = "alice-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_alice_source", "wav");
+    audio.sampling_frequency_in_hz = 48000;
+    audio.sync_group = "alice-media";
+    alice->SetAudioConfig(std::move(audio));
+    alice->SetVideoCodecs({VideoCodecConfig(cricket::kH264CodecName)});
+    alice->SetVideoEncoderFactory(webrtc::test::CreateObjCEncoderFactory());
+    alice->SetVideoDecoderFactory(webrtc::test::CreateObjCDecoderFactory());
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* charlie) {
+    charlie->SetName("charlie");
+    VideoConfig video(160, 120, 15);
+    video.stream_label = "charlie-video";
+    video.temporal_layers_count = 2;
+    charlie->AddVideoConfig(std::move(video));
+
+    AudioConfig audio;
+    audio.stream_label = "charlie-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_bob_source", "wav");
+    charlie->SetAudioConfig(std::move(audio));
+    charlie->SetVideoCodecs({VideoCodecConfig(cricket::kH264CodecName)});
+    charlie->SetVideoEncoderFactory(webrtc::test::CreateObjCEncoderFactory());
+    charlie->SetVideoDecoderFactory(webrtc::test::CreateObjCDecoderFactory());
+  });
+
+  fixture()->AddQualityMetricsReporter(
+      std::make_unique<StatsBasedNetworkQualityMetricsReporter>(
+          std::map<std::string, std::vector<EmulatedEndpoint*>>(
+              {{"alice", network_links.first->endpoints()},
+               {"charlie", network_links.second->endpoints()}}),
+          network_emulation(), test::GetGlobalMetricsLogger()));
+  RunParams run_params(TimeDelta::Seconds(2));
+  run_params.enable_flex_fec_support = true;
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+#endif
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_ChangeNetworkConditions DISABLED_ChangeNetworkConditions
+#else
+#define MAYBE_ChangeNetworkConditions ChangeNetworkConditions
+#endif
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_ChangeNetworkConditions) {
+  NetworkEmulationManager::SimulatedNetworkNode alice_node =
+      network_emulation()
+          ->NodeBuilder()
+          .config(BuiltInNetworkBehaviorConfig())
+          .Build();
+  NetworkEmulationManager::SimulatedNetworkNode bob_node =
+      network_emulation()
+          ->NodeBuilder()
+          .config(BuiltInNetworkBehaviorConfig())
+          .Build();
+
+  EmulatedEndpoint* alice_endpoint =
+      network_emulation()->CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_emulation()->CreateEndpoint(EmulatedEndpointConfig());
+
+  network_emulation()->CreateRoute(alice_endpoint, {alice_node.node},
+                                   bob_endpoint);
+  network_emulation()->CreateRoute(bob_endpoint, {bob_node.node},
+                                   alice_endpoint);
+
+  EmulatedNetworkManagerInterface* alice_network =
+      network_emulation()->CreateEmulatedNetworkManagerInterface(
+          {alice_endpoint});
+  EmulatedNetworkManagerInterface* bob_network =
+      network_emulation()->CreateEmulatedNetworkManagerInterface(
+          {bob_endpoint});
+
+  AddPeer(alice_network, [](PeerConfigurer* alice) {
+    VideoConfig video(160, 120, 15);
+    video.stream_label = "alice-video";
+    video.sync_group = "alice-media";
+    alice->AddVideoConfig(std::move(video));
+    alice->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+
+    alice->SetUseFlexFEC(true);
+    alice->SetUseUlpFEC(true);
+    alice->SetVideoEncoderBitrateMultiplier(1.1);
+  });
+  AddPeer(bob_network, [](PeerConfigurer* bob) {
+    bob->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+
+    bob->SetUseFlexFEC(true);
+    bob->SetUseUlpFEC(true);
+    bob->SetVideoEncoderBitrateMultiplier(1.1);
+  });
+  fixture()->AddQualityMetricsReporter(
+      std::make_unique<StatsBasedNetworkQualityMetricsReporter>(
+          std::map<std::string, std::vector<EmulatedEndpoint*>>(
+              {{"alice", alice_network->endpoints()},
+               {"bob", bob_network->endpoints()}}),
+          network_emulation(), test::GetGlobalMetricsLogger()));
+
+  fixture()->ExecuteAt(TimeDelta::Seconds(1), [alice_node](TimeDelta) {
+    BuiltInNetworkBehaviorConfig config;
+    config.loss_percent = 5;
+    alice_node.simulation->SetConfig(config);
+  });
+
+  RunParams run_params(TimeDelta::Seconds(2));
+  run_params.enable_flex_fec_support = true;
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_Screenshare DISABLED_Screenshare
+#else
+#define MAYBE_Screenshare Screenshare
+#endif
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Screenshare) {
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    VideoConfig screenshare(320, 180, 30);
+    screenshare.stream_label = "alice-screenshare";
+    screenshare.content_hint = VideoTrackInterface::ContentHint::kText;
+    ScreenShareConfig screen_share_config =
+        ScreenShareConfig(TimeDelta::Seconds(2));
+    screen_share_config.scrolling_params =
+        ScrollingParams{.duration = TimeDelta::Millis(1800)};
+    auto screen_share_frame_generator =
+        CreateScreenShareFrameGenerator(screenshare, screen_share_config);
+    alice->AddVideoConfig(std::move(screenshare),
+                          std::move(screen_share_frame_generator));
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* bob) {});
+  RunAndCheckEachVideoStreamReceivedFrames(RunParams(TimeDelta::Seconds(2)));
+}
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_Echo DISABLED_Echo
+#else
+#define MAYBE_Echo Echo
+#endif
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Echo) {
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    AudioConfig audio;
+    audio.stream_label = "alice-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_alice_source", "wav");
+    audio.sampling_frequency_in_hz = 48000;
+    alice->SetAudioConfig(std::move(audio));
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* bob) {
+    AudioConfig audio;
+    audio.stream_label = "bob-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_bob_source", "wav");
+    bob->SetAudioConfig(std::move(audio));
+  });
+  RunParams run_params(TimeDelta::Seconds(2));
+  run_params.echo_emulation_config = EchoEmulationConfig();
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_Simulcast DISABLED_Simulcast
+#else
+#define MAYBE_Simulcast Simulcast
+#endif
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Simulcast) {
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    VideoConfig simulcast(1280, 720, 15);
+    simulcast.stream_label = "alice-simulcast";
+    simulcast.simulcast_config = VideoSimulcastConfig(2);
+    simulcast.emulated_sfu_config = EmulatedSFUConfig(0);
+    alice->AddVideoConfig(std::move(simulcast));
+
+    AudioConfig audio;
+    audio.stream_label = "alice-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_alice_source", "wav");
+    alice->SetAudioConfig(std::move(audio));
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* bob) {});
+  RunParams run_params(TimeDelta::Seconds(2));
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_Svc DISABLED_Svc
+#else
+#define MAYBE_Svc Svc
+#endif
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_Svc) {
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    VideoConfig simulcast("alice-svc", 1280, 720, 15);
+    // Because we have network with packets loss we can analyze only the
+    // highest spatial layer in SVC mode.
+    simulcast.simulcast_config = VideoSimulcastConfig(2);
+    simulcast.emulated_sfu_config = EmulatedSFUConfig(1);
+    alice->AddVideoConfig(std::move(simulcast));
+
+    AudioConfig audio("alice-audio");
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_alice_source", "wav");
+    alice->SetAudioConfig(std::move(audio));
+    alice->SetVideoCodecs({VideoCodecConfig(cricket::kVp9CodecName)});
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* bob) {
+    bob->SetVideoCodecs({VideoCodecConfig(cricket::kVp9CodecName)});
+  });
+  RunParams run_params(TimeDelta::Seconds(2));
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+
+// IOS debug builds can be quite slow, disabling to avoid issues with timeouts.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_HighBitrate DISABLED_HighBitrate
+#else
+#define MAYBE_HighBitrate HighBitrate
+#endif
+TEST_F(PeerConnectionE2EQualityTestSmokeTest, MAYBE_HighBitrate) {
+  std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+      network_links = CreateNetwork();
+  AddPeer(network_links.first, [](PeerConfigurer* alice) {
+    BitrateSettings bitrate_settings;
+    bitrate_settings.start_bitrate_bps = 3'000'000;
+    bitrate_settings.max_bitrate_bps = 3'000'000;
+    alice->SetBitrateSettings(bitrate_settings);
+    VideoConfig video(800, 600, 15);
+    video.stream_label = "alice-video";
+    RtpEncodingParameters encoding_parameters;
+    encoding_parameters.min_bitrate_bps = 500'000;
+    encoding_parameters.max_bitrate_bps = 3'000'000;
+    video.encoding_params.push_back(std::move(encoding_parameters));
+    alice->AddVideoConfig(std::move(video));
+
+    AudioConfig audio;
+    audio.stream_label = "alice-audio";
+    audio.mode = AudioConfig::Mode::kFile;
+    audio.input_file_name =
+        test::ResourcePath("pc_quality_smoke_test_alice_source", "wav");
+    audio.sampling_frequency_in_hz = 48000;
+    alice->SetAudioConfig(std::move(audio));
+    alice->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+  });
+  AddPeer(network_links.second, [](PeerConfigurer* bob) {
+    bob->SetVideoCodecs(
+        {VideoCodecConfig(cricket::kVp9CodecName, {{"profile-id", "0"}})});
+  });
+  RunParams run_params(TimeDelta::Seconds(2));
+  RunAndCheckEachVideoStreamReceivedFrames(run_params);
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
new file mode 100644
index 0000000000..83613118f9
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
@@ -0,0 +1,763 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/peer_connection_quality_test.h"
+
+#include <algorithm>
+#include <memory>
+#include <set>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "api/jsep.h"
+#include "api/media_stream_interface.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/rtc_event_log_output_file.h"
+#include "api/scoped_refptr.h"
+#include "api/test/metrics/metric.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/time_controller.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "pc/sdp_utils.h"
+#include "pc/test/mock_peer_connection_observers.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "system_wrappers/include/cpu_info.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+#include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h"
+#include "test/pc/e2e/analyzer/video/default_video_quality_analyzer.h"
+#include "test/pc/e2e/analyzer/video/video_frame_tracking_id_injector.h"
+#include "test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h"
+#include "test/pc/e2e/cross_media_metrics_reporter.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+#include "test/pc/e2e/peer_params_preprocessor.h"
+#include "test/pc/e2e/stats_poller.h"
+#include "test/pc/e2e/test_peer_factory.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+constexpr TimeDelta kDefaultTimeout = TimeDelta::Seconds(10);
+constexpr char kSignalThreadName[] = "signaling_thread";
+// 1 signaling, 2 network, 2 worker and 2 extra for codecs etc.
+constexpr int kPeerConnectionUsedThreads = 7;
+// Framework has extra thread for network layer and extra thread for peer
+// connection stats polling.
+constexpr int kFrameworkUsedThreads = 2;
+constexpr int kMaxVideoAnalyzerThreads = 8;
+
+constexpr TimeDelta kStatsUpdateInterval = TimeDelta::Seconds(1);
+
+constexpr TimeDelta kAliveMessageLogInterval = TimeDelta::Seconds(30);
+
+constexpr TimeDelta kQuickTestModeRunDuration = TimeDelta::Millis(100);
+
+// Field trials to enable Flex FEC advertising and receiving.
+constexpr char kFlexFecEnabledFieldTrials[] =
+    "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/";
+constexpr char kUseStandardsBytesStats[] =
+    "WebRTC-UseStandardBytesStats/Enabled/";
+
+class FixturePeerConnectionObserver : public MockPeerConnectionObserver {
+ public:
+  // `on_track_callback` will be called when any new track will be added to peer
+  // connection.
+  // `on_connected_callback` will be called when peer connection will come to
+  // either connected or completed state. Client should notice that in the case
+  // of reconnect this callback can be called again, so it should be tolerant
+  // to such behavior.
+  FixturePeerConnectionObserver(
+      std::function<void(rtc::scoped_refptr<RtpTransceiverInterface>)>
+          on_track_callback,
+      std::function<void()> on_connected_callback)
+      : on_track_callback_(std::move(on_track_callback)),
+        on_connected_callback_(std::move(on_connected_callback)) {}
+
+  void OnTrack(
+      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override {
+    MockPeerConnectionObserver::OnTrack(transceiver);
+    on_track_callback_(transceiver);
+  }
+
+  void OnIceConnectionChange(
+      PeerConnectionInterface::IceConnectionState new_state) override {
+    MockPeerConnectionObserver::OnIceConnectionChange(new_state);
+    if (ice_connected_) {
+      on_connected_callback_();
+    }
+  }
+
+ private:
+  std::function<void(rtc::scoped_refptr<RtpTransceiverInterface>)>
+      on_track_callback_;
+  std::function<void()> on_connected_callback_;
+};
+
+void ValidateP2PSimulcastParams(
+    const std::vector<std::unique_ptr<PeerConfigurer>>& peers) {
+  for (size_t i = 0; i < peers.size(); ++i) {
+    Params* params = peers[i]->params();
+    ConfigurableParams* configurable_params = peers[i]->configurable_params();
+    for (const VideoConfig& video_config : configurable_params->video_configs) {
+      if (video_config.simulcast_config) {
+        // When we simulate SFU we support only one video codec.
+        RTC_CHECK_EQ(params->video_codecs.size(), 1)
+            << "Only 1 video codec is supported when simulcast is enabled in "
+            << "at least 1 video config";
+      }
+    }
+  }
+}
+
+}  // namespace
+
+PeerConnectionE2EQualityTest::PeerConnectionE2EQualityTest(
+    std::string test_case_name,
+    TimeController& time_controller,
+    std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer,
+    std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer)
+    : PeerConnectionE2EQualityTest(std::move(test_case_name),
+                                   time_controller,
+                                   std::move(audio_quality_analyzer),
+                                   std::move(video_quality_analyzer),
+                                   /*metrics_logger_=*/nullptr) {}
+
+PeerConnectionE2EQualityTest::PeerConnectionE2EQualityTest(
+    std::string test_case_name,
+    TimeController& time_controller,
+    std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer,
+    std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer,
+    test::MetricsLogger* metrics_logger)
+    : time_controller_(time_controller),
+      task_queue_factory_(time_controller_.CreateTaskQueueFactory()),
+      test_case_name_(std::move(test_case_name)),
+      executor_(std::make_unique<TestActivitiesExecutor>(
+          time_controller_.GetClock())),
+      metrics_logger_(metrics_logger) {
+  // Create default video quality analyzer. We will always create an analyzer,
+  // even if there are no video streams, because it will be installed into video
+  // encoder/decoder factories.
+  if (video_quality_analyzer == nullptr) {
+    video_quality_analyzer = std::make_unique<DefaultVideoQualityAnalyzer>(
+        time_controller_.GetClock(), metrics_logger_);
+  }
+  if (field_trial::IsEnabled("WebRTC-VideoFrameTrackingIdAdvertised")) {
+    encoded_image_data_propagator_ =
+        std::make_unique<VideoFrameTrackingIdInjector>();
+  } else {
+    encoded_image_data_propagator_ =
+        std::make_unique<SingleProcessEncodedImageDataInjector>();
+  }
+  video_quality_analyzer_injection_helper_ =
+      std::make_unique<VideoQualityAnalyzerInjectionHelper>(
+          time_controller_.GetClock(), std::move(video_quality_analyzer),
+          encoded_image_data_propagator_.get(),
+          encoded_image_data_propagator_.get());
+
+  if (audio_quality_analyzer == nullptr) {
+    audio_quality_analyzer =
+        std::make_unique<DefaultAudioQualityAnalyzer>(metrics_logger_);
+  }
+  audio_quality_analyzer_.swap(audio_quality_analyzer);
+}
+
+void PeerConnectionE2EQualityTest::ExecuteAt(
+    TimeDelta target_time_since_start,
+    std::function<void(TimeDelta)> func) {
+  executor_->ScheduleActivity(target_time_since_start, absl::nullopt, func);
+}
+
+void PeerConnectionE2EQualityTest::ExecuteEvery(
+    TimeDelta initial_delay_since_start,
+    TimeDelta interval,
+    std::function<void(TimeDelta)> func) {
+  executor_->ScheduleActivity(initial_delay_since_start, interval, func);
+}
+
+void PeerConnectionE2EQualityTest::AddQualityMetricsReporter(
+    std::unique_ptr<QualityMetricsReporter> quality_metrics_reporter) {
+  quality_metrics_reporters_.push_back(std::move(quality_metrics_reporter));
+}
+
+PeerConnectionE2EQualityTest::PeerHandle* PeerConnectionE2EQualityTest::AddPeer(
+    std::unique_ptr<PeerConfigurer> configurer) {
+  peer_configurations_.push_back(std::move(configurer));
+  peer_handles_.push_back(PeerHandleImpl());
+  return &peer_handles_.back();
+}
+
+void PeerConnectionE2EQualityTest::Run(RunParams run_params) {
+  webrtc::webrtc_pc_e2e::PeerParamsPreprocessor params_preprocessor;
+  for (auto& peer_configuration : peer_configurations_) {
+    params_preprocessor.SetDefaultValuesForMissingParams(*peer_configuration);
+    params_preprocessor.ValidateParams(*peer_configuration);
+  }
+  ValidateP2PSimulcastParams(peer_configurations_);
+  RTC_CHECK_EQ(peer_configurations_.size(), 2)
+      << "Only peer to peer calls are allowed, please add 2 peers";
+
+  std::unique_ptr<PeerConfigurer> alice_configurer =
+      std::move(peer_configurations_[0]);
+  std::unique_ptr<PeerConfigurer> bob_configurer =
+      std::move(peer_configurations_[1]);
+  peer_configurations_.clear();
+
+  for (size_t i = 0;
+       i < bob_configurer->configurable_params()->video_configs.size(); ++i) {
+    // We support simulcast only from caller.
+    RTC_CHECK(!bob_configurer->configurable_params()
+                   ->video_configs[i]
+                   .simulcast_config)
+        << "Only simulcast stream from first peer is supported";
+  }
+
+  test::ScopedFieldTrials field_trials(GetFieldTrials(run_params));
+
+  // Print test summary
+  RTC_LOG(LS_INFO)
+      << "Media quality test: " << *alice_configurer->params()->name
+      << " will make a call to " << *bob_configurer->params()->name
+      << " with media video="
+      << !alice_configurer->configurable_params()->video_configs.empty()
+      << "; audio=" << alice_configurer->params()->audio_config.has_value()
+      << ". " << *bob_configurer->params()->name
+      << " will respond with media video="
+      << !bob_configurer->configurable_params()->video_configs.empty()
+      << "; audio=" << bob_configurer->params()->audio_config.has_value();
+
+  const std::unique_ptr<rtc::Thread> signaling_thread =
+      time_controller_.CreateThread(kSignalThreadName);
+  media_helper_ = std::make_unique<MediaHelper>(
+      video_quality_analyzer_injection_helper_.get(), task_queue_factory_.get(),
+      time_controller_.GetClock());
+
+  // Create a `task_queue_`.
+  task_queue_ = std::make_unique<webrtc::TaskQueueForTest>(
+      time_controller_.GetTaskQueueFactory()->CreateTaskQueue(
+          "pc_e2e_quality_test", webrtc::TaskQueueFactory::Priority::NORMAL));
+
+  // Create call participants: Alice and Bob.
+  // Audio streams are intercepted in AudioDeviceModule, so if it is required to
+  // catch output of Alice's stream, Alice's output_dump_file_name should be
+  // passed to Bob's TestPeer setup as audio output file name.
+  absl::optional<RemotePeerAudioConfig> alice_remote_audio_config =
+      RemotePeerAudioConfig::Create(bob_configurer->params()->audio_config);
+  absl::optional<RemotePeerAudioConfig> bob_remote_audio_config =
+      RemotePeerAudioConfig::Create(alice_configurer->params()->audio_config);
+  // Copy Alice and Bob video configs, subscriptions and names to correctly pass
+  // them into lambdas.
+  VideoSubscription alice_subscription =
+      alice_configurer->configurable_params()->video_subscription;
+  std::vector<VideoConfig> alice_video_configs =
+      alice_configurer->configurable_params()->video_configs;
+  std::string alice_name = alice_configurer->params()->name.value();
+  VideoSubscription bob_subscription =
+      alice_configurer->configurable_params()->video_subscription;
+  std::vector<VideoConfig> bob_video_configs =
+      bob_configurer->configurable_params()->video_configs;
+  std::string bob_name = bob_configurer->params()->name.value();
+
+  TestPeerFactory test_peer_factory(
+      signaling_thread.get(), time_controller_,
+      video_quality_analyzer_injection_helper_.get(), task_queue_.get());
+  alice_ = test_peer_factory.CreateTestPeer(
+      std::move(alice_configurer),
+      std::make_unique<FixturePeerConnectionObserver>(
+          [this, alice_name, alice_subscription, bob_video_configs](
+              rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+            OnTrackCallback(alice_name, alice_subscription, transceiver,
+                            bob_video_configs);
+          },
+          [this]() { StartVideo(alice_video_sources_); }),
+      alice_remote_audio_config, run_params.echo_emulation_config);
+  bob_ = test_peer_factory.CreateTestPeer(
+      std::move(bob_configurer),
+      std::make_unique<FixturePeerConnectionObserver>(
+          [this, bob_name, bob_subscription, alice_video_configs](
+              rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+            OnTrackCallback(bob_name, bob_subscription, transceiver,
+                            alice_video_configs);
+          },
+          [this]() { StartVideo(bob_video_sources_); }),
+      bob_remote_audio_config, run_params.echo_emulation_config);
+
+  int num_cores = CpuInfo::DetectNumberOfCores();
+  RTC_DCHECK_GE(num_cores, 1);
+
+  int video_analyzer_threads =
+      num_cores - kPeerConnectionUsedThreads - kFrameworkUsedThreads;
+  if (video_analyzer_threads <= 0) {
+    video_analyzer_threads = 1;
+  }
+  video_analyzer_threads =
+      std::min(video_analyzer_threads, kMaxVideoAnalyzerThreads);
+  RTC_LOG(LS_INFO) << "video_analyzer_threads=" << video_analyzer_threads;
+  quality_metrics_reporters_.push_back(
+      std::make_unique<VideoQualityMetricsReporter>(time_controller_.GetClock(),
+                                                    metrics_logger_));
+  quality_metrics_reporters_.push_back(
+      std::make_unique<CrossMediaMetricsReporter>(metrics_logger_));
+
+  video_quality_analyzer_injection_helper_->Start(
+      test_case_name_,
+      std::vector<std::string>{alice_->params().name.value(),
+                               bob_->params().name.value()},
+      video_analyzer_threads);
+  audio_quality_analyzer_->Start(test_case_name_, &analyzer_helper_);
+  for (auto& reporter : quality_metrics_reporters_) {
+    reporter->Start(test_case_name_, &analyzer_helper_);
+  }
+
+  // Start RTCEventLog recording if requested.
+  if (alice_->params().rtc_event_log_path) {
+    auto alice_rtc_event_log = std::make_unique<webrtc::RtcEventLogOutputFile>(
+        alice_->params().rtc_event_log_path.value());
+    alice_->pc()->StartRtcEventLog(std::move(alice_rtc_event_log),
+                                   webrtc::RtcEventLog::kImmediateOutput);
+  }
+  if (bob_->params().rtc_event_log_path) {
+    auto bob_rtc_event_log = std::make_unique<webrtc::RtcEventLogOutputFile>(
+        bob_->params().rtc_event_log_path.value());
+    bob_->pc()->StartRtcEventLog(std::move(bob_rtc_event_log),
+                                 webrtc::RtcEventLog::kImmediateOutput);
+  }
+
+  // Setup alive logging. It is done to prevent test infra to think that test is
+  // dead.
+  RepeatingTaskHandle::DelayedStart(task_queue_->Get(),
+                                    kAliveMessageLogInterval, []() {
+                                      std::printf("Test is still running...\n");
+                                      return kAliveMessageLogInterval;
+                                    });
+
+  RTC_LOG(LS_INFO) << "Configuration is done. Now " << *alice_->params().name
+                   << " is calling to " << *bob_->params().name << "...";
+
+  // Setup stats poller.
+  std::vector<StatsObserverInterface*> observers = {
+      audio_quality_analyzer_.get(),
+      video_quality_analyzer_injection_helper_.get()};
+  for (auto& reporter : quality_metrics_reporters_) {
+    observers.push_back(reporter.get());
+  }
+  StatsPoller stats_poller(observers,
+                           std::map<std::string, StatsProvider*>{
+                               {*alice_->params().name, alice_.get()},
+                               {*bob_->params().name, bob_.get()}});
+  executor_->ScheduleActivity(TimeDelta::Zero(), kStatsUpdateInterval,
+                              [&stats_poller](TimeDelta) {
+                                stats_poller.PollStatsAndNotifyObservers();
+                              });
+
+  // Setup call.
+  SendTask(signaling_thread.get(),
+           [this, &run_params] { SetupCallOnSignalingThread(run_params); });
+  std::unique_ptr<SignalingInterceptor> signaling_interceptor =
+      CreateSignalingInterceptor(run_params);
+  // Connect peers.
+  SendTask(signaling_thread.get(), [this, &signaling_interceptor] {
+    ExchangeOfferAnswer(signaling_interceptor.get());
+  });
+  WaitUntilIceCandidatesGathered(signaling_thread.get());
+
+  SendTask(signaling_thread.get(), [this, &signaling_interceptor] {
+    ExchangeIceCandidates(signaling_interceptor.get());
+  });
+  WaitUntilPeersAreConnected(signaling_thread.get());
+
+  executor_->Start(task_queue_.get());
+  Timestamp start_time = Now();
+
+  bool is_quick_test_enabled = field_trial::IsEnabled("WebRTC-QuickPerfTest");
+  if (is_quick_test_enabled) {
+    time_controller_.AdvanceTime(kQuickTestModeRunDuration);
+  } else {
+    time_controller_.AdvanceTime(run_params.run_duration);
+  }
+
+  RTC_LOG(LS_INFO) << "Test is done, initiating disconnect sequence.";
+
+  // Stop all client started tasks to prevent their access to any call related
+  // objects after these objects will be destroyed during call tear down.
+  executor_->Stop();
+  // There is no guarantee, that last stats collection will happen at the end
+  // of the call, so we force it after executor, which is among others is doing
+  // stats collection, was stopped.
+  task_queue_->SendTask([&stats_poller]() {
+    // Get final end-of-call stats.
+    stats_poller.PollStatsAndNotifyObservers();
+  });
+  // We need to detach AEC dumping from peers, because dump uses `task_queue_`
+  // inside.
+  alice_->DetachAecDump();
+  bob_->DetachAecDump();
+  // Tear down the call.
+  SendTask(signaling_thread.get(), [this] { TearDownCallOnSignalingThread(); });
+
+  Timestamp end_time = Now();
+  RTC_LOG(LS_INFO) << "All peers are disconnected.";
+  {
+    MutexLock lock(&lock_);
+    real_test_duration_ = end_time - start_time;
+  }
+
+  ReportGeneralTestResults();
+  audio_quality_analyzer_->Stop();
+  video_quality_analyzer_injection_helper_->Stop();
+  for (auto& reporter : quality_metrics_reporters_) {
+    reporter->StopAndReportResults();
+  }
+
+  // Reset `task_queue_` after test to cleanup.
+  task_queue_.reset();
+
+  alice_ = nullptr;
+  bob_ = nullptr;
+  // Ensuring that TestVideoCapturerVideoTrackSource are destroyed on the right
+  // thread.
+  RTC_CHECK(alice_video_sources_.empty());
+  RTC_CHECK(bob_video_sources_.empty());
+}
+
+std::string PeerConnectionE2EQualityTest::GetFieldTrials(
+    const RunParams& run_params) {
+  std::vector<absl::string_view> default_field_trials = {
+      kUseStandardsBytesStats};
+  if (run_params.enable_flex_fec_support) {
+    default_field_trials.push_back(kFlexFecEnabledFieldTrials);
+  }
+  rtc::StringBuilder sb;
+  sb << field_trial::GetFieldTrialString();
+  for (const absl::string_view& field_trial : default_field_trials) {
+    sb << field_trial;
+  }
+  return sb.Release();
+}
+
+void PeerConnectionE2EQualityTest::OnTrackCallback(
+    absl::string_view peer_name,
+    VideoSubscription peer_subscription,
+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver,
+    std::vector<VideoConfig> remote_video_configs) {
+  const rtc::scoped_refptr<MediaStreamTrackInterface>& track =
+      transceiver->receiver()->track();
+  RTC_CHECK_EQ(transceiver->receiver()->stream_ids().size(), 2)
+      << "Expected 2 stream ids: 1st - sync group, 2nd - unique stream label";
+  std::string sync_group = transceiver->receiver()->stream_ids()[0];
+  std::string stream_label = transceiver->receiver()->stream_ids()[1];
+  analyzer_helper_.AddTrackToStreamMapping(track->id(), peer_name, stream_label,
+                                           sync_group);
+  if (track->kind() != MediaStreamTrackInterface::kVideoKind) {
+    return;
+  }
+
+  // It is safe to cast here, because it is checked above that
+  // track->kind() is kVideoKind.
+  auto* video_track = static_cast<VideoTrackInterface*>(track.get());
+  std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> video_sink =
+      video_quality_analyzer_injection_helper_->CreateVideoSink(
+          peer_name, peer_subscription, /*report_infra_stats=*/false);
+  video_track->AddOrUpdateSink(video_sink.get(), rtc::VideoSinkWants());
+  output_video_sinks_.push_back(std::move(video_sink));
+}
+
+void PeerConnectionE2EQualityTest::SetupCallOnSignalingThread(
+    const RunParams& run_params) {
+  // We need receive-only transceivers for Bob's media stream, so there will
+  // be media section in SDP for that streams in Alice's offer, because it is
+  // forbidden to add new media sections in answer in Unified Plan.
+  RtpTransceiverInit receive_only_transceiver_init;
+  receive_only_transceiver_init.direction = RtpTransceiverDirection::kRecvOnly;
+  int alice_transceivers_counter = 0;
+  if (bob_->params().audio_config) {
+    // Setup receive audio transceiver if Bob has audio to send. If we'll need
+    // multiple audio streams, then we need transceiver for each Bob's audio
+    // stream.
+    RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
+        alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_AUDIO,
+                               receive_only_transceiver_init);
+    RTC_CHECK(result.ok());
+    alice_transceivers_counter++;
+  }
+
+  size_t alice_video_transceivers_non_simulcast_counter = 0;
+  for (auto& video_config : alice_->configurable_params().video_configs) {
+    RtpTransceiverInit transceiver_params;
+    if (video_config.simulcast_config) {
+      transceiver_params.direction = RtpTransceiverDirection::kSendOnly;
+      // Because simulcast enabled `alice_->params().video_codecs` has only 1
+      // element.
+      if (alice_->params().video_codecs[0].name == cricket::kVp8CodecName) {
+        // For Vp8 simulcast we need to add as many RtpEncodingParameters to the
+        // track as many simulcast streams requested. If they specified in
+        // `video_config.simulcast_config` it should be copied from there.
+        for (int i = 0;
+             i < video_config.simulcast_config->simulcast_streams_count; ++i) {
+          RtpEncodingParameters enc_params;
+          if (!video_config.encoding_params.empty()) {
+            enc_params = video_config.encoding_params[i];
+          }
+          // We need to be sure, that all rids will be unique with all mids.
+          enc_params.rid = std::to_string(alice_transceivers_counter) + "000" +
+                           std::to_string(i);
+          transceiver_params.send_encodings.push_back(enc_params);
+        }
+      }
+    } else {
+      transceiver_params.direction = RtpTransceiverDirection::kSendRecv;
+      RtpEncodingParameters enc_params;
+      if (video_config.encoding_params.size() == 1) {
+        enc_params = video_config.encoding_params[0];
+      }
+      transceiver_params.send_encodings.push_back(enc_params);
+
+      alice_video_transceivers_non_simulcast_counter++;
+    }
+    RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
+        alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO,
+                               transceiver_params);
+    RTC_CHECK(result.ok());
+
+    alice_transceivers_counter++;
+  }
+
+  // Add receive only transceivers in case Bob has more video_configs than
+  // Alice.
+  for (size_t i = alice_video_transceivers_non_simulcast_counter;
+       i < bob_->configurable_params().video_configs.size(); ++i) {
+    RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
+        alice_->AddTransceiver(cricket::MediaType::MEDIA_TYPE_VIDEO,
+                               receive_only_transceiver_init);
+    RTC_CHECK(result.ok());
+    alice_transceivers_counter++;
+  }
+
+  // Then add media for Alice and Bob
+  media_helper_->MaybeAddAudio(alice_.get());
+  alice_video_sources_ = media_helper_->MaybeAddVideo(alice_.get());
+  media_helper_->MaybeAddAudio(bob_.get());
+  bob_video_sources_ = media_helper_->MaybeAddVideo(bob_.get());
+
+  SetPeerCodecPreferences(alice_.get());
+  SetPeerCodecPreferences(bob_.get());
+}
+
+void PeerConnectionE2EQualityTest::TearDownCallOnSignalingThread() {
+  TearDownCall();
+}
+
+void PeerConnectionE2EQualityTest::SetPeerCodecPreferences(TestPeer* peer) {
+  std::vector<RtpCodecCapability> with_rtx_video_capabilities =
+      FilterVideoCodecCapabilities(
+          peer->params().video_codecs, true, peer->params().use_ulp_fec,
+          peer->params().use_flex_fec,
+          peer->pc_factory()
+              ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO)
+              .codecs);
+  std::vector<RtpCodecCapability> without_rtx_video_capabilities =
+      FilterVideoCodecCapabilities(
+          peer->params().video_codecs, false, peer->params().use_ulp_fec,
+          peer->params().use_flex_fec,
+          peer->pc_factory()
+              ->GetRtpSenderCapabilities(cricket::MediaType::MEDIA_TYPE_VIDEO)
+              .codecs);
+
+  // Set codecs for transceivers
+  for (auto transceiver : peer->pc()->GetTransceivers()) {
+    if (transceiver->media_type() == cricket::MediaType::MEDIA_TYPE_VIDEO) {
+      if (transceiver->sender()->init_send_encodings().size() > 1) {
+        // If transceiver's sender has more then 1 send encodings, it means it
+        // has multiple simulcast streams, so we need disable RTX on it.
+        RTCError result =
+            transceiver->SetCodecPreferences(without_rtx_video_capabilities);
+        RTC_CHECK(result.ok());
+      } else {
+        RTCError result =
+            transceiver->SetCodecPreferences(with_rtx_video_capabilities);
+        RTC_CHECK(result.ok());
+      }
+    }
+  }
+}
+
+std::unique_ptr<SignalingInterceptor>
+PeerConnectionE2EQualityTest::CreateSignalingInterceptor(
+    const RunParams& run_params) {
+  std::map<std::string, int> stream_label_to_simulcast_streams_count;
+  // We add only Alice here, because simulcast/svc is supported only from the
+  // first peer.
+  for (auto& video_config : alice_->configurable_params().video_configs) {
+    if (video_config.simulcast_config) {
+      stream_label_to_simulcast_streams_count.insert(
+          {*video_config.stream_label,
+           video_config.simulcast_config->simulcast_streams_count});
+    }
+  }
+  PatchingParams patching_params(run_params.use_conference_mode,
+                                 stream_label_to_simulcast_streams_count);
+  return std::make_unique<SignalingInterceptor>(patching_params);
+}
+
+void PeerConnectionE2EQualityTest::WaitUntilIceCandidatesGathered(
+    rtc::Thread* signaling_thread) {
+  ASSERT_TRUE(time_controller_.Wait(
+      [&]() {
+        bool result;
+        SendTask(signaling_thread, [&]() {
+          result = alice_->IsIceGatheringDone() && bob_->IsIceGatheringDone();
+        });
+        return result;
+      },
+      2 * kDefaultTimeout));
+}
+
+void PeerConnectionE2EQualityTest::WaitUntilPeersAreConnected(
+    rtc::Thread* signaling_thread) {
+  // This means that ICE and DTLS are connected.
+  alice_connected_ = time_controller_.Wait(
+      [&]() {
+        bool result;
+        SendTask(signaling_thread, [&] { result = alice_->IsIceConnected(); });
+        return result;
+      },
+      kDefaultTimeout);
+  bob_connected_ = time_controller_.Wait(
+      [&]() {
+        bool result;
+        SendTask(signaling_thread, [&] { result = bob_->IsIceConnected(); });
+        return result;
+      },
+      kDefaultTimeout);
+}
+
+void PeerConnectionE2EQualityTest::ExchangeOfferAnswer(
+    SignalingInterceptor* signaling_interceptor) {
+  std::string log_output;
+
+  auto offer = alice_->CreateOffer();
+  RTC_CHECK(offer);
+  offer->ToString(&log_output);
+  RTC_LOG(LS_INFO) << "Original offer: " << log_output;
+  LocalAndRemoteSdp patch_result = signaling_interceptor->PatchOffer(
+      std::move(offer), alice_->params().video_codecs[0]);
+  patch_result.local_sdp->ToString(&log_output);
+  RTC_LOG(LS_INFO) << "Offer to set as local description: " << log_output;
+  patch_result.remote_sdp->ToString(&log_output);
+  RTC_LOG(LS_INFO) << "Offer to set as remote description: " << log_output;
+
+  bool set_local_offer =
+      alice_->SetLocalDescription(std::move(patch_result.local_sdp));
+  RTC_CHECK(set_local_offer);
+  bool set_remote_offer =
+      bob_->SetRemoteDescription(std::move(patch_result.remote_sdp));
+  RTC_CHECK(set_remote_offer);
+  auto answer = bob_->CreateAnswer();
+  RTC_CHECK(answer);
+  answer->ToString(&log_output);
+  RTC_LOG(LS_INFO) << "Original answer: " << log_output;
+  patch_result = signaling_interceptor->PatchAnswer(
+      std::move(answer), bob_->params().video_codecs[0]);
+  patch_result.local_sdp->ToString(&log_output);
+  RTC_LOG(LS_INFO) << "Answer to set as local description: " << log_output;
+  patch_result.remote_sdp->ToString(&log_output);
+  RTC_LOG(LS_INFO) << "Answer to set as remote description: " << log_output;
+
+  bool set_local_answer =
+      bob_->SetLocalDescription(std::move(patch_result.local_sdp));
+  RTC_CHECK(set_local_answer);
+  bool set_remote_answer =
+      alice_->SetRemoteDescription(std::move(patch_result.remote_sdp));
+  RTC_CHECK(set_remote_answer);
+}
+
+void PeerConnectionE2EQualityTest::ExchangeIceCandidates(
+    SignalingInterceptor* signaling_interceptor) {
+  // Connect an ICE candidate pairs.
+  std::vector<std::unique_ptr<IceCandidateInterface>> alice_candidates =
+      signaling_interceptor->PatchOffererIceCandidates(
+          alice_->observer()->GetAllCandidates());
+  for (auto& candidate : alice_candidates) {
+    std::string candidate_str;
+    RTC_CHECK(candidate->ToString(&candidate_str));
+    RTC_LOG(LS_INFO) << *alice_->params().name
+                     << " ICE candidate(mid= " << candidate->sdp_mid()
+                     << "): " << candidate_str;
+  }
+  ASSERT_TRUE(bob_->AddIceCandidates(std::move(alice_candidates)));
+  std::vector<std::unique_ptr<IceCandidateInterface>> bob_candidates =
+      signaling_interceptor->PatchAnswererIceCandidates(
+          bob_->observer()->GetAllCandidates());
+  for (auto& candidate : bob_candidates) {
+    std::string candidate_str;
+    RTC_CHECK(candidate->ToString(&candidate_str));
+    RTC_LOG(LS_INFO) << *bob_->params().name
+                     << " ICE candidate(mid= " << candidate->sdp_mid()
+                     << "): " << candidate_str;
+  }
+  ASSERT_TRUE(alice_->AddIceCandidates(std::move(bob_candidates)));
+}
+
+void PeerConnectionE2EQualityTest::StartVideo(
+    const std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>>&
+        sources) {
+  for (auto& source : sources) {
+    if (source->state() != MediaSourceInterface::SourceState::kLive) {
+      source->Start();
+    }
+  }
+}
+
+void PeerConnectionE2EQualityTest::TearDownCall() {
+  for (const auto& video_source : alice_video_sources_) {
+    video_source->Stop();
+  }
+  for (const auto& video_source : bob_video_sources_) {
+    video_source->Stop();
+  }
+
+  alice_video_sources_.clear();
+  bob_video_sources_.clear();
+
+  alice_->Close();
+  bob_->Close();
+
+  media_helper_ = nullptr;
+}
+
+void PeerConnectionE2EQualityTest::ReportGeneralTestResults() {
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  metrics_logger_->LogSingleValueMetric(
+      *alice_->params().name + "_connected", test_case_name_, alice_connected_,
+      Unit::kUnitless, ImprovementDirection::kBiggerIsBetter,
+      {{MetricMetadataKey::kPeerMetadataKey, *alice_->params().name},
+       {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}});
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  metrics_logger_->LogSingleValueMetric(
+      *bob_->params().name + "_connected", test_case_name_, bob_connected_,
+      Unit::kUnitless, ImprovementDirection::kBiggerIsBetter,
+      {{MetricMetadataKey::kPeerMetadataKey, *bob_->params().name},
+       {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}});
+}
+
+Timestamp PeerConnectionE2EQualityTest::Now() const {
+  return time_controller_.GetClock()->CurrentTime();
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.h b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.h
new file mode 100644
index 0000000000..6cbf232874
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.h
@@ -0,0 +1,155 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PC_E2E_PEER_CONNECTION_QUALITY_TEST_H_
+#define TEST_PC_E2E_PEER_CONNECTION_QUALITY_TEST_H_
+
+#include <memory>
+#include <queue>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/audio_quality_analyzer_interface.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/test/time_controller.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+#include "test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.h"
+#include "test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h"
+#include "test/pc/e2e/analyzer_helper.h"
+#include "test/pc/e2e/media/media_helper.h"
+#include "test/pc/e2e/sdp/sdp_changer.h"
+#include "test/pc/e2e/test_activities_executor.h"
+#include "test/pc/e2e/test_peer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class PeerConnectionE2EQualityTest
+    : public PeerConnectionE2EQualityTestFixture {
+ public:
+  using QualityMetricsReporter =
+      PeerConnectionE2EQualityTestFixture::QualityMetricsReporter;
+
+  PeerConnectionE2EQualityTest(
+      std::string test_case_name,
+      TimeController& time_controller,
+      std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer,
+      std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer);
+  PeerConnectionE2EQualityTest(
+      std::string test_case_name,
+      TimeController& time_controller,
+      std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer,
+      std::unique_ptr<VideoQualityAnalyzerInterface> video_quality_analyzer,
+      test::MetricsLogger* metrics_logger);
+
+  ~PeerConnectionE2EQualityTest() override = default;
+
+  void ExecuteAt(TimeDelta target_time_since_start,
+                 std::function<void(TimeDelta)> func) override;
+  void ExecuteEvery(TimeDelta initial_delay_since_start,
+                    TimeDelta interval,
+                    std::function<void(TimeDelta)> func) override;
+
+  void AddQualityMetricsReporter(std::unique_ptr<QualityMetricsReporter>
+                                     quality_metrics_reporter) override;
+
+  PeerHandle* AddPeer(std::unique_ptr<PeerConfigurer> configurer) override;
+  void Run(RunParams run_params) override;
+
+  TimeDelta GetRealTestDuration() const override {
+    MutexLock lock(&lock_);
+    RTC_CHECK_NE(real_test_duration_, TimeDelta::Zero());
+    return real_test_duration_;
+  }
+
+ private:
+  class PeerHandleImpl : public PeerHandle {
+   public:
+    ~PeerHandleImpl() override = default;
+  };
+
+  // For some functionality some field trials have to be enabled, they will be
+  // enabled in Run().
+  std::string GetFieldTrials(const RunParams& run_params);
+  void OnTrackCallback(absl::string_view peer_name,
+                       VideoSubscription peer_subscription,
+                       rtc::scoped_refptr<RtpTransceiverInterface> transceiver,
+                       std::vector<VideoConfig> remote_video_configs);
+  // Have to be run on the signaling thread.
+  void SetupCallOnSignalingThread(const RunParams& run_params);
+  void TearDownCallOnSignalingThread();
+  void SetPeerCodecPreferences(TestPeer* peer);
+  std::unique_ptr<SignalingInterceptor> CreateSignalingInterceptor(
+      const RunParams& run_params);
+  void WaitUntilIceCandidatesGathered(rtc::Thread* signaling_thread);
+  void WaitUntilPeersAreConnected(rtc::Thread* signaling_thread);
+  void ExchangeOfferAnswer(SignalingInterceptor* signaling_interceptor);
+  void ExchangeIceCandidates(SignalingInterceptor* signaling_interceptor);
+  void StartVideo(
+      const std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>>&
+          sources);
+  void TearDownCall();
+  void ReportGeneralTestResults();
+  Timestamp Now() const;
+
+  TimeController& time_controller_;
+  const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  std::string test_case_name_;
+  std::unique_ptr<VideoQualityAnalyzerInjectionHelper>
+      video_quality_analyzer_injection_helper_;
+  std::unique_ptr<MediaHelper> media_helper_;
+  std::unique_ptr<EncodedImageDataPropagator> encoded_image_data_propagator_;
+  std::unique_ptr<AudioQualityAnalyzerInterface> audio_quality_analyzer_;
+  std::unique_ptr<TestActivitiesExecutor> executor_;
+  test::MetricsLogger* const metrics_logger_;
+
+  std::vector<std::unique_ptr<PeerConfigurer>> peer_configurations_;
+  std::vector<PeerHandleImpl> peer_handles_;
+
+  std::unique_ptr<TestPeer> alice_;
+  std::unique_ptr<TestPeer> bob_;
+  std::vector<std::unique_ptr<QualityMetricsReporter>>
+      quality_metrics_reporters_;
+
+  std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>>
+      alice_video_sources_;
+  std::vector<rtc::scoped_refptr<TestVideoCapturerVideoTrackSource>>
+      bob_video_sources_;
+  std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>>
+      output_video_sinks_;
+  AnalyzerHelper analyzer_helper_;
+
+  mutable Mutex lock_;
+  TimeDelta real_test_duration_ RTC_GUARDED_BY(lock_) = TimeDelta::Zero();
+
+  // Task queue, that is used for running activities during test call.
+  // This task queue will be created before call set up and will be destroyed
+  // immediately before call tear down.
+  std::unique_ptr<TaskQueueForTest> task_queue_;
+
+  bool alice_connected_ = false;
+  bool bob_connected_ = false;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_PEER_CONNECTION_QUALITY_TEST_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_metric_names_test.cc b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_metric_names_test.cc
new file mode 100644
index 0000000000..8a47e108e0
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_metric_names_test.cc
@@ -0,0 +1,1102 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <map>
+#include <memory>
+#include <string>
+
+#include "api/test/create_network_emulation_manager.h"
+#include "api/test/create_peer_connection_quality_test_frame_generator.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/metrics/stdout_metrics_exporter.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/units/time_delta.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+#include "test/pc/e2e/peer_connection_quality_test.h"
+#include "test/pc/e2e/stats_based_network_quality_metrics_reporter.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::testing::UnorderedElementsAre;
+
+using ::webrtc::test::DefaultMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Metric;
+using ::webrtc::test::MetricsExporter;
+using ::webrtc::test::StdoutMetricsExporter;
+using ::webrtc::test::Unit;
+using ::webrtc::webrtc_pc_e2e::PeerConfigurer;
+
+// Adds a peer with some audio and video (the client should not care about
+// details about audio and video configs).
+void AddDefaultAudioVideoPeer(
+    absl::string_view peer_name,
+    absl::string_view audio_stream_label,
+    absl::string_view video_stream_label,
+    const PeerNetworkDependencies& network_dependencies,
+    PeerConnectionE2EQualityTestFixture& fixture) {
+  AudioConfig audio{std::string(audio_stream_label)};
+  audio.sync_group = std::string(peer_name);
+  VideoConfig video(std::string(video_stream_label), 320, 180, 15);
+  video.sync_group = std::string(peer_name);
+  auto peer = std::make_unique<PeerConfigurer>(network_dependencies);
+  peer->SetName(peer_name);
+  peer->SetAudioConfig(std::move(audio));
+  peer->AddVideoConfig(std::move(video));
+  peer->SetVideoCodecs({VideoCodecConfig(cricket::kVp8CodecName)});
+  fixture.AddPeer(std::move(peer));
+}
+
+// Metric fields to assert on
+struct MetricValidationInfo {
+  std::string test_case;
+  std::string name;
+  Unit unit;
+  ImprovementDirection improvement_direction;
+  std::map<std::string, std::string> metadata;
+};
+
+bool operator==(const MetricValidationInfo& a, const MetricValidationInfo& b) {
+  return a.name == b.name && a.test_case == b.test_case && a.unit == b.unit &&
+         a.improvement_direction == b.improvement_direction &&
+         a.metadata == b.metadata;
+}
+
+std::ostream& operator<<(std::ostream& os, const MetricValidationInfo& m) {
+  os << "{ test_case=" << m.test_case << "; name=" << m.name
+     << "; unit=" << test::ToString(m.unit)
+     << "; improvement_direction=" << test::ToString(m.improvement_direction)
+     << "; metadata={ ";
+  for (const auto& [key, value] : m.metadata) {
+    os << "{ key=" << key << "; value=" << value << " }";
+  }
+  os << " }}";
+  return os;
+}
+
+std::vector<MetricValidationInfo> ToValidationInfo(
+    const std::vector<Metric>& metrics) {
+  std::vector<MetricValidationInfo> out;
+  for (const Metric& m : metrics) {
+    out.push_back(
+        MetricValidationInfo{.test_case = m.test_case,
+                             .name = m.name,
+                             .unit = m.unit,
+                             .improvement_direction = m.improvement_direction,
+                             .metadata = m.metric_metadata});
+  }
+  return out;
+}
+
+TEST(PeerConnectionE2EQualityTestMetricNamesTest,
+     ExportedMetricsHasCorrectNamesAndAnnotation) {
+  std::unique_ptr<NetworkEmulationManager> network_emulation =
+      CreateNetworkEmulationManager(TimeMode::kSimulated);
+  DefaultMetricsLogger metrics_logger(
+      network_emulation->time_controller()->GetClock());
+  PeerConnectionE2EQualityTest fixture(
+      "test_case", *network_emulation->time_controller(),
+      /*audio_quality_analyzer=*/nullptr, /*video_quality_analyzer=*/nullptr,
+      &metrics_logger);
+
+  EmulatedEndpoint* alice_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+
+  network_emulation->CreateRoute(
+      alice_endpoint, {network_emulation->CreateUnconstrainedEmulatedNode()},
+      bob_endpoint);
+  network_emulation->CreateRoute(
+      bob_endpoint, {network_emulation->CreateUnconstrainedEmulatedNode()},
+      alice_endpoint);
+
+  EmulatedNetworkManagerInterface* alice_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface(
+          {alice_endpoint});
+  EmulatedNetworkManagerInterface* bob_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  AddDefaultAudioVideoPeer("alice", "alice_audio", "alice_video",
+                           alice_network->network_dependencies(), fixture);
+  AddDefaultAudioVideoPeer("bob", "bob_audio", "bob_video",
+                           bob_network->network_dependencies(), fixture);
+  fixture.AddQualityMetricsReporter(
+      std::make_unique<StatsBasedNetworkQualityMetricsReporter>(
+          std::map<std::string, std::vector<EmulatedEndpoint*>>(
+              {{"alice", alice_network->endpoints()},
+               {"bob", bob_network->endpoints()}}),
+          network_emulation.get(), &metrics_logger));
+
+  // Run for at least 7 seconds, so AV-sync metrics will be collected.
+  fixture.Run(RunParams(TimeDelta::Seconds(7)));
+
+  std::vector<MetricValidationInfo> metrics =
+      ToValidationInfo(metrics_logger.GetCollectedMetrics());
+  EXPECT_THAT(
+      metrics,
+      UnorderedElementsAre(
+          // Metrics from PeerConnectionE2EQualityTest
+          MetricValidationInfo{
+              .test_case = "test_case",
+              .name = "alice_connected",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case",
+              .name = "bob_connected",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+
+          // Metrics from DefaultAudioQualityAnalyzer
+          MetricValidationInfo{
+              .test_case = "test_case/alice_audio",
+              .name = "expand_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "alice_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_audio",
+              .name = "accelerate_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "alice_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_audio",
+              .name = "preemptive_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "alice_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_audio",
+              .name = "speech_expand_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "alice_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_audio",
+              .name = "average_jitter_buffer_delay_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "alice_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_audio",
+              .name = "preferred_buffer_size_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "alice_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_audio",
+              .name = "expand_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "bob_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_audio",
+              .name = "accelerate_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "bob_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_audio",
+              .name = "preemptive_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "bob_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_audio",
+              .name = "speech_expand_rate",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "bob_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_audio",
+              .name = "average_jitter_buffer_delay_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "bob_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_audio",
+              .name = "preferred_buffer_size_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kAudioStreamMetadataKey,
+                            "bob_audio"},
+                           {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+
+          // Metrics from DefaultVideoQualityAnalyzer
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "ssim",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "transport_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "total_delay_incl_transport",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "time_between_rendered_frames",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "harmonic_framerate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "encode_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "encode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "time_between_freezes",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "freeze_time_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "pixels_per_frame",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "min_psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "decode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "receive_to_render_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "dropped_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "frames_in_flight",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "rendered_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "max_skipped",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "target_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "qp_sl0",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kSpatialLayerMetadataKey, "0"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_video",
+              .name = "actual_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "alice_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "alice"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "ssim",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "transport_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "total_delay_incl_transport",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "time_between_rendered_frames",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "harmonic_framerate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "encode_frame_rate",
+              .unit = Unit::kHertz,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "encode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "time_between_freezes",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "freeze_time_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "pixels_per_frame",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "min_psnr_dB",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "decode_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "receive_to_render_time",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "dropped_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "frames_in_flight",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "rendered_frames",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kBiggerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "max_skipped",
+              .unit = Unit::kCount,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "target_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "actual_encode_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_video",
+              .name = "qp_sl0",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kVideoStreamMetadataKey,
+                            "bob_video"},
+                           {MetricMetadataKey::kSenderMetadataKey, "bob"},
+                           {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                           {MetricMetadataKey::kSpatialLayerMetadataKey, "0"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case",
+              .name = "cpu_usage_%",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {{MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+
+          // Metrics from StatsBasedNetworkQualityMetricsReporter
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "bytes_discarded_no_receiver",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "packets_discarded_no_receiver",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "payload_bytes_received",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "payload_bytes_sent",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "bytes_sent",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "packets_sent",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "average_send_rate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "bytes_received",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "packets_received",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "average_receive_rate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "sent_packets_loss",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "bytes_discarded_no_receiver",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "packets_discarded_no_receiver",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "payload_bytes_received",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "payload_bytes_sent",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "bytes_sent",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "packets_sent",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "average_send_rate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "bytes_received",
+              .unit = Unit::kBytes,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "packets_received",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "average_receive_rate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "sent_packets_loss",
+              .unit = Unit::kUnitless,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+
+          // Metrics from VideoQualityMetricsReporter
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "available_send_bandwidth",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "transmission_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice",
+              .name = "retransmission_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "alice"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "available_send_bandwidth",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "transmission_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob",
+              .name = "retransmission_bitrate",
+              .unit = Unit::kKilobitsPerSecond,
+              .improvement_direction = ImprovementDirection::kNeitherIsBetter,
+              .metadata = {{MetricMetadataKey::kPeerMetadataKey, "bob"},
+                           {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                            "test_case"}}},
+
+          // Metrics from CrossMediaMetricsReporter
+          MetricValidationInfo{
+              .test_case = "test_case/alice_alice_audio",
+              .name = "audio_ahead_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata =
+                  {{MetricMetadataKey::kAudioStreamMetadataKey, "alice_audio"},
+                   {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                   {MetricMetadataKey::kPeerSyncGroupMetadataKey, "alice"},
+                   {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                   {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                    "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/alice_alice_video",
+              .name = "video_ahead_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata =
+                  {{MetricMetadataKey::kAudioStreamMetadataKey, "alice_video"},
+                   {MetricMetadataKey::kPeerMetadataKey, "bob"},
+                   {MetricMetadataKey::kPeerSyncGroupMetadataKey, "alice"},
+                   {MetricMetadataKey::kReceiverMetadataKey, "bob"},
+                   {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                    "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_bob_audio",
+              .name = "audio_ahead_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata =
+                  {{MetricMetadataKey::kAudioStreamMetadataKey, "bob_audio"},
+                   {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                   {MetricMetadataKey::kPeerSyncGroupMetadataKey, "bob"},
+                   {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                   {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                    "test_case"}}},
+          MetricValidationInfo{
+              .test_case = "test_case/bob_bob_video",
+              .name = "video_ahead_ms",
+              .unit = Unit::kMilliseconds,
+              .improvement_direction = ImprovementDirection::kSmallerIsBetter,
+              .metadata = {
+                  {MetricMetadataKey::kAudioStreamMetadataKey, "bob_video"},
+                  {MetricMetadataKey::kPeerMetadataKey, "alice"},
+                  {MetricMetadataKey::kPeerSyncGroupMetadataKey, "bob"},
+                  {MetricMetadataKey::kReceiverMetadataKey, "alice"},
+                  {MetricMetadataKey::kExperimentalTestNameMetadataKey,
+                   "test_case"}}}));
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_test.cc b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_test.cc
new file mode 100644
index 0000000000..066fe7d8ee
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_test.cc
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/peer_connection_quality_test.h"
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "api/test/create_network_emulation_manager.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/time_utils.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::testing::Eq;
+using ::testing::Test;
+
+using ::webrtc::webrtc_pc_e2e::PeerConfigurer;
+
+// Remove files and directories in a directory non-recursively.
+void CleanDir(absl::string_view dir, size_t expected_output_files_count) {
+  absl::optional<std::vector<std::string>> dir_content =
+      test::ReadDirectory(dir);
+  if (expected_output_files_count == 0) {
+    ASSERT_FALSE(dir_content.has_value()) << "Empty directory is expected";
+  } else {
+    ASSERT_TRUE(dir_content.has_value()) << "Test directory is empty!";
+    EXPECT_EQ(dir_content->size(), expected_output_files_count);
+    for (const auto& entry : *dir_content) {
+      if (test::DirExists(entry)) {
+        EXPECT_TRUE(test::RemoveDir(entry))
+            << "Failed to remove sub directory: " << entry;
+      } else if (test::FileExists(entry)) {
+        EXPECT_TRUE(test::RemoveFile(entry))
+            << "Failed to remove file: " << entry;
+      } else {
+        FAIL() << "Can't remove unknown file type: " << entry;
+      }
+    }
+  }
+  EXPECT_TRUE(test::RemoveDir(dir)) << "Failed to remove directory: " << dir;
+}
+
+class PeerConnectionE2EQualityTestTest : public Test {
+ protected:
+  ~PeerConnectionE2EQualityTestTest() override = default;
+
+  void SetUp() override {
+    // Create an empty temporary directory for this test.
+    test_directory_ = test::JoinFilename(
+        test::OutputPath(),
+        "TestDir_PeerConnectionE2EQualityTestTest_" +
+            std::string(
+                testing::UnitTest::GetInstance()->current_test_info()->name()));
+    test::CreateDir(test_directory_);
+  }
+
+  void TearDown() override {
+    CleanDir(test_directory_, expected_output_files_count_);
+  }
+
+  void ExpectOutputFilesCount(size_t count) {
+    expected_output_files_count_ = count;
+  }
+
+  std::string test_directory_;
+  size_t expected_output_files_count_ = 0;
+};
+
+TEST_F(PeerConnectionE2EQualityTestTest, OutputVideoIsDumpedWhenRequested) {
+  std::unique_ptr<NetworkEmulationManager> network_emulation =
+      CreateNetworkEmulationManager(TimeMode::kSimulated);
+  PeerConnectionE2EQualityTest fixture(
+      "test_case", *network_emulation->time_controller(),
+      /*audio_quality_analyzer=*/nullptr, /*video_quality_analyzer=*/nullptr,
+      test::GetGlobalMetricsLogger());
+
+  EmulatedEndpoint* alice_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+
+  network_emulation->CreateRoute(
+      alice_endpoint, {network_emulation->CreateUnconstrainedEmulatedNode()},
+      bob_endpoint);
+  network_emulation->CreateRoute(
+      bob_endpoint, {network_emulation->CreateUnconstrainedEmulatedNode()},
+      alice_endpoint);
+
+  EmulatedNetworkManagerInterface* alice_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface(
+          {alice_endpoint});
+  EmulatedNetworkManagerInterface* bob_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  VideoConfig alice_video("alice_video", 320, 180, 15);
+  alice_video.output_dump_options = VideoDumpOptions(test_directory_);
+  PeerConfigurer alice(alice_network->network_dependencies());
+  alice.SetName("alice");
+  alice.AddVideoConfig(std::move(alice_video));
+  fixture.AddPeer(std::make_unique<PeerConfigurer>(std::move(alice)));
+
+  PeerConfigurer bob(bob_network->network_dependencies());
+  bob.SetName("bob");
+  fixture.AddPeer(std::make_unique<PeerConfigurer>(std::move(bob)));
+
+  fixture.Run(RunParams(TimeDelta::Seconds(2)));
+
+  auto frame_reader = test::CreateY4mFrameReader(
+      test::JoinFilename(test_directory_, "alice_video_bob_320x180_15.y4m"));
+  EXPECT_THAT(frame_reader->num_frames(), Eq(31));  // 2 seconds 15 fps + 1
+
+  ExpectOutputFilesCount(1);
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.cc b/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.cc
new file mode 100644
index 0000000000..05372125d2
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.cc
@@ -0,0 +1,217 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/peer_params_preprocessor.h"
+
+#include <set>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/peer_network_dependencies.h"
+#include "modules/video_coding/svc/create_scalability_structure.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "rtc_base/arraysize.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+// List of default names of generic participants according to
+// https://en.wikipedia.org/wiki/Alice_and_Bob
+constexpr absl::string_view kDefaultNames[] = {"alice", "bob",  "charlie",
+                                               "david", "erin", "frank"};
+
+}  // namespace
+
+class PeerParamsPreprocessor::DefaultNamesProvider {
+ public:
+  // Caller have to ensure that default names array will outlive names provider
+  // instance.
+  explicit DefaultNamesProvider(
+      absl::string_view prefix,
+      rtc::ArrayView<const absl::string_view> default_names = {})
+      : prefix_(prefix), default_names_(default_names) {}
+
+  void MaybeSetName(absl::optional<std::string>& name) {
+    if (name.has_value()) {
+      known_names_.insert(name.value());
+    } else {
+      name = GenerateName();
+    }
+  }
+
+ private:
+  std::string GenerateName() {
+    std::string name;
+    do {
+      name = GenerateNameInternal();
+    } while (!known_names_.insert(name).second);
+    return name;
+  }
+
+  std::string GenerateNameInternal() {
+    if (counter_ < default_names_.size()) {
+      return std::string(default_names_[counter_++]);
+    }
+    return prefix_ + std::to_string(counter_++);
+  }
+
+  const std::string prefix_;
+  const rtc::ArrayView<const absl::string_view> default_names_;
+
+  std::set<std::string> known_names_;
+  size_t counter_ = 0;
+};
+
+PeerParamsPreprocessor::PeerParamsPreprocessor()
+    : peer_names_provider_(
+          std::make_unique<DefaultNamesProvider>("peer_", kDefaultNames)) {}
+PeerParamsPreprocessor::~PeerParamsPreprocessor() = default;
+
+void PeerParamsPreprocessor::SetDefaultValuesForMissingParams(
+    PeerConfigurer& peer) {
+  Params* params = peer.params();
+  ConfigurableParams* configurable_params = peer.configurable_params();
+  peer_names_provider_->MaybeSetName(params->name);
+  DefaultNamesProvider video_stream_names_provider(*params->name +
+                                                   "_auto_video_stream_label_");
+  for (VideoConfig& config : configurable_params->video_configs) {
+    video_stream_names_provider.MaybeSetName(config.stream_label);
+  }
+  if (params->audio_config) {
+    DefaultNamesProvider audio_stream_names_provider(
+        *params->name + "_auto_audio_stream_label_");
+    audio_stream_names_provider.MaybeSetName(
+        params->audio_config->stream_label);
+  }
+
+  if (params->video_codecs.empty()) {
+    params->video_codecs.push_back(VideoCodecConfig(cricket::kVp8CodecName));
+  }
+}
+
+void PeerParamsPreprocessor::ValidateParams(const PeerConfigurer& peer) {
+  const Params& p = peer.params();
+  RTC_CHECK_GT(p.video_encoder_bitrate_multiplier, 0.0);
+  // Each peer should at least support 1 video codec.
+  RTC_CHECK_GE(p.video_codecs.size(), 1);
+
+  {
+    RTC_CHECK(p.name);
+    bool inserted = peer_names_.insert(p.name.value()).second;
+    RTC_CHECK(inserted) << "Duplicate name=" << p.name.value();
+  }
+
+  // Validate that all video stream labels are unique and sync groups are
+  // valid.
+  for (const VideoConfig& video_config :
+       peer.configurable_params().video_configs) {
+    RTC_CHECK(video_config.stream_label);
+    bool inserted =
+        video_labels_.insert(video_config.stream_label.value()).second;
+    RTC_CHECK(inserted) << "Duplicate video_config.stream_label="
+                        << video_config.stream_label.value();
+
+    // TODO(bugs.webrtc.org/4762): remove this check after synchronization of
+    // more than two streams is supported.
+    if (video_config.sync_group.has_value()) {
+      bool sync_group_inserted =
+          video_sync_groups_.insert(video_config.sync_group.value()).second;
+      RTC_CHECK(sync_group_inserted)
+          << "Sync group shouldn't consist of more than two streams (one "
+             "video and one audio). Duplicate video_config.sync_group="
+          << video_config.sync_group.value();
+    }
+
+    if (video_config.simulcast_config) {
+      if (!video_config.encoding_params.empty()) {
+        RTC_CHECK_EQ(video_config.simulcast_config->simulcast_streams_count,
+                     video_config.encoding_params.size())
+            << "|encoding_params| have to be specified for each simulcast "
+            << "stream in |video_config|.";
+      }
+    } else {
+      RTC_CHECK_LE(video_config.encoding_params.size(), 1)
+          << "|encoding_params| has multiple values but simulcast is not "
+             "enabled.";
+    }
+
+    if (video_config.emulated_sfu_config) {
+      if (video_config.simulcast_config &&
+          video_config.emulated_sfu_config->target_layer_index) {
+        RTC_CHECK_LT(*video_config.emulated_sfu_config->target_layer_index,
+                     video_config.simulcast_config->simulcast_streams_count);
+      }
+      if (!video_config.encoding_params.empty()) {
+        bool is_svc = false;
+        for (const auto& encoding_param : video_config.encoding_params) {
+          if (!encoding_param.scalability_mode)
+            continue;
+
+          absl::optional<ScalabilityMode> scalability_mode =
+              ScalabilityModeFromString(*encoding_param.scalability_mode);
+          RTC_CHECK(scalability_mode) << "Unknown scalability_mode requested";
+
+          absl::optional<ScalableVideoController::StreamLayersConfig>
+              stream_layers_config =
+                  ScalabilityStructureConfig(*scalability_mode);
+          is_svc |= stream_layers_config->num_spatial_layers > 1;
+          RTC_CHECK(stream_layers_config->num_spatial_layers == 1 ||
+                    video_config.encoding_params.size() == 1)
+              << "Can't enable SVC modes with multiple spatial layers ("
+              << stream_layers_config->num_spatial_layers
+              << " layers) or simulcast ("
+              << video_config.encoding_params.size() << " layers)";
+          if (video_config.emulated_sfu_config->target_layer_index) {
+            RTC_CHECK_LT(*video_config.emulated_sfu_config->target_layer_index,
+                         stream_layers_config->num_spatial_layers);
+          }
+        }
+        if (!is_svc && video_config.emulated_sfu_config->target_layer_index) {
+          RTC_CHECK_LT(*video_config.emulated_sfu_config->target_layer_index,
+                       video_config.encoding_params.size());
+        }
+      }
+    }
+  }
+  if (p.audio_config) {
+    bool inserted =
+        audio_labels_.insert(p.audio_config->stream_label.value()).second;
+    RTC_CHECK(inserted) << "Duplicate audio_config.stream_label="
+                        << p.audio_config->stream_label.value();
+    // TODO(bugs.webrtc.org/4762): remove this check after synchronization of
+    // more than two streams is supported.
+    if (p.audio_config->sync_group.has_value()) {
+      bool sync_group_inserted =
+          audio_sync_groups_.insert(p.audio_config->sync_group.value()).second;
+      RTC_CHECK(sync_group_inserted)
+          << "Sync group shouldn't consist of more than two streams (one "
+             "video and one audio). Duplicate audio_config.sync_group="
+          << p.audio_config->sync_group.value();
+    }
+    // Check that if mode input file name specified only if mode is kFile.
+    if (p.audio_config.value().mode == AudioConfig::Mode::kGenerated) {
+      RTC_CHECK(!p.audio_config.value().input_file_name);
+    }
+    if (p.audio_config.value().mode == AudioConfig::Mode::kFile) {
+      RTC_CHECK(p.audio_config.value().input_file_name);
+      RTC_CHECK(
+          test::FileExists(p.audio_config.value().input_file_name.value()))
+          << p.audio_config.value().input_file_name.value() << " doesn't exist";
+    }
+  }
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.h b/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.h
new file mode 100644
index 0000000000..c222811546
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_params_preprocessor.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PC_E2E_PEER_PARAMS_PREPROCESSOR_H_
+#define TEST_PC_E2E_PEER_PARAMS_PREPROCESSOR_H_
+
+#include <memory>
+#include <set>
+#include <string>
+
+#include "api/test/pclf/peer_configurer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class PeerParamsPreprocessor {
+ public:
+  PeerParamsPreprocessor();
+  ~PeerParamsPreprocessor();
+
+  // Set missing params to default values if it is required:
+  //  * Generate video stream labels if some of them are missing
+  //  * Generate audio stream labels if some of them are missing
+  //  * Set video source generation mode if it is not specified
+  //  * Video codecs under test
+  void SetDefaultValuesForMissingParams(PeerConfigurer& peer);
+
+  // Validate peer's parameters, also ensure uniqueness of all video stream
+  // labels.
+  void ValidateParams(const PeerConfigurer& peer);
+
+ private:
+  class DefaultNamesProvider;
+  std::unique_ptr<DefaultNamesProvider> peer_names_provider_;
+
+  std::set<std::string> peer_names_;
+  std::set<std::string> video_labels_;
+  std::set<std::string> audio_labels_;
+  std::set<std::string> video_sync_groups_;
+  std::set<std::string> audio_sync_groups_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_PEER_PARAMS_PREPROCESSOR_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.cc b/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.cc
new file mode 100644
index 0000000000..af55f29175
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.cc
@@ -0,0 +1,601 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/sdp/sdp_changer.h"
+
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "api/jsep_session_description.h"
+#include "api/test/pclf/media_configuration.h"
+#include "media/base/media_constants.h"
+#include "p2p/base/p2p_constants.h"
+#include "pc/sdp_utils.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+std::string CodecRequiredParamsToString(
+    const std::map<std::string, std::string>& codec_required_params) {
+  rtc::StringBuilder out;
+  for (const auto& entry : codec_required_params) {
+    out << entry.first << "=" << entry.second << ";";
+  }
+  return out.str();
+}
+
+std::string SupportedCodecsToString(
+    rtc::ArrayView<const RtpCodecCapability> supported_codecs) {
+  rtc::StringBuilder out;
+  for (const auto& codec : supported_codecs) {
+    out << codec.name;
+    if (!codec.parameters.empty()) {
+      out << "(";
+      for (const auto& param : codec.parameters) {
+        out << param.first << "=" << param.second << ";";
+      }
+      out << ")";
+    }
+    out << "; ";
+  }
+  return out.str();
+}
+
+}  // namespace
+
+std::vector<RtpCodecCapability> FilterVideoCodecCapabilities(
+    rtc::ArrayView<const VideoCodecConfig> video_codecs,
+    bool use_rtx,
+    bool use_ulpfec,
+    bool use_flexfec,
+    rtc::ArrayView<const RtpCodecCapability> supported_codecs) {
+  std::vector<RtpCodecCapability> output_codecs;
+  // Find requested codecs among supported and add them to output in the order
+  // they were requested.
+  for (auto& codec_request : video_codecs) {
+    size_t size_before = output_codecs.size();
+    for (auto& codec : supported_codecs) {
+      if (codec.name != codec_request.name) {
+        continue;
+      }
+      bool parameters_matched = true;
+      for (const auto& item : codec_request.required_params) {
+        auto it = codec.parameters.find(item.first);
+        if (it == codec.parameters.end()) {
+          parameters_matched = false;
+          break;
+        }
+        if (item.second != it->second) {
+          parameters_matched = false;
+          break;
+        }
+      }
+      if (parameters_matched) {
+        output_codecs.push_back(codec);
+      }
+    }
+    RTC_CHECK_GT(output_codecs.size(), size_before)
+        << "Codec with name=" << codec_request.name << " and params {"
+        << CodecRequiredParamsToString(codec_request.required_params)
+        << "} is unsupported for this peer connection. Supported codecs are: "
+        << SupportedCodecsToString(supported_codecs);
+  }
+
+  // Add required FEC and RTX codecs to output.
+  for (auto& codec : supported_codecs) {
+    if (codec.name == cricket::kRtxCodecName && use_rtx) {
+      output_codecs.push_back(codec);
+    } else if (codec.name == cricket::kFlexfecCodecName && use_flexfec) {
+      output_codecs.push_back(codec);
+    } else if ((codec.name == cricket::kRedCodecName ||
+                codec.name == cricket::kUlpfecCodecName) &&
+               use_ulpfec) {
+      // Red and ulpfec should be enabled or disabled together.
+      output_codecs.push_back(codec);
+    }
+  }
+  return output_codecs;
+}
+
+// If offer has no simulcast video sections - do nothing.
+//
+// If offer has simulcast video sections - for each section creates
+// SimulcastSectionInfo and put it into `context_`.
+void SignalingInterceptor::FillSimulcastContext(
+    SessionDescriptionInterface* offer) {
+  for (auto& content : offer->description()->contents()) {
+    cricket::MediaContentDescription* media_desc = content.media_description();
+    if (media_desc->type() != cricket::MediaType::MEDIA_TYPE_VIDEO) {
+      continue;
+    }
+    if (media_desc->HasSimulcast()) {
+      // We support only single stream simulcast sections with rids.
+      RTC_CHECK_EQ(media_desc->mutable_streams().size(), 1);
+      RTC_CHECK(media_desc->mutable_streams()[0].has_rids());
+
+      // Create SimulcastSectionInfo for this video section.
+      SimulcastSectionInfo info(content.mid(), content.type,
+                                media_desc->mutable_streams()[0].rids());
+
+      // Set new rids basing on created SimulcastSectionInfo.
+      std::vector<cricket::RidDescription> rids;
+      cricket::SimulcastDescription simulcast_description;
+      for (std::string& rid : info.rids) {
+        rids.emplace_back(rid, cricket::RidDirection::kSend);
+        simulcast_description.send_layers().AddLayer(
+            cricket::SimulcastLayer(rid, false));
+      }
+      media_desc->mutable_streams()[0].set_rids(rids);
+      media_desc->set_simulcast_description(simulcast_description);
+
+      info.simulcast_description = media_desc->simulcast_description();
+      for (const auto& extension : media_desc->rtp_header_extensions()) {
+        if (extension.uri == RtpExtension::kMidUri) {
+          info.mid_extension = extension;
+        } else if (extension.uri == RtpExtension::kRidUri) {
+          info.rid_extension = extension;
+        } else if (extension.uri == RtpExtension::kRepairedRidUri) {
+          info.rrid_extension = extension;
+        }
+      }
+      RTC_CHECK_NE(info.rid_extension.id, 0);
+      RTC_CHECK_NE(info.mid_extension.id, 0);
+      bool transport_description_found = false;
+      for (auto& transport_info : offer->description()->transport_infos()) {
+        if (transport_info.content_name == info.mid) {
+          info.transport_description = transport_info.description;
+          transport_description_found = true;
+          break;
+        }
+      }
+      RTC_CHECK(transport_description_found);
+
+      context_.AddSimulcastInfo(info);
+    }
+  }
+}
+
+LocalAndRemoteSdp SignalingInterceptor::PatchOffer(
+    std::unique_ptr<SessionDescriptionInterface> offer,
+    const VideoCodecConfig& first_codec) {
+  for (auto& content : offer->description()->contents()) {
+    context_.mids_order.push_back(content.mid());
+    cricket::MediaContentDescription* media_desc = content.media_description();
+    if (media_desc->type() != cricket::MediaType::MEDIA_TYPE_VIDEO) {
+      continue;
+    }
+    if (content.media_description()->streams().empty()) {
+      // It means that this media section describes receive only media section
+      // in SDP.
+      RTC_CHECK_EQ(content.media_description()->direction(),
+                   RtpTransceiverDirection::kRecvOnly);
+      continue;
+    }
+    media_desc->set_conference_mode(params_.use_conference_mode);
+  }
+
+  if (!params_.stream_label_to_simulcast_streams_count.empty()) {
+    // Because simulcast enabled `params_.video_codecs` has only 1 element.
+    if (first_codec.name == cricket::kVp8CodecName) {
+      return PatchVp8Offer(std::move(offer));
+    }
+
+    if (first_codec.name == cricket::kVp9CodecName) {
+      return PatchVp9Offer(std::move(offer));
+    }
+  }
+
+  auto offer_for_remote = CloneSessionDescription(offer.get());
+  return LocalAndRemoteSdp(std::move(offer), std::move(offer_for_remote));
+}
+
+LocalAndRemoteSdp SignalingInterceptor::PatchVp8Offer(
+    std::unique_ptr<SessionDescriptionInterface> offer) {
+  FillSimulcastContext(offer.get());
+  if (!context_.HasSimulcast()) {
+    auto offer_for_remote = CloneSessionDescription(offer.get());
+    return LocalAndRemoteSdp(std::move(offer), std::move(offer_for_remote));
+  }
+
+  // Clone original offer description. We mustn't access original offer after
+  // this point.
+  std::unique_ptr<cricket::SessionDescription> desc =
+      offer->description()->Clone();
+
+  for (auto& info : context_.simulcast_infos) {
+    // For each simulcast section we have to perform:
+    //   1. Swap MID and RID header extensions
+    //   2. Remove RIDs from streams and remove SimulcastDescription
+    //   3. For each RID duplicate media section
+    cricket::ContentInfo* simulcast_content = desc->GetContentByName(info.mid);
+
+    // Now we need to prepare common prototype for "m=video" sections, in which
+    // single simulcast section will be converted. Do it before removing content
+    // because otherwise description will be deleted.
+    std::unique_ptr<cricket::MediaContentDescription> prototype_media_desc =
+        simulcast_content->media_description()->Clone();
+
+    // Remove simulcast video section from offer.
+    RTC_CHECK(desc->RemoveContentByName(simulcast_content->mid()));
+    // Clear `simulcast_content`, because now it is pointing to removed object.
+    simulcast_content = nullptr;
+
+    // Swap mid and rid extensions, so remote peer will understand rid as mid.
+    // Also remove rid extension.
+    std::vector<webrtc::RtpExtension> extensions =
+        prototype_media_desc->rtp_header_extensions();
+    for (auto ext_it = extensions.begin(); ext_it != extensions.end();) {
+      if (ext_it->uri == RtpExtension::kRidUri) {
+        // We don't need rid extension for remote peer.
+        ext_it = extensions.erase(ext_it);
+        continue;
+      }
+      if (ext_it->uri == RtpExtension::kRepairedRidUri) {
+        // We don't support RTX in simulcast.
+        ext_it = extensions.erase(ext_it);
+        continue;
+      }
+      if (ext_it->uri == RtpExtension::kMidUri) {
+        ext_it->id = info.rid_extension.id;
+      }
+      ++ext_it;
+    }
+
+    prototype_media_desc->ClearRtpHeaderExtensions();
+    prototype_media_desc->set_rtp_header_extensions(extensions);
+
+    // We support only single stream inside video section with simulcast
+    RTC_CHECK_EQ(prototype_media_desc->mutable_streams().size(), 1);
+    // This stream must have rids.
+    RTC_CHECK(prototype_media_desc->mutable_streams()[0].has_rids());
+
+    // Remove rids and simulcast description from media description.
+    prototype_media_desc->mutable_streams()[0].set_rids({});
+    prototype_media_desc->set_simulcast_description(
+        cricket::SimulcastDescription());
+
+    // For each rid add separate video section.
+    for (std::string& rid : info.rids) {
+      desc->AddContent(rid, info.media_protocol_type,
+                       prototype_media_desc->Clone());
+    }
+  }
+
+  // Now we need to add bundle line to have all media bundled together.
+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
+  for (auto& content : desc->contents()) {
+    bundle_group.AddContentName(content.mid());
+  }
+  if (desc->HasGroup(cricket::GROUP_TYPE_BUNDLE)) {
+    desc->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
+  }
+  desc->AddGroup(bundle_group);
+
+  // Update transport_infos to add TransportInfo for each new media section.
+  std::vector<cricket::TransportInfo> transport_infos = desc->transport_infos();
+  transport_infos.erase(std::remove_if(
+      transport_infos.begin(), transport_infos.end(),
+      [this](const cricket::TransportInfo& ti) {
+        // Remove transport infos that correspond to simulcast video sections.
+        return context_.simulcast_infos_by_mid.find(ti.content_name) !=
+               context_.simulcast_infos_by_mid.end();
+      }));
+  for (auto& info : context_.simulcast_infos) {
+    for (auto& rid : info.rids) {
+      transport_infos.emplace_back(rid, info.transport_description);
+    }
+  }
+  desc->set_transport_infos(transport_infos);
+
+  // Create patched offer.
+  auto patched_offer =
+      std::make_unique<JsepSessionDescription>(SdpType::kOffer);
+  patched_offer->Initialize(std::move(desc), offer->session_id(),
+                            offer->session_version());
+  return LocalAndRemoteSdp(std::move(offer), std::move(patched_offer));
+}
+
+LocalAndRemoteSdp SignalingInterceptor::PatchVp9Offer(
+    std::unique_ptr<SessionDescriptionInterface> offer) {
+  rtc::UniqueRandomIdGenerator ssrcs_generator;
+  for (auto& content : offer->description()->contents()) {
+    for (auto& stream : content.media_description()->streams()) {
+      for (auto& ssrc : stream.ssrcs) {
+        ssrcs_generator.AddKnownId(ssrc);
+      }
+    }
+  }
+
+  for (auto& content : offer->description()->contents()) {
+    if (content.media_description()->type() !=
+        cricket::MediaType::MEDIA_TYPE_VIDEO) {
+      // We are interested in only video tracks
+      continue;
+    }
+    if (content.media_description()->direction() ==
+        RtpTransceiverDirection::kRecvOnly) {
+      // If direction is receive only, then there is no media in this track from
+      // sender side, so we needn't to do anything with this track.
+      continue;
+    }
+    RTC_CHECK_EQ(content.media_description()->streams().size(), 1);
+    cricket::StreamParams& stream =
+        content.media_description()->mutable_streams()[0];
+    RTC_CHECK_EQ(stream.stream_ids().size(), 2)
+        << "Expected 2 stream ids in video stream: 1st - sync_group, 2nd - "
+           "unique label";
+    std::string stream_label = stream.stream_ids()[1];
+
+    auto it =
+        params_.stream_label_to_simulcast_streams_count.find(stream_label);
+    if (it == params_.stream_label_to_simulcast_streams_count.end()) {
+      continue;
+    }
+    int svc_layers_count = it->second;
+
+    RTC_CHECK(stream.has_ssrc_groups()) << "Only SVC with RTX is supported";
+    RTC_CHECK_EQ(stream.ssrc_groups.size(), 1)
+        << "Too many ssrc groups in the track";
+    std::vector<uint32_t> primary_ssrcs;
+    stream.GetPrimarySsrcs(&primary_ssrcs);
+    RTC_CHECK(primary_ssrcs.size() == 1);
+    for (int i = 1; i < svc_layers_count; ++i) {
+      uint32_t ssrc = ssrcs_generator.GenerateId();
+      primary_ssrcs.push_back(ssrc);
+      stream.add_ssrc(ssrc);
+      stream.AddFidSsrc(ssrc, ssrcs_generator.GenerateId());
+    }
+    stream.ssrc_groups.push_back(
+        cricket::SsrcGroup(cricket::kSimSsrcGroupSemantics, primary_ssrcs));
+  }
+  auto offer_for_remote = CloneSessionDescription(offer.get());
+  return LocalAndRemoteSdp(std::move(offer), std::move(offer_for_remote));
+}
+
+LocalAndRemoteSdp SignalingInterceptor::PatchAnswer(
+    std::unique_ptr<SessionDescriptionInterface> answer,
+    const VideoCodecConfig& first_codec) {
+  for (auto& content : answer->description()->contents()) {
+    cricket::MediaContentDescription* media_desc = content.media_description();
+    if (media_desc->type() != cricket::MediaType::MEDIA_TYPE_VIDEO) {
+      continue;
+    }
+    if (content.media_description()->direction() !=
+        RtpTransceiverDirection::kRecvOnly) {
+      continue;
+    }
+    media_desc->set_conference_mode(params_.use_conference_mode);
+  }
+
+  if (!params_.stream_label_to_simulcast_streams_count.empty()) {
+    // Because simulcast enabled `params_.video_codecs` has only 1 element.
+    if (first_codec.name == cricket::kVp8CodecName) {
+      return PatchVp8Answer(std::move(answer));
+    }
+
+    if (first_codec.name == cricket::kVp9CodecName) {
+      return PatchVp9Answer(std::move(answer));
+    }
+  }
+
+  auto answer_for_remote = CloneSessionDescription(answer.get());
+  return LocalAndRemoteSdp(std::move(answer), std::move(answer_for_remote));
+}
+
+LocalAndRemoteSdp SignalingInterceptor::PatchVp8Answer(
+    std::unique_ptr<SessionDescriptionInterface> answer) {
+  if (!context_.HasSimulcast()) {
+    auto answer_for_remote = CloneSessionDescription(answer.get());
+    return LocalAndRemoteSdp(std::move(answer), std::move(answer_for_remote));
+  }
+
+  std::unique_ptr<cricket::SessionDescription> desc =
+      answer->description()->Clone();
+
+  for (auto& info : context_.simulcast_infos) {
+    cricket::ContentInfo* simulcast_content =
+        desc->GetContentByName(info.rids[0]);
+    RTC_CHECK(simulcast_content);
+
+    // Get media description, which will be converted to simulcast answer.
+    std::unique_ptr<cricket::MediaContentDescription> media_desc =
+        simulcast_content->media_description()->Clone();
+    // Set `simulcast_content` to nullptr, because then it will be removed, so
+    // it will point to deleted object.
+    simulcast_content = nullptr;
+
+    // Remove separate media sections for simulcast streams.
+    for (auto& rid : info.rids) {
+      RTC_CHECK(desc->RemoveContentByName(rid));
+    }
+
+    // Patch `media_desc` to make it simulcast answer description.
+    // Restore mid/rid rtp header extensions
+    std::vector<webrtc::RtpExtension> extensions =
+        media_desc->rtp_header_extensions();
+    // First remove existing rid/mid header extensions.
+    extensions.erase(std::remove_if(extensions.begin(), extensions.end(),
+                                    [](const webrtc::RtpExtension& e) {
+                                      return e.uri == RtpExtension::kMidUri ||
+                                             e.uri == RtpExtension::kRidUri ||
+                                             e.uri ==
+                                                 RtpExtension::kRepairedRidUri;
+                                    }));
+
+    // Then add right ones.
+    extensions.push_back(info.mid_extension);
+    extensions.push_back(info.rid_extension);
+    // extensions.push_back(info.rrid_extension);
+    media_desc->ClearRtpHeaderExtensions();
+    media_desc->set_rtp_header_extensions(extensions);
+
+    // Add StreamParams with rids for receive.
+    RTC_CHECK_EQ(media_desc->mutable_streams().size(), 0);
+    std::vector<cricket::RidDescription> rids;
+    for (auto& rid : info.rids) {
+      rids.emplace_back(rid, cricket::RidDirection::kReceive);
+    }
+    cricket::StreamParams stream_params;
+    stream_params.set_rids(rids);
+    media_desc->mutable_streams().push_back(stream_params);
+
+    // Restore SimulcastDescription. It should correspond to one from offer,
+    // but it have to have receive layers instead of send. So we need to put
+    // send layers from offer to receive layers in answer.
+    cricket::SimulcastDescription simulcast_description;
+    for (const auto& layer : info.simulcast_description.send_layers()) {
+      simulcast_description.receive_layers().AddLayerWithAlternatives(layer);
+    }
+    media_desc->set_simulcast_description(simulcast_description);
+
+    // Add simulcast media section.
+    desc->AddContent(info.mid, info.media_protocol_type, std::move(media_desc));
+  }
+
+  desc = RestoreMediaSectionsOrder(std::move(desc));
+
+  // Now we need to add bundle line to have all media bundled together.
+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
+  for (auto& content : desc->contents()) {
+    bundle_group.AddContentName(content.mid());
+  }
+  if (desc->HasGroup(cricket::GROUP_TYPE_BUNDLE)) {
+    desc->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
+  }
+  desc->AddGroup(bundle_group);
+
+  // Fix transport_infos: it have to have single info for simulcast section.
+  std::vector<cricket::TransportInfo> transport_infos = desc->transport_infos();
+  std::map<std::string, cricket::TransportDescription>
+      mid_to_transport_description;
+  for (auto info_it = transport_infos.begin();
+       info_it != transport_infos.end();) {
+    auto it = context_.simulcast_infos_by_rid.find(info_it->content_name);
+    if (it != context_.simulcast_infos_by_rid.end()) {
+      // This transport info correspond to some extra added media section.
+      mid_to_transport_description.insert(
+          {it->second->mid, info_it->description});
+      info_it = transport_infos.erase(info_it);
+    } else {
+      ++info_it;
+    }
+  }
+  for (auto& info : context_.simulcast_infos) {
+    transport_infos.emplace_back(info.mid,
+                                 mid_to_transport_description.at(info.mid));
+  }
+  desc->set_transport_infos(transport_infos);
+
+  auto patched_answer =
+      std::make_unique<JsepSessionDescription>(SdpType::kAnswer);
+  patched_answer->Initialize(std::move(desc), answer->session_id(),
+                             answer->session_version());
+  return LocalAndRemoteSdp(std::move(answer), std::move(patched_answer));
+}
+
+std::unique_ptr<cricket::SessionDescription>
+SignalingInterceptor::RestoreMediaSectionsOrder(
+    std::unique_ptr<cricket::SessionDescription> source) {
+  std::unique_ptr<cricket::SessionDescription> out = source->Clone();
+  for (auto& mid : context_.mids_order) {
+    RTC_CHECK(out->RemoveContentByName(mid));
+  }
+  RTC_CHECK_EQ(out->contents().size(), 0);
+  for (auto& mid : context_.mids_order) {
+    cricket::ContentInfo* content = source->GetContentByName(mid);
+    RTC_CHECK(content);
+    out->AddContent(mid, content->type, content->media_description()->Clone());
+  }
+  return out;
+}
+
+LocalAndRemoteSdp SignalingInterceptor::PatchVp9Answer(
+    std::unique_ptr<SessionDescriptionInterface> answer) {
+  auto answer_for_remote = CloneSessionDescription(answer.get());
+  return LocalAndRemoteSdp(std::move(answer), std::move(answer_for_remote));
+}
+
+std::vector<std::unique_ptr<IceCandidateInterface>>
+SignalingInterceptor::PatchOffererIceCandidates(
+    rtc::ArrayView<const IceCandidateInterface* const> candidates) {
+  std::vector<std::unique_ptr<IceCandidateInterface>> out;
+  for (auto* candidate : candidates) {
+    auto simulcast_info_it =
+        context_.simulcast_infos_by_mid.find(candidate->sdp_mid());
+    if (simulcast_info_it != context_.simulcast_infos_by_mid.end()) {
+      // This is candidate for simulcast section, so it should be transformed
+      // into candidates for replicated sections. The sdpMLineIndex is set to
+      // -1 and ignored if the rid is present.
+      for (const std::string& rid : simulcast_info_it->second->rids) {
+        out.push_back(CreateIceCandidate(rid, -1, candidate->candidate()));
+      }
+    } else {
+      out.push_back(CreateIceCandidate(candidate->sdp_mid(),
+                                       candidate->sdp_mline_index(),
+                                       candidate->candidate()));
+    }
+  }
+  RTC_CHECK_GT(out.size(), 0);
+  return out;
+}
+
+std::vector<std::unique_ptr<IceCandidateInterface>>
+SignalingInterceptor::PatchAnswererIceCandidates(
+    rtc::ArrayView<const IceCandidateInterface* const> candidates) {
+  std::vector<std::unique_ptr<IceCandidateInterface>> out;
+  for (auto* candidate : candidates) {
+    auto simulcast_info_it =
+        context_.simulcast_infos_by_rid.find(candidate->sdp_mid());
+    if (simulcast_info_it != context_.simulcast_infos_by_rid.end()) {
+      // This is candidate for replicated section, created from single simulcast
+      // section, so it should be transformed into candidates for simulcast
+      // section.
+      out.push_back(CreateIceCandidate(simulcast_info_it->second->mid, 0,
+                                       candidate->candidate()));
+    } else if (!context_.simulcast_infos_by_rid.empty()) {
+      // When using simulcast and bundle, put everything on the first m-line.
+      out.push_back(CreateIceCandidate("", 0, candidate->candidate()));
+    } else {
+      out.push_back(CreateIceCandidate(candidate->sdp_mid(),
+                                       candidate->sdp_mline_index(),
+                                       candidate->candidate()));
+    }
+  }
+  RTC_CHECK_GT(out.size(), 0);
+  return out;
+}
+
+SignalingInterceptor::SimulcastSectionInfo::SimulcastSectionInfo(
+    const std::string& mid,
+    cricket::MediaProtocolType media_protocol_type,
+    const std::vector<cricket::RidDescription>& rids_desc)
+    : mid(mid), media_protocol_type(media_protocol_type) {
+  for (auto& rid : rids_desc) {
+    rids.push_back(rid.rid);
+  }
+}
+
+void SignalingInterceptor::SignalingContext::AddSimulcastInfo(
+    const SimulcastSectionInfo& info) {
+  simulcast_infos.push_back(info);
+  bool inserted =
+      simulcast_infos_by_mid.insert({info.mid, &simulcast_infos.back()}).second;
+  RTC_CHECK(inserted);
+  for (auto& rid : info.rids) {
+    inserted =
+        simulcast_infos_by_rid.insert({rid, &simulcast_infos.back()}).second;
+    RTC_CHECK(inserted);
+  }
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.h b/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.h
new file mode 100644
index 0000000000..6f68d03f52
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.h
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_SDP_SDP_CHANGER_H_
+#define TEST_PC_E2E_SDP_SDP_CHANGER_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/jsep.h"
+#include "api/rtp_parameters.h"
+#include "api/test/pclf/media_configuration.h"
+#include "media/base/rid_description.h"
+#include "pc/session_description.h"
+#include "pc/simulcast_description.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Creates list of capabilities, which can be set on RtpTransceiverInterface via
+// RtpTransceiverInterface::SetCodecPreferences(...) to negotiate use of codecs
+// from list of `supported_codecs` which will match `video_codecs`. If flags
+// `ulpfec` or `flexfec` set to true corresponding FEC codec will be added.
+// FEC and RTX codecs will be added after required codecs.
+//
+// All codecs will be added only if they exists in the list of
+// `supported_codecs`. If multiple codecs from this list will match
+// `video_codecs`, then all of them will be added to the output
+// vector and they will be added in the same order, as they were in
+// `supported_codecs`.
+std::vector<RtpCodecCapability> FilterVideoCodecCapabilities(
+    rtc::ArrayView<const VideoCodecConfig> video_codecs,
+    bool use_rtx,
+    bool use_ulpfec,
+    bool use_flexfec,
+    rtc::ArrayView<const RtpCodecCapability> supported_codecs);
+
+struct LocalAndRemoteSdp {
+  LocalAndRemoteSdp(std::unique_ptr<SessionDescriptionInterface> local_sdp,
+                    std::unique_ptr<SessionDescriptionInterface> remote_sdp)
+      : local_sdp(std::move(local_sdp)), remote_sdp(std::move(remote_sdp)) {}
+
+  // Sdp, that should be as local description on the peer, that created it.
+  std::unique_ptr<SessionDescriptionInterface> local_sdp;
+  // Sdp, that should be set as remote description on the peer opposite to the
+  // one, who created it.
+  std::unique_ptr<SessionDescriptionInterface> remote_sdp;
+};
+
+struct PatchingParams {
+  PatchingParams(
+      bool use_conference_mode,
+      std::map<std::string, int> stream_label_to_simulcast_streams_count)
+      : use_conference_mode(use_conference_mode),
+        stream_label_to_simulcast_streams_count(
+            stream_label_to_simulcast_streams_count) {}
+
+  bool use_conference_mode;
+  std::map<std::string, int> stream_label_to_simulcast_streams_count;
+};
+
+class SignalingInterceptor {
+ public:
+  explicit SignalingInterceptor(PatchingParams params) : params_(params) {}
+
+  LocalAndRemoteSdp PatchOffer(
+      std::unique_ptr<SessionDescriptionInterface> offer,
+      const VideoCodecConfig& first_codec);
+  LocalAndRemoteSdp PatchAnswer(
+      std::unique_ptr<SessionDescriptionInterface> answer,
+      const VideoCodecConfig& first_codec);
+
+  std::vector<std::unique_ptr<IceCandidateInterface>> PatchOffererIceCandidates(
+      rtc::ArrayView<const IceCandidateInterface* const> candidates);
+  std::vector<std::unique_ptr<IceCandidateInterface>>
+  PatchAnswererIceCandidates(
+      rtc::ArrayView<const IceCandidateInterface* const> candidates);
+
+ private:
+  // Contains information about simulcast section, that is required to perform
+  // modified offer/answer and ice candidates exchange.
+  struct SimulcastSectionInfo {
+    SimulcastSectionInfo(const std::string& mid,
+                         cricket::MediaProtocolType media_protocol_type,
+                         const std::vector<cricket::RidDescription>& rids_desc);
+
+    const std::string mid;
+    const cricket::MediaProtocolType media_protocol_type;
+    std::vector<std::string> rids;
+    cricket::SimulcastDescription simulcast_description;
+    webrtc::RtpExtension mid_extension;
+    webrtc::RtpExtension rid_extension;
+    webrtc::RtpExtension rrid_extension;
+    cricket::TransportDescription transport_description;
+  };
+
+  struct SignalingContext {
+    SignalingContext() = default;
+    // SignalingContext is not copyable and movable.
+    SignalingContext(SignalingContext&) = delete;
+    SignalingContext& operator=(SignalingContext&) = delete;
+    SignalingContext(SignalingContext&&) = delete;
+    SignalingContext& operator=(SignalingContext&&) = delete;
+
+    void AddSimulcastInfo(const SimulcastSectionInfo& info);
+    bool HasSimulcast() const { return !simulcast_infos.empty(); }
+
+    std::vector<SimulcastSectionInfo> simulcast_infos;
+    std::map<std::string, SimulcastSectionInfo*> simulcast_infos_by_mid;
+    std::map<std::string, SimulcastSectionInfo*> simulcast_infos_by_rid;
+
+    std::vector<std::string> mids_order;
+  };
+
+  LocalAndRemoteSdp PatchVp8Offer(
+      std::unique_ptr<SessionDescriptionInterface> offer);
+  LocalAndRemoteSdp PatchVp9Offer(
+      std::unique_ptr<SessionDescriptionInterface> offer);
+  LocalAndRemoteSdp PatchVp8Answer(
+      std::unique_ptr<SessionDescriptionInterface> answer);
+  LocalAndRemoteSdp PatchVp9Answer(
+      std::unique_ptr<SessionDescriptionInterface> answer);
+
+  void FillSimulcastContext(SessionDescriptionInterface* offer);
+  std::unique_ptr<cricket::SessionDescription> RestoreMediaSectionsOrder(
+      std::unique_ptr<cricket::SessionDescription> source);
+
+  PatchingParams params_;
+  SignalingContext context_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_SDP_SDP_CHANGER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
new file mode 100644
index 0000000000..65dca5b518
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
@@ -0,0 +1,592 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/stats_based_network_quality_metrics_reporter.h"
+
+#include <cstdint>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <type_traits>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/stats/rtc_stats.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/test/metrics/metric.h"
+#include "api/test/network_emulation/network_emulation_interfaces.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/ip_address.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+using NetworkLayerStats =
+    StatsBasedNetworkQualityMetricsReporter::NetworkLayerStats;
+
+constexpr TimeDelta kStatsWaitTimeout = TimeDelta::Seconds(1);
+
+// Field trial which controls whether to report standard-compliant bytes
+// sent/received per stream.  If enabled, padding and headers are not included
+// in bytes sent or received.
+constexpr char kUseStandardBytesStats[] = "WebRTC-UseStandardBytesStats";
+
+EmulatedNetworkStats PopulateStats(std::vector<EmulatedEndpoint*> endpoints,
+                                   NetworkEmulationManager* network_emulation) {
+  rtc::Event stats_loaded;
+  EmulatedNetworkStats stats;
+  network_emulation->GetStats(endpoints, [&](EmulatedNetworkStats s) {
+    stats = std::move(s);
+    stats_loaded.Set();
+  });
+  bool stats_received = stats_loaded.Wait(kStatsWaitTimeout);
+  RTC_CHECK(stats_received);
+  return stats;
+}
+
+std::map<rtc::IPAddress, std::string> PopulateIpToPeer(
+    const std::map<std::string, std::vector<EmulatedEndpoint*>>&
+        peer_endpoints) {
+  std::map<rtc::IPAddress, std::string> out;
+  for (const auto& entry : peer_endpoints) {
+    for (const EmulatedEndpoint* const endpoint : entry.second) {
+      RTC_CHECK(out.find(endpoint->GetPeerLocalAddress()) == out.end())
+          << "Two peers can't share the same endpoint";
+      out.emplace(endpoint->GetPeerLocalAddress(), entry.first);
+    }
+  }
+  return out;
+}
+
+// Accumulates emulated network stats being executed on the network thread.
+// When all stats are collected stores it in thread safe variable.
+class EmulatedNetworkStatsAccumulator {
+ public:
+  // `expected_stats_count` - the number of calls to
+  // AddEndpointStats/AddUplinkStats/AddDownlinkStats the accumulator is going
+  // to wait. If called more than expected, the program will crash.
+  explicit EmulatedNetworkStatsAccumulator(size_t expected_stats_count)
+      : not_collected_stats_count_(expected_stats_count) {
+    RTC_DCHECK_GE(not_collected_stats_count_, 0);
+    if (not_collected_stats_count_ == 0) {
+      all_stats_collected_.Set();
+    }
+    sequence_checker_.Detach();
+  }
+
+  // Has to be executed on network thread.
+  void AddEndpointStats(std::string peer_name, EmulatedNetworkStats stats) {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    n_stats_[peer_name].endpoints_stats = std::move(stats);
+    DecrementNotCollectedStatsCount();
+  }
+
+  // Has to be executed on network thread.
+  void AddUplinkStats(std::string peer_name, EmulatedNetworkNodeStats stats) {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    n_stats_[peer_name].uplink_stats = std::move(stats);
+    DecrementNotCollectedStatsCount();
+  }
+
+  // Has to be executed on network thread.
+  void AddDownlinkStats(std::string peer_name, EmulatedNetworkNodeStats stats) {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    n_stats_[peer_name].downlink_stats = std::move(stats);
+    DecrementNotCollectedStatsCount();
+  }
+
+  // Can be executed on any thread.
+  // Returns true if count down was completed and false if timeout elapsed
+  // before.
+  bool Wait(TimeDelta timeout) { return all_stats_collected_.Wait(timeout); }
+
+  // Can be called once. Returns all collected stats by moving underlying
+  // object.
+  std::map<std::string, NetworkLayerStats> ReleaseStats() {
+    RTC_DCHECK(!stats_released_);
+    stats_released_ = true;
+    MutexLock lock(&mutex_);
+    return std::move(stats_);
+  }
+
+ private:
+  void DecrementNotCollectedStatsCount() {
+    RTC_DCHECK_RUN_ON(&sequence_checker_);
+    RTC_CHECK_GT(not_collected_stats_count_, 0)
+        << "All stats are already collected";
+    not_collected_stats_count_--;
+    if (not_collected_stats_count_ == 0) {
+      MutexLock lock(&mutex_);
+      stats_ = std::move(n_stats_);
+      all_stats_collected_.Set();
+    }
+  }
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+  size_t not_collected_stats_count_ RTC_GUARDED_BY(sequence_checker_);
+  // Collected on the network thread. Moved into `stats_` after all stats are
+  // collected.
+  std::map<std::string, NetworkLayerStats> n_stats_
+      RTC_GUARDED_BY(sequence_checker_);
+
+  rtc::Event all_stats_collected_;
+  Mutex mutex_;
+  std::map<std::string, NetworkLayerStats> stats_ RTC_GUARDED_BY(mutex_);
+  bool stats_released_ = false;
+};
+
+}  // namespace
+
+StatsBasedNetworkQualityMetricsReporter::
+    StatsBasedNetworkQualityMetricsReporter(
+        std::map<std::string, std::vector<EmulatedEndpoint*>> peer_endpoints,
+        NetworkEmulationManager* network_emulation,
+        test::MetricsLogger* metrics_logger)
+    : collector_(std::move(peer_endpoints), network_emulation),
+      clock_(network_emulation->time_controller()->GetClock()),
+      metrics_logger_(metrics_logger) {
+  RTC_CHECK(metrics_logger_);
+}
+
+StatsBasedNetworkQualityMetricsReporter::NetworkLayerStatsCollector::
+    NetworkLayerStatsCollector(
+        std::map<std::string, std::vector<EmulatedEndpoint*>> peer_endpoints,
+        NetworkEmulationManager* network_emulation)
+    : peer_endpoints_(std::move(peer_endpoints)),
+      ip_to_peer_(PopulateIpToPeer(peer_endpoints_)),
+      network_emulation_(network_emulation) {}
+
+void StatsBasedNetworkQualityMetricsReporter::NetworkLayerStatsCollector::
+    Start() {
+  MutexLock lock(&mutex_);
+  // Check that network stats are clean before test execution.
+  for (const auto& entry : peer_endpoints_) {
+    EmulatedNetworkStats stats =
+        PopulateStats(entry.second, network_emulation_);
+    RTC_CHECK_EQ(stats.overall_outgoing_stats.packets_sent, 0);
+    RTC_CHECK_EQ(stats.overall_incoming_stats.packets_received, 0);
+  }
+}
+
+void StatsBasedNetworkQualityMetricsReporter::NetworkLayerStatsCollector::
+    AddPeer(absl::string_view peer_name,
+            std::vector<EmulatedEndpoint*> endpoints,
+            std::vector<EmulatedNetworkNode*> uplink,
+            std::vector<EmulatedNetworkNode*> downlink) {
+  MutexLock lock(&mutex_);
+  // When new peer is added not in the constructor, don't check if it has empty
+  // stats, because their endpoint could be used for traffic before.
+  peer_endpoints_.emplace(peer_name, std::move(endpoints));
+  peer_uplinks_.emplace(peer_name, std::move(uplink));
+  peer_downlinks_.emplace(peer_name, std::move(downlink));
+  for (const EmulatedEndpoint* const endpoint : endpoints) {
+    RTC_CHECK(ip_to_peer_.find(endpoint->GetPeerLocalAddress()) ==
+              ip_to_peer_.end())
+        << "Two peers can't share the same endpoint";
+    ip_to_peer_.emplace(endpoint->GetPeerLocalAddress(), peer_name);
+  }
+}
+
+std::map<std::string, NetworkLayerStats>
+StatsBasedNetworkQualityMetricsReporter::NetworkLayerStatsCollector::
+    GetStats() {
+  MutexLock lock(&mutex_);
+  EmulatedNetworkStatsAccumulator stats_accumulator(
+      peer_endpoints_.size() + peer_uplinks_.size() + peer_downlinks_.size());
+  for (const auto& entry : peer_endpoints_) {
+    network_emulation_->GetStats(
+        entry.second, [&stats_accumulator,
+                       peer = entry.first](EmulatedNetworkStats s) mutable {
+          stats_accumulator.AddEndpointStats(std::move(peer), std::move(s));
+        });
+  }
+  for (const auto& entry : peer_uplinks_) {
+    network_emulation_->GetStats(
+        entry.second, [&stats_accumulator,
+                       peer = entry.first](EmulatedNetworkNodeStats s) mutable {
+          stats_accumulator.AddUplinkStats(std::move(peer), std::move(s));
+        });
+  }
+  for (const auto& entry : peer_downlinks_) {
+    network_emulation_->GetStats(
+        entry.second, [&stats_accumulator,
+                       peer = entry.first](EmulatedNetworkNodeStats s) mutable {
+          stats_accumulator.AddDownlinkStats(std::move(peer), std::move(s));
+        });
+  }
+  bool stats_collected = stats_accumulator.Wait(kStatsWaitTimeout);
+  RTC_CHECK(stats_collected);
+  std::map<std::string, NetworkLayerStats> peer_to_stats =
+      stats_accumulator.ReleaseStats();
+  std::map<std::string, std::vector<std::string>> sender_to_receivers;
+  for (const auto& entry : peer_endpoints_) {
+    const std::string& peer_name = entry.first;
+    const NetworkLayerStats& stats = peer_to_stats[peer_name];
+    for (const auto& income_stats_entry :
+         stats.endpoints_stats.incoming_stats_per_source) {
+      const rtc::IPAddress& source_ip = income_stats_entry.first;
+      auto it = ip_to_peer_.find(source_ip);
+      if (it == ip_to_peer_.end()) {
+        // Source IP is unknown for this collector, so will be skipped.
+        continue;
+      }
+      sender_to_receivers[it->second].push_back(peer_name);
+    }
+  }
+  for (auto& entry : peer_to_stats) {
+    const std::vector<std::string>& receivers =
+        sender_to_receivers[entry.first];
+    entry.second.receivers =
+        std::set<std::string>(receivers.begin(), receivers.end());
+  }
+  return peer_to_stats;
+}
+
+void StatsBasedNetworkQualityMetricsReporter::AddPeer(
+    absl::string_view peer_name,
+    std::vector<EmulatedEndpoint*> endpoints) {
+  collector_.AddPeer(peer_name, std::move(endpoints), /*uplink=*/{},
+                     /*downlink=*/{});
+}
+
+void StatsBasedNetworkQualityMetricsReporter::AddPeer(
+    absl::string_view peer_name,
+    std::vector<EmulatedEndpoint*> endpoints,
+    std::vector<EmulatedNetworkNode*> uplink,
+    std::vector<EmulatedNetworkNode*> downlink) {
+  collector_.AddPeer(peer_name, std::move(endpoints), std::move(uplink),
+                     std::move(downlink));
+}
+
+void StatsBasedNetworkQualityMetricsReporter::Start(
+    absl::string_view test_case_name,
+    const TrackIdStreamInfoMap* reporter_helper) {
+  test_case_name_ = std::string(test_case_name);
+  collector_.Start();
+  start_time_ = clock_->CurrentTime();
+}
+
+void StatsBasedNetworkQualityMetricsReporter::OnStatsReports(
+    absl::string_view pc_label,
+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
+  PCStats cur_stats;
+
+  auto inbound_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
+  for (const auto& stat : inbound_stats) {
+    cur_stats.payload_received +=
+        DataSize::Bytes(stat->bytes_received.ValueOrDefault(0ul) +
+                        stat->header_bytes_received.ValueOrDefault(0ul));
+  }
+
+  auto outbound_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
+  for (const auto& stat : outbound_stats) {
+    cur_stats.payload_sent +=
+        DataSize::Bytes(stat->bytes_sent.ValueOrDefault(0ul) +
+                        stat->header_bytes_sent.ValueOrDefault(0ul));
+  }
+
+  auto candidate_pairs_stats = report->GetStatsOfType<RTCTransportStats>();
+  for (const auto& stat : candidate_pairs_stats) {
+    cur_stats.total_received +=
+        DataSize::Bytes(stat->bytes_received.ValueOrDefault(0ul));
+    cur_stats.total_sent +=
+        DataSize::Bytes(stat->bytes_sent.ValueOrDefault(0ul));
+    cur_stats.packets_received += stat->packets_received.ValueOrDefault(0ul);
+    cur_stats.packets_sent += stat->packets_sent.ValueOrDefault(0ul);
+  }
+
+  MutexLock lock(&mutex_);
+  pc_stats_[std::string(pc_label)] = cur_stats;
+}
+
+void StatsBasedNetworkQualityMetricsReporter::StopAndReportResults() {
+  Timestamp end_time = clock_->CurrentTime();
+
+  if (!webrtc::field_trial::IsEnabled(kUseStandardBytesStats)) {
+    RTC_LOG(LS_ERROR)
+        << "Non-standard GetStats; \"payload\" counts include RTP headers";
+  }
+
+  std::map<std::string, NetworkLayerStats> stats = collector_.GetStats();
+  for (const auto& entry : stats) {
+    LogNetworkLayerStats(entry.first, entry.second);
+  }
+  MutexLock lock(&mutex_);
+  for (const auto& pair : pc_stats_) {
+    auto it = stats.find(pair.first);
+    RTC_CHECK(it != stats.end())
+        << "Peer name used for PeerConnection stats collection and peer name "
+           "used for endpoints naming doesn't match. No endpoints found for "
+           "peer "
+        << pair.first;
+    const NetworkLayerStats& network_layer_stats = it->second;
+    int64_t total_packets_received = 0;
+    bool found = false;
+    for (const auto& dest_peer : network_layer_stats.receivers) {
+      auto pc_stats_it = pc_stats_.find(dest_peer);
+      if (pc_stats_it == pc_stats_.end()) {
+        continue;
+      }
+      found = true;
+      total_packets_received += pc_stats_it->second.packets_received;
+    }
+    int64_t packet_loss = -1;
+    if (found) {
+      packet_loss = pair.second.packets_sent - total_packets_received;
+    }
+    ReportStats(pair.first, pair.second, network_layer_stats, packet_loss,
+                end_time);
+  }
+}
+
+void StatsBasedNetworkQualityMetricsReporter::ReportStats(
+    const std::string& pc_label,
+    const PCStats& pc_stats,
+    const NetworkLayerStats& network_layer_stats,
+    int64_t packet_loss,
+    const Timestamp& end_time) {
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  std::map<std::string, std::string> metric_metadata{
+      {MetricMetadataKey::kPeerMetadataKey, pc_label},
+      {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}};
+  metrics_logger_->LogSingleValueMetric(
+      "bytes_discarded_no_receiver", GetTestCaseName(pc_label),
+      network_layer_stats.endpoints_stats.overall_incoming_stats
+          .bytes_discarded_no_receiver.bytes(),
+      Unit::kBytes, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "packets_discarded_no_receiver", GetTestCaseName(pc_label),
+      network_layer_stats.endpoints_stats.overall_incoming_stats
+          .packets_discarded_no_receiver,
+      Unit::kUnitless, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+
+  metrics_logger_->LogSingleValueMetric(
+      "payload_bytes_received", GetTestCaseName(pc_label),
+      pc_stats.payload_received.bytes(), Unit::kBytes,
+      ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "payload_bytes_sent", GetTestCaseName(pc_label),
+      pc_stats.payload_sent.bytes(), Unit::kBytes,
+      ImprovementDirection::kNeitherIsBetter, metric_metadata);
+
+  metrics_logger_->LogSingleValueMetric(
+      "bytes_sent", GetTestCaseName(pc_label), pc_stats.total_sent.bytes(),
+      Unit::kBytes, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "packets_sent", GetTestCaseName(pc_label), pc_stats.packets_sent,
+      Unit::kUnitless, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "average_send_rate", GetTestCaseName(pc_label),
+      (pc_stats.total_sent / (end_time - start_time_)).kbps<double>(),
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter,
+      metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "bytes_received", GetTestCaseName(pc_label),
+      pc_stats.total_received.bytes(), Unit::kBytes,
+      ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "packets_received", GetTestCaseName(pc_label), pc_stats.packets_received,
+      Unit::kUnitless, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "average_receive_rate", GetTestCaseName(pc_label),
+      (pc_stats.total_received / (end_time - start_time_)).kbps<double>(),
+      Unit::kKilobitsPerSecond, ImprovementDirection::kNeitherIsBetter,
+      metric_metadata);
+  metrics_logger_->LogSingleValueMetric(
+      "sent_packets_loss", GetTestCaseName(pc_label), packet_loss,
+      Unit::kUnitless, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+}
+
+std::string StatsBasedNetworkQualityMetricsReporter::GetTestCaseName(
+    absl::string_view network_label) const {
+  rtc::StringBuilder builder;
+  builder << test_case_name_ << "/" << network_label.data();
+  return builder.str();
+}
+
+void StatsBasedNetworkQualityMetricsReporter::LogNetworkLayerStats(
+    const std::string& peer_name,
+    const NetworkLayerStats& stats) const {
+  DataRate average_send_rate =
+      stats.endpoints_stats.overall_outgoing_stats.packets_sent >= 2
+          ? stats.endpoints_stats.overall_outgoing_stats.AverageSendRate()
+          : DataRate::Zero();
+  DataRate average_receive_rate =
+      stats.endpoints_stats.overall_incoming_stats.packets_received >= 2
+          ? stats.endpoints_stats.overall_incoming_stats.AverageReceiveRate()
+          : DataRate::Zero();
+  // TODO(bugs.webrtc.org/14757): Remove kExperimentalTestNameMetadataKey.
+  std::map<std::string, std::string> metric_metadata{
+      {MetricMetadataKey::kPeerMetadataKey, peer_name},
+      {MetricMetadataKey::kExperimentalTestNameMetadataKey, test_case_name_}};
+  rtc::StringBuilder log;
+  log << "Raw network layer statistic for [" << peer_name << "]:\n"
+      << "Local IPs:\n";
+  for (size_t i = 0; i < stats.endpoints_stats.local_addresses.size(); ++i) {
+    log << "  " << stats.endpoints_stats.local_addresses[i].ToString() << "\n";
+  }
+  if (!stats.endpoints_stats.overall_outgoing_stats.sent_packets_size
+           .IsEmpty()) {
+    metrics_logger_->LogMetric(
+        "sent_packets_size", GetTestCaseName(peer_name),
+        stats.endpoints_stats.overall_outgoing_stats.sent_packets_size,
+        Unit::kBytes, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+  if (!stats.endpoints_stats.overall_incoming_stats.received_packets_size
+           .IsEmpty()) {
+    metrics_logger_->LogMetric(
+        "received_packets_size", GetTestCaseName(peer_name),
+        stats.endpoints_stats.overall_incoming_stats.received_packets_size,
+        Unit::kBytes, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+  if (!stats.endpoints_stats.overall_incoming_stats
+           .packets_discarded_no_receiver_size.IsEmpty()) {
+    metrics_logger_->LogMetric(
+        "packets_discarded_no_receiver_size", GetTestCaseName(peer_name),
+        stats.endpoints_stats.overall_incoming_stats
+            .packets_discarded_no_receiver_size,
+        Unit::kBytes, ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+  if (!stats.endpoints_stats.sent_packets_queue_wait_time_us.IsEmpty()) {
+    metrics_logger_->LogMetric(
+        "sent_packets_queue_wait_time_us", GetTestCaseName(peer_name),
+        stats.endpoints_stats.sent_packets_queue_wait_time_us, Unit::kUnitless,
+        ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+
+  log << "Send statistic:\n"
+      << "  packets: "
+      << stats.endpoints_stats.overall_outgoing_stats.packets_sent << " bytes: "
+      << stats.endpoints_stats.overall_outgoing_stats.bytes_sent.bytes()
+      << " avg_rate (bytes/sec): " << average_send_rate.bytes_per_sec()
+      << " avg_rate (bps): " << average_send_rate.bps() << "\n"
+      << "Send statistic per destination:\n";
+
+  for (const auto& entry :
+       stats.endpoints_stats.outgoing_stats_per_destination) {
+    DataRate source_average_send_rate = entry.second.packets_sent >= 2
+                                            ? entry.second.AverageSendRate()
+                                            : DataRate::Zero();
+    log << "(" << entry.first.ToString() << "):\n"
+        << "  packets: " << entry.second.packets_sent
+        << " bytes: " << entry.second.bytes_sent.bytes()
+        << " avg_rate (bytes/sec): " << source_average_send_rate.bytes_per_sec()
+        << " avg_rate (bps): " << source_average_send_rate.bps() << "\n";
+    if (!entry.second.sent_packets_size.IsEmpty()) {
+      metrics_logger_->LogMetric(
+          "sent_packets_size",
+          GetTestCaseName(peer_name + "/" + entry.first.ToString()),
+          entry.second.sent_packets_size, Unit::kBytes,
+          ImprovementDirection::kNeitherIsBetter, metric_metadata);
+    }
+  }
+
+  if (!stats.uplink_stats.packet_transport_time.IsEmpty()) {
+    log << "[Debug stats] packet_transport_time=("
+        << stats.uplink_stats.packet_transport_time.GetAverage() << ", "
+        << stats.uplink_stats.packet_transport_time.GetStandardDeviation()
+        << ")\n";
+    metrics_logger_->LogMetric(
+        "uplink_packet_transport_time", GetTestCaseName(peer_name),
+        stats.uplink_stats.packet_transport_time, Unit::kMilliseconds,
+        ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+  if (!stats.uplink_stats.size_to_packet_transport_time.IsEmpty()) {
+    log << "[Debug stats] size_to_packet_transport_time=("
+        << stats.uplink_stats.size_to_packet_transport_time.GetAverage() << ", "
+        << stats.uplink_stats.size_to_packet_transport_time
+               .GetStandardDeviation()
+        << ")\n";
+    metrics_logger_->LogMetric(
+        "uplink_size_to_packet_transport_time", GetTestCaseName(peer_name),
+        stats.uplink_stats.size_to_packet_transport_time, Unit::kUnitless,
+        ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+
+  log << "Receive statistic:\n"
+      << "  packets: "
+      << stats.endpoints_stats.overall_incoming_stats.packets_received
+      << " bytes: "
+      << stats.endpoints_stats.overall_incoming_stats.bytes_received.bytes()
+      << " avg_rate (bytes/sec): " << average_receive_rate.bytes_per_sec()
+      << " avg_rate (bps): " << average_receive_rate.bps() << "\n"
+      << "Receive statistic per source:\n";
+
+  for (const auto& entry : stats.endpoints_stats.incoming_stats_per_source) {
+    DataRate source_average_receive_rate =
+        entry.second.packets_received >= 2 ? entry.second.AverageReceiveRate()
+                                           : DataRate::Zero();
+    log << "(" << entry.first.ToString() << "):\n"
+        << "  packets: " << entry.second.packets_received
+        << " bytes: " << entry.second.bytes_received.bytes()
+        << " avg_rate (bytes/sec): "
+        << source_average_receive_rate.bytes_per_sec()
+        << " avg_rate (bps): " << source_average_receive_rate.bps() << "\n";
+    if (!entry.second.received_packets_size.IsEmpty()) {
+      metrics_logger_->LogMetric(
+          "received_packets_size",
+          GetTestCaseName(peer_name + "/" + entry.first.ToString()),
+          entry.second.received_packets_size, Unit::kBytes,
+          ImprovementDirection::kNeitherIsBetter, metric_metadata);
+    }
+    if (!entry.second.packets_discarded_no_receiver_size.IsEmpty()) {
+      metrics_logger_->LogMetric(
+          "packets_discarded_no_receiver_size",
+          GetTestCaseName(peer_name + "/" + entry.first.ToString()),
+          entry.second.packets_discarded_no_receiver_size, Unit::kBytes,
+          ImprovementDirection::kNeitherIsBetter, metric_metadata);
+    }
+  }
+  if (!stats.downlink_stats.packet_transport_time.IsEmpty()) {
+    log << "[Debug stats] packet_transport_time=("
+        << stats.downlink_stats.packet_transport_time.GetAverage() << ", "
+        << stats.downlink_stats.packet_transport_time.GetStandardDeviation()
+        << ")\n";
+    metrics_logger_->LogMetric(
+        "downlink_packet_transport_time", GetTestCaseName(peer_name),
+        stats.downlink_stats.packet_transport_time, Unit::kMilliseconds,
+        ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+  if (!stats.downlink_stats.size_to_packet_transport_time.IsEmpty()) {
+    log << "[Debug stats] size_to_packet_transport_time=("
+        << stats.downlink_stats.size_to_packet_transport_time.GetAverage()
+        << ", "
+        << stats.downlink_stats.size_to_packet_transport_time
+               .GetStandardDeviation()
+        << ")\n";
+    metrics_logger_->LogMetric(
+        "downlink_size_to_packet_transport_time", GetTestCaseName(peer_name),
+        stats.downlink_stats.size_to_packet_transport_time, Unit::kUnitless,
+        ImprovementDirection::kNeitherIsBetter, metric_metadata);
+  }
+
+  RTC_LOG(LS_INFO) << log.str();
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
new file mode 100644
index 0000000000..60daf40c8c
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_STATS_BASED_NETWORK_QUALITY_METRICS_REPORTER_H_
+#define TEST_PC_E2E_STATS_BASED_NETWORK_QUALITY_METRICS_REPORTER_H_
+
+#include <cstdint>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/network_emulation/network_emulation_interfaces.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/ip_address.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// TODO(titovartem): make this class testable and add tests.
+class StatsBasedNetworkQualityMetricsReporter
+    : public PeerConnectionE2EQualityTestFixture::QualityMetricsReporter {
+ public:
+  // Emulated network layer stats for single peer.
+  struct NetworkLayerStats {
+    EmulatedNetworkStats endpoints_stats;
+    EmulatedNetworkNodeStats uplink_stats;
+    EmulatedNetworkNodeStats downlink_stats;
+    std::set<std::string> receivers;
+  };
+
+  // `networks` map peer name to network to report network layer stability stats
+  // and to log network layer metrics.
+  StatsBasedNetworkQualityMetricsReporter(
+      std::map<std::string, std::vector<EmulatedEndpoint*>> peer_endpoints,
+      NetworkEmulationManager* network_emulation,
+      test::MetricsLogger* metrics_logger);
+  ~StatsBasedNetworkQualityMetricsReporter() override = default;
+
+  void AddPeer(absl::string_view peer_name,
+               std::vector<EmulatedEndpoint*> endpoints);
+  void AddPeer(absl::string_view peer_name,
+               std::vector<EmulatedEndpoint*> endpoints,
+               std::vector<EmulatedNetworkNode*> uplink,
+               std::vector<EmulatedNetworkNode*> downlink);
+
+  // Network stats must be empty when this method will be invoked.
+  void Start(absl::string_view test_case_name,
+             const TrackIdStreamInfoMap* reporter_helper) override;
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override;
+  void StopAndReportResults() override;
+
+ private:
+  struct PCStats {
+    // TODO(bugs.webrtc.org/10525): Separate audio and video counters. Depends
+    // on standard stat counters, enabled by field trial
+    // "WebRTC-UseStandardBytesStats".
+    DataSize payload_received = DataSize::Zero();
+    DataSize payload_sent = DataSize::Zero();
+
+    // Total bytes/packets sent/received in all RTCTransport's.
+    DataSize total_received = DataSize::Zero();
+    DataSize total_sent = DataSize::Zero();
+    int64_t packets_received = 0;
+    int64_t packets_sent = 0;
+  };
+
+  class NetworkLayerStatsCollector {
+   public:
+    NetworkLayerStatsCollector(
+        std::map<std::string, std::vector<EmulatedEndpoint*>> peer_endpoints,
+        NetworkEmulationManager* network_emulation);
+
+    void Start();
+
+    void AddPeer(absl::string_view peer_name,
+                 std::vector<EmulatedEndpoint*> endpoints,
+                 std::vector<EmulatedNetworkNode*> uplink,
+                 std::vector<EmulatedNetworkNode*> downlink);
+
+    std::map<std::string, NetworkLayerStats> GetStats();
+
+   private:
+    Mutex mutex_;
+    std::map<std::string, std::vector<EmulatedEndpoint*>> peer_endpoints_
+        RTC_GUARDED_BY(mutex_);
+    std::map<std::string, std::vector<EmulatedNetworkNode*>> peer_uplinks_
+        RTC_GUARDED_BY(mutex_);
+    std::map<std::string, std::vector<EmulatedNetworkNode*>> peer_downlinks_
+        RTC_GUARDED_BY(mutex_);
+    std::map<rtc::IPAddress, std::string> ip_to_peer_ RTC_GUARDED_BY(mutex_);
+    NetworkEmulationManager* const network_emulation_;
+  };
+
+  void ReportStats(const std::string& pc_label,
+                   const PCStats& pc_stats,
+                   const NetworkLayerStats& network_layer_stats,
+                   int64_t packet_loss,
+                   const Timestamp& end_time);
+  std::string GetTestCaseName(absl::string_view network_label) const;
+  void LogNetworkLayerStats(const std::string& peer_name,
+                            const NetworkLayerStats& stats) const;
+
+  NetworkLayerStatsCollector collector_;
+  Clock* const clock_;
+  test::MetricsLogger* const metrics_logger_;
+
+  std::string test_case_name_;
+  Timestamp start_time_ = Timestamp::MinusInfinity();
+
+  Mutex mutex_;
+  std::map<std::string, PCStats> pc_stats_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_STATS_BASED_NETWORK_QUALITY_METRICS_REPORTER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
new file mode 100644
index 0000000000..be55149482
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter_test.cc
@@ -0,0 +1,150 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/stats_based_network_quality_metrics_reporter.h"
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/test/create_network_emulation_manager.h"
+#include "api/test/create_peer_connection_quality_test_frame_generator.h"
+#include "api/test/metrics/metrics_logger.h"
+#include "api/test/metrics/stdout_metrics_exporter.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/units/time_delta.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/metric_metadata_keys.h"
+#include "test/pc/e2e/peer_connection_quality_test.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::testing::UnorderedElementsAre;
+
+using ::webrtc::test::DefaultMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Metric;
+using ::webrtc::test::Unit;
+using ::webrtc::webrtc_pc_e2e::PeerConfigurer;
+
+// Adds a peer with some audio and video (the client should not care about
+// details about audio and video configs).
+void AddDefaultAudioVideoPeer(
+    absl::string_view peer_name,
+    absl::string_view audio_stream_label,
+    absl::string_view video_stream_label,
+    const PeerNetworkDependencies& network_dependencies,
+    PeerConnectionE2EQualityTestFixture& fixture) {
+  AudioConfig audio{std::string(audio_stream_label)};
+  audio.sync_group = std::string(peer_name);
+  VideoConfig video(std::string(video_stream_label), 320, 180, 15);
+  video.sync_group = std::string(peer_name);
+  auto peer = std::make_unique<PeerConfigurer>(network_dependencies);
+  peer->SetName(peer_name);
+  peer->SetAudioConfig(std::move(audio));
+  peer->AddVideoConfig(std::move(video));
+  peer->SetVideoCodecs({VideoCodecConfig(cricket::kVp8CodecName)});
+  fixture.AddPeer(std::move(peer));
+}
+
+absl::optional<Metric> FindMeetricByName(absl::string_view name,
+                                         rtc::ArrayView<const Metric> metrics) {
+  for (const Metric& metric : metrics) {
+    if (metric.name == name) {
+      return metric;
+    }
+  }
+  return absl::nullopt;
+}
+
+TEST(StatsBasedNetworkQualityMetricsReporterTest, DebugStatsAreCollected) {
+  std::unique_ptr<NetworkEmulationManager> network_emulation =
+      CreateNetworkEmulationManager(TimeMode::kSimulated,
+                                    EmulatedNetworkStatsGatheringMode::kDebug);
+  DefaultMetricsLogger metrics_logger(
+      network_emulation->time_controller()->GetClock());
+  PeerConnectionE2EQualityTest fixture(
+      "test_case", *network_emulation->time_controller(),
+      /*audio_quality_analyzer=*/nullptr, /*video_quality_analyzer=*/nullptr,
+      &metrics_logger);
+
+  EmulatedEndpoint* alice_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+  EmulatedEndpoint* bob_endpoint =
+      network_emulation->CreateEndpoint(EmulatedEndpointConfig());
+
+  EmulatedNetworkNode* alice_link = network_emulation->CreateEmulatedNode(
+      BuiltInNetworkBehaviorConfig{.link_capacity_kbps = 500});
+  network_emulation->CreateRoute(alice_endpoint, {alice_link}, bob_endpoint);
+  EmulatedNetworkNode* bob_link = network_emulation->CreateEmulatedNode(
+      BuiltInNetworkBehaviorConfig{.link_capacity_kbps = 500});
+  network_emulation->CreateRoute(bob_endpoint, {bob_link}, alice_endpoint);
+
+  EmulatedNetworkManagerInterface* alice_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface(
+          {alice_endpoint});
+  EmulatedNetworkManagerInterface* bob_network =
+      network_emulation->CreateEmulatedNetworkManagerInterface({bob_endpoint});
+
+  AddDefaultAudioVideoPeer("alice", "alice_audio", "alice_video",
+                           alice_network->network_dependencies(), fixture);
+  AddDefaultAudioVideoPeer("bob", "bob_audio", "bob_video",
+                           bob_network->network_dependencies(), fixture);
+
+  auto network_stats_reporter =
+      std::make_unique<StatsBasedNetworkQualityMetricsReporter>(
+          /*peer_endpoints=*/std::map<std::string,
+                                      std::vector<EmulatedEndpoint*>>{},
+          network_emulation.get(), &metrics_logger);
+  network_stats_reporter->AddPeer("alice", alice_network->endpoints(),
+                                  /*uplink=*/{alice_link},
+                                  /*downlink=*/{bob_link});
+  network_stats_reporter->AddPeer("bob", bob_network->endpoints(),
+                                  /*uplink=*/{bob_link},
+                                  /*downlink=*/{alice_link});
+  fixture.AddQualityMetricsReporter(std::move(network_stats_reporter));
+
+  fixture.Run(RunParams(TimeDelta::Seconds(4)));
+
+  std::vector<Metric> metrics = metrics_logger.GetCollectedMetrics();
+  absl::optional<Metric> uplink_packet_transport_time =
+      FindMeetricByName("uplink_packet_transport_time", metrics);
+  ASSERT_TRUE(uplink_packet_transport_time.has_value());
+  ASSERT_FALSE(uplink_packet_transport_time->time_series.samples.empty());
+  absl::optional<Metric> uplink_size_to_packet_transport_time =
+      FindMeetricByName("uplink_size_to_packet_transport_time", metrics);
+  ASSERT_TRUE(uplink_size_to_packet_transport_time.has_value());
+  ASSERT_FALSE(
+      uplink_size_to_packet_transport_time->time_series.samples.empty());
+  absl::optional<Metric> downlink_packet_transport_time =
+      FindMeetricByName("downlink_packet_transport_time", metrics);
+  ASSERT_TRUE(downlink_packet_transport_time.has_value());
+  ASSERT_FALSE(downlink_packet_transport_time->time_series.samples.empty());
+  absl::optional<Metric> downlink_size_to_packet_transport_time =
+      FindMeetricByName("downlink_size_to_packet_transport_time", metrics);
+  ASSERT_TRUE(downlink_size_to_packet_transport_time.has_value());
+  ASSERT_FALSE(
+      downlink_size_to_packet_transport_time->time_series.samples.empty());
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_poller.cc b/third_party/libwebrtc/test/pc/e2e/stats_poller.cc
new file mode 100644
index 0000000000..c04805fb20
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_poller.cc
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/stats_poller.h"
+
+#include <utility>
+
+#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+void InternalStatsObserver::PollStats() {
+  peer_->GetStats(this);
+}
+
+void InternalStatsObserver::OnStatsDelivered(
+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
+  for (auto* observer : observers_) {
+    observer->OnStatsReports(pc_label_, report);
+  }
+}
+
+StatsPoller::StatsPoller(std::vector<StatsObserverInterface*> observers,
+                         std::map<std::string, StatsProvider*> peers)
+    : observers_(std::move(observers)) {
+  webrtc::MutexLock lock(&mutex_);
+  for (auto& peer : peers) {
+    pollers_.push_back(rtc::make_ref_counted<InternalStatsObserver>(
+        peer.first, peer.second, observers_));
+  }
+}
+
+StatsPoller::StatsPoller(std::vector<StatsObserverInterface*> observers,
+                         std::map<std::string, TestPeer*> peers)
+    : observers_(std::move(observers)) {
+  webrtc::MutexLock lock(&mutex_);
+  for (auto& peer : peers) {
+    pollers_.push_back(rtc::make_ref_counted<InternalStatsObserver>(
+        peer.first, peer.second, observers_));
+  }
+}
+
+void StatsPoller::PollStatsAndNotifyObservers() {
+  webrtc::MutexLock lock(&mutex_);
+  for (auto& poller : pollers_) {
+    poller->PollStats();
+  }
+}
+
+void StatsPoller::RegisterParticipantInCall(absl::string_view peer_name,
+                                            StatsProvider* peer) {
+  webrtc::MutexLock lock(&mutex_);
+  pollers_.push_back(rtc::make_ref_counted<InternalStatsObserver>(
+      peer_name, peer, observers_));
+}
+
+bool StatsPoller::UnregisterParticipantInCall(absl::string_view peer_name) {
+  webrtc::MutexLock lock(&mutex_);
+  for (auto it = pollers_.begin(); it != pollers_.end(); ++it) {
+    if ((*it)->pc_label() == peer_name) {
+      pollers_.erase(it);
+      return true;
+    }
+  }
+  return false;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_poller.h b/third_party/libwebrtc/test/pc/e2e/stats_poller.h
new file mode 100644
index 0000000000..3576f1bf05
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_poller.h
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_STATS_POLLER_H_
+#define TEST_PC_E2E_STATS_POLLER_H_
+
+#include <map>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/peer_connection_interface.h"
+#include "api/stats/rtc_stats_collector_callback.h"
+#include "api/test/stats_observer_interface.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "test/pc/e2e/stats_provider.h"
+#include "test/pc/e2e/test_peer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Helper class that will notify all the webrtc::test::StatsObserverInterface
+// objects subscribed.
+class InternalStatsObserver : public RTCStatsCollectorCallback {
+ public:
+  InternalStatsObserver(absl::string_view pc_label,
+                        StatsProvider* peer,
+                        std::vector<StatsObserverInterface*> observers)
+      : pc_label_(pc_label), peer_(peer), observers_(std::move(observers)) {}
+
+  std::string pc_label() const { return pc_label_; }
+
+  void PollStats();
+
+  void OnStatsDelivered(
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override;
+
+ private:
+  std::string pc_label_;
+  StatsProvider* peer_;
+  std::vector<StatsObserverInterface*> observers_;
+};
+
+// Helper class to invoke GetStats on a PeerConnection by passing a
+// webrtc::StatsObserver that will notify all the
+// webrtc::test::StatsObserverInterface subscribed.
+class StatsPoller {
+ public:
+  StatsPoller(std::vector<StatsObserverInterface*> observers,
+              std::map<std::string, StatsProvider*> peers_to_observe);
+  StatsPoller(std::vector<StatsObserverInterface*> observers,
+              std::map<std::string, TestPeer*> peers_to_observe);
+
+  void PollStatsAndNotifyObservers();
+
+  void RegisterParticipantInCall(absl::string_view peer_name,
+                                 StatsProvider* peer);
+  // Unregister participant from stats poller. Returns true if participant was
+  // removed and false if participant wasn't found.
+  bool UnregisterParticipantInCall(absl::string_view peer_name);
+
+ private:
+  const std::vector<StatsObserverInterface*> observers_;
+  webrtc::Mutex mutex_;
+  std::vector<rtc::scoped_refptr<InternalStatsObserver>> pollers_
+      RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_STATS_POLLER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_poller_test.cc b/third_party/libwebrtc/test/pc/e2e/stats_poller_test.cc
new file mode 100644
index 0000000000..02a323127b
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_poller_test.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/stats_poller.h"
+
+#include "api/stats/rtc_stats_collector_callback.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using ::testing::Eq;
+
+class TestStatsProvider : public StatsProvider {
+ public:
+  ~TestStatsProvider() override = default;
+
+  void GetStats(RTCStatsCollectorCallback* callback) override {
+    stats_collections_count_++;
+  }
+
+  int stats_collections_count() const { return stats_collections_count_; }
+
+ private:
+  int stats_collections_count_ = 0;
+};
+
+class MockStatsObserver : public StatsObserverInterface {
+ public:
+  ~MockStatsObserver() override = default;
+
+  MOCK_METHOD(void,
+              OnStatsReports,
+              (absl::string_view pc_label,
+               const rtc::scoped_refptr<const RTCStatsReport>& report));
+};
+
+TEST(StatsPollerTest, UnregisterParticipantAddedInCtor) {
+  TestStatsProvider alice;
+  TestStatsProvider bob;
+
+  MockStatsObserver stats_observer;
+
+  StatsPoller poller(/*observers=*/{&stats_observer},
+                     /*peers_to_observe=*/{{"alice", &alice}, {"bob", &bob}});
+  poller.PollStatsAndNotifyObservers();
+
+  EXPECT_THAT(alice.stats_collections_count(), Eq(1));
+  EXPECT_THAT(bob.stats_collections_count(), Eq(1));
+
+  poller.UnregisterParticipantInCall("bob");
+  poller.PollStatsAndNotifyObservers();
+
+  EXPECT_THAT(alice.stats_collections_count(), Eq(2));
+  EXPECT_THAT(bob.stats_collections_count(), Eq(1));
+}
+
+TEST(StatsPollerTest, UnregisterParticipantRegisteredInCall) {
+  TestStatsProvider alice;
+  TestStatsProvider bob;
+
+  MockStatsObserver stats_observer;
+
+  StatsPoller poller(/*observers=*/{&stats_observer},
+                     /*peers_to_observe=*/{{"alice", &alice}});
+  poller.RegisterParticipantInCall("bob", &bob);
+  poller.PollStatsAndNotifyObservers();
+
+  EXPECT_THAT(alice.stats_collections_count(), Eq(1));
+  EXPECT_THAT(bob.stats_collections_count(), Eq(1));
+
+  poller.UnregisterParticipantInCall("bob");
+  poller.PollStatsAndNotifyObservers();
+
+  EXPECT_THAT(alice.stats_collections_count(), Eq(2));
+  EXPECT_THAT(bob.stats_collections_count(), Eq(1));
+}
+
+}  // namespace
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/stats_provider.h b/third_party/libwebrtc/test/pc/e2e/stats_provider.h
new file mode 100644
index 0000000000..eef62d779c
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_provider.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_STATS_PROVIDER_H_
+#define TEST_PC_E2E_STATS_PROVIDER_H_
+
+#include "api/stats/rtc_stats_collector_callback.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class StatsProvider {
+ public:
+  virtual ~StatsProvider() = default;
+
+  virtual void GetStats(RTCStatsCollectorCallback* callback) = 0;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_STATS_PROVIDER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/test_activities_executor.cc b/third_party/libwebrtc/test/pc/e2e/test_activities_executor.cc
new file mode 100644
index 0000000000..7bcf7dd6c3
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_activities_executor.cc
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/pc/e2e/test_activities_executor.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/task_queue_for_test.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+void TestActivitiesExecutor::Start(TaskQueueBase* task_queue) {
+  RTC_DCHECK(task_queue);
+  task_queue_ = task_queue;
+  MutexLock lock(&lock_);
+  start_time_ = Now();
+  while (!scheduled_activities_.empty()) {
+    PostActivity(std::move(scheduled_activities_.front()));
+    scheduled_activities_.pop();
+  }
+}
+
+void TestActivitiesExecutor::Stop() {
+  if (task_queue_ == nullptr) {
+    // Already stopped or not started.
+    return;
+  }
+  SendTask(task_queue_, [this]() {
+    MutexLock lock(&lock_);
+    for (auto& handle : repeating_task_handles_) {
+      handle.Stop();
+    }
+  });
+  task_queue_ = nullptr;
+}
+
+void TestActivitiesExecutor::ScheduleActivity(
+    TimeDelta initial_delay_since_start,
+    absl::optional<TimeDelta> interval,
+    std::function<void(TimeDelta)> func) {
+  RTC_CHECK(initial_delay_since_start.IsFinite() &&
+            initial_delay_since_start >= TimeDelta::Zero());
+  RTC_CHECK(!interval ||
+            (interval->IsFinite() && *interval > TimeDelta::Zero()));
+  MutexLock lock(&lock_);
+  ScheduledActivity activity(initial_delay_since_start, interval, func);
+  if (start_time_.IsInfinite()) {
+    scheduled_activities_.push(std::move(activity));
+  } else {
+    PostActivity(std::move(activity));
+  }
+}
+
+void TestActivitiesExecutor::PostActivity(ScheduledActivity activity) {
+  // Because start_time_ will never change at this point copy it to local
+  // variable to capture in in lambda without requirement to hold a lock.
+  Timestamp start_time = start_time_;
+
+  TimeDelta remaining_delay =
+      activity.initial_delay_since_start == TimeDelta::Zero()
+          ? TimeDelta::Zero()
+          : activity.initial_delay_since_start - (Now() - start_time);
+  if (remaining_delay < TimeDelta::Zero()) {
+    RTC_LOG(LS_WARNING) << "Executing late task immediately, late by="
+                        << ToString(remaining_delay.Abs());
+    remaining_delay = TimeDelta::Zero();
+  }
+
+  if (activity.interval) {
+    if (remaining_delay == TimeDelta::Zero()) {
+      repeating_task_handles_.push_back(RepeatingTaskHandle::Start(
+          task_queue_, [activity, start_time, this]() {
+            activity.func(Now() - start_time);
+            return *activity.interval;
+          }));
+      return;
+    }
+    repeating_task_handles_.push_back(RepeatingTaskHandle::DelayedStart(
+        task_queue_, remaining_delay, [activity, start_time, this]() {
+          activity.func(Now() - start_time);
+          return *activity.interval;
+        }));
+    return;
+  }
+
+  if (remaining_delay == TimeDelta::Zero()) {
+    task_queue_->PostTask(
+        [activity, start_time, this]() { activity.func(Now() - start_time); });
+    return;
+  }
+
+  task_queue_->PostDelayedTask(
+      [activity, start_time, this]() { activity.func(Now() - start_time); },
+      remaining_delay);
+}
+
+Timestamp TestActivitiesExecutor::Now() const {
+  return clock_->CurrentTime();
+}
+
+TestActivitiesExecutor::ScheduledActivity::ScheduledActivity(
+    TimeDelta initial_delay_since_start,
+    absl::optional<TimeDelta> interval,
+    std::function<void(TimeDelta)> func)
+    : initial_delay_since_start(initial_delay_since_start),
+      interval(interval),
+      func(std::move(func)) {}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/test_activities_executor.h b/third_party/libwebrtc/test/pc/e2e/test_activities_executor.h
new file mode 100644
index 0000000000..2469ac7f36
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_activities_executor.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_TEST_ACTIVITIES_EXECUTOR_H_
+#define TEST_PC_E2E_TEST_ACTIVITIES_EXECUTOR_H_
+
+#include <queue>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class TestActivitiesExecutor {
+ public:
+  explicit TestActivitiesExecutor(Clock* clock) : clock_(clock) {}
+  ~TestActivitiesExecutor() { Stop(); }
+
+  // Starts scheduled activities according to their schedule. All activities
+  // that will be scheduled after Start(...) was invoked will be executed
+  // immediately according to their schedule.
+  void Start(TaskQueueForTest* task_queue) { Start(task_queue->Get()); }
+  void Start(TaskQueueBase* task_queue);
+  void Stop();
+
+  // Schedule activity to be executed. If test isn't started yet, then activity
+  // will be executed according to its schedule after Start() will be invoked.
+  // If test is started, then it will be executed immediately according to its
+  // schedule.
+  void ScheduleActivity(TimeDelta initial_delay_since_start,
+                        absl::optional<TimeDelta> interval,
+                        std::function<void(TimeDelta)> func);
+
+ private:
+  struct ScheduledActivity {
+    ScheduledActivity(TimeDelta initial_delay_since_start,
+                      absl::optional<TimeDelta> interval,
+                      std::function<void(TimeDelta)> func);
+
+    TimeDelta initial_delay_since_start;
+    absl::optional<TimeDelta> interval;
+    std::function<void(TimeDelta)> func;
+  };
+
+  void PostActivity(ScheduledActivity activity)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
+  Timestamp Now() const;
+
+  Clock* const clock_;
+
+  TaskQueueBase* task_queue_;
+
+  Mutex lock_;
+  // Time when test was started. Minus infinity means that it wasn't started
+  // yet.
+  Timestamp start_time_ RTC_GUARDED_BY(lock_) = Timestamp::MinusInfinity();
+  // Queue of activities that were added before test was started.
+  // Activities from this queue will be posted on the `task_queue_` after test
+  // will be set up and then this queue will be unused.
+  std::queue<ScheduledActivity> scheduled_activities_ RTC_GUARDED_BY(lock_);
+  // List of task handles for activities, that are posted on `task_queue_` as
+  // repeated during the call.
+  std::vector<RepeatingTaskHandle> repeating_task_handles_
+      RTC_GUARDED_BY(lock_);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_TEST_ACTIVITIES_EXECUTOR_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/test_peer.cc b/third_party/libwebrtc/test/pc/e2e/test_peer.cc
new file mode 100644
index 0000000000..b3a9e1c164
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/test_peer.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "api/scoped_refptr.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+class SetRemoteDescriptionCallback
+    : public webrtc::SetRemoteDescriptionObserverInterface {
+ public:
+  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
+    is_called_ = true;
+    error_ = error;
+  }
+
+  bool is_called() const { return is_called_; }
+
+  webrtc::RTCError error() const { return error_; }
+
+ private:
+  bool is_called_ = false;
+  webrtc::RTCError error_;
+};
+
+}  // namespace
+
+ConfigurableParams TestPeer::configurable_params() const {
+  MutexLock lock(&mutex_);
+  return configurable_params_;
+}
+
+void TestPeer::AddVideoConfig(VideoConfig config) {
+  MutexLock lock(&mutex_);
+  configurable_params_.video_configs.push_back(std::move(config));
+}
+
+void TestPeer::RemoveVideoConfig(absl::string_view stream_label) {
+  MutexLock lock(&mutex_);
+  bool config_removed = false;
+  for (auto it = configurable_params_.video_configs.begin();
+       it != configurable_params_.video_configs.end(); ++it) {
+    if (*it->stream_label == stream_label) {
+      configurable_params_.video_configs.erase(it);
+      config_removed = true;
+      break;
+    }
+  }
+  RTC_CHECK(config_removed) << *params_.name << ": No video config with label ["
+                            << stream_label << "] was found";
+}
+
+void TestPeer::SetVideoSubscription(VideoSubscription subscription) {
+  MutexLock lock(&mutex_);
+  configurable_params_.video_subscription = std::move(subscription);
+}
+
+void TestPeer::GetStats(RTCStatsCollectorCallback* callback) {
+  pc()->signaling_thread()->PostTask(
+      SafeTask(signaling_thread_task_safety_,
+               [this, callback]() { pc()->GetStats(callback); }));
+}
+
+bool TestPeer::SetRemoteDescription(
+    std::unique_ptr<SessionDescriptionInterface> desc,
+    std::string* error_out) {
+  RTC_CHECK(wrapper_) << "TestPeer is already closed";
+
+  auto observer = rtc::make_ref_counted<SetRemoteDescriptionCallback>();
+  // We're assuming (and asserting) that the PeerConnection implementation of
+  // SetRemoteDescription is synchronous when called on the signaling thread.
+  pc()->SetRemoteDescription(std::move(desc), observer);
+  RTC_CHECK(observer->is_called());
+  if (!observer->error().ok()) {
+    RTC_LOG(LS_ERROR) << *params_.name << ": Failed to set remote description: "
+                      << observer->error().message();
+    if (error_out) {
+      *error_out = observer->error().message();
+    }
+  }
+  return observer->error().ok();
+}
+
+bool TestPeer::AddIceCandidates(
+    std::vector<std::unique_ptr<IceCandidateInterface>> candidates) {
+  RTC_CHECK(wrapper_) << "TestPeer is already closed";
+  bool success = true;
+  for (auto& candidate : candidates) {
+    if (!pc()->AddIceCandidate(candidate.get())) {
+      std::string candidate_str;
+      bool res = candidate->ToString(&candidate_str);
+      RTC_CHECK(res);
+      RTC_LOG(LS_ERROR) << "Failed to add ICE candidate, candidate_str="
+                        << candidate_str;
+      success = false;
+    } else {
+      remote_ice_candidates_.push_back(std::move(candidate));
+    }
+  }
+  return success;
+}
+
+void TestPeer::Close() {
+  signaling_thread_task_safety_->SetNotAlive();
+  wrapper_->pc()->Close();
+  remote_ice_candidates_.clear();
+  audio_processing_ = nullptr;
+  video_sources_.clear();
+  wrapper_ = nullptr;
+  worker_thread_ = nullptr;
+}
+
+TestPeer::TestPeer(
+    rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
+    rtc::scoped_refptr<PeerConnectionInterface> pc,
+    std::unique_ptr<MockPeerConnectionObserver> observer,
+    Params params,
+    ConfigurableParams configurable_params,
+    std::vector<PeerConfigurer::VideoSource> video_sources,
+    rtc::scoped_refptr<AudioProcessing> audio_processing,
+    std::unique_ptr<rtc::Thread> worker_thread)
+    : params_(std::move(params)),
+      configurable_params_(std::move(configurable_params)),
+      worker_thread_(std::move(worker_thread)),
+      wrapper_(std::make_unique<PeerConnectionWrapper>(std::move(pc_factory),
+                                                       std::move(pc),
+                                                       std::move(observer))),
+      video_sources_(std::move(video_sources)),
+      audio_processing_(audio_processing) {
+  signaling_thread_task_safety_ = PendingTaskSafetyFlag::CreateDetached();
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/test_peer.h b/third_party/libwebrtc/test/pc/e2e/test_peer.h
new file mode 100644
index 0000000000..1088871817
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer.h
@@ -0,0 +1,188 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_TEST_PEER_H_
+#define TEST_PC_E2E_TEST_PEER_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "api/function_view.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/set_remote_description_observer_interface.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "pc/peer_connection_wrapper.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/pc/e2e/stats_provider.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Describes a single participant in the call.
+class TestPeer final : public StatsProvider {
+ public:
+  ~TestPeer() override = default;
+
+  const Params& params() const { return params_; }
+
+  ConfigurableParams configurable_params() const;
+  void AddVideoConfig(VideoConfig config);
+  // Removes video config with specified name. Crashes if the config with
+  // specified name isn't found.
+  void RemoveVideoConfig(absl::string_view stream_label);
+  void SetVideoSubscription(VideoSubscription subscription);
+
+  void GetStats(RTCStatsCollectorCallback* callback) override;
+
+  PeerConfigurer::VideoSource ReleaseVideoSource(size_t i) {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return std::move(video_sources_[i]);
+  }
+
+  PeerConnectionFactoryInterface* pc_factory() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->pc_factory();
+  }
+  PeerConnectionInterface* pc() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->pc();
+  }
+  MockPeerConnectionObserver* observer() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->observer();
+  }
+
+  // Tell underlying `PeerConnection` to create an Offer.
+  // `observer` will be invoked on the signaling thread when offer is created.
+  void CreateOffer(
+      rtc::scoped_refptr<CreateSessionDescriptionObserver> observer) {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    pc()->CreateOffer(observer.get(), params_.rtc_offer_answer_options);
+  }
+  std::unique_ptr<SessionDescriptionInterface> CreateOffer() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->CreateOffer(params_.rtc_offer_answer_options);
+  }
+
+  std::unique_ptr<SessionDescriptionInterface> CreateAnswer() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->CreateAnswer();
+  }
+
+  bool SetLocalDescription(std::unique_ptr<SessionDescriptionInterface> desc,
+                           std::string* error_out = nullptr) {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->SetLocalDescription(std::move(desc), error_out);
+  }
+
+  // `error_out` will be set only if returned value is false.
+  bool SetRemoteDescription(std::unique_ptr<SessionDescriptionInterface> desc,
+                            std::string* error_out = nullptr);
+
+  rtc::scoped_refptr<RtpTransceiverInterface> AddTransceiver(
+      cricket::MediaType media_type,
+      const RtpTransceiverInit& init) {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->AddTransceiver(media_type, init);
+  }
+
+  rtc::scoped_refptr<RtpSenderInterface> AddTrack(
+      rtc::scoped_refptr<MediaStreamTrackInterface> track,
+      const std::vector<std::string>& stream_ids = {}) {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->AddTrack(track, stream_ids);
+  }
+
+  rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
+      const std::string& label) {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->CreateDataChannel(label);
+  }
+
+  PeerConnectionInterface::SignalingState signaling_state() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->signaling_state();
+  }
+
+  bool IsIceGatheringDone() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->IsIceGatheringDone();
+  }
+
+  bool IsIceConnected() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->IsIceConnected();
+  }
+
+  rtc::scoped_refptr<const RTCStatsReport> GetStats() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    return wrapper_->GetStats();
+  }
+
+  void DetachAecDump() {
+    RTC_CHECK(wrapper_) << "TestPeer is already closed";
+    if (audio_processing_) {
+      audio_processing_->DetachAecDump();
+    }
+  }
+
+  // Adds provided `candidates` to the owned peer connection.
+  bool AddIceCandidates(
+      std::vector<std::unique_ptr<IceCandidateInterface>> candidates);
+
+  // Closes underlying peer connection and destroys all related objects freeing
+  // up related resources.
+  void Close();
+
+ protected:
+  friend class TestPeerFactory;
+  TestPeer(rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
+           rtc::scoped_refptr<PeerConnectionInterface> pc,
+           std::unique_ptr<MockPeerConnectionObserver> observer,
+           Params params,
+           ConfigurableParams configurable_params,
+           std::vector<PeerConfigurer::VideoSource> video_sources,
+           rtc::scoped_refptr<AudioProcessing> audio_processing,
+           std::unique_ptr<rtc::Thread> worker_thread);
+
+ private:
+  const Params params_;
+
+  mutable Mutex mutex_;
+  ConfigurableParams configurable_params_ RTC_GUARDED_BY(mutex_);
+
+  // Safety flag to protect all tasks posted on the signaling thread to not be
+  // executed after `wrapper_` object is destructed.
+  rtc::scoped_refptr<PendingTaskSafetyFlag> signaling_thread_task_safety_ =
+      nullptr;
+
+  // Keeps ownership of worker thread. It has to be destroyed after `wrapper_`.
+  // `worker_thread_`can be null if the Peer use only one thread as both the
+  // worker thread and network thread.
+  std::unique_ptr<rtc::Thread> worker_thread_;
+  std::unique_ptr<PeerConnectionWrapper> wrapper_;
+  std::vector<PeerConfigurer::VideoSource> video_sources_;
+  rtc::scoped_refptr<AudioProcessing> audio_processing_;
+
+  std::vector<std::unique_ptr<IceCandidateInterface>> remote_ice_candidates_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_TEST_PEER_H_
diff --git a/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc
new file mode 100644
index 0000000000..7fc12f2c11
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc
@@ -0,0 +1,374 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/pc/e2e/test_peer_factory.h"
+
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/test/create_time_controller.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/time_controller.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video_codecs/builtin_video_decoder_factory.h"
+#include "api/video_codecs/builtin_video_encoder_factory.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "media/engine/webrtc_media_engine_defaults.h"
+#include "modules/audio_processing/aec_dump/aec_dump_factory.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "rtc_base/thread.h"
+#include "test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h"
+#include "test/pc/e2e/echo/echo_emulation.h"
+#include "test/testsupport/copy_to_file_audio_capturer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using EmulatedSFUConfigMap =
+    ::webrtc::webrtc_pc_e2e::QualityAnalyzingVideoEncoder::EmulatedSFUConfigMap;
+
+constexpr int16_t kGeneratedAudioMaxAmplitude = 32000;
+constexpr int kDefaultSamplingFrequencyInHz = 48000;
+
+// Sets mandatory entities in injectable components like `pcf_dependencies`
+// and `pc_dependencies` if they are omitted. Also setup required
+// dependencies, that won't be specially provided by factory and will be just
+// transferred to peer connection creation code.
+void SetMandatoryEntities(InjectableComponents* components,
+                          TimeController& time_controller) {
+  RTC_DCHECK(components->pcf_dependencies);
+  RTC_DCHECK(components->pc_dependencies);
+
+  // Setup required peer connection factory dependencies.
+  if (components->pcf_dependencies->task_queue_factory == nullptr) {
+    components->pcf_dependencies->task_queue_factory =
+        time_controller.CreateTaskQueueFactory();
+  }
+  if (components->pcf_dependencies->call_factory == nullptr) {
+    components->pcf_dependencies->call_factory =
+        CreateTimeControllerBasedCallFactory(&time_controller);
+  }
+  if (components->pcf_dependencies->event_log_factory == nullptr) {
+    components->pcf_dependencies->event_log_factory =
+        std::make_unique<RtcEventLogFactory>(
+            components->pcf_dependencies->task_queue_factory.get());
+  }
+  if (!components->pcf_dependencies->trials) {
+    components->pcf_dependencies->trials =
+        std::make_unique<FieldTrialBasedConfig>();
+  }
+}
+
+// Returns mapping from stream label to optional spatial index.
+// If we have stream label "Foo" and mapping contains
+// 1. `absl::nullopt` means all simulcast/SVC streams are required
+// 2. Concrete value means that particular simulcast/SVC stream have to be
+//    analyzed.
+EmulatedSFUConfigMap CalculateRequiredSpatialIndexPerStream(
+    const std::vector<VideoConfig>& video_configs) {
+  EmulatedSFUConfigMap result;
+  for (auto& video_config : video_configs) {
+    // Stream label should be set by fixture implementation here.
+    RTC_DCHECK(video_config.stream_label);
+    bool res = result
+                   .insert({*video_config.stream_label,
+                            video_config.emulated_sfu_config})
+                   .second;
+    RTC_DCHECK(res) << "Duplicate video_config.stream_label="
+                    << *video_config.stream_label;
+  }
+  return result;
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer> CreateAudioRenderer(
+    const absl::optional<RemotePeerAudioConfig>& config) {
+  if (!config) {
+    // Return default renderer because we always require some renderer.
+    return TestAudioDeviceModule::CreateDiscardRenderer(
+        kDefaultSamplingFrequencyInHz);
+  }
+  if (config->output_file_name) {
+    return TestAudioDeviceModule::CreateBoundedWavFileWriter(
+        config->output_file_name.value(), config->sampling_frequency_in_hz);
+  }
+  return TestAudioDeviceModule::CreateDiscardRenderer(
+      config->sampling_frequency_in_hz);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer> CreateAudioCapturer(
+    const absl::optional<AudioConfig>& audio_config) {
+  if (!audio_config) {
+    // If we have no audio config we still need to provide some audio device.
+    // In such case use generated capturer. Despite of we provided audio here,
+    // in test media setup audio stream won't be added into peer connection.
+    return TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+        kGeneratedAudioMaxAmplitude, kDefaultSamplingFrequencyInHz);
+  }
+
+  switch (audio_config->mode) {
+    case AudioConfig::Mode::kGenerated:
+      return TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+          kGeneratedAudioMaxAmplitude, audio_config->sampling_frequency_in_hz);
+    case AudioConfig::Mode::kFile:
+      RTC_DCHECK(audio_config->input_file_name);
+      return TestAudioDeviceModule::CreateWavFileReader(
+          audio_config->input_file_name.value(), /*repeat=*/true);
+  }
+}
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+    absl::optional<AudioConfig> audio_config,
+    absl::optional<RemotePeerAudioConfig> remote_audio_config,
+    absl::optional<EchoEmulationConfig> echo_emulation_config,
+    TaskQueueFactory* task_queue_factory) {
+  std::unique_ptr<TestAudioDeviceModule::Renderer> renderer =
+      CreateAudioRenderer(remote_audio_config);
+  std::unique_ptr<TestAudioDeviceModule::Capturer> capturer =
+      CreateAudioCapturer(audio_config);
+  RTC_DCHECK(renderer);
+  RTC_DCHECK(capturer);
+
+  // Setup echo emulation if required.
+  if (echo_emulation_config) {
+    capturer = std::make_unique<EchoEmulatingCapturer>(std::move(capturer),
+                                                       *echo_emulation_config);
+    renderer = std::make_unique<EchoEmulatingRenderer>(
+        std::move(renderer),
+        static_cast<EchoEmulatingCapturer*>(capturer.get()));
+  }
+
+  // Setup input stream dumping if required.
+  if (audio_config && audio_config->input_dump_file_name) {
+    capturer = std::make_unique<test::CopyToFileAudioCapturer>(
+        std::move(capturer), audio_config->input_dump_file_name.value());
+  }
+
+  return TestAudioDeviceModule::Create(task_queue_factory, std::move(capturer),
+                                       std::move(renderer), /*speed=*/1.f);
+}
+
+std::unique_ptr<cricket::MediaEngineInterface> CreateMediaEngine(
+    PeerConnectionFactoryComponents* pcf_dependencies,
+    rtc::scoped_refptr<AudioDeviceModule> audio_device_module) {
+  cricket::MediaEngineDependencies media_deps;
+  media_deps.task_queue_factory = pcf_dependencies->task_queue_factory.get();
+  media_deps.adm = audio_device_module;
+  media_deps.audio_processing = pcf_dependencies->audio_processing;
+  media_deps.audio_mixer = pcf_dependencies->audio_mixer;
+  media_deps.video_encoder_factory =
+      std::move(pcf_dependencies->video_encoder_factory);
+  media_deps.video_decoder_factory =
+      std::move(pcf_dependencies->video_decoder_factory);
+  webrtc::SetMediaEngineDefaults(&media_deps);
+  RTC_DCHECK(pcf_dependencies->trials);
+  media_deps.trials = pcf_dependencies->trials.get();
+
+  return cricket::CreateMediaEngine(std::move(media_deps));
+}
+
+void WrapVideoEncoderFactory(
+    absl::string_view peer_name,
+    double bitrate_multiplier,
+    EmulatedSFUConfigMap stream_to_sfu_config,
+    PeerConnectionFactoryComponents* pcf_dependencies,
+    VideoQualityAnalyzerInjectionHelper* video_analyzer_helper) {
+  std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
+  if (pcf_dependencies->video_encoder_factory != nullptr) {
+    video_encoder_factory = std::move(pcf_dependencies->video_encoder_factory);
+  } else {
+    video_encoder_factory = CreateBuiltinVideoEncoderFactory();
+  }
+  pcf_dependencies->video_encoder_factory =
+      video_analyzer_helper->WrapVideoEncoderFactory(
+          peer_name, std::move(video_encoder_factory), bitrate_multiplier,
+          std::move(stream_to_sfu_config));
+}
+
+void WrapVideoDecoderFactory(
+    absl::string_view peer_name,
+    PeerConnectionFactoryComponents* pcf_dependencies,
+    VideoQualityAnalyzerInjectionHelper* video_analyzer_helper) {
+  std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
+  if (pcf_dependencies->video_decoder_factory != nullptr) {
+    video_decoder_factory = std::move(pcf_dependencies->video_decoder_factory);
+  } else {
+    video_decoder_factory = CreateBuiltinVideoDecoderFactory();
+  }
+  pcf_dependencies->video_decoder_factory =
+      video_analyzer_helper->WrapVideoDecoderFactory(
+          peer_name, std::move(video_decoder_factory));
+}
+
+// Creates PeerConnectionFactoryDependencies objects, providing entities
+// from InjectableComponents::PeerConnectionFactoryComponents.
+PeerConnectionFactoryDependencies CreatePCFDependencies(
+    std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies,
+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
+    rtc::Thread* signaling_thread,
+    rtc::Thread* worker_thread,
+    rtc::Thread* network_thread) {
+  PeerConnectionFactoryDependencies pcf_deps;
+  pcf_deps.signaling_thread = signaling_thread;
+  pcf_deps.worker_thread = worker_thread;
+  pcf_deps.network_thread = network_thread;
+  pcf_deps.media_engine = std::move(media_engine);
+
+  pcf_deps.call_factory = std::move(pcf_dependencies->call_factory);
+  pcf_deps.event_log_factory = std::move(pcf_dependencies->event_log_factory);
+  pcf_deps.task_queue_factory = std::move(pcf_dependencies->task_queue_factory);
+
+  if (pcf_dependencies->fec_controller_factory != nullptr) {
+    pcf_deps.fec_controller_factory =
+        std::move(pcf_dependencies->fec_controller_factory);
+  }
+  if (pcf_dependencies->network_controller_factory != nullptr) {
+    pcf_deps.network_controller_factory =
+        std::move(pcf_dependencies->network_controller_factory);
+  }
+  if (pcf_dependencies->neteq_factory != nullptr) {
+    pcf_deps.neteq_factory = std::move(pcf_dependencies->neteq_factory);
+  }
+  if (pcf_dependencies->trials != nullptr) {
+    pcf_deps.trials = std::move(pcf_dependencies->trials);
+  }
+
+  return pcf_deps;
+}
+
+// Creates PeerConnectionDependencies objects, providing entities
+// from InjectableComponents::PeerConnectionComponents.
+PeerConnectionDependencies CreatePCDependencies(
+    MockPeerConnectionObserver* observer,
+    uint32_t port_allocator_extra_flags,
+    std::unique_ptr<PeerConnectionComponents> pc_dependencies) {
+  PeerConnectionDependencies pc_deps(observer);
+
+  auto port_allocator = std::make_unique<cricket::BasicPortAllocator>(
+      pc_dependencies->network_manager, pc_dependencies->packet_socket_factory);
+
+  // This test does not support TCP
+  int flags = port_allocator_extra_flags | cricket::PORTALLOCATOR_DISABLE_TCP;
+  port_allocator->set_flags(port_allocator->flags() | flags);
+
+  pc_deps.allocator = std::move(port_allocator);
+
+  if (pc_dependencies->async_resolver_factory != nullptr) {
+    pc_deps.async_resolver_factory =
+        std::move(pc_dependencies->async_resolver_factory);
+  }
+  if (pc_dependencies->cert_generator != nullptr) {
+    pc_deps.cert_generator = std::move(pc_dependencies->cert_generator);
+  }
+  if (pc_dependencies->tls_cert_verifier != nullptr) {
+    pc_deps.tls_cert_verifier = std::move(pc_dependencies->tls_cert_verifier);
+  }
+  if (pc_dependencies->ice_transport_factory != nullptr) {
+    pc_deps.ice_transport_factory =
+        std::move(pc_dependencies->ice_transport_factory);
+  }
+  return pc_deps;
+}
+
+}  // namespace
+
+absl::optional<RemotePeerAudioConfig> RemotePeerAudioConfig::Create(
+    absl::optional<AudioConfig> config) {
+  if (!config) {
+    return absl::nullopt;
+  }
+  return RemotePeerAudioConfig(config.value());
+}
+
+std::unique_ptr<TestPeer> TestPeerFactory::CreateTestPeer(
+    std::unique_ptr<PeerConfigurer> configurer,
+    std::unique_ptr<MockPeerConnectionObserver> observer,
+    absl::optional<RemotePeerAudioConfig> remote_audio_config,
+    absl::optional<EchoEmulationConfig> echo_emulation_config) {
+  std::unique_ptr<InjectableComponents> components =
+      configurer->ReleaseComponents();
+  std::unique_ptr<Params> params = configurer->ReleaseParams();
+  std::unique_ptr<ConfigurableParams> configurable_params =
+      configurer->ReleaseConfigurableParams();
+  std::vector<PeerConfigurer::VideoSource> video_sources =
+      configurer->ReleaseVideoSources();
+  RTC_DCHECK(components);
+  RTC_DCHECK(params);
+  RTC_DCHECK(configurable_params);
+  RTC_DCHECK_EQ(configurable_params->video_configs.size(),
+                video_sources.size());
+  SetMandatoryEntities(components.get(), time_controller_);
+  params->rtc_configuration.sdp_semantics = SdpSemantics::kUnifiedPlan;
+
+  // Create peer connection factory.
+  if (components->pcf_dependencies->audio_processing == nullptr) {
+    components->pcf_dependencies->audio_processing =
+        webrtc::AudioProcessingBuilder().Create();
+  }
+  if (params->aec_dump_path) {
+    components->pcf_dependencies->audio_processing->CreateAndAttachAecDump(
+        *params->aec_dump_path, -1, task_queue_);
+  }
+  rtc::scoped_refptr<AudioDeviceModule> audio_device_module =
+      CreateAudioDeviceModule(
+          params->audio_config, remote_audio_config, echo_emulation_config,
+          components->pcf_dependencies->task_queue_factory.get());
+  WrapVideoEncoderFactory(
+      params->name.value(), params->video_encoder_bitrate_multiplier,
+      CalculateRequiredSpatialIndexPerStream(
+          configurable_params->video_configs),
+      components->pcf_dependencies.get(), video_analyzer_helper_);
+  WrapVideoDecoderFactory(params->name.value(),
+                          components->pcf_dependencies.get(),
+                          video_analyzer_helper_);
+  std::unique_ptr<cricket::MediaEngineInterface> media_engine =
+      CreateMediaEngine(components->pcf_dependencies.get(),
+                        audio_device_module);
+
+  std::unique_ptr<rtc::Thread> owned_worker_thread =
+      components->worker_thread != nullptr
+          ? nullptr
+          : time_controller_.CreateThread("worker_thread");
+  if (components->worker_thread == nullptr) {
+    components->worker_thread = owned_worker_thread.get();
+  }
+
+  // Store `webrtc::AudioProcessing` into local variable before move of
+  // `components->pcf_dependencies`
+  rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing =
+      components->pcf_dependencies->audio_processing;
+  PeerConnectionFactoryDependencies pcf_deps = CreatePCFDependencies(
+      std::move(components->pcf_dependencies), std::move(media_engine),
+      signaling_thread_, components->worker_thread, components->network_thread);
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> peer_connection_factory =
+      CreateModularPeerConnectionFactory(std::move(pcf_deps));
+
+  // Create peer connection.
+  PeerConnectionDependencies pc_deps =
+      CreatePCDependencies(observer.get(), params->port_allocator_extra_flags,
+                           std::move(components->pc_dependencies));
+  rtc::scoped_refptr<PeerConnectionInterface> peer_connection =
+      peer_connection_factory
+          ->CreatePeerConnectionOrError(params->rtc_configuration,
+                                        std::move(pc_deps))
+          .MoveValue();
+  peer_connection->SetBitrate(params->bitrate_settings);
+
+  return absl::WrapUnique(
+      new TestPeer(peer_connection_factory, peer_connection,
+                   std::move(observer), std::move(*params),
+                   std::move(*configurable_params), std::move(video_sources),
+                   audio_processing, std::move(owned_worker_thread)));
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/pc/e2e/test_peer_factory.h b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.h
new file mode 100644
index 0000000000..f2698e2a15
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.h
@@ -0,0 +1,84 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_E2E_TEST_PEER_FACTORY_H_
+#define TEST_PC_E2E_TEST_PEER_FACTORY_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/pclf/peer_configurer.h"
+#include "api/test/time_controller.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "rtc_base/task_queue.h"
+#include "test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h"
+#include "test/pc/e2e/test_peer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+struct RemotePeerAudioConfig {
+  explicit RemotePeerAudioConfig(AudioConfig config)
+      : sampling_frequency_in_hz(config.sampling_frequency_in_hz),
+        output_file_name(config.output_dump_file_name) {}
+
+  static absl::optional<RemotePeerAudioConfig> Create(
+      absl::optional<AudioConfig> config);
+
+  int sampling_frequency_in_hz;
+  absl::optional<std::string> output_file_name;
+};
+
+class TestPeerFactory {
+ public:
+  // Creates a test peer factory.
+  // `signaling_thread` will be used as a signaling thread for all peers created
+  // by this factory.
+  // `time_controller` will be used to create required threads, task queue
+  // factories and call factory.
+  // `video_analyzer_helper` will be used to setup video quality analysis for
+  // created peers.
+  // `task_queue` will be used for AEC dump if it is requested.
+  TestPeerFactory(rtc::Thread* signaling_thread,
+                  TimeController& time_controller,
+                  VideoQualityAnalyzerInjectionHelper* video_analyzer_helper,
+                  rtc::TaskQueue* task_queue)
+      : signaling_thread_(signaling_thread),
+        time_controller_(time_controller),
+        video_analyzer_helper_(video_analyzer_helper),
+        task_queue_(task_queue) {}
+
+  // Setups all components, that should be provided to WebRTC
+  // PeerConnectionFactory and PeerConnection creation methods,
+  // also will setup dependencies, that are required for media analyzers
+  // injection.
+  std::unique_ptr<TestPeer> CreateTestPeer(
+      std::unique_ptr<PeerConfigurer> configurer,
+      std::unique_ptr<MockPeerConnectionObserver> observer,
+      absl::optional<RemotePeerAudioConfig> remote_audio_config,
+      absl::optional<EchoEmulationConfig> echo_emulation_config);
+
+ private:
+  rtc::Thread* signaling_thread_;
+  TimeController& time_controller_;
+  VideoQualityAnalyzerInjectionHelper* video_analyzer_helper_;
+  rtc::TaskQueue* task_queue_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_TEST_PEER_FACTORY_H_
diff --git a/third_party/libwebrtc/test/pc/sctp/BUILD.gn b/third_party/libwebrtc/test/pc/sctp/BUILD.gn
new file mode 100644
index 0000000000..f088a5b20c
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/sctp/BUILD.gn
@@ -0,0 +1,18 @@
+# Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("fake_sctp_transport") {
+  visibility = [ "*" ]
+  sources = [ "fake_sctp_transport.h" ]
+  deps = [
+    "../../../api/transport:sctp_transport_factory_interface",
+    "../../../media:rtc_data_sctp_transport_internal",
+  ]
+}
diff --git a/third_party/libwebrtc/test/pc/sctp/fake_sctp_transport.h b/third_party/libwebrtc/test/pc/sctp/fake_sctp_transport.h
new file mode 100644
index 0000000000..a1bb0e219c
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/sctp/fake_sctp_transport.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_PC_SCTP_FAKE_SCTP_TRANSPORT_H_
+#define TEST_PC_SCTP_FAKE_SCTP_TRANSPORT_H_
+
+#include <memory>
+
+#include "api/transport/sctp_transport_factory_interface.h"
+#include "media/sctp/sctp_transport_internal.h"
+
+// Used for tests in this file to verify that PeerConnection responds to signals
+// from the SctpTransport correctly, and calls Start with the correct
+// local/remote ports.
+class FakeSctpTransport : public cricket::SctpTransportInternal {
+ public:
+  void SetOnConnectedCallback(std::function<void()> callback) override {}
+  void SetDataChannelSink(webrtc::DataChannelSink* sink) override {}
+  void SetDtlsTransport(rtc::PacketTransportInternal* transport) override {}
+  bool Start(int local_port, int remote_port, int max_message_size) override {
+    local_port_.emplace(local_port);
+    remote_port_.emplace(remote_port);
+    max_message_size_ = max_message_size;
+    return true;
+  }
+  bool OpenStream(int sid) override { return true; }
+  bool ResetStream(int sid) override { return true; }
+  bool SendData(int sid,
+                const webrtc::SendDataParams& params,
+                const rtc::CopyOnWriteBuffer& payload,
+                cricket::SendDataResult* result = nullptr) override {
+    return true;
+  }
+  bool ReadyToSendData() override { return true; }
+  void set_debug_name_for_testing(const char* debug_name) override {}
+
+  int max_message_size() const { return max_message_size_; }
+  absl::optional<int> max_outbound_streams() const { return absl::nullopt; }
+  absl::optional<int> max_inbound_streams() const { return absl::nullopt; }
+  int local_port() const {
+    RTC_DCHECK(local_port_);
+    return *local_port_;
+  }
+  int remote_port() const {
+    RTC_DCHECK(remote_port_);
+    return *remote_port_;
+  }
+
+ private:
+  absl::optional<int> local_port_;
+  absl::optional<int> remote_port_;
+  int max_message_size_;
+};
+
+class FakeSctpTransportFactory : public webrtc::SctpTransportFactoryInterface {
+ public:
+  std::unique_ptr<cricket::SctpTransportInternal> CreateSctpTransport(
+      rtc::PacketTransportInternal*) override {
+    last_fake_sctp_transport_ = new FakeSctpTransport();
+    return std::unique_ptr<cricket::SctpTransportInternal>(
+        last_fake_sctp_transport_);
+  }
+
+  FakeSctpTransport* last_fake_sctp_transport() {
+    return last_fake_sctp_transport_;
+  }
+
+ private:
+  FakeSctpTransport* last_fake_sctp_transport_ = nullptr;
+};
+
+#endif  // TEST_PC_SCTP_FAKE_SCTP_TRANSPORT_H_
diff --git a/third_party/libwebrtc/test/peer_scenario/BUILD.gn b/third_party/libwebrtc/test/peer_scenario/BUILD.gn
new file mode 100644
index 0000000000..00492a18a9
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/BUILD.gn
@@ -0,0 +1,68 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+if (rtc_include_tests) {
+  rtc_library("peer_scenario") {
+    testonly = true
+    sources = [
+      "peer_scenario.cc",
+      "peer_scenario.h",
+      "peer_scenario_client.cc",
+      "peer_scenario_client.h",
+      "scenario_connection.cc",
+      "scenario_connection.h",
+      "signaling_route.cc",
+      "signaling_route.h",
+    ]
+    deps = [
+      "..:fake_video_codecs",
+      "..:fileutils",
+      "..:test_support",
+      "../:video_test_common",
+      "../../api:candidate",
+      "../../api:create_time_controller",
+      "../../api:libjingle_peerconnection_api",
+      "../../api:network_emulation_manager_api",
+      "../../api:rtc_stats_api",
+      "../../api:time_controller",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../api/audio_codecs:builtin_audio_encoder_factory",
+      "../../api/rtc_event_log:rtc_event_log_factory",
+      "../../api/task_queue:default_task_queue_factory",
+      "../../api/transport:field_trial_based_config",
+      "../../api/video_codecs:builtin_video_decoder_factory",
+      "../../api/video_codecs:builtin_video_encoder_factory",
+      "../../media:rtc_audio_video",
+      "../../media:rtc_media_base",
+      "../../media:rtp_utils",
+      "../../modules/audio_device:audio_device_impl",
+      "../../modules/rtp_rtcp:rtp_rtcp_format",
+      "../../p2p:rtc_p2p",
+      "../../pc:channel",
+      "../../pc:jsep_transport_controller",
+      "../../pc:pc_test_utils",
+      "../../pc:rtp_transport_internal",
+      "../../pc:session_description",
+      "../../rtc_base:null_socket_server",
+      "../../rtc_base:stringutils",
+      "../../rtc_base:task_queue_for_test",
+      "../../test:explicit_key_value_config",
+      "../../test:scoped_key_value_config",
+      "../logging:log_writer",
+      "../network:emulated_network",
+      "../scenario",
+      "../time_controller",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/memory",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/test/peer_scenario/DEPS b/third_party/libwebrtc/test/peer_scenario/DEPS
new file mode 100644
index 0000000000..68e9f46087
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+    "+pc",
+    "+p2p",
+]
+
diff --git a/third_party/libwebrtc/test/peer_scenario/peer_scenario.cc b/third_party/libwebrtc/test/peer_scenario/peer_scenario.cc
new file mode 100644
index 0000000000..485e33f67f
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/peer_scenario.cc
@@ -0,0 +1,127 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/peer_scenario/peer_scenario.h"
+
+#include "absl/flags/flag.h"
+#include "absl/memory/memory.h"
+#include "rtc_base/null_socket_server.h"
+#include "rtc_base/string_encode.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/logging/file_log_writer.h"
+#include "test/testsupport/file_utils.h"
+#include "test/time_controller/real_time_controller.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+ABSL_FLAG(bool, peer_logs, false, "Save logs from peer scenario framework.");
+ABSL_FLAG(std::string,
+          peer_logs_root,
+          "",
+          "Output root path, based on project root if unset.");
+
+namespace webrtc {
+namespace test {
+namespace {
+std::unique_ptr<FileLogWriterFactory> GetPeerScenarioLogManager(
+    std::string file_name) {
+  if (absl::GetFlag(FLAGS_peer_logs) && !file_name.empty()) {
+    std::string output_root = absl::GetFlag(FLAGS_peer_logs_root);
+    if (output_root.empty())
+      output_root = OutputPath() + "output_data/";
+
+    auto base_filename = output_root + file_name + ".";
+    RTC_LOG(LS_INFO) << "Saving peer scenario logs to: " << base_filename;
+    return std::make_unique<FileLogWriterFactory>(base_filename);
+  }
+  return nullptr;
+}
+}  // namespace
+
+PeerScenario::PeerScenario(const testing::TestInfo& test_info, TimeMode mode)
+    : PeerScenario(
+          std::string(test_info.test_suite_name()) + "/" + test_info.name(),
+          mode) {}
+
+PeerScenario::PeerScenario(std::string file_name, TimeMode mode)
+    : PeerScenario(GetPeerScenarioLogManager(file_name), mode) {}
+
+PeerScenario::PeerScenario(
+    std::unique_ptr<LogWriterFactoryInterface> log_writer_manager,
+    TimeMode mode)
+    : log_writer_manager_(std::move(log_writer_manager)),
+      net_(mode, EmulatedNetworkStatsGatheringMode::kDefault),
+      signaling_thread_(net_.time_controller()->GetMainThread()) {}
+
+PeerScenarioClient* PeerScenario::CreateClient(
+    PeerScenarioClient::Config config) {
+  return CreateClient(
+      std::string("client_") + rtc::ToString(peer_clients_.size() + 1), config);
+}
+
+PeerScenarioClient* PeerScenario::CreateClient(
+    std::string name,
+    PeerScenarioClient::Config config) {
+  peer_clients_.emplace_back(net(), signaling_thread_,
+                             GetLogWriterFactory(name), config);
+  return &peer_clients_.back();
+}
+
+SignalingRoute PeerScenario::ConnectSignaling(
+    PeerScenarioClient* caller,
+    PeerScenarioClient* callee,
+    std::vector<EmulatedNetworkNode*> send_link,
+    std::vector<EmulatedNetworkNode*> ret_link) {
+  return SignalingRoute(caller, callee, net_.CreateCrossTrafficRoute(send_link),
+                        net_.CreateCrossTrafficRoute(ret_link));
+}
+
+void PeerScenario::SimpleConnection(
+    PeerScenarioClient* caller,
+    PeerScenarioClient* callee,
+    std::vector<EmulatedNetworkNode*> send_link,
+    std::vector<EmulatedNetworkNode*> ret_link) {
+  net()->CreateRoute(caller->endpoint(), send_link, callee->endpoint());
+  net()->CreateRoute(callee->endpoint(), ret_link, caller->endpoint());
+  auto signaling = ConnectSignaling(caller, callee, send_link, ret_link);
+  signaling.StartIceSignaling();
+  std::atomic<bool> done(false);
+  signaling.NegotiateSdp(
+      [&](const SessionDescriptionInterface&) { done = true; });
+  RTC_CHECK(WaitAndProcess(&done));
+}
+
+void PeerScenario::AttachVideoQualityAnalyzer(VideoQualityAnalyzer* analyzer,
+                                              VideoTrackInterface* send_track,
+                                              PeerScenarioClient* receiver) {
+  video_quality_pairs_.emplace_back(clock(), analyzer);
+  auto pair = &video_quality_pairs_.back();
+  send_track->AddOrUpdateSink(&pair->capture_tap_, rtc::VideoSinkWants());
+  receiver->AddVideoReceiveSink(send_track->id(), &pair->decode_tap_);
+}
+
+bool PeerScenario::WaitAndProcess(std::atomic<bool>* event,
+                                  TimeDelta max_duration) {
+  return net_.time_controller()->Wait([event] { return event->load(); },
+                                      max_duration);
+}
+
+void PeerScenario::ProcessMessages(TimeDelta duration) {
+  net_.time_controller()->AdvanceTime(duration);
+}
+
+std::unique_ptr<LogWriterFactoryInterface> PeerScenario::GetLogWriterFactory(
+    std::string name) {
+  if (!log_writer_manager_ || name.empty())
+    return nullptr;
+  return std::make_unique<LogWriterFactoryAddPrefix>(log_writer_manager_.get(),
+                                                     name);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/peer_scenario/peer_scenario.h b/third_party/libwebrtc/test/peer_scenario/peer_scenario.h
new file mode 100644
index 0000000000..a177eeaac6
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/peer_scenario.h
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PEER_SCENARIO_PEER_SCENARIO_H_
+#define TEST_PEER_SCENARIO_PEER_SCENARIO_H_
+
+// The peer connection scenario test framework enables writing end to end unit
+// tests on the peer connection level. It's similar to the Scenario test but
+// uses the full stack, including SDP and ICE negotiation. This ensures that
+// features work end to end. It's also diffferent from the other tests on peer
+// connection level in that it does not rely on any mocks or fakes other than
+// for media input and networking. Additionally it provides direct access to the
+// underlying peer connection class.
+
+#include <list>
+#include <vector>
+
+#include "api/test/time_controller.h"
+#include "test/gtest.h"
+#include "test/logging/log_writer.h"
+#include "test/network/network_emulation_manager.h"
+#include "test/peer_scenario/peer_scenario_client.h"
+#include "test/peer_scenario/signaling_route.h"
+#include "test/scenario/stats_collection.h"
+#include "test/scenario/video_frame_matcher.h"
+
+namespace webrtc {
+namespace test {
+// The PeerScenario class represents a PeerConnection simulation scenario. The
+// main purpose is to maintain ownership and ensure safe destruction order of
+// clients and network emulation. Additionally it reduces the amount of boiler
+// plate requited for some actions. For example usage see the existing tests
+// using this class. Note that it should be used from a single calling thread.
+// This thread will also be assigned as the signaling thread for all peer
+// connections that are created. This means that the process methods must be
+// used when waiting to ensure that messages are processed on the signaling
+// thread.
+class PeerScenario {
+ public:
+  // The name is used for log output when those are enabled by the --peer_logs
+  // command line flag. Optionally, the TestInfo struct available in gtest can
+  // be used to automatically generate a path based on the test name.
+  explicit PeerScenario(const testing::TestInfo& test_info,
+                        TimeMode mode = TimeMode::kSimulated);
+  explicit PeerScenario(std::string file_name,
+                        TimeMode mode = TimeMode::kSimulated);
+  explicit PeerScenario(
+      std::unique_ptr<LogWriterFactoryInterface> log_writer_manager,
+      TimeMode mode = TimeMode::kSimulated);
+
+  NetworkEmulationManagerImpl* net() { return &net_; }
+
+  // Creates a client wrapping a peer connection conforming to the given config.
+  // The client  will share the signaling thread with the scenario. To maintain
+  // control of destruction order, ownership is kept within the scenario.
+  PeerScenarioClient* CreateClient(PeerScenarioClient::Config config);
+  PeerScenarioClient* CreateClient(std::string name,
+                                   PeerScenarioClient::Config config);
+
+  // Sets up a signaling route that can be used for SDP and ICE.
+  SignalingRoute ConnectSignaling(PeerScenarioClient* caller,
+                                  PeerScenarioClient* callee,
+                                  std::vector<EmulatedNetworkNode*> send_link,
+                                  std::vector<EmulatedNetworkNode*> ret_link);
+
+  // Connects two clients over given links. This will also start ICE signaling
+  // and SDP negotiation with default behavior. For customized behavior,
+  // ConnectSignaling should be used to allow more detailed control, for
+  // instance to allow different signaling and media routes.
+  void SimpleConnection(PeerScenarioClient* caller,
+                        PeerScenarioClient* callee,
+                        std::vector<EmulatedNetworkNode*> send_link,
+                        std::vector<EmulatedNetworkNode*> ret_link);
+
+  // Starts feeding the results of comparing captured frames from `send_track`
+  // with decoded frames on `receiver` to `analyzer`.
+  // TODO(srte): Provide a way to detach to allow removal of tracks.
+  void AttachVideoQualityAnalyzer(VideoQualityAnalyzer* analyzer,
+                                  VideoTrackInterface* send_track,
+                                  PeerScenarioClient* receiver);
+
+  // Waits on `event` while processing messages on the signaling thread.
+  bool WaitAndProcess(std::atomic<bool>* event,
+                      TimeDelta max_duration = TimeDelta::Seconds(5));
+
+  // Process messages on the signaling thread for the given duration.
+  void ProcessMessages(TimeDelta duration);
+
+ private:
+  // Helper struct to maintain ownership of the matcher and taps.
+  struct PeerVideoQualityPair {
+   public:
+    PeerVideoQualityPair(Clock* capture_clock, VideoQualityAnalyzer* analyzer)
+        : matcher_({analyzer->Handler()}),
+          capture_tap_(capture_clock, &matcher_),
+          decode_tap_(capture_clock, &matcher_, 0) {}
+    VideoFrameMatcher matcher_;
+    CapturedFrameTap capture_tap_;
+    DecodedFrameTap decode_tap_;
+  };
+
+  Clock* clock() { return Clock::GetRealTimeClock(); }
+
+  std::unique_ptr<LogWriterFactoryInterface> GetLogWriterFactory(
+      std::string name);
+
+  const std::unique_ptr<LogWriterFactoryInterface> log_writer_manager_;
+  NetworkEmulationManagerImpl net_;
+  rtc::Thread* const signaling_thread_;
+  std::list<PeerVideoQualityPair> video_quality_pairs_;
+  std::list<PeerScenarioClient> peer_clients_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_PEER_SCENARIO_PEER_SCENARIO_H_
diff --git a/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc
new file mode 100644
index 0000000000..5d77f17561
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.cc
@@ -0,0 +1,428 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/peer_scenario/peer_scenario_client.h"
+
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/memory/memory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/test/create_time_controller.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/video_codecs/builtin_video_decoder_factory.h"
+#include "api/video_codecs/builtin_video_encoder_factory.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "test/fake_decoder.h"
+#include "test/fake_vp8_encoder.h"
+#include "test/frame_generator_capturer.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+constexpr char kCommonStreamId[] = "stream_id";
+
+std::map<int, EmulatedEndpoint*> CreateEndpoints(
+    NetworkEmulationManager* net,
+    std::map<int, EmulatedEndpointConfig> endpoint_configs) {
+  std::map<int, EmulatedEndpoint*> endpoints;
+  for (const auto& kv : endpoint_configs)
+    endpoints[kv.first] = net->CreateEndpoint(kv.second);
+  return endpoints;
+}
+
+class LambdaPeerConnectionObserver final : public PeerConnectionObserver {
+ public:
+  explicit LambdaPeerConnectionObserver(
+      PeerScenarioClient::CallbackHandlers* handlers)
+      : handlers_(handlers) {}
+  void OnSignalingChange(
+      PeerConnectionInterface::SignalingState new_state) override {
+    for (const auto& handler : handlers_->on_signaling_change)
+      handler(new_state);
+  }
+  void OnDataChannel(
+      rtc::scoped_refptr<DataChannelInterface> data_channel) override {
+    for (const auto& handler : handlers_->on_data_channel)
+      handler(data_channel);
+  }
+  void OnRenegotiationNeeded() override {
+    for (const auto& handler : handlers_->on_renegotiation_needed)
+      handler();
+  }
+  void OnStandardizedIceConnectionChange(
+      PeerConnectionInterface::IceConnectionState new_state) override {
+    for (const auto& handler : handlers_->on_standardized_ice_connection_change)
+      handler(new_state);
+  }
+  void OnConnectionChange(
+      PeerConnectionInterface::PeerConnectionState new_state) override {
+    for (const auto& handler : handlers_->on_connection_change)
+      handler(new_state);
+  }
+  void OnIceGatheringChange(
+      PeerConnectionInterface::IceGatheringState new_state) override {
+    for (const auto& handler : handlers_->on_ice_gathering_change)
+      handler(new_state);
+  }
+  void OnIceCandidate(const IceCandidateInterface* candidate) override {
+    for (const auto& handler : handlers_->on_ice_candidate)
+      handler(candidate);
+  }
+  void OnIceCandidateError(const std::string& address,
+                           int port,
+                           const std::string& url,
+                           int error_code,
+                           const std::string& error_text) override {
+    for (const auto& handler : handlers_->on_ice_candidate_error)
+      handler(address, port, url, error_code, error_text);
+  }
+  void OnIceCandidatesRemoved(
+      const std::vector<cricket::Candidate>& candidates) override {
+    for (const auto& handler : handlers_->on_ice_candidates_removed)
+      handler(candidates);
+  }
+  void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
+                  const std::vector<rtc::scoped_refptr<MediaStreamInterface> >&
+                      streams) override {
+    for (const auto& handler : handlers_->on_add_track)
+      handler(receiver, streams);
+  }
+  void OnTrack(
+      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override {
+    for (const auto& handler : handlers_->on_track)
+      handler(transceiver);
+  }
+  void OnRemoveTrack(
+      rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
+    for (const auto& handler : handlers_->on_remove_track)
+      handler(receiver);
+  }
+
+ private:
+  PeerScenarioClient::CallbackHandlers* handlers_;
+};
+
+class LambdaCreateSessionDescriptionObserver
+    : public CreateSessionDescriptionObserver {
+ public:
+  explicit LambdaCreateSessionDescriptionObserver(
+      std::function<void(std::unique_ptr<SessionDescriptionInterface> desc)>
+          on_success)
+      : on_success_(on_success) {}
+  void OnSuccess(SessionDescriptionInterface* desc) override {
+    // Takes ownership of answer, according to CreateSessionDescriptionObserver
+    // convention.
+    on_success_(absl::WrapUnique(desc));
+  }
+  void OnFailure(RTCError error) override {
+    RTC_DCHECK_NOTREACHED() << error.message();
+  }
+
+ private:
+  std::function<void(std::unique_ptr<SessionDescriptionInterface> desc)>
+      on_success_;
+};
+
+class LambdaSetLocalDescriptionObserver
+    : public SetLocalDescriptionObserverInterface {
+ public:
+  explicit LambdaSetLocalDescriptionObserver(
+      std::function<void(RTCError)> on_complete)
+      : on_complete_(on_complete) {}
+  void OnSetLocalDescriptionComplete(RTCError error) override {
+    on_complete_(error);
+  }
+
+ private:
+  std::function<void(RTCError)> on_complete_;
+};
+
+class LambdaSetRemoteDescriptionObserver
+    : public SetRemoteDescriptionObserverInterface {
+ public:
+  explicit LambdaSetRemoteDescriptionObserver(
+      std::function<void(RTCError)> on_complete)
+      : on_complete_(on_complete) {}
+  void OnSetRemoteDescriptionComplete(RTCError error) override {
+    on_complete_(error);
+  }
+
+ private:
+  std::function<void(RTCError)> on_complete_;
+};
+
+class FakeVideoEncoderFactory : public VideoEncoderFactory {
+ public:
+  FakeVideoEncoderFactory(Clock* clock) : clock_(clock) {}
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
+    return {SdpVideoFormat("VP8")};
+  }
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override {
+    RTC_CHECK_EQ(format.name, "VP8");
+    return std::make_unique<FakeVp8Encoder>(clock_);
+  }
+
+ private:
+  Clock* const clock_;
+};
+class FakeVideoDecoderFactory : public VideoDecoderFactory {
+ public:
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
+    return {SdpVideoFormat("VP8")};
+  }
+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
+      const SdpVideoFormat& format) override {
+    return std::make_unique<FakeDecoder>();
+  }
+};
+}  // namespace
+
+PeerScenarioClient::PeerScenarioClient(
+    NetworkEmulationManager* net,
+    rtc::Thread* signaling_thread,
+    std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
+    PeerScenarioClient::Config config)
+    : endpoints_(CreateEndpoints(net, config.endpoints)),
+      task_queue_factory_(net->time_controller()->GetTaskQueueFactory()),
+      signaling_thread_(signaling_thread),
+      log_writer_factory_(std::move(log_writer_factory)),
+      worker_thread_(net->time_controller()->CreateThread("worker")),
+      handlers_(config.handlers),
+      observer_(new LambdaPeerConnectionObserver(&handlers_)) {
+  handlers_.on_track.push_back(
+      [this](rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+        auto track = transceiver->receiver()->track().get();
+        if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
+          auto* video = static_cast<VideoTrackInterface*>(track);
+          RTC_DCHECK_RUN_ON(signaling_thread_);
+          for (auto* sink : track_id_to_video_sinks_[track->id()]) {
+            video->AddOrUpdateSink(sink, rtc::VideoSinkWants());
+          }
+        }
+      });
+  handlers_.on_signaling_change.push_back(
+      [this](PeerConnectionInterface::SignalingState state) {
+        RTC_DCHECK_RUN_ON(signaling_thread_);
+        if (state == PeerConnectionInterface::SignalingState::kStable &&
+            peer_connection_->current_remote_description()) {
+          for (const auto& candidate : pending_ice_candidates_) {
+            RTC_CHECK(peer_connection_->AddIceCandidate(candidate.get()));
+          }
+          pending_ice_candidates_.clear();
+        }
+      });
+
+  std::vector<EmulatedEndpoint*> endpoints_vector;
+  for (const auto& kv : endpoints_)
+    endpoints_vector.push_back(kv.second);
+  auto* manager = net->CreateEmulatedNetworkManagerInterface(endpoints_vector);
+
+  PeerConnectionFactoryDependencies pcf_deps;
+  pcf_deps.network_thread = manager->network_thread();
+  pcf_deps.signaling_thread = signaling_thread_;
+  pcf_deps.worker_thread = worker_thread_.get();
+  pcf_deps.call_factory =
+      CreateTimeControllerBasedCallFactory(net->time_controller());
+  pcf_deps.task_queue_factory =
+      net->time_controller()->CreateTaskQueueFactory();
+  pcf_deps.event_log_factory =
+      std::make_unique<RtcEventLogFactory>(task_queue_factory_);
+  pcf_deps.trials = std::make_unique<FieldTrialBasedConfig>();
+
+  cricket::MediaEngineDependencies media_deps;
+  media_deps.task_queue_factory = task_queue_factory_;
+  media_deps.adm = TestAudioDeviceModule::Create(
+      task_queue_factory_,
+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(
+          config.audio.pulsed_noise->amplitude *
+              std::numeric_limits<int16_t>::max(),
+          config.audio.sample_rate, config.audio.channels),
+      TestAudioDeviceModule::CreateDiscardRenderer(config.audio.sample_rate));
+
+  media_deps.audio_processing = AudioProcessingBuilder().Create();
+  if (config.video.use_fake_codecs) {
+    media_deps.video_encoder_factory =
+        std::make_unique<FakeVideoEncoderFactory>(
+            net->time_controller()->GetClock());
+    media_deps.video_decoder_factory =
+        std::make_unique<FakeVideoDecoderFactory>();
+  } else {
+    media_deps.video_encoder_factory = CreateBuiltinVideoEncoderFactory();
+    media_deps.video_decoder_factory = CreateBuiltinVideoDecoderFactory();
+  }
+  media_deps.audio_encoder_factory = CreateBuiltinAudioEncoderFactory();
+  media_deps.audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
+  media_deps.trials = pcf_deps.trials.get();
+
+  pcf_deps.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+  pcf_deps.fec_controller_factory = nullptr;
+  pcf_deps.network_controller_factory = nullptr;
+  pcf_deps.network_state_predictor_factory = nullptr;
+
+  pc_factory_ = CreateModularPeerConnectionFactory(std::move(pcf_deps));
+  PeerConnectionFactoryInterface::Options pc_options;
+  pc_options.disable_encryption = config.disable_encryption;
+  pc_factory_->SetOptions(pc_options);
+
+  PeerConnectionDependencies pc_deps(observer_.get());
+  pc_deps.allocator = std::make_unique<cricket::BasicPortAllocator>(
+      manager->network_manager(), manager->packet_socket_factory());
+  pc_deps.allocator->set_flags(pc_deps.allocator->flags() |
+                               cricket::PORTALLOCATOR_DISABLE_TCP);
+  peer_connection_ =
+      pc_factory_
+          ->CreatePeerConnectionOrError(config.rtc_config, std::move(pc_deps))
+          .MoveValue();
+  if (log_writer_factory_) {
+    peer_connection_->StartRtcEventLog(log_writer_factory_->Create(".rtc.dat"),
+                                       /*output_period_ms=*/1000);
+  }
+}
+
+EmulatedEndpoint* PeerScenarioClient::endpoint(int index) {
+  RTC_CHECK_GT(endpoints_.size(), index);
+  return endpoints_.at(index);
+}
+
+PeerScenarioClient::AudioSendTrack PeerScenarioClient::CreateAudio(
+    std::string track_id,
+    cricket::AudioOptions options) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  AudioSendTrack res;
+  auto source = pc_factory_->CreateAudioSource(options);
+  auto track = pc_factory_->CreateAudioTrack(track_id, source.get());
+  res.track = track;
+  res.sender = peer_connection_->AddTrack(track, {kCommonStreamId}).value();
+  return res;
+}
+
+PeerScenarioClient::VideoSendTrack PeerScenarioClient::CreateVideo(
+    std::string track_id,
+    VideoSendTrackConfig config) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  VideoSendTrack res;
+  auto capturer = FrameGeneratorCapturer::Create(clock(), *task_queue_factory_,
+                                                 config.generator);
+  res.capturer = capturer.get();
+  capturer->Init();
+  res.source = rtc::make_ref_counted<FrameGeneratorCapturerVideoTrackSource>(
+      std::move(capturer), config.screencast);
+  auto track = pc_factory_->CreateVideoTrack(track_id, res.source.get());
+  res.track = track.get();
+  res.sender =
+      peer_connection_->AddTrack(track, {kCommonStreamId}).MoveValue().get();
+  return res;
+}
+
+void PeerScenarioClient::AddVideoReceiveSink(
+    std::string track_id,
+    rtc::VideoSinkInterface<VideoFrame>* video_sink) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  track_id_to_video_sinks_[track_id].push_back(video_sink);
+}
+
+void PeerScenarioClient::CreateAndSetSdp(
+    std::function<void(SessionDescriptionInterface*)> munge_offer,
+    std::function<void(std::string)> offer_handler) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  peer_connection_->CreateOffer(
+      rtc::make_ref_counted<LambdaCreateSessionDescriptionObserver>(
+          [=](std::unique_ptr<SessionDescriptionInterface> offer) {
+            RTC_DCHECK_RUN_ON(signaling_thread_);
+            if (munge_offer) {
+              munge_offer(offer.get());
+            }
+            std::string sdp_offer;
+            RTC_CHECK(offer->ToString(&sdp_offer));
+            peer_connection_->SetLocalDescription(
+                std::move(offer),
+                rtc::make_ref_counted<LambdaSetLocalDescriptionObserver>(
+                    [sdp_offer, offer_handler](RTCError) {
+                      offer_handler(sdp_offer);
+                    }));
+          })
+          .get(),
+      PeerConnectionInterface::RTCOfferAnswerOptions());
+}
+
+void PeerScenarioClient::SetSdpOfferAndGetAnswer(
+    std::string remote_offer,
+    std::function<void(std::string)> answer_handler) {
+  if (!signaling_thread_->IsCurrent()) {
+    signaling_thread_->PostTask(
+        [=] { SetSdpOfferAndGetAnswer(remote_offer, answer_handler); });
+    return;
+  }
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  peer_connection_->SetRemoteDescription(
+      CreateSessionDescription(SdpType::kOffer, remote_offer),
+      rtc::make_ref_counted<LambdaSetRemoteDescriptionObserver>([=](RTCError) {
+        RTC_DCHECK_RUN_ON(signaling_thread_);
+        peer_connection_->CreateAnswer(
+            rtc::make_ref_counted<LambdaCreateSessionDescriptionObserver>(
+                [=](std::unique_ptr<SessionDescriptionInterface> answer) {
+                  RTC_DCHECK_RUN_ON(signaling_thread_);
+                  std::string sdp_answer;
+                  answer->ToString(&sdp_answer);
+                  RTC_LOG(LS_INFO) << sdp_answer;
+                  peer_connection_->SetLocalDescription(
+                      std::move(answer),
+                      rtc::make_ref_counted<LambdaSetLocalDescriptionObserver>(
+                          [answer_handler, sdp_answer](RTCError) {
+                            answer_handler(sdp_answer);
+                          }));
+                })
+                .get(),
+            PeerConnectionInterface::RTCOfferAnswerOptions());
+      }));
+}
+
+void PeerScenarioClient::SetSdpAnswer(
+    std::string remote_answer,
+    std::function<void(const SessionDescriptionInterface&)> done_handler) {
+  if (!signaling_thread_->IsCurrent()) {
+    signaling_thread_->PostTask(
+        [=] { SetSdpAnswer(remote_answer, done_handler); });
+    return;
+  }
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  peer_connection_->SetRemoteDescription(
+      CreateSessionDescription(SdpType::kAnswer, remote_answer),
+      rtc::make_ref_counted<LambdaSetRemoteDescriptionObserver>(
+          [remote_answer, done_handler](RTCError) {
+            auto answer =
+                CreateSessionDescription(SdpType::kAnswer, remote_answer);
+            done_handler(*answer);
+          }));
+}
+
+void PeerScenarioClient::AddIceCandidate(
+    std::unique_ptr<IceCandidateInterface> candidate) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  if (peer_connection_->signaling_state() ==
+          PeerConnectionInterface::SignalingState::kStable &&
+      peer_connection_->current_remote_description()) {
+    RTC_CHECK(peer_connection_->AddIceCandidate(candidate.get()));
+  } else {
+    pending_ice_candidates_.push_back(std::move(candidate));
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.h b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.h
new file mode 100644
index 0000000000..ab6aac9cf8
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/peer_scenario_client.h
@@ -0,0 +1,179 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PEER_SCENARIO_PEER_SCENARIO_CLIENT_H_
+#define TEST_PEER_SCENARIO_PEER_SCENARIO_CLIENT_H_
+
+#include <functional>
+#include <list>
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "api/peer_connection_interface.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/time_controller.h"
+#include "pc/test/frame_generator_capturer_video_track_source.h"
+#include "test/logging/log_writer.h"
+
+namespace webrtc {
+namespace test {
+
+// Wrapper for a PeerConnection for use in PeerScenario tests. It's intended to
+// be a minimal wrapper for a peer connection that's simple to use in testing.
+// In particular the constructor hides a lot of the required setup for a peer
+// connection.
+class PeerScenarioClient {
+ public:
+  struct CallbackHandlers {
+    std::vector<std::function<void(PeerConnectionInterface::SignalingState)>>
+        on_signaling_change;
+    std::vector<std::function<void(rtc::scoped_refptr<DataChannelInterface>)>>
+        on_data_channel;
+    std::vector<std::function<void()>> on_renegotiation_needed;
+    std::vector<
+        std::function<void(PeerConnectionInterface::IceConnectionState)>>
+        on_standardized_ice_connection_change;
+    std::vector<
+        std::function<void(PeerConnectionInterface::PeerConnectionState)>>
+        on_connection_change;
+    std::vector<std::function<void(PeerConnectionInterface::IceGatheringState)>>
+        on_ice_gathering_change;
+    std::vector<std::function<void(const IceCandidateInterface*)>>
+        on_ice_candidate;
+    std::vector<std::function<void(const std::string&,
+                                   int,
+                                   const std::string&,
+                                   int,
+                                   const std::string&)>>
+        on_ice_candidate_error;
+    std::vector<std::function<void(const std::vector<cricket::Candidate>&)>>
+        on_ice_candidates_removed;
+    std::vector<std::function<void(
+        rtc::scoped_refptr<RtpReceiverInterface>,
+        const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&)>>
+        on_add_track;
+    std::vector<
+        std::function<void(rtc::scoped_refptr<RtpTransceiverInterface>)>>
+        on_track;
+    std::vector<std::function<void(rtc::scoped_refptr<RtpReceiverInterface>)>>
+        on_remove_track;
+  };
+  struct Config {
+    // WebRTC only support one audio device that is setup up on construction, so
+    // we provide the audio generator configuration here rather than on creation
+    // of the tracks. This is unlike video, where multiple capture sources can
+    // be used at the same time.
+    struct AudioSource {
+      int sample_rate = 48000;
+      int channels = 1;
+      struct PulsedNoise {
+        double amplitude = 0.1;
+      };
+      absl::optional<PulsedNoise> pulsed_noise = PulsedNoise();
+    } audio;
+    struct Video {
+      bool use_fake_codecs = false;
+    } video;
+    // The created endpoints can be accessed using the map key as `index` in
+    // PeerScenarioClient::endpoint(index).
+    std::map<int, EmulatedEndpointConfig> endpoints = {
+        {0, EmulatedEndpointConfig()}};
+    CallbackHandlers handlers;
+    PeerConnectionInterface::RTCConfiguration rtc_config;
+    bool disable_encryption = false;
+    Config() { rtc_config.sdp_semantics = SdpSemantics::kUnifiedPlan; }
+  };
+
+  struct VideoSendTrackConfig {
+    FrameGeneratorCapturerConfig generator;
+    bool screencast = false;
+  };
+
+  struct AudioSendTrack {
+    rtc::scoped_refptr<AudioTrackInterface> track;
+    rtc::scoped_refptr<RtpSenderInterface> sender;
+  };
+
+  struct VideoSendTrack {
+    // Raw pointer to the capturer owned by `source`.
+    FrameGeneratorCapturer* capturer;
+    rtc::scoped_refptr<FrameGeneratorCapturerVideoTrackSource> source;
+    rtc::scoped_refptr<VideoTrackInterface> track;
+    rtc::scoped_refptr<RtpSenderInterface> sender;
+  };
+
+  PeerScenarioClient(
+      NetworkEmulationManager* net,
+      rtc::Thread* signaling_thread,
+      std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
+      Config config);
+
+  PeerConnectionFactoryInterface* factory() { return pc_factory_.get(); }
+  PeerConnectionInterface* pc() {
+    RTC_DCHECK_RUN_ON(signaling_thread_);
+    return peer_connection_.get();
+  }
+  rtc::Thread* thread() { return signaling_thread_; }
+  Clock* clock() { return Clock::GetRealTimeClock(); }
+
+  // Returns the endpoint created from the EmulatedEndpointConfig with the same
+  // index in PeerScenarioClient::config.
+  EmulatedEndpoint* endpoint(int index = 0);
+
+  AudioSendTrack CreateAudio(std::string track_id,
+                             cricket::AudioOptions options);
+  VideoSendTrack CreateVideo(std::string track_id, VideoSendTrackConfig config);
+
+  void AddVideoReceiveSink(std::string track_id,
+                           rtc::VideoSinkInterface<VideoFrame>* video_sink);
+
+  CallbackHandlers* handlers() { return &handlers_; }
+
+  // The `munge_offer` function can be used to munge the SDP, i.e. modify a
+  // local description afer creating it but before setting it. Note that this is
+  // legacy behavior. It's added here only to be able to have test coverage for
+  // scenarios even if they are not spec compliant.
+  void CreateAndSetSdp(
+      std::function<void(SessionDescriptionInterface*)> munge_offer,
+      std::function<void(std::string)> offer_handler);
+  void SetSdpOfferAndGetAnswer(std::string remote_offer,
+                               std::function<void(std::string)> answer_handler);
+  void SetSdpAnswer(
+      std::string remote_answer,
+      std::function<void(const SessionDescriptionInterface& answer)>
+          done_handler);
+
+  // Adds the given ice candidate when the peer connection is ready.
+  void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate);
+
+ private:
+  const std::map<int, EmulatedEndpoint*> endpoints_;
+  TaskQueueFactory* const task_queue_factory_;
+  rtc::Thread* const signaling_thread_;
+  const std::unique_ptr<LogWriterFactoryInterface> log_writer_factory_;
+  const std::unique_ptr<rtc::Thread> worker_thread_;
+  CallbackHandlers handlers_ RTC_GUARDED_BY(signaling_thread_);
+  const std::unique_ptr<PeerConnectionObserver> observer_;
+  std::map<std::string, std::vector<rtc::VideoSinkInterface<VideoFrame>*>>
+      track_id_to_video_sinks_ RTC_GUARDED_BY(signaling_thread_);
+  std::list<std::unique_ptr<IceCandidateInterface>> pending_ice_candidates_
+      RTC_GUARDED_BY(signaling_thread_);
+
+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
+  rtc::scoped_refptr<PeerConnectionInterface> peer_connection_
+      RTC_GUARDED_BY(signaling_thread_);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_PEER_SCENARIO_PEER_SCENARIO_CLIENT_H_
diff --git a/third_party/libwebrtc/test/peer_scenario/scenario_connection.cc b/third_party/libwebrtc/test/peer_scenario/scenario_connection.cc
new file mode 100644
index 0000000000..66eca275d1
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/scenario_connection.cc
@@ -0,0 +1,242 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/peer_scenario/scenario_connection.h"
+
+#include "absl/memory/memory.h"
+#include "media/base/rtp_utils.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "p2p/client/basic_port_allocator.h"
+#include "pc/channel.h"
+#include "pc/jsep_transport_controller.h"
+#include "pc/rtp_transport_internal.h"
+#include "pc/session_description.h"
+#include "rtc_base/task_queue_for_test.h"
+
+namespace webrtc {
+class ScenarioIceConnectionImpl : public ScenarioIceConnection,
+                                  public sigslot::has_slots<>,
+                                  private JsepTransportController::Observer,
+                                  private RtpPacketSinkInterface {
+ public:
+  ScenarioIceConnectionImpl(test::NetworkEmulationManagerImpl* net,
+                            IceConnectionObserver* observer);
+  ~ScenarioIceConnectionImpl() override;
+
+  void SendRtpPacket(rtc::ArrayView<const uint8_t> packet_view) override;
+  void SendRtcpPacket(rtc::ArrayView<const uint8_t> packet_view) override;
+
+  void SetRemoteSdp(SdpType type, const std::string& remote_sdp) override;
+  void SetLocalSdp(SdpType type, const std::string& local_sdp) override;
+
+  EmulatedEndpoint* endpoint() override { return endpoint_; }
+  const cricket::TransportDescription& transport_description() const override {
+    return transport_description_;
+  }
+
+ private:
+  JsepTransportController::Config CreateJsepConfig();
+  bool OnTransportChanged(
+      const std::string& mid,
+      RtpTransportInternal* rtp_transport,
+      rtc::scoped_refptr<DtlsTransport> dtls_transport,
+      DataChannelTransportInterface* data_channel_transport) override;
+
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+  void OnCandidates(const std::string& mid,
+                    const std::vector<cricket::Candidate>& candidates);
+
+  IceConnectionObserver* const observer_;
+  EmulatedEndpoint* const endpoint_;
+  EmulatedNetworkManagerInterface* const manager_;
+  rtc::Thread* const signaling_thread_;
+  rtc::Thread* const network_thread_;
+  rtc::scoped_refptr<rtc::RTCCertificate> const certificate_
+      RTC_GUARDED_BY(network_thread_);
+  cricket::TransportDescription const transport_description_
+      RTC_GUARDED_BY(signaling_thread_);
+  std::unique_ptr<cricket::BasicPortAllocator> port_allocator_
+      RTC_GUARDED_BY(network_thread_);
+  std::unique_ptr<JsepTransportController> jsep_controller_;
+  RtpTransportInternal* rtp_transport_ RTC_GUARDED_BY(network_thread_) =
+      nullptr;
+  std::unique_ptr<SessionDescriptionInterface> remote_description_
+      RTC_GUARDED_BY(signaling_thread_);
+  std::unique_ptr<SessionDescriptionInterface> local_description_
+      RTC_GUARDED_BY(signaling_thread_);
+};
+
+std::unique_ptr<ScenarioIceConnection> ScenarioIceConnection::Create(
+    webrtc::test::NetworkEmulationManagerImpl* net,
+    IceConnectionObserver* observer) {
+  return std::make_unique<ScenarioIceConnectionImpl>(net, observer);
+}
+
+ScenarioIceConnectionImpl::ScenarioIceConnectionImpl(
+    test::NetworkEmulationManagerImpl* net,
+    IceConnectionObserver* observer)
+    : observer_(observer),
+      endpoint_(net->CreateEndpoint(EmulatedEndpointConfig())),
+      manager_(net->CreateEmulatedNetworkManagerInterface({endpoint_})),
+      signaling_thread_(rtc::Thread::Current()),
+      network_thread_(manager_->network_thread()),
+      certificate_(rtc::RTCCertificate::Create(
+          rtc::SSLIdentity::Create("", ::rtc::KT_DEFAULT))),
+      transport_description_(
+          /*transport_options*/ {},
+          rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH),
+          rtc::CreateRandomString(cricket::ICE_PWD_LENGTH),
+          cricket::IceMode::ICEMODE_FULL,
+          cricket::ConnectionRole::CONNECTIONROLE_PASSIVE,
+          rtc::SSLFingerprint::CreateFromCertificate(*certificate_.get())
+              .get()),
+      port_allocator_(
+          new cricket::BasicPortAllocator(manager_->network_manager(),
+                                          manager_->packet_socket_factory())),
+      jsep_controller_(
+          new JsepTransportController(network_thread_,
+                                      port_allocator_.get(),
+                                      /*async_resolver_factory*/ nullptr,
+                                      CreateJsepConfig())) {
+  SendTask(network_thread_, [this] {
+    RTC_DCHECK_RUN_ON(network_thread_);
+    uint32_t flags = cricket::PORTALLOCATOR_DISABLE_TCP;
+    port_allocator_->set_flags(port_allocator_->flags() | flags);
+    port_allocator_->Initialize();
+    RTC_CHECK(port_allocator_->SetConfiguration(/*stun_servers*/ {},
+                                                /*turn_servers*/ {}, 0,
+                                                webrtc::NO_PRUNE));
+    jsep_controller_->SetLocalCertificate(certificate_);
+  });
+}
+
+ScenarioIceConnectionImpl::~ScenarioIceConnectionImpl() {
+  SendTask(network_thread_, [this] {
+    RTC_DCHECK_RUN_ON(network_thread_);
+    jsep_controller_.reset();
+    port_allocator_.reset();
+    rtp_transport_ = nullptr;
+  });
+}
+
+JsepTransportController::Config ScenarioIceConnectionImpl::CreateJsepConfig() {
+  JsepTransportController::Config config;
+  config.transport_observer = this;
+  config.bundle_policy =
+      PeerConnectionInterface::BundlePolicy::kBundlePolicyMaxBundle;
+  config.rtcp_handler = [this](const rtc::CopyOnWriteBuffer& packet,
+                               int64_t packet_time_us) {
+    RTC_DCHECK_RUN_ON(network_thread_);
+    observer_->OnPacketReceived(packet);
+  };
+  config.field_trials = &field_trials;
+  return config;
+}
+
+void ScenarioIceConnectionImpl::SendRtpPacket(
+    rtc::ArrayView<const uint8_t> packet_view) {
+  rtc::CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
+                                ::cricket::kMaxRtpPacketLen);
+  network_thread_->PostTask([this, packet = std::move(packet)]() mutable {
+    RTC_DCHECK_RUN_ON(network_thread_);
+    if (rtp_transport_ != nullptr)
+      rtp_transport_->SendRtpPacket(&packet, rtc::PacketOptions(),
+                                    cricket::PF_SRTP_BYPASS);
+  });
+}
+
+void ScenarioIceConnectionImpl::SendRtcpPacket(
+    rtc::ArrayView<const uint8_t> packet_view) {
+  rtc::CopyOnWriteBuffer packet(packet_view.data(), packet_view.size(),
+                                ::cricket::kMaxRtpPacketLen);
+  network_thread_->PostTask([this, packet = std::move(packet)]() mutable {
+    RTC_DCHECK_RUN_ON(network_thread_);
+    if (rtp_transport_ != nullptr)
+      rtp_transport_->SendRtcpPacket(&packet, rtc::PacketOptions(),
+                                     cricket::PF_SRTP_BYPASS);
+  });
+}
+void ScenarioIceConnectionImpl::SetRemoteSdp(SdpType type,
+                                             const std::string& remote_sdp) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  remote_description_ = webrtc::CreateSessionDescription(type, remote_sdp);
+  jsep_controller_->SubscribeIceCandidateGathered(
+      [this](const std::string& transport,
+             const std::vector<cricket::Candidate>& candidate) {
+        ScenarioIceConnectionImpl::OnCandidates(transport, candidate);
+      });
+
+  auto res = jsep_controller_->SetRemoteDescription(
+      remote_description_->GetType(), remote_description_->description());
+  RTC_CHECK(res.ok()) << res.message();
+  RtpDemuxerCriteria criteria;
+  for (const auto& content : remote_description_->description()->contents()) {
+    if (content.media_description()->as_audio()) {
+      for (const auto& codec :
+           content.media_description()->as_audio()->codecs()) {
+        criteria.payload_types().insert(codec.id);
+      }
+    }
+    if (content.media_description()->as_video()) {
+      for (const auto& codec :
+           content.media_description()->as_video()->codecs()) {
+        criteria.payload_types().insert(codec.id);
+      }
+    }
+  }
+
+  network_thread_->PostTask([this, criteria]() {
+    RTC_DCHECK_RUN_ON(network_thread_);
+    RTC_DCHECK(rtp_transport_);
+    rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
+  });
+}
+
+void ScenarioIceConnectionImpl::SetLocalSdp(SdpType type,
+                                            const std::string& local_sdp) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  local_description_ = webrtc::CreateSessionDescription(type, local_sdp);
+  auto res = jsep_controller_->SetLocalDescription(
+      local_description_->GetType(), local_description_->description());
+  RTC_CHECK(res.ok()) << res.message();
+  jsep_controller_->MaybeStartGathering();
+}
+
+bool ScenarioIceConnectionImpl::OnTransportChanged(
+    const std::string& mid,
+    RtpTransportInternal* rtp_transport,
+    rtc::scoped_refptr<DtlsTransport> dtls_transport,
+    DataChannelTransportInterface* data_channel_transport) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  if (rtp_transport == nullptr) {
+    rtp_transport_->UnregisterRtpDemuxerSink(this);
+  } else {
+    RTC_DCHECK(rtp_transport_ == nullptr || rtp_transport_ == rtp_transport);
+    if (rtp_transport_ != rtp_transport) {
+      rtp_transport_ = rtp_transport;
+    }
+    RtpDemuxerCriteria criteria(mid);
+    rtp_transport_->RegisterRtpDemuxerSink(criteria, this);
+  }
+  return true;
+}
+
+void ScenarioIceConnectionImpl::OnRtpPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  observer_->OnPacketReceived(packet.Buffer());
+}
+
+void ScenarioIceConnectionImpl::OnCandidates(
+    const std::string& mid,
+    const std::vector<cricket::Candidate>& candidates) {
+  RTC_DCHECK_RUN_ON(signaling_thread_);
+  observer_->OnIceCandidates(mid, candidates);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/peer_scenario/scenario_connection.h b/third_party/libwebrtc/test/peer_scenario/scenario_connection.h
new file mode 100644
index 0000000000..e8cef527c5
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/scenario_connection.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PEER_SCENARIO_SCENARIO_CONNECTION_H_
+#define TEST_PEER_SCENARIO_SCENARIO_CONNECTION_H_
+
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/candidate.h"
+#include "api/jsep.h"
+#include "p2p/base/transport_description.h"
+#include "test/network/network_emulation_manager.h"
+#include "test/scoped_key_value_config.h"
+
+namespace webrtc {
+
+// ScenarioIceConnection provides the transport level functionality of a
+// PeerConnection for use in peer connection scenario tests. This allows
+// implementing custom server side behavior in tests.
+class ScenarioIceConnection {
+ public:
+  class IceConnectionObserver {
+   public:
+    // Called on network thread.
+    virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet) = 0;
+    // Called on signaling thread.
+    virtual void OnIceCandidates(
+        const std::string& mid,
+        const std::vector<cricket::Candidate>& candidates) = 0;
+
+   protected:
+    ~IceConnectionObserver() = default;
+  };
+  static std::unique_ptr<ScenarioIceConnection> Create(
+      test::NetworkEmulationManagerImpl* net,
+      IceConnectionObserver* observer);
+
+  virtual ~ScenarioIceConnection() = default;
+
+  // Posts tasks to send packets to network thread.
+  virtual void SendRtpPacket(rtc::ArrayView<const uint8_t> packet_view) = 0;
+  virtual void SendRtcpPacket(rtc::ArrayView<const uint8_t> packet_view) = 0;
+
+  // Used for ICE configuration, called on signaling thread.
+  virtual void SetRemoteSdp(SdpType type, const std::string& remote_sdp) = 0;
+  virtual void SetLocalSdp(SdpType type, const std::string& local_sdp) = 0;
+
+  virtual EmulatedEndpoint* endpoint() = 0;
+  virtual const cricket::TransportDescription& transport_description()
+      const = 0;
+
+  webrtc::test::ScopedKeyValueConfig field_trials;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_PEER_SCENARIO_SCENARIO_CONNECTION_H_
diff --git a/third_party/libwebrtc/test/peer_scenario/signaling_route.cc b/third_party/libwebrtc/test/peer_scenario/signaling_route.cc
new file mode 100644
index 0000000000..eeec7c8657
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/signaling_route.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/peer_scenario/signaling_route.h"
+
+#include <memory>
+
+#include "test/network/network_emulation_manager.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+constexpr size_t kIcePacketSize = 400;
+constexpr size_t kSdpPacketSize = 1200;
+
+struct IceMessage {
+  IceMessage() = default;
+  explicit IceMessage(const IceCandidateInterface* candidate)
+      : sdp_mid(candidate->sdp_mid()),
+        sdp_mline_index(candidate->sdp_mline_index()) {
+    RTC_CHECK(candidate->ToString(&sdp_line));
+  }
+  std::unique_ptr<IceCandidateInterface> AsCandidate() const {
+    SdpParseError err;
+    std::unique_ptr<IceCandidateInterface> candidate(
+        CreateIceCandidate(sdp_mid, sdp_mline_index, sdp_line, &err));
+    RTC_CHECK(candidate) << "Failed to parse: \"" << err.line
+                         << "\". Reason: " << err.description;
+    return candidate;
+  }
+  std::string sdp_mid;
+  int sdp_mline_index;
+  std::string sdp_line;
+};
+
+void StartIceSignalingForRoute(PeerScenarioClient* caller,
+                               PeerScenarioClient* callee,
+                               CrossTrafficRoute* send_route) {
+  caller->handlers()->on_ice_candidate.push_back(
+      [=](const IceCandidateInterface* candidate) {
+        IceMessage msg(candidate);
+        send_route->NetworkDelayedAction(kIcePacketSize, [callee, msg]() {
+          callee->thread()->PostTask(
+              [callee, msg]() { callee->AddIceCandidate(msg.AsCandidate()); });
+        });
+      });
+}
+
+void StartSdpNegotiation(
+    PeerScenarioClient* caller,
+    PeerScenarioClient* callee,
+    CrossTrafficRoute* send_route,
+    CrossTrafficRoute* ret_route,
+    std::function<void(SessionDescriptionInterface* offer)> munge_offer,
+    std::function<void(SessionDescriptionInterface*)> modify_offer,
+    std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
+  caller->CreateAndSetSdp(munge_offer, [=](std::string sdp_offer) {
+    if (modify_offer) {
+      auto offer = CreateSessionDescription(SdpType::kOffer, sdp_offer);
+      modify_offer(offer.get());
+      RTC_CHECK(offer->ToString(&sdp_offer));
+    }
+    send_route->NetworkDelayedAction(kSdpPacketSize, [=] {
+      callee->SetSdpOfferAndGetAnswer(sdp_offer, [=](std::string answer) {
+        ret_route->NetworkDelayedAction(kSdpPacketSize, [=] {
+          caller->SetSdpAnswer(std::move(answer), std::move(exchange_finished));
+        });
+      });
+    });
+  });
+}
+}  // namespace
+
+SignalingRoute::SignalingRoute(PeerScenarioClient* caller,
+                               PeerScenarioClient* callee,
+                               CrossTrafficRoute* send_route,
+                               CrossTrafficRoute* ret_route)
+    : caller_(caller),
+      callee_(callee),
+      send_route_(send_route),
+      ret_route_(ret_route) {}
+
+void SignalingRoute::StartIceSignaling() {
+  StartIceSignalingForRoute(caller_, callee_, send_route_);
+  StartIceSignalingForRoute(callee_, caller_, ret_route_);
+}
+
+void SignalingRoute::NegotiateSdp(
+    std::function<void(SessionDescriptionInterface*)> munge_offer,
+    std::function<void(SessionDescriptionInterface*)> modify_offer,
+    std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
+  StartSdpNegotiation(caller_, callee_, send_route_, ret_route_, munge_offer,
+                      modify_offer, exchange_finished);
+}
+
+void SignalingRoute::NegotiateSdp(
+    std::function<void(SessionDescriptionInterface*)> modify_offer,
+    std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
+  NegotiateSdp({}, modify_offer, exchange_finished);
+}
+
+void SignalingRoute::NegotiateSdp(
+    std::function<void(const SessionDescriptionInterface&)> exchange_finished) {
+  NegotiateSdp({}, {}, exchange_finished);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/peer_scenario/signaling_route.h b/third_party/libwebrtc/test/peer_scenario/signaling_route.h
new file mode 100644
index 0000000000..a95ae5c9f7
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/signaling_route.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PEER_SCENARIO_SIGNALING_ROUTE_H_
+#define TEST_PEER_SCENARIO_SIGNALING_ROUTE_H_
+
+#include <string>
+#include <utility>
+
+#include "test/network/network_emulation_manager.h"
+#include "test/peer_scenario/peer_scenario_client.h"
+
+namespace webrtc {
+namespace test {
+
+// Helper class to reduce the amount of boilerplate required for ICE signalling
+// ad SDP negotiation.
+class SignalingRoute {
+ public:
+  SignalingRoute(PeerScenarioClient* caller,
+                 PeerScenarioClient* callee,
+                 CrossTrafficRoute* send_route,
+                 CrossTrafficRoute* ret_route);
+
+  void StartIceSignaling();
+
+  // The `modify_offer` callback is used to modify an offer after the local
+  // description has been set. This is legal (but odd) behavior.
+  // The `munge_offer` callback is used to modify an offer between its creation
+  // and set local description. This behavior is forbidden according to the spec
+  // but available here in order to allow test coverage on corner cases.
+  // The `exchange_finished` callback is called with the answer produced after
+  // SDP negotations has completed.
+  // TODO(srte): Handle lossy links.
+  void NegotiateSdp(
+      std::function<void(SessionDescriptionInterface* offer)> munge_offer,
+      std::function<void(SessionDescriptionInterface* offer)> modify_offer,
+      std::function<void(const SessionDescriptionInterface& answer)>
+          exchange_finished);
+  void NegotiateSdp(
+      std::function<void(SessionDescriptionInterface* offer)> modify_offer,
+      std::function<void(const SessionDescriptionInterface& answer)>
+          exchange_finished);
+  void NegotiateSdp(
+      std::function<void(const SessionDescriptionInterface& answer)>
+          exchange_finished);
+  SignalingRoute reverse() {
+    return SignalingRoute(callee_, caller_, ret_route_, send_route_);
+  }
+
+ private:
+  PeerScenarioClient* const caller_;
+  PeerScenarioClient* const callee_;
+  CrossTrafficRoute* const send_route_;
+  CrossTrafficRoute* const ret_route_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_PEER_SCENARIO_SIGNALING_ROUTE_H_
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/BUILD.gn b/third_party/libwebrtc/test/peer_scenario/tests/BUILD.gn
new file mode 100644
index 0000000000..ba6ec20e84
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/tests/BUILD.gn
@@ -0,0 +1,30 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+if (rtc_include_tests) {
+  rtc_library("tests") {
+    testonly = true
+    sources = [
+      "peer_scenario_quality_test.cc",
+      "remote_estimate_test.cc",
+      "unsignaled_stream_test.cc",
+    ]
+    deps = [
+      "..:peer_scenario",
+      "../../:field_trial",
+      "../../:test_support",
+      "../../../media:rtc_media_base",
+      "../../../media:stream_params",
+      "../../../modules/rtp_rtcp:rtp_rtcp_format",
+      "../../../pc:media_session",
+      "../../../pc:session_description",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/peer_scenario_quality_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/peer_scenario_quality_test.cc
new file mode 100644
index 0000000000..911a68720f
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/tests/peer_scenario_quality_test.cc
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/gtest.h"
+#include "test/peer_scenario/peer_scenario.h"
+#include "test/peer_scenario/peer_scenario_client.h"
+
+namespace webrtc {
+namespace test {
+#if defined(WEBRTC_WIN)
+#define MAYBE_PsnrIsCollected DISABLED_PsnrIsCollected
+#else
+#define MAYBE_PsnrIsCollected PsnrIsCollected
+#endif
+TEST(PeerScenarioQualityTest, MAYBE_PsnrIsCollected) {
+  VideoQualityAnalyzer analyzer;
+  {
+    PeerScenario s(*test_info_);
+    auto caller = s.CreateClient(PeerScenarioClient::Config());
+    auto callee = s.CreateClient(PeerScenarioClient::Config());
+    PeerScenarioClient::VideoSendTrackConfig video_conf;
+    video_conf.generator.squares_video->framerate = 20;
+    auto video = caller->CreateVideo("VIDEO", video_conf);
+    auto link_builder = s.net()->NodeBuilder().delay_ms(100).capacity_kbps(600);
+    s.AttachVideoQualityAnalyzer(&analyzer, video.track.get(), callee);
+    s.SimpleConnection(caller, callee, {link_builder.Build().node},
+                       {link_builder.Build().node});
+    s.ProcessMessages(TimeDelta::Seconds(2));
+    // Exit scope to ensure that there's no pending tasks reporting to analyzer.
+  }
+
+  // We expect ca 40 frames to be produced, but to avoid flakiness on slow
+  // machines we only test for 10.
+  EXPECT_GT(analyzer.stats().render.count, 10);
+  EXPECT_GT(analyzer.stats().psnr_with_freeze.Mean(), 20);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc
new file mode 100644
index 0000000000..2dfbfdd3c9
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/tests/remote_estimate_test.cc
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "pc/media_session.h"
+#include "pc/session_description.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/peer_scenario/peer_scenario.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+RtpHeaderExtensionMap AudioExtensions(
+    const SessionDescriptionInterface& session) {
+  auto* audio_desc =
+      cricket::GetFirstAudioContentDescription(session.description());
+  return RtpHeaderExtensionMap(audio_desc->rtp_header_extensions());
+}
+
+}  // namespace
+
+TEST(RemoteEstimateEndToEnd, OfferedCapabilityIsInAnswer) {
+  PeerScenario s(*test_info_);
+
+  auto* caller = s.CreateClient(PeerScenarioClient::Config());
+  auto* callee = s.CreateClient(PeerScenarioClient::Config());
+
+  auto send_link = {s.net()->NodeBuilder().Build().node};
+  auto ret_link = {s.net()->NodeBuilder().Build().node};
+
+  s.net()->CreateRoute(caller->endpoint(), send_link, callee->endpoint());
+  s.net()->CreateRoute(callee->endpoint(), ret_link, caller->endpoint());
+
+  auto signaling = s.ConnectSignaling(caller, callee, send_link, ret_link);
+  caller->CreateVideo("VIDEO", PeerScenarioClient::VideoSendTrackConfig());
+  std::atomic<bool> offer_exchange_done(false);
+  signaling.NegotiateSdp(
+      [](SessionDescriptionInterface* offer) {
+        for (auto& cont : offer->description()->contents()) {
+          cont.media_description()->set_remote_estimate(true);
+        }
+      },
+      [&](const SessionDescriptionInterface& answer) {
+        for (auto& cont : answer.description()->contents()) {
+          EXPECT_TRUE(cont.media_description()->remote_estimate());
+        }
+        offer_exchange_done = true;
+      });
+  RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
+}
+
+TEST(RemoteEstimateEndToEnd, AudioUsesAbsSendTimeExtension) {
+  // Defined before PeerScenario so it gets destructed after, to avoid use after free.
+  std::atomic<bool> received_abs_send_time(false);
+  PeerScenario s(*test_info_);
+
+  auto* caller = s.CreateClient(PeerScenarioClient::Config());
+  auto* callee = s.CreateClient(PeerScenarioClient::Config());
+
+  auto send_node = s.net()->NodeBuilder().Build().node;
+  auto ret_node = s.net()->NodeBuilder().Build().node;
+
+  s.net()->CreateRoute(caller->endpoint(), {send_node}, callee->endpoint());
+  s.net()->CreateRoute(callee->endpoint(), {ret_node}, caller->endpoint());
+
+  auto signaling = s.ConnectSignaling(caller, callee, {send_node}, {ret_node});
+  caller->CreateAudio("AUDIO", cricket::AudioOptions());
+  signaling.StartIceSignaling();
+  RtpHeaderExtensionMap extension_map;
+  std::atomic<bool> offer_exchange_done(false);
+  signaling.NegotiateSdp(
+      [&extension_map](SessionDescriptionInterface* offer) {
+        extension_map = AudioExtensions(*offer);
+        EXPECT_TRUE(extension_map.IsRegistered(kRtpExtensionAbsoluteSendTime));
+      },
+      [&](const SessionDescriptionInterface& answer) {
+        EXPECT_TRUE(AudioExtensions(answer).IsRegistered(
+            kRtpExtensionAbsoluteSendTime));
+        offer_exchange_done = true;
+      });
+  RTC_CHECK(s.WaitAndProcess(&offer_exchange_done));
+  send_node->router()->SetWatcher(
+      [extension_map, &received_abs_send_time](const EmulatedIpPacket& packet) {
+        // The dummy packets used by the fake signaling are filled with 0. We
+        // want to ignore those and we can do that on the basis that the first
+        // byte of RTP packets are guaranteed to not be 0.
+        RtpPacket rtp_packet(&extension_map);
+        // TODO(bugs.webrtc.org/14525): Look why there are RTP packets with
+        // payload 72 or 73 (these don't have the RTP AbsoluteSendTime
+        // Extension).
+        if (rtp_packet.Parse(packet.data) && rtp_packet.PayloadType() == 111) {
+          EXPECT_TRUE(rtp_packet.HasExtension<AbsoluteSendTime>());
+          received_abs_send_time = true;
+        }
+      });
+  RTC_CHECK(s.WaitAndProcess(&received_abs_send_time));
+  caller->pc()->Close();
+  callee->pc()->Close();
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc b/third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc
new file mode 100644
index 0000000000..4f478b4b2a
--- /dev/null
+++ b/third_party/libwebrtc/test/peer_scenario/tests/unsignaled_stream_test.cc
@@ -0,0 +1,270 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "media/base/stream_params.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "pc/media_session.h"
+#include "pc/session_description.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/peer_scenario/peer_scenario.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+enum class MidTestConfiguration {
+  // Legacy endpoint setup where PT demuxing is used.
+  kMidNotNegotiated,
+  // MID is negotiated but missing from packets. PT demuxing is disabled, so
+  // SSRCs have to be added to the SDP for WebRTC to forward packets correctly.
+  // Happens when client is spec compliant but the SFU isn't. Popular legacy.
+  kMidNegotiatedButMissingFromPackets,
+  // Fully spec-compliant: MID is present so we can safely drop packets with
+  // unknown MIDs.
+  kMidNegotiatedAndPresentInPackets,
+};
+
+// Gives the parameterized test a readable suffix.
+std::string TestParametersMidTestConfigurationToString(
+    testing::TestParamInfo<MidTestConfiguration> info) {
+  switch (info.param) {
+    case MidTestConfiguration::kMidNotNegotiated:
+      return "MidNotNegotiated";
+    case MidTestConfiguration::kMidNegotiatedButMissingFromPackets:
+      return "MidNegotiatedButMissingFromPackets";
+    case MidTestConfiguration::kMidNegotiatedAndPresentInPackets:
+      return "MidNegotiatedAndPresentInPackets";
+  }
+}
+
+class FrameObserver : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  FrameObserver() : frame_observed_(false) {}
+  void OnFrame(const VideoFrame&) override { frame_observed_ = true; }
+
+  std::atomic<bool> frame_observed_;
+};
+
+uint32_t get_ssrc(SessionDescriptionInterface* offer, size_t track_index) {
+  EXPECT_LT(track_index, offer->description()->contents().size());
+  return offer->description()
+      ->contents()[track_index]
+      .media_description()
+      ->streams()[0]
+      .ssrcs[0];
+}
+
+void set_ssrc(SessionDescriptionInterface* offer, size_t index, uint32_t ssrc) {
+  EXPECT_LT(index, offer->description()->contents().size());
+  cricket::StreamParams& new_stream_params = offer->description()
+                                                 ->contents()[index]
+                                                 .media_description()
+                                                 ->mutable_streams()[0];
+  new_stream_params.ssrcs[0] = ssrc;
+  new_stream_params.ssrc_groups[0].ssrcs[0] = ssrc;
+}
+
+}  // namespace
+
+class UnsignaledStreamTest
+    : public ::testing::Test,
+      public ::testing::WithParamInterface<MidTestConfiguration> {};
+
+TEST_P(UnsignaledStreamTest, ReplacesUnsignaledStreamOnCompletedSignaling) {
+  // This test covers a scenario that might occur if a remote client starts
+  // sending media packets before negotiation has completed. Depending on setup,
+  // these packets either get dropped or trigger an unsignalled default stream
+  // to be created, and connects that to a default video sink.
+  // In some edge cases using Unified Plan and PT demuxing, the default stream
+  // is create in a different transceiver to where the media SSRC will actually
+  // be used. This test verifies that the default stream is removed properly,
+  // and that packets are demuxed and video frames reach the desired sink.
+  const MidTestConfiguration kMidTestConfiguration = GetParam();
+
+  // Defined before PeerScenario so it gets destructed after, to avoid use after
+  // free.
+  PeerScenario s(*::testing::UnitTest::GetInstance()->current_test_info());
+
+  PeerScenarioClient::Config config = PeerScenarioClient::Config();
+  // Disable encryption so that we can inject a fake early media packet without
+  // triggering srtp failures.
+  config.disable_encryption = true;
+  auto* caller = s.CreateClient(config);
+  auto* callee = s.CreateClient(config);
+
+  auto send_node = s.net()->NodeBuilder().Build().node;
+  auto ret_node = s.net()->NodeBuilder().Build().node;
+
+  s.net()->CreateRoute(caller->endpoint(), {send_node}, callee->endpoint());
+  s.net()->CreateRoute(callee->endpoint(), {ret_node}, caller->endpoint());
+
+  auto signaling = s.ConnectSignaling(caller, callee, {send_node}, {ret_node});
+  PeerScenarioClient::VideoSendTrackConfig video_conf;
+  video_conf.generator.squares_video->framerate = 15;
+
+  auto first_track = caller->CreateVideo("VIDEO", video_conf);
+  FrameObserver first_sink;
+  callee->AddVideoReceiveSink(first_track.track->id(), &first_sink);
+
+  signaling.StartIceSignaling();
+  std::atomic<bool> offer_exchange_done(false);
+  std::atomic<bool> got_unsignaled_packet(false);
+
+  // We will capture the media ssrc of the first added stream, and preemptively
+  // inject a new media packet using a different ssrc. What happens depends on
+  // the test configuration.
+  //
+  // MidTestConfiguration::kMidNotNegotiated:
+  // - MID is not negotiated which means PT-based demuxing is enabled. Because
+  //   the packets have no MID, the second ssrc packet gets forwarded to the
+  //   first m= section. This will create a "default stream" for the second ssrc
+  //   and connect it to the default video sink (not set in this test). The test
+  //   verifies we can recover from this when we later get packets for the first
+  //   ssrc.
+  //
+  // MidTestConfiguration::kMidNegotiatedButMissingFromPackets:
+  // - MID is negotiated wich means PT-based demuxing is disabled. Because we
+  //   modify the packets not to contain the MID anyway (simulating a legacy SFU
+  //   that does not negotiate properly) unknown SSRCs are dropped but do not
+  //   otherwise cause any issues.
+  //
+  // MidTestConfiguration::kMidNegotiatedAndPresentInPackets:
+  // - MID is negotiated which means PT-based demuxing is enabled. In this case
+  //   the packets have the MID so they either get forwarded or dropped
+  //   depending on if the MID is known. The spec-compliant way is also the most
+  //   straight-forward one.
+
+  uint32_t first_ssrc = 0;
+  uint32_t second_ssrc = 0;
+  absl::optional<int> mid_header_extension_id = absl::nullopt;
+
+  signaling.NegotiateSdp(
+      /* munge_sdp = */
+      [&](SessionDescriptionInterface* offer) {
+        // Obtain the MID header extension ID and if we want the
+        // MidTestConfiguration::kMidNotNegotiated setup then we remove the MID
+        // header extension through SDP munging (otherwise SDP is not modified).
+        for (cricket::ContentInfo& content_info :
+             offer->description()->contents()) {
+          std::vector<RtpExtension> header_extensions =
+              content_info.media_description()->rtp_header_extensions();
+          for (auto it = header_extensions.begin();
+               it != header_extensions.end(); ++it) {
+            if (it->uri == RtpExtension::kMidUri) {
+              // MID header extension found!
+              mid_header_extension_id = it->id;
+              if (kMidTestConfiguration ==
+                  MidTestConfiguration::kMidNotNegotiated) {
+                // Munge away the extension.
+                header_extensions.erase(it);
+              }
+              break;
+            }
+          }
+          content_info.media_description()->set_rtp_header_extensions(
+              std::move(header_extensions));
+        }
+        ASSERT_TRUE(mid_header_extension_id.has_value());
+      },
+      /* modify_sdp = */
+      [&](SessionDescriptionInterface* offer) {
+        first_ssrc = get_ssrc(offer, 0);
+        second_ssrc = first_ssrc + 1;
+
+        send_node->router()->SetWatcher([&](const EmulatedIpPacket& packet) {
+          if (IsRtpPacket(packet.data) &&
+              ByteReader<uint32_t>::ReadBigEndian(&(packet.cdata()[8])) ==
+                  first_ssrc &&
+              !got_unsignaled_packet) {
+            // Parse packet and modify the SSRC to simulate a second m=
+            // section that has not been negotiated yet.
+            std::vector<RtpExtension> extensions;
+            extensions.emplace_back(RtpExtension::kMidUri,
+                                    mid_header_extension_id.value());
+            RtpHeaderExtensionMap extensions_map(extensions);
+            RtpPacket parsed_packet;
+            parsed_packet.IdentifyExtensions(extensions_map);
+            ASSERT_TRUE(parsed_packet.Parse(packet.data));
+            parsed_packet.SetSsrc(second_ssrc);
+            // The MID extension is present if and only if it was negotiated.
+            // If present, we either want to remove it or modify it depending
+            // on setup.
+            switch (kMidTestConfiguration) {
+              case MidTestConfiguration::kMidNotNegotiated:
+                EXPECT_FALSE(parsed_packet.HasExtension<RtpMid>());
+                break;
+              case MidTestConfiguration::kMidNegotiatedButMissingFromPackets:
+                EXPECT_TRUE(parsed_packet.HasExtension<RtpMid>());
+                ASSERT_TRUE(parsed_packet.RemoveExtension(RtpMid::kId));
+                break;
+              case MidTestConfiguration::kMidNegotiatedAndPresentInPackets:
+                EXPECT_TRUE(parsed_packet.HasExtension<RtpMid>());
+                // The simulated second m= section would have a different MID.
+                // If we don't modify it here then `second_ssrc` would end up
+                // being mapped to the first m= section which would cause SSRC
+                // conflicts if we later add the same SSRC to a second m=
+                // section. Hidden assumption: first m= section does not use
+                // MID:1.
+                ASSERT_TRUE(parsed_packet.SetExtension<RtpMid>("1"));
+                break;
+            }
+            // Inject the modified packet.
+            rtc::CopyOnWriteBuffer updated_buffer = parsed_packet.Buffer();
+            EmulatedIpPacket updated_packet(
+                packet.from, packet.to, updated_buffer, packet.arrival_time);
+            send_node->OnPacketReceived(std::move(updated_packet));
+            got_unsignaled_packet = true;
+          }
+        });
+      },
+      [&](const SessionDescriptionInterface& answer) {
+        EXPECT_EQ(answer.description()->contents().size(), 1u);
+        offer_exchange_done = true;
+      });
+  EXPECT_TRUE(s.WaitAndProcess(&offer_exchange_done));
+  EXPECT_TRUE(s.WaitAndProcess(&got_unsignaled_packet));
+  EXPECT_TRUE(s.WaitAndProcess(&first_sink.frame_observed_));
+
+  auto second_track = caller->CreateVideo("VIDEO2", video_conf);
+  FrameObserver second_sink;
+  callee->AddVideoReceiveSink(second_track.track->id(), &second_sink);
+
+  // Create a second video stream, munge the sdp to force it to use our fake
+  // early media ssrc.
+  offer_exchange_done = false;
+  signaling.NegotiateSdp(
+      /* munge_sdp = */
+      [&](SessionDescriptionInterface* offer) {
+        set_ssrc(offer, 1, second_ssrc);
+      },
+      /* modify_sdp = */ {},
+      [&](const SessionDescriptionInterface& answer) {
+        EXPECT_EQ(answer.description()->contents().size(), 2u);
+        offer_exchange_done = true;
+      });
+  EXPECT_TRUE(s.WaitAndProcess(&offer_exchange_done));
+  EXPECT_TRUE(s.WaitAndProcess(&second_sink.frame_observed_));
+  caller->pc()->Close();
+  callee->pc()->Close();
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    All,
+    UnsignaledStreamTest,
+    ::testing::Values(MidTestConfiguration::kMidNotNegotiated,
+                      MidTestConfiguration::kMidNegotiatedButMissingFromPackets,
+                      MidTestConfiguration::kMidNegotiatedAndPresentInPackets),
+    TestParametersMidTestConfigurationToString);
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/platform_video_capturer.cc b/third_party/libwebrtc/test/platform_video_capturer.cc
new file mode 100644
index 0000000000..fb3392a052
--- /dev/null
+++ b/third_party/libwebrtc/test/platform_video_capturer.cc
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/platform_video_capturer.h"
+
+#include "absl/memory/memory.h"
+#if defined(WEBRTC_MAC)
+#include "test/mac_capturer.h"
+#else
+#include "test/vcm_capturer.h"
+#endif
+
+namespace webrtc {
+namespace test {
+
+std::unique_ptr<TestVideoCapturer> CreateVideoCapturer(
+    size_t width,
+    size_t height,
+    size_t target_fps,
+    size_t capture_device_index) {
+#if defined(WEBRTC_MAC)
+  return absl::WrapUnique<TestVideoCapturer>(test::MacCapturer::Create(
+      width, height, target_fps, capture_device_index));
+#else
+  return absl::WrapUnique<TestVideoCapturer>(test::VcmCapturer::Create(
+      width, height, target_fps, capture_device_index));
+#endif
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/platform_video_capturer.h b/third_party/libwebrtc/test/platform_video_capturer.h
new file mode 100644
index 0000000000..241ba87df8
--- /dev/null
+++ b/third_party/libwebrtc/test/platform_video_capturer.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_PLATFORM_VIDEO_CAPTURER_H_
+#define TEST_PLATFORM_VIDEO_CAPTURER_H_
+
+#include <memory>
+
+#include "test/test_video_capturer.h"
+
+namespace webrtc {
+namespace test {
+
+std::unique_ptr<TestVideoCapturer> CreateVideoCapturer(
+    size_t width,
+    size_t height,
+    size_t target_fps,
+    size_t capture_device_index);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_PLATFORM_VIDEO_CAPTURER_H_
diff --git a/third_party/libwebrtc/test/rtcp_packet_parser.cc b/third_party/libwebrtc/test/rtcp_packet_parser.cc
new file mode 100644
index 0000000000..e286ec5a36
--- /dev/null
+++ b/third_party/libwebrtc/test/rtcp_packet_parser.cc
@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/rtcp_packet_parser.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace test {
+
+RtcpPacketParser::RtcpPacketParser() = default;
+RtcpPacketParser::~RtcpPacketParser() = default;
+
+bool RtcpPacketParser::Parse(const void* data, size_t length) {
+  ++processed_rtcp_packets_;
+
+  const uint8_t* const buffer = static_cast<const uint8_t*>(data);
+  const uint8_t* const buffer_end = buffer + length;
+
+  rtcp::CommonHeader header;
+  for (const uint8_t* next_packet = buffer; next_packet != buffer_end;
+       next_packet = header.NextPacket()) {
+    RTC_DCHECK_GT(buffer_end - next_packet, 0);
+    if (!header.Parse(next_packet, buffer_end - next_packet)) {
+      RTC_LOG(LS_WARNING)
+          << "Invalid rtcp header or unaligned rtcp packet at position "
+          << (next_packet - buffer);
+      return false;
+    }
+    switch (header.type()) {
+      case rtcp::App::kPacketType:
+        app_.Parse(header);
+        break;
+      case rtcp::Bye::kPacketType:
+        bye_.Parse(header, &sender_ssrc_);
+        break;
+      case rtcp::ExtendedReports::kPacketType:
+        xr_.Parse(header, &sender_ssrc_);
+        break;
+      case rtcp::Psfb::kPacketType:
+        switch (header.fmt()) {
+          case rtcp::Fir::kFeedbackMessageType:
+            fir_.Parse(header, &sender_ssrc_);
+            break;
+          case rtcp::Pli::kFeedbackMessageType:
+            pli_.Parse(header, &sender_ssrc_);
+            break;
+          case rtcp::Psfb::kAfbMessageType:
+            if (!loss_notification_.Parse(header, &sender_ssrc_) &&
+                !remb_.Parse(header, &sender_ssrc_)) {
+              RTC_LOG(LS_WARNING) << "Unknown application layer FB message.";
+            }
+            break;
+          default:
+            RTC_LOG(LS_WARNING)
+                << "Unknown rtcp payload specific feedback type "
+                << header.fmt();
+            break;
+        }
+        break;
+      case rtcp::ReceiverReport::kPacketType:
+        receiver_report_.Parse(header, &sender_ssrc_);
+        break;
+      case rtcp::Rtpfb::kPacketType:
+        switch (header.fmt()) {
+          case rtcp::Nack::kFeedbackMessageType:
+            nack_.Parse(header, &sender_ssrc_);
+            break;
+          case rtcp::RapidResyncRequest::kFeedbackMessageType:
+            rrr_.Parse(header, &sender_ssrc_);
+            break;
+          case rtcp::Tmmbn::kFeedbackMessageType:
+            tmmbn_.Parse(header, &sender_ssrc_);
+            break;
+          case rtcp::Tmmbr::kFeedbackMessageType:
+            tmmbr_.Parse(header, &sender_ssrc_);
+            break;
+          case rtcp::TransportFeedback::kFeedbackMessageType:
+            transport_feedback_.Parse(header, &sender_ssrc_);
+            break;
+          default:
+            RTC_LOG(LS_WARNING)
+                << "Unknown rtcp transport feedback type " << header.fmt();
+            break;
+        }
+        break;
+      case rtcp::Sdes::kPacketType:
+        sdes_.Parse(header);
+        break;
+      case rtcp::SenderReport::kPacketType:
+        sender_report_.Parse(header, &sender_ssrc_);
+        break;
+      default:
+        RTC_LOG(LS_WARNING) << "Unknown rtcp packet type " << header.type();
+        break;
+    }
+  }
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/rtcp_packet_parser.h b/third_party/libwebrtc/test/rtcp_packet_parser.h
new file mode 100644
index 0000000000..9e8c9685e9
--- /dev/null
+++ b/third_party/libwebrtc/test/rtcp_packet_parser.h
@@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef TEST_RTCP_PACKET_PARSER_H_
+#define TEST_RTCP_PACKET_PARSER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/app.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/fir.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/loss_notification.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sdes.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbn.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/tmmbr.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+// Parse RTCP packet of given type. Assumes RTCP header is valid and that there
+// is excatly one packet of correct type in the buffer.
+template <typename Packet>
+bool ParseSinglePacket(const uint8_t* buffer, size_t size, Packet* packet) {
+  rtcp::CommonHeader header;
+  RTC_CHECK(header.Parse(buffer, size));
+  RTC_CHECK_EQ(size, header.NextPacket() - buffer);
+  return packet->Parse(header);
+}
+// Same function, but takes raw buffer as single argument instead of pair.
+template <typename Packet>
+bool ParseSinglePacket(rtc::ArrayView<const uint8_t> buffer, Packet* packet) {
+  return ParseSinglePacket(buffer.data(), buffer.size(), packet);
+}
+
+class RtcpPacketParser {
+ public:
+  // Keeps last parsed packet, count number of parsed packets of given type.
+  template <typename TypedRtcpPacket>
+  class PacketCounter : public TypedRtcpPacket {
+   public:
+    int num_packets() const { return num_packets_; }
+    void Parse(const rtcp::CommonHeader& header) {
+      if (TypedRtcpPacket::Parse(header))
+        ++num_packets_;
+    }
+    bool Parse(const rtcp::CommonHeader& header, uint32_t* sender_ssrc) {
+      const bool result = TypedRtcpPacket::Parse(header);
+      if (result) {
+        ++num_packets_;
+        if (*sender_ssrc == 0)  // Use first sender ssrc in compound packet.
+          *sender_ssrc = TypedRtcpPacket::sender_ssrc();
+      }
+      return result;
+    }
+
+   private:
+    int num_packets_ = 0;
+  };
+
+  RtcpPacketParser();
+  ~RtcpPacketParser();
+
+  bool Parse(const void* packet, size_t packet_len);
+
+  PacketCounter<rtcp::App>* app() { return &app_; }
+  PacketCounter<rtcp::Bye>* bye() { return &bye_; }
+  PacketCounter<rtcp::ExtendedReports>* xr() { return &xr_; }
+  PacketCounter<rtcp::Fir>* fir() { return &fir_; }
+  PacketCounter<rtcp::Nack>* nack() { return &nack_; }
+  PacketCounter<rtcp::Pli>* pli() { return &pli_; }
+  PacketCounter<rtcp::RapidResyncRequest>* rrr() { return &rrr_; }
+  PacketCounter<rtcp::ReceiverReport>* receiver_report() {
+    return &receiver_report_;
+  }
+  PacketCounter<rtcp::LossNotification>* loss_notification() {
+    return &loss_notification_;
+  }
+  PacketCounter<rtcp::Remb>* remb() { return &remb_; }
+  PacketCounter<rtcp::Sdes>* sdes() { return &sdes_; }
+  PacketCounter<rtcp::SenderReport>* sender_report() { return &sender_report_; }
+  PacketCounter<rtcp::Tmmbn>* tmmbn() { return &tmmbn_; }
+  PacketCounter<rtcp::Tmmbr>* tmmbr() { return &tmmbr_; }
+  PacketCounter<rtcp::TransportFeedback>* transport_feedback() {
+    return &transport_feedback_;
+  }
+  uint32_t sender_ssrc() const { return sender_ssrc_; }
+  size_t processed_rtcp_packets() const { return processed_rtcp_packets_; }
+
+ private:
+  PacketCounter<rtcp::App> app_;
+  PacketCounter<rtcp::Bye> bye_;
+  PacketCounter<rtcp::ExtendedReports> xr_;
+  PacketCounter<rtcp::Fir> fir_;
+  PacketCounter<rtcp::Nack> nack_;
+  PacketCounter<rtcp::Pli> pli_;
+  PacketCounter<rtcp::RapidResyncRequest> rrr_;
+  PacketCounter<rtcp::ReceiverReport> receiver_report_;
+  PacketCounter<rtcp::LossNotification> loss_notification_;
+  PacketCounter<rtcp::Remb> remb_;
+  PacketCounter<rtcp::Sdes> sdes_;
+  PacketCounter<rtcp::SenderReport> sender_report_;
+  PacketCounter<rtcp::Tmmbn> tmmbn_;
+  PacketCounter<rtcp::Tmmbr> tmmbr_;
+  PacketCounter<rtcp::TransportFeedback> transport_feedback_;
+  uint32_t sender_ssrc_ = 0;
+  size_t processed_rtcp_packets_ = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_RTCP_PACKET_PARSER_H_
diff --git a/third_party/libwebrtc/test/rtp_file_reader.cc b/third_party/libwebrtc/test/rtp_file_reader.cc
new file mode 100644
index 0000000000..b6f3cbbe5b
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_file_reader.cc
@@ -0,0 +1,691 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/rtp_file_reader.h"
+
+#include <stdio.h>
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/arch.h"
+
+namespace webrtc {
+namespace test {
+
+static const size_t kFirstLineLength = 80;
+static uint16_t kPacketHeaderSize = 8;
+
+#define TRY(expr)                           \
+  do {                                      \
+    if (!(expr)) {                          \
+      RTC_LOG(LS_INFO) << "Failed to read"; \
+      return false;                         \
+    }                                       \
+  } while (0)
+
+bool ReadUint32(uint32_t* out, FILE* file) {
+  *out = 0;
+  for (size_t i = 0; i < 4; ++i) {
+    *out <<= 8;
+    uint8_t tmp;
+    if (fread(&tmp, 1, sizeof(uint8_t), file) != sizeof(uint8_t))
+      return false;
+    *out |= tmp;
+  }
+  return true;
+}
+
+bool ReadUint16(uint16_t* out, FILE* file) {
+  *out = 0;
+  for (size_t i = 0; i < 2; ++i) {
+    *out <<= 8;
+    uint8_t tmp;
+    if (fread(&tmp, 1, sizeof(uint8_t), file) != sizeof(uint8_t))
+      return false;
+    *out |= tmp;
+  }
+  return true;
+}
+
+class RtpFileReaderImpl : public RtpFileReader {
+ public:
+  virtual bool Init(FILE* file, const std::set<uint32_t>& ssrc_filter) = 0;
+};
+
+class InterleavedRtpFileReader : public RtpFileReaderImpl {
+ public:
+  ~InterleavedRtpFileReader() override {
+    if (file_ != nullptr) {
+      fclose(file_);
+      file_ = nullptr;
+    }
+  }
+
+  bool Init(FILE* file, const std::set<uint32_t>& ssrc_filter) override {
+    file_ = file;
+    return true;
+  }
+
+  bool NextPacket(RtpPacket* packet) override {
+    RTC_DCHECK(file_);
+    packet->length = RtpPacket::kMaxPacketBufferSize;
+    uint32_t len = 0;
+    TRY(ReadUint32(&len, file_));
+    if (packet->length < len) {
+      RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
+                  << packet->length
+                  << " bytes allocated. Consider increasing the buffer "
+                  << "size";
+    }
+    if (fread(packet->data, 1, len, file_) != len)
+      return false;
+
+    packet->length = len;
+    packet->original_length = len;
+    packet->time_ms = time_ms_;
+    time_ms_ += 5;
+    return true;
+  }
+
+ private:
+  FILE* file_ = nullptr;
+  int64_t time_ms_ = 0;
+};
+
+// Read RTP packets from file in rtpdump format, as documented at:
+// http://www.cs.columbia.edu/irt/software/rtptools/
+class RtpDumpReader : public RtpFileReaderImpl {
+ public:
+  RtpDumpReader() : file_(nullptr) {}
+  ~RtpDumpReader() override {
+    if (file_ != nullptr) {
+      fclose(file_);
+      file_ = nullptr;
+    }
+  }
+
+  RtpDumpReader(const RtpDumpReader&) = delete;
+  RtpDumpReader& operator=(const RtpDumpReader&) = delete;
+
+  bool Init(FILE* file, const std::set<uint32_t>& ssrc_filter) override {
+    file_ = file;
+
+    char firstline[kFirstLineLength + 1] = {0};
+    if (fgets(firstline, kFirstLineLength, file_) == nullptr) {
+      RTC_LOG(LS_INFO) << "Can't read from file";
+      return false;
+    }
+    if (strncmp(firstline, "#!rtpplay", 9) == 0) {
+      if (strncmp(firstline, "#!rtpplay1.0", 12) != 0) {
+        RTC_LOG(LS_INFO) << "Wrong rtpplay version, must be 1.0";
+        return false;
+      }
+    } else if (strncmp(firstline, "#!RTPencode", 11) == 0) {
+      if (strncmp(firstline, "#!RTPencode1.0", 14) != 0) {
+        RTC_LOG(LS_INFO) << "Wrong RTPencode version, must be 1.0";
+        return false;
+      }
+    } else {
+      RTC_LOG(LS_INFO) << "Wrong file format of input file";
+      return false;
+    }
+
+    uint32_t start_sec;
+    uint32_t start_usec;
+    uint32_t source;
+    uint16_t port;
+    uint16_t padding;
+    TRY(ReadUint32(&start_sec, file_));
+    TRY(ReadUint32(&start_usec, file_));
+    TRY(ReadUint32(&source, file_));
+    TRY(ReadUint16(&port, file_));
+    TRY(ReadUint16(&padding, file_));
+
+    return true;
+  }
+
+  bool NextPacket(RtpPacket* packet) override {
+    uint8_t* rtp_data = packet->data;
+    packet->length = RtpPacket::kMaxPacketBufferSize;
+
+    uint16_t len;
+    uint16_t plen;
+    uint32_t offset;
+    TRY(ReadUint16(&len, file_));
+    TRY(ReadUint16(&plen, file_));
+    TRY(ReadUint32(&offset, file_));
+
+    // Use 'len' here because a 'plen' of 0 specifies rtcp.
+    len -= kPacketHeaderSize;
+    if (packet->length < len) {
+      RTC_LOG(LS_ERROR) << "Packet is too large to fit: " << len << " bytes vs "
+                        << packet->length
+                        << " bytes allocated. Consider increasing the buffer "
+                           "size";
+      return false;
+    }
+    if (fread(rtp_data, 1, len, file_) != len) {
+      return false;
+    }
+
+    packet->length = len;
+    packet->original_length = plen;
+    packet->time_ms = offset;
+    return true;
+  }
+
+ private:
+  FILE* file_;
+};
+
+enum {
+  kResultFail = -1,
+  kResultSuccess = 0,
+  kResultSkip = 1,
+
+  kPcapVersionMajor = 2,
+  kPcapVersionMinor = 4,
+  kLinktypeNull = 0,
+  kLinktypeEthernet = 1,
+  kBsdNullLoopback1 = 0x00000002,
+  kBsdNullLoopback2 = 0x02000000,
+  kEthernetIIHeaderMacSkip = 12,
+  kEthertypeIp = 0x0800,
+  kIpVersion4 = 4,
+  kMinIpHeaderLength = 20,
+  kFragmentOffsetClear = 0x0000,
+  kFragmentOffsetDoNotFragment = 0x4000,
+  kProtocolTcp = 0x06,
+  kProtocolUdp = 0x11,
+  kUdpHeaderLength = 8,
+  kMaxReadBufferSize = 4096
+};
+
+const uint32_t kPcapBOMSwapOrder = 0xd4c3b2a1UL;
+const uint32_t kPcapBOMNoSwapOrder = 0xa1b2c3d4UL;
+
+#define TRY_PCAP(expr)                                               \
+  do {                                                               \
+    int r = (expr);                                                  \
+    if (r == kResultFail) {                                          \
+      RTC_LOG(LS_INFO) << "FAIL at " << __FILE__ << ":" << __LINE__; \
+      return kResultFail;                                            \
+    } else if (r == kResultSkip) {                                   \
+      return kResultSkip;                                            \
+    }                                                                \
+  } while (0)
+
+// Read RTP packets from file in tcpdump/libpcap format, as documented at:
+// http://wiki.wireshark.org/Development/LibpcapFileFormat
+class PcapReader : public RtpFileReaderImpl {
+ public:
+  PcapReader()
+      : file_(nullptr),
+        swap_pcap_byte_order_(false),
+#ifdef WEBRTC_ARCH_BIG_ENDIAN
+        swap_network_byte_order_(false),
+#else
+        swap_network_byte_order_(true),
+#endif
+        read_buffer_(),
+        packets_by_ssrc_(),
+        packets_(),
+        next_packet_it_() {
+  }
+
+  ~PcapReader() override {
+    if (file_ != nullptr) {
+      fclose(file_);
+      file_ = nullptr;
+    }
+  }
+
+  PcapReader(const PcapReader&) = delete;
+  PcapReader& operator=(const PcapReader&) = delete;
+
+  bool Init(FILE* file, const std::set<uint32_t>& ssrc_filter) override {
+    return Initialize(file, ssrc_filter) == kResultSuccess;
+  }
+
+  int Initialize(FILE* file, const std::set<uint32_t>& ssrc_filter) {
+    file_ = file;
+
+    if (ReadGlobalHeader() < 0) {
+      return kResultFail;
+    }
+
+    int total_packet_count = 0;
+    uint32_t stream_start_ms = 0;
+    int32_t next_packet_pos = ftell(file_);
+    for (;;) {
+      TRY_PCAP(fseek(file_, next_packet_pos, SEEK_SET));
+      int result = ReadPacket(&next_packet_pos, stream_start_ms,
+                              ++total_packet_count, ssrc_filter);
+      if (result == kResultFail) {
+        break;
+      } else if (result == kResultSuccess && packets_.size() == 1) {
+        RTC_DCHECK_EQ(stream_start_ms, 0);
+        PacketIterator it = packets_.begin();
+        stream_start_ms = it->time_offset_ms;
+        it->time_offset_ms = 0;
+      }
+    }
+
+    if (feof(file_) == 0) {
+      printf("Failed reading file!\n");
+      return kResultFail;
+    }
+
+    printf("Total packets in file: %d\n", total_packet_count);
+    printf("Total RTP/RTCP packets: %zu\n", packets_.size());
+
+    for (SsrcMapIterator mit = packets_by_ssrc_.begin();
+         mit != packets_by_ssrc_.end(); ++mit) {
+      uint32_t ssrc = mit->first;
+      const std::vector<uint32_t>& packet_indices = mit->second;
+      int pt = packets_[packet_indices[0]].payload_type;
+      printf("SSRC: %08x, %zu packets, pt=%d\n", ssrc, packet_indices.size(),
+             pt);
+    }
+
+    // TODO(solenberg): Better validation of identified SSRC streams.
+    //
+    // Since we're dealing with raw network data here, we will wrongly identify
+    // some packets as RTP. When these packets are consumed by RtpPlayer, they
+    // are unlikely to cause issues as they will ultimately be filtered out by
+    // the RtpRtcp module. However, we should really do better filtering here,
+    // which we can accomplish in a number of ways, e.g.:
+    //
+    // - Verify that the time stamps and sequence numbers for RTP packets are
+    //   both increasing/decreasing. If they move in different directions, the
+    //   SSRC is likely bogus and can be dropped. (Normally they should be inc-
+    //   reasing but we must allow packet reordering).
+    // - If RTP sequence number is not changing, drop the stream.
+    // - Can also use srcip:port->dstip:port pairs, assuming few SSRC collisions
+    //   for up/down streams.
+
+    next_packet_it_ = packets_.begin();
+    return kResultSuccess;
+  }
+
+  bool NextPacket(RtpPacket* packet) override {
+    uint32_t length = RtpPacket::kMaxPacketBufferSize;
+    if (NextPcap(packet->data, &length, &packet->time_ms) != kResultSuccess)
+      return false;
+    packet->length = static_cast<size_t>(length);
+    packet->original_length = packet->length;
+    return true;
+  }
+
+  virtual int NextPcap(uint8_t* data, uint32_t* length, uint32_t* time_ms) {
+    RTC_DCHECK(data);
+    RTC_DCHECK(length);
+    RTC_DCHECK(time_ms);
+
+    if (next_packet_it_ == packets_.end()) {
+      return -1;
+    }
+    if (*length < next_packet_it_->payload_length) {
+      return -1;
+    }
+    TRY_PCAP(fseek(file_, next_packet_it_->pos_in_file, SEEK_SET));
+    TRY_PCAP(Read(data, next_packet_it_->payload_length));
+    *length = next_packet_it_->payload_length;
+    *time_ms = next_packet_it_->time_offset_ms;
+    next_packet_it_++;
+
+    return 0;
+  }
+
+ private:
+  // A marker of an RTP packet within the file.
+  struct RtpPacketMarker {
+    uint32_t packet_number;  // One-based index (like in WireShark)
+    uint32_t time_offset_ms;
+    uint32_t source_ip;
+    uint32_t dest_ip;
+    uint16_t source_port;
+    uint16_t dest_port;
+    // Payload type of the RTP packet,
+    // or RTCP packet type of the first RTCP packet in a compound RTCP packet.
+    int payload_type;
+    int32_t pos_in_file;  // Byte offset of payload from start of file.
+    uint32_t payload_length;
+  };
+
+  typedef std::vector<RtpPacketMarker>::iterator PacketIterator;
+  typedef std::map<uint32_t, std::vector<uint32_t> > SsrcMap;
+  typedef std::map<uint32_t, std::vector<uint32_t> >::iterator SsrcMapIterator;
+
+  int ReadGlobalHeader() {
+    uint32_t magic;
+    TRY_PCAP(Read(&magic, false));
+    if (magic == kPcapBOMSwapOrder) {
+      swap_pcap_byte_order_ = true;
+    } else if (magic == kPcapBOMNoSwapOrder) {
+      swap_pcap_byte_order_ = false;
+    } else {
+      return kResultFail;
+    }
+
+    uint16_t version_major;
+    uint16_t version_minor;
+    TRY_PCAP(Read(&version_major, false));
+    TRY_PCAP(Read(&version_minor, false));
+    if (version_major != kPcapVersionMajor ||
+        version_minor != kPcapVersionMinor) {
+      return kResultFail;
+    }
+
+    int32_t this_zone;  // GMT to local correction.
+    uint32_t sigfigs;   // Accuracy of timestamps.
+    uint32_t snaplen;   // Max length of captured packets, in octets.
+    uint32_t network;   // Data link type.
+    TRY_PCAP(Read(&this_zone, false));
+    TRY_PCAP(Read(&sigfigs, false));
+    TRY_PCAP(Read(&snaplen, false));
+    TRY_PCAP(Read(&network, false));
+
+    // Accept only LINKTYPE_NULL and LINKTYPE_ETHERNET.
+    // See: http://www.tcpdump.org/linktypes.html
+    if (network != kLinktypeNull && network != kLinktypeEthernet) {
+      return kResultFail;
+    }
+
+    return kResultSuccess;
+  }
+
+  int ReadPacket(int32_t* next_packet_pos,
+                 uint32_t stream_start_ms,
+                 uint32_t number,
+                 const std::set<uint32_t>& ssrc_filter) {
+    RTC_DCHECK(next_packet_pos);
+
+    uint32_t ts_sec;    // Timestamp seconds.
+    uint32_t ts_usec;   // Timestamp microseconds.
+    uint32_t incl_len;  // Number of octets of packet saved in file.
+    uint32_t orig_len;  // Actual length of packet.
+    TRY_PCAP(Read(&ts_sec, false));
+    TRY_PCAP(Read(&ts_usec, false));
+    TRY_PCAP(Read(&incl_len, false));
+    TRY_PCAP(Read(&orig_len, false));
+
+    *next_packet_pos = ftell(file_) + incl_len;
+
+    RtpPacketMarker marker = {0};
+    marker.packet_number = number;
+    marker.time_offset_ms = CalcTimeDelta(ts_sec, ts_usec, stream_start_ms);
+    TRY_PCAP(ReadPacketHeader(&marker));
+    marker.pos_in_file = ftell(file_);
+
+    if (marker.payload_length > sizeof(read_buffer_)) {
+      printf("Packet too large!\n");
+      return kResultFail;
+    }
+    TRY_PCAP(Read(read_buffer_, marker.payload_length));
+
+    rtc::ArrayView<const uint8_t> packet(read_buffer_, marker.payload_length);
+    if (IsRtcpPacket(packet)) {
+      marker.payload_type = packet[1];
+      packets_.push_back(marker);
+    } else if (IsRtpPacket(packet)) {
+      uint32_t ssrc = ParseRtpSsrc(packet);
+      marker.payload_type = ParseRtpPayloadType(packet);
+      if (ssrc_filter.empty() || ssrc_filter.find(ssrc) != ssrc_filter.end()) {
+        packets_by_ssrc_[ssrc].push_back(
+            static_cast<uint32_t>(packets_.size()));
+        packets_.push_back(marker);
+      } else {
+        return kResultSkip;
+      }
+    } else {
+      RTC_LOG(LS_INFO) << "Not recognized as RTP/RTCP";
+      return kResultSkip;
+    }
+
+    return kResultSuccess;
+  }
+
+  int ReadPacketHeader(RtpPacketMarker* marker) {
+    int32_t file_pos = ftell(file_);
+
+    // Check for BSD null/loopback frame header. The header is just 4 bytes in
+    // native byte order, so we check for both versions as we don't care about
+    // the header as such and will likely fail reading the IP header if this is
+    // something else than null/loopback.
+    uint32_t protocol;
+    TRY_PCAP(Read(&protocol, true));
+    if (protocol == kBsdNullLoopback1 || protocol == kBsdNullLoopback2) {
+      int result = ReadXxpIpHeader(marker);
+      RTC_LOG(LS_INFO) << "Recognized loopback frame";
+      if (result != kResultSkip) {
+        return result;
+      }
+    }
+
+    TRY_PCAP(fseek(file_, file_pos, SEEK_SET));
+
+    // Check for Ethernet II, IP frame header.
+    uint16_t type;
+    TRY_PCAP(Skip(kEthernetIIHeaderMacSkip));  // Source+destination MAC.
+    TRY_PCAP(Read(&type, true));
+    if (type == kEthertypeIp) {
+      int result = ReadXxpIpHeader(marker);
+      RTC_LOG(LS_INFO) << "Recognized ethernet 2 frame";
+      if (result != kResultSkip) {
+        return result;
+      }
+    }
+
+    return kResultSkip;
+  }
+
+  uint32_t CalcTimeDelta(uint32_t ts_sec, uint32_t ts_usec, uint32_t start_ms) {
+    // Round to nearest ms.
+    uint64_t t2_ms =
+        ((static_cast<uint64_t>(ts_sec) * 1000000) + ts_usec + 500) / 1000;
+    uint64_t t1_ms = static_cast<uint64_t>(start_ms);
+    if (t2_ms < t1_ms) {
+      return 0;
+    } else {
+      return t2_ms - t1_ms;
+    }
+  }
+
+  int ReadXxpIpHeader(RtpPacketMarker* marker) {
+    RTC_DCHECK(marker);
+
+    uint16_t version;
+    uint16_t length;
+    uint16_t id;
+    uint16_t fragment;
+    uint16_t protocol;
+    uint16_t checksum;
+    TRY_PCAP(Read(&version, true));
+    TRY_PCAP(Read(&length, true));
+    TRY_PCAP(Read(&id, true));
+    TRY_PCAP(Read(&fragment, true));
+    TRY_PCAP(Read(&protocol, true));
+    TRY_PCAP(Read(&checksum, true));
+    TRY_PCAP(Read(&marker->source_ip, true));
+    TRY_PCAP(Read(&marker->dest_ip, true));
+
+    if (((version >> 12) & 0x000f) != kIpVersion4) {
+      RTC_LOG(LS_INFO) << "IP header is not IPv4";
+      return kResultSkip;
+    }
+
+    if (fragment != kFragmentOffsetClear &&
+        fragment != kFragmentOffsetDoNotFragment) {
+      RTC_LOG(LS_INFO) << "IP fragments cannot be handled";
+      return kResultSkip;
+    }
+
+    // Skip remaining fields of IP header.
+    uint16_t header_length = (version & 0x0f00) >> (8 - 2);
+    RTC_DCHECK_GE(header_length, kMinIpHeaderLength);
+    TRY_PCAP(Skip(header_length - kMinIpHeaderLength));
+
+    protocol = protocol & 0x00ff;
+    if (protocol == kProtocolTcp) {
+      RTC_LOG(LS_INFO) << "TCP packets are not handled";
+      return kResultSkip;
+    } else if (protocol == kProtocolUdp) {
+      uint16_t length;
+      uint16_t checksum;
+      TRY_PCAP(Read(&marker->source_port, true));
+      TRY_PCAP(Read(&marker->dest_port, true));
+      TRY_PCAP(Read(&length, true));
+      TRY_PCAP(Read(&checksum, true));
+      marker->payload_length = length - kUdpHeaderLength;
+    } else {
+      RTC_LOG(LS_INFO) << "Unknown transport (expected UDP or TCP)";
+      return kResultSkip;
+    }
+
+    return kResultSuccess;
+  }
+
+  int Read(uint32_t* out, bool expect_network_order) {
+    uint32_t tmp = 0;
+    if (fread(&tmp, 1, sizeof(uint32_t), file_) != sizeof(uint32_t)) {
+      return kResultFail;
+    }
+    if ((!expect_network_order && swap_pcap_byte_order_) ||
+        (expect_network_order && swap_network_byte_order_)) {
+      tmp = ((tmp >> 24) & 0x000000ff) | (tmp << 24) |
+            ((tmp >> 8) & 0x0000ff00) | ((tmp << 8) & 0x00ff0000);
+    }
+    *out = tmp;
+    return kResultSuccess;
+  }
+
+  int Read(uint16_t* out, bool expect_network_order) {
+    uint16_t tmp = 0;
+    if (fread(&tmp, 1, sizeof(uint16_t), file_) != sizeof(uint16_t)) {
+      return kResultFail;
+    }
+    if ((!expect_network_order && swap_pcap_byte_order_) ||
+        (expect_network_order && swap_network_byte_order_)) {
+      tmp = ((tmp >> 8) & 0x00ff) | (tmp << 8);
+    }
+    *out = tmp;
+    return kResultSuccess;
+  }
+
+  int Read(uint8_t* out, uint32_t count) {
+    if (fread(out, 1, count, file_) != count) {
+      return kResultFail;
+    }
+    return kResultSuccess;
+  }
+
+  int Read(int32_t* out, bool expect_network_order) {
+    int32_t tmp = 0;
+    if (fread(&tmp, 1, sizeof(uint32_t), file_) != sizeof(uint32_t)) {
+      return kResultFail;
+    }
+    if ((!expect_network_order && swap_pcap_byte_order_) ||
+        (expect_network_order && swap_network_byte_order_)) {
+      tmp = ((tmp >> 24) & 0x000000ff) | (tmp << 24) |
+            ((tmp >> 8) & 0x0000ff00) | ((tmp << 8) & 0x00ff0000);
+    }
+    *out = tmp;
+    return kResultSuccess;
+  }
+
+  int Skip(uint32_t length) {
+    if (fseek(file_, length, SEEK_CUR) != 0) {
+      return kResultFail;
+    }
+    return kResultSuccess;
+  }
+
+  FILE* file_;
+  bool swap_pcap_byte_order_;
+  const bool swap_network_byte_order_;
+  uint8_t read_buffer_[kMaxReadBufferSize];
+
+  SsrcMap packets_by_ssrc_;
+  std::vector<RtpPacketMarker> packets_;
+  PacketIterator next_packet_it_;
+};
+
+RtpFileReaderImpl* CreateReaderForFormat(RtpFileReader::FileFormat format) {
+  RtpFileReaderImpl* reader = nullptr;
+  switch (format) {
+    case RtpFileReader::kPcap:
+      reader = new PcapReader();
+      break;
+    case RtpFileReader::kRtpDump:
+      reader = new RtpDumpReader();
+      break;
+    case RtpFileReader::kLengthPacketInterleaved:
+      reader = new InterleavedRtpFileReader();
+      break;
+  }
+  return reader;
+}
+
+RtpFileReader* RtpFileReader::Create(FileFormat format,
+                                     const uint8_t* data,
+                                     size_t size,
+                                     const std::set<uint32_t>& ssrc_filter) {
+  std::unique_ptr<RtpFileReaderImpl> reader(CreateReaderForFormat(format));
+
+  FILE* file = tmpfile();
+  if (file == nullptr) {
+    printf("ERROR: Can't open file from memory buffer\n");
+    return nullptr;
+  }
+
+  if (fwrite(reinterpret_cast<const void*>(data), sizeof(uint8_t), size,
+             file) != size) {
+    return nullptr;
+  }
+  rewind(file);
+
+  if (!reader->Init(file, ssrc_filter)) {
+    return nullptr;
+  }
+  return reader.release();
+}
+
+RtpFileReader* RtpFileReader::Create(FileFormat format,
+                                     absl::string_view filename,
+                                     const std::set<uint32_t>& ssrc_filter) {
+  RtpFileReaderImpl* reader = CreateReaderForFormat(format);
+  std::string filename_str = std::string(filename);
+  FILE* file = fopen(filename_str.c_str(), "rb");
+  if (file == nullptr) {
+    printf("ERROR: Can't open file: %s\n", filename_str.c_str());
+    return nullptr;
+  }
+
+  if (!reader->Init(file, ssrc_filter)) {
+    delete reader;
+    return nullptr;
+  }
+  return reader;
+}
+
+RtpFileReader* RtpFileReader::Create(FileFormat format,
+                                     absl::string_view filename) {
+  return RtpFileReader::Create(format, filename, std::set<uint32_t>());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/rtp_file_reader.h b/third_party/libwebrtc/test/rtp_file_reader.h
new file mode 100644
index 0000000000..4e26c71baa
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_file_reader.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_RTP_FILE_READER_H_
+#define TEST_RTP_FILE_READER_H_
+
+#include <set>
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+struct RtpPacket {
+  // Accommodate for 50 ms packets of 32 kHz PCM16 samples (3200 bytes) plus
+  // some overhead.
+  static const size_t kMaxPacketBufferSize = 3500;
+  uint8_t data[kMaxPacketBufferSize];
+  size_t length;
+  // The length the packet had on wire. Will be different from `length` when
+  // reading a header-only RTP dump.
+  size_t original_length;
+
+  uint32_t time_ms;
+};
+
+class RtpFileReader {
+ public:
+  enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved };
+
+  virtual ~RtpFileReader() {}
+  static RtpFileReader* Create(FileFormat format,
+                               const uint8_t* data,
+                               size_t size,
+                               const std::set<uint32_t>& ssrc_filter);
+  static RtpFileReader* Create(FileFormat format, absl::string_view filename);
+  static RtpFileReader* Create(FileFormat format,
+                               absl::string_view filename,
+                               const std::set<uint32_t>& ssrc_filter);
+  virtual bool NextPacket(RtpPacket* packet) = 0;
+};
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_RTP_FILE_READER_H_
diff --git a/third_party/libwebrtc/test/rtp_file_reader_unittest.cc b/third_party/libwebrtc/test/rtp_file_reader_unittest.cc
new file mode 100644
index 0000000000..995d9fbc9d
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_file_reader_unittest.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/rtp_file_reader.h"
+
+#include <map>
+#include <memory>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+class TestRtpFileReader : public ::testing::Test {
+ public:
+  void Init(const std::string& filename, bool headers_only_file) {
+    std::string filepath =
+        test::ResourcePath("video_coding/" + filename, "rtp");
+    rtp_packet_source_.reset(
+        test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filepath));
+    ASSERT_TRUE(rtp_packet_source_.get() != NULL);
+    headers_only_file_ = headers_only_file;
+  }
+
+  int CountRtpPackets() {
+    test::RtpPacket packet;
+    int c = 0;
+    while (rtp_packet_source_->NextPacket(&packet)) {
+      if (headers_only_file_)
+        EXPECT_LT(packet.length, packet.original_length);
+      else
+        EXPECT_EQ(packet.length, packet.original_length);
+      c++;
+    }
+    return c;
+  }
+
+ private:
+  std::unique_ptr<test::RtpFileReader> rtp_packet_source_;
+  bool headers_only_file_;
+};
+
+TEST_F(TestRtpFileReader, Test60Packets) {
+  Init("pltype103", false);
+  EXPECT_EQ(60, CountRtpPackets());
+}
+
+TEST_F(TestRtpFileReader, Test60PacketsHeaderOnly) {
+  Init("pltype103_header_only", true);
+  EXPECT_EQ(60, CountRtpPackets());
+}
+
+typedef std::map<uint32_t, int> PacketsPerSsrc;
+
+class TestPcapFileReader : public ::testing::Test {
+ public:
+  void Init(const std::string& filename) {
+    std::string filepath =
+        test::ResourcePath("video_coding/" + filename, "pcap");
+    rtp_packet_source_.reset(
+        test::RtpFileReader::Create(test::RtpFileReader::kPcap, filepath));
+    ASSERT_TRUE(rtp_packet_source_.get() != NULL);
+  }
+
+  int CountRtpPackets() {
+    int c = 0;
+    test::RtpPacket packet;
+    while (rtp_packet_source_->NextPacket(&packet)) {
+      EXPECT_EQ(packet.length, packet.original_length);
+      c++;
+    }
+    return c;
+  }
+
+  PacketsPerSsrc CountRtpPacketsPerSsrc() {
+    PacketsPerSsrc pps;
+    test::RtpPacket packet;
+    while (rtp_packet_source_->NextPacket(&packet)) {
+      rtc::ArrayView<const uint8_t> raw(packet.data, packet.length);
+      if (IsRtpPacket(raw)) {
+        pps[ParseRtpSsrc(raw)]++;
+      }
+    }
+    return pps;
+  }
+
+ private:
+  std::unique_ptr<test::RtpFileReader> rtp_packet_source_;
+};
+
+TEST_F(TestPcapFileReader, TestEthernetIIFrame) {
+  Init("frame-ethernet-ii");
+  EXPECT_EQ(368, CountRtpPackets());
+}
+
+TEST_F(TestPcapFileReader, TestLoopbackFrame) {
+  Init("frame-loopback");
+  EXPECT_EQ(491, CountRtpPackets());
+}
+
+TEST_F(TestPcapFileReader, TestTwoSsrc) {
+  Init("ssrcs-2");
+  PacketsPerSsrc pps = CountRtpPacketsPerSsrc();
+  EXPECT_EQ(2UL, pps.size());
+  EXPECT_EQ(370, pps[0x78d48f61]);
+  EXPECT_EQ(60, pps[0xae94130b]);
+}
+
+TEST_F(TestPcapFileReader, TestThreeSsrc) {
+  Init("ssrcs-3");
+  PacketsPerSsrc pps = CountRtpPacketsPerSsrc();
+  EXPECT_EQ(3UL, pps.size());
+  EXPECT_EQ(162, pps[0x938c5eaa]);
+  EXPECT_EQ(113, pps[0x59fe6ef0]);
+  EXPECT_EQ(61, pps[0xed2bd2ac]);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/rtp_file_writer.cc b/third_party/libwebrtc/test/rtp_file_writer.cc
new file mode 100644
index 0000000000..22f664abc8
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_file_writer.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/rtp_file_writer.h"
+
+#include <stdint.h>
+#include <stdio.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+static const uint16_t kPacketHeaderSize = 8;
+static const char kFirstLine[] = "#!rtpplay1.0 0.0.0.0/0\n";
+
+// Write RTP packets to file in rtpdump format, as documented at:
+// http://www.cs.columbia.edu/irt/software/rtptools/
+class RtpDumpWriter : public RtpFileWriter {
+ public:
+  explicit RtpDumpWriter(FILE* file) : file_(file) {
+    RTC_CHECK(file_ != NULL);
+    Init();
+  }
+  ~RtpDumpWriter() override {
+    if (file_ != NULL) {
+      fclose(file_);
+      file_ = NULL;
+    }
+  }
+
+  RtpDumpWriter(const RtpDumpWriter&) = delete;
+  RtpDumpWriter& operator=(const RtpDumpWriter&) = delete;
+
+  bool WritePacket(const RtpPacket* packet) override {
+    if (!first_packet_time_) {
+      first_packet_time_ = packet->time_ms;
+    }
+    uint16_t len = static_cast<uint16_t>(packet->length + kPacketHeaderSize);
+    uint16_t plen = static_cast<uint16_t>(packet->original_length);
+    uint32_t offset = packet->time_ms - *first_packet_time_;
+    RTC_CHECK(WriteUint16(len));
+    RTC_CHECK(WriteUint16(plen));
+    RTC_CHECK(WriteUint32(offset));
+    return fwrite(packet->data, sizeof(uint8_t), packet->length, file_) ==
+           packet->length;
+  }
+
+ private:
+  bool Init() {
+    fprintf(file_, "%s", kFirstLine);
+
+    RTC_CHECK(WriteUint32(0));
+    RTC_CHECK(WriteUint32(0));
+    RTC_CHECK(WriteUint32(0));
+    RTC_CHECK(WriteUint16(0));
+    RTC_CHECK(WriteUint16(0));
+
+    return true;
+  }
+
+  bool WriteUint32(uint32_t in) {
+    // Loop through shifts = {24, 16, 8, 0}.
+    for (int shifts = 24; shifts >= 0; shifts -= 8) {
+      uint8_t tmp = static_cast<uint8_t>((in >> shifts) & 0xFF);
+      if (fwrite(&tmp, sizeof(uint8_t), 1, file_) != 1)
+        return false;
+    }
+    return true;
+  }
+
+  bool WriteUint16(uint16_t in) {
+    // Write 8 MSBs.
+    uint8_t tmp = static_cast<uint8_t>((in >> 8) & 0xFF);
+    if (fwrite(&tmp, sizeof(uint8_t), 1, file_) != 1)
+      return false;
+    // Write 8 LSBs.
+    tmp = static_cast<uint8_t>(in & 0xFF);
+    if (fwrite(&tmp, sizeof(uint8_t), 1, file_) != 1)
+      return false;
+    return true;
+  }
+
+  FILE* file_;
+  absl::optional<uint32_t> first_packet_time_;
+};
+
+RtpFileWriter* RtpFileWriter::Create(FileFormat format,
+                                     const std::string& filename) {
+  FILE* file = fopen(filename.c_str(), "wb");
+  if (file == NULL) {
+    printf("ERROR: Can't open file: %s\n", filename.c_str());
+    return NULL;
+  }
+  switch (format) {
+    case kRtpDump:
+      return new RtpDumpWriter(file);
+  }
+  fclose(file);
+  return NULL;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/rtp_file_writer.h b/third_party/libwebrtc/test/rtp_file_writer.h
new file mode 100644
index 0000000000..5e560d7375
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_file_writer.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_RTP_FILE_WRITER_H_
+#define TEST_RTP_FILE_WRITER_H_
+
+#include <string>
+
+#include "test/rtp_file_reader.h"
+
+namespace webrtc {
+namespace test {
+class RtpFileWriter {
+ public:
+  enum FileFormat {
+    kRtpDump,
+  };
+
+  virtual ~RtpFileWriter() {}
+  static RtpFileWriter* Create(FileFormat format, const std::string& filename);
+
+  virtual bool WritePacket(const RtpPacket* packet) = 0;
+};
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_RTP_FILE_WRITER_H_
diff --git a/third_party/libwebrtc/test/rtp_file_writer_unittest.cc b/third_party/libwebrtc/test/rtp_file_writer_unittest.cc
new file mode 100644
index 0000000000..2396d7c346
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_file_writer_unittest.cc
@@ -0,0 +1,85 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/rtp_file_writer.h"
+
+#include <stdint.h>
+#include <string.h>
+
+#include <memory>
+
+#include "test/gtest.h"
+#include "test/rtp_file_reader.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+
+class RtpFileWriterTest : public ::testing::Test {
+ public:
+  void Init(const std::string& filename) {
+    filename_ = test::OutputPath() + filename;
+    rtp_writer_.reset(
+        test::RtpFileWriter::Create(test::RtpFileWriter::kRtpDump, filename_));
+  }
+
+  void WriteRtpPackets(int num_packets, int time_ms_offset = 0) {
+    ASSERT_TRUE(rtp_writer_.get() != NULL);
+    test::RtpPacket packet;
+    for (int i = 1; i <= num_packets; ++i) {
+      packet.length = i;
+      packet.original_length = i;
+      packet.time_ms = i + time_ms_offset;
+      memset(packet.data, i, packet.length);
+      EXPECT_TRUE(rtp_writer_->WritePacket(&packet));
+    }
+  }
+
+  void CloseOutputFile() { rtp_writer_.reset(); }
+
+  void VerifyFileContents(int expected_packets) {
+    ASSERT_TRUE(rtp_writer_.get() == NULL)
+        << "Must call CloseOutputFile before VerifyFileContents";
+    std::unique_ptr<test::RtpFileReader> rtp_reader(
+        test::RtpFileReader::Create(test::RtpFileReader::kRtpDump, filename_));
+    ASSERT_TRUE(rtp_reader.get() != NULL);
+    test::RtpPacket packet;
+    int i = 0;
+    while (rtp_reader->NextPacket(&packet)) {
+      ++i;
+      EXPECT_EQ(static_cast<size_t>(i), packet.length);
+      EXPECT_EQ(static_cast<size_t>(i), packet.original_length);
+      EXPECT_EQ(static_cast<uint32_t>(i - 1), packet.time_ms);
+      for (int j = 0; j < i; ++j) {
+        EXPECT_EQ(i, packet.data[j]);
+      }
+    }
+    EXPECT_EQ(expected_packets, i);
+  }
+
+ private:
+  std::unique_ptr<test::RtpFileWriter> rtp_writer_;
+  std::string filename_;
+};
+
+TEST_F(RtpFileWriterTest, WriteToRtpDump) {
+  Init("test_rtp_file_writer.rtp");
+  WriteRtpPackets(10);
+  CloseOutputFile();
+  VerifyFileContents(10);
+}
+
+TEST_F(RtpFileWriterTest, WriteToRtpDumpWithOffset) {
+  Init("test_rtp_file_writer.rtp");
+  WriteRtpPackets(10, 100);
+  CloseOutputFile();
+  VerifyFileContents(10);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/rtp_rtcp_observer.h b/third_party/libwebrtc/test/rtp_rtcp_observer.h
new file mode 100644
index 0000000000..06a438e712
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_rtcp_observer.h
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_RTP_RTCP_OBSERVER_H_
+#define TEST_RTP_RTCP_OBSERVER_H_
+
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/test/simulated_network.h"
+#include "api/units/time_delta.h"
+#include "call/simulated_packet_receiver.h"
+#include "call/video_send_stream.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+#include "rtc_base/event.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/direct_transport.h"
+#include "test/gtest.h"
+
+namespace {
+constexpr webrtc::TimeDelta kShortTimeout = webrtc::TimeDelta::Millis(500);
+}
+
+namespace webrtc {
+namespace test {
+
+class PacketTransport;
+
+class RtpRtcpObserver {
+ public:
+  enum Action {
+    SEND_PACKET,
+    DROP_PACKET,
+  };
+
+  virtual ~RtpRtcpObserver() {}
+
+  virtual bool Wait() {
+    if (field_trial::IsEnabled("WebRTC-QuickPerfTest")) {
+      observation_complete_.Wait(kShortTimeout);
+      return true;
+    }
+    return observation_complete_.Wait(timeout_);
+  }
+
+  virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
+    return SEND_PACKET;
+  }
+
+  virtual Action OnSendRtcp(const uint8_t* packet, size_t length) {
+    return SEND_PACKET;
+  }
+
+  virtual Action OnReceiveRtp(const uint8_t* packet, size_t length) {
+    return SEND_PACKET;
+  }
+
+  virtual Action OnReceiveRtcp(const uint8_t* packet, size_t length) {
+    return SEND_PACKET;
+  }
+
+ protected:
+  RtpRtcpObserver() : RtpRtcpObserver(TimeDelta::Zero()) {}
+  explicit RtpRtcpObserver(TimeDelta event_timeout) : timeout_(event_timeout) {}
+
+  rtc::Event observation_complete_;
+
+ private:
+  const TimeDelta timeout_;
+};
+
+class PacketTransport : public test::DirectTransport {
+ public:
+  enum TransportType { kReceiver, kSender };
+
+  PacketTransport(TaskQueueBase* task_queue,
+                  Call* send_call,
+                  RtpRtcpObserver* observer,
+                  TransportType transport_type,
+                  const std::map<uint8_t, MediaType>& payload_type_map,
+                  std::unique_ptr<SimulatedPacketReceiverInterface> nw_pipe,
+                  rtc::ArrayView<const RtpExtension> audio_extensions,
+                  rtc::ArrayView<const RtpExtension> video_extensions)
+      : test::DirectTransport(task_queue,
+                              std::move(nw_pipe),
+                              send_call,
+                              payload_type_map,
+                              audio_extensions,
+                              video_extensions),
+        observer_(observer),
+        transport_type_(transport_type) {}
+
+ private:
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override {
+    EXPECT_TRUE(IsRtpPacket(rtc::MakeArrayView(packet, length)));
+    RtpRtcpObserver::Action action = RtpRtcpObserver::SEND_PACKET;
+    if (observer_) {
+      if (transport_type_ == kSender) {
+        action = observer_->OnSendRtp(packet, length);
+      } else {
+        action = observer_->OnReceiveRtp(packet, length);
+      }
+    }
+    switch (action) {
+      case RtpRtcpObserver::DROP_PACKET:
+        // Drop packet silently.
+        return true;
+      case RtpRtcpObserver::SEND_PACKET:
+        return test::DirectTransport::SendRtp(packet, length, options);
+    }
+    return true;  // Will never happen, makes compiler happy.
+  }
+
+  bool SendRtcp(const uint8_t* packet, size_t length) override {
+    EXPECT_TRUE(IsRtcpPacket(rtc::MakeArrayView(packet, length)));
+    RtpRtcpObserver::Action action = RtpRtcpObserver::SEND_PACKET;
+    if (observer_) {
+      if (transport_type_ == kSender) {
+        action = observer_->OnSendRtcp(packet, length);
+      } else {
+        action = observer_->OnReceiveRtcp(packet, length);
+      }
+    }
+    switch (action) {
+      case RtpRtcpObserver::DROP_PACKET:
+        // Drop packet silently.
+        return true;
+      case RtpRtcpObserver::SEND_PACKET:
+        return test::DirectTransport::SendRtcp(packet, length);
+    }
+    return true;  // Will never happen, makes compiler happy.
+  }
+
+  RtpRtcpObserver* const observer_;
+  TransportType transport_type_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_RTP_RTCP_OBSERVER_H_
diff --git a/third_party/libwebrtc/test/rtp_test_utils_gn/moz.build b/third_party/libwebrtc/test/rtp_test_utils_gn/moz.build
new file mode 100644
index 0000000000..25bb2ee0a1
--- /dev/null
+++ b/third_party/libwebrtc/test/rtp_test_utils_gn/moz.build
@@ -0,0 +1,216 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("rtp_test_utils_gn")
diff --git a/third_party/libwebrtc/test/run_loop.cc b/third_party/libwebrtc/test/run_loop.cc
new file mode 100644
index 0000000000..7cc80ab481
--- /dev/null
+++ b/third_party/libwebrtc/test/run_loop.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/run_loop.h"
+
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+namespace test {
+
+RunLoop::RunLoop() {
+  worker_thread_.WrapCurrent();
+}
+
+RunLoop::~RunLoop() {
+  worker_thread_.UnwrapCurrent();
+}
+
+TaskQueueBase* RunLoop::task_queue() {
+  return &worker_thread_;
+}
+
+void RunLoop::Run() {
+  worker_thread_.ProcessMessages(WorkerThread::kForever);
+}
+
+void RunLoop::Quit() {
+  socket_server_.FailNextWait();
+}
+
+void RunLoop::Flush() {
+  worker_thread_.PostTask([this]() { socket_server_.FailNextWait(); });
+  // If a test clock is used, like with GlobalSimulatedTimeController then the
+  // thread will loop forever since time never increases. Since the clock is
+  // simulated, 0ms can be used as the loop delay, which will process all
+  // messages ready for execution.
+  int cms = rtc::GetClockForTesting() ? 0 : 1000;
+  worker_thread_.ProcessMessages(cms);
+}
+
+RunLoop::FakeSocketServer::FakeSocketServer() = default;
+RunLoop::FakeSocketServer::~FakeSocketServer() = default;
+
+void RunLoop::FakeSocketServer::FailNextWait() {
+  fail_next_wait_ = true;
+}
+
+bool RunLoop::FakeSocketServer::Wait(webrtc::TimeDelta max_wait_duration,
+                                     bool process_io) {
+  if (fail_next_wait_) {
+    fail_next_wait_ = false;
+    return false;
+  }
+  return true;
+}
+
+void RunLoop::FakeSocketServer::WakeUp() {}
+
+rtc::Socket* RunLoop::FakeSocketServer::CreateSocket(int family, int type) {
+  return nullptr;
+}
+
+RunLoop::WorkerThread::WorkerThread(rtc::SocketServer* ss)
+    : rtc::Thread(ss), tq_setter_(this) {}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/run_loop.h b/third_party/libwebrtc/test/run_loop.h
new file mode 100644
index 0000000000..8a2bf54402
--- /dev/null
+++ b/third_party/libwebrtc/test/run_loop.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_RUN_LOOP_H_
+#define TEST_RUN_LOOP_H_
+
+#include <utility>
+
+#include "absl/functional/any_invocable.h"
+#include "api/task_queue/task_queue_base.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace test {
+
+// This utility class allows you to run a TaskQueue supported interface on the
+// main test thread, call Run() while doing things asynchonously and break
+// the loop (from the same thread) from a callback by calling Quit().
+class RunLoop {
+ public:
+  RunLoop();
+  ~RunLoop();
+
+  TaskQueueBase* task_queue();
+
+  void Run();
+  void Quit();
+
+  void Flush();
+
+  void PostTask(absl::AnyInvocable<void() &&> task) {
+    task_queue()->PostTask(std::move(task));
+  }
+
+ private:
+  class FakeSocketServer : public rtc::SocketServer {
+   public:
+    FakeSocketServer();
+    ~FakeSocketServer();
+
+    void FailNextWait();
+
+   private:
+    bool Wait(webrtc::TimeDelta max_wait_duration, bool process_io) override;
+    void WakeUp() override;
+
+    rtc::Socket* CreateSocket(int family, int type) override;
+
+   private:
+    bool fail_next_wait_ = false;
+  };
+
+  class WorkerThread : public rtc::Thread {
+   public:
+    explicit WorkerThread(rtc::SocketServer* ss);
+
+   private:
+    CurrentTaskQueueSetter tq_setter_;
+  };
+
+  FakeSocketServer socket_server_;
+  WorkerThread worker_thread_{&socket_server_};
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_RUN_LOOP_H_
diff --git a/third_party/libwebrtc/test/run_loop_unittest.cc b/third_party/libwebrtc/test/run_loop_unittest.cc
new file mode 100644
index 0000000000..80f0bcbdcc
--- /dev/null
+++ b/third_party/libwebrtc/test/run_loop_unittest.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/run_loop.h"
+
+#include "api/units/time_delta.h"
+#include "rtc_base/task_queue.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+TEST(RunLoopTest, TaskQueueOnThread) {
+  test::RunLoop loop;
+  EXPECT_EQ(TaskQueueBase::Current(), loop.task_queue());
+  EXPECT_TRUE(loop.task_queue()->IsCurrent());
+}
+
+TEST(RunLoopTest, Flush) {
+  test::RunLoop loop;
+  int counter = 0;
+  loop.PostTask([&counter]() { ++counter; });
+  EXPECT_EQ(counter, 0);
+  loop.Flush();
+  EXPECT_EQ(counter, 1);
+}
+
+TEST(RunLoopTest, Delayed) {
+  test::RunLoop loop;
+  bool ran = false;
+  loop.task_queue()->PostDelayedTask(
+      [&ran, &loop]() {
+        ran = true;
+        loop.Quit();
+      },
+      TimeDelta::Millis(100));
+  loop.Flush();
+  EXPECT_FALSE(ran);
+  loop.Run();
+  EXPECT_TRUE(ran);
+}
+
+TEST(RunLoopTest, PostAndQuit) {
+  test::RunLoop loop;
+  bool ran = false;
+  loop.PostTask([&ran, &loop]() {
+    ran = true;
+    loop.Quit();
+  });
+  loop.Run();
+  EXPECT_TRUE(ran);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/run_test.cc b/third_party/libwebrtc/test/run_test.cc
new file mode 100644
index 0000000000..7af47b9faf
--- /dev/null
+++ b/third_party/libwebrtc/test/run_test.cc
@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/run_test.h"
+
+namespace webrtc {
+namespace test {
+
+void RunTest(void (*test)()) {
+  (*test)();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/run_test.h b/third_party/libwebrtc/test/run_test.h
new file mode 100644
index 0000000000..bd05584365
--- /dev/null
+++ b/third_party/libwebrtc/test/run_test.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_RUN_TEST_H_
+#define TEST_RUN_TEST_H_
+
+namespace webrtc {
+namespace test {
+
+// Running a test function on a separate thread, if required by the OS.
+void RunTest(void (*test)());
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_RUN_TEST_H_
diff --git a/third_party/libwebrtc/test/scenario/BUILD.gn b/third_party/libwebrtc/test/scenario/BUILD.gn
new file mode 100644
index 0000000000..5da6dce87d
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/BUILD.gn
@@ -0,0 +1,200 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("column_printer") {
+  testonly = true
+  sources = [
+    "column_printer.cc",
+    "column_printer.h",
+  ]
+  deps = [
+    "../../rtc_base:macromagic",
+    "../../rtc_base:stringutils",
+    "../logging:log_writer",
+  ]
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  scenario_resources = [
+    "../../resources/difficult_photo_1850_1110.yuv",
+    "../../resources/photo_1850_1110.yuv",
+    "../../resources/presentation_1850_1110.yuv",
+    "../../resources/web_screenshot_1850_1110.yuv",
+  ]
+  scenario_unittest_resources = [ "../../resources/foreman_cif.yuv" ]
+
+  if (is_ios) {
+    bundle_data("scenario_resources_bundle_data") {
+      testonly = true
+      sources = scenario_resources
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+    bundle_data("scenario_unittest_resources_bundle_data") {
+      testonly = true
+      sources = scenario_unittest_resources
+      outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+    }
+  }
+
+  rtc_library("scenario") {
+    testonly = true
+    sources = [
+      "audio_stream.cc",
+      "audio_stream.h",
+      "call_client.cc",
+      "call_client.h",
+      "hardware_codecs.cc",
+      "hardware_codecs.h",
+      "network_node.cc",
+      "network_node.h",
+      "performance_stats.cc",
+      "performance_stats.h",
+      "scenario.cc",
+      "scenario.h",
+      "scenario_config.cc",
+      "scenario_config.h",
+      "stats_collection.cc",
+      "stats_collection.h",
+      "video_frame_matcher.cc",
+      "video_frame_matcher.h",
+      "video_stream.cc",
+      "video_stream.h",
+    ]
+    deps = [
+      ":column_printer",
+      "../:fake_video_codecs",
+      "../:fileutils",
+      "../:test_common",
+      "../:test_support",
+      "../:video_test_common",
+      "../../api:array_view",
+      "../../api:create_frame_generator",
+      "../../api:fec_controller_api",
+      "../../api:frame_generator_api",
+      "../../api:libjingle_peerconnection_api",
+      "../../api:rtc_event_log_output_file",
+      "../../api:rtp_parameters",
+      "../../api:sequence_checker",
+      "../../api:time_controller",
+      "../../api:time_controller",
+      "../../api:transport_api",
+      "../../api/audio_codecs:builtin_audio_decoder_factory",
+      "../../api/audio_codecs:builtin_audio_encoder_factory",
+      "../../api/rtc_event_log",
+      "../../api/rtc_event_log:rtc_event_log_factory",
+      "../../api/task_queue",
+      "../../api/test/video:function_video_factory",
+      "../../api/transport:network_control",
+      "../../api/units:data_rate",
+      "../../api/units:data_size",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../api/video:builtin_video_bitrate_allocator_factory",
+      "../../api/video:video_frame",
+      "../../api/video:video_rtp_headers",
+      "../../api/video_codecs:scalability_mode",
+      "../../api/video_codecs:video_codecs_api",
+      "../../audio",
+      "../../call",
+      "../../call:call_interfaces",
+      "../../call:rtp_sender",
+      "../../call:simulated_network",
+      "../../call:video_stream_api",
+      "../../common_video",
+      "../../media:media_constants",
+      "../../media:rtc_audio_video",
+      "../../media:rtc_internal_video_codecs",
+      "../../media:rtc_media_base",
+      "../../modules/audio_device",
+      "../../modules/audio_device:audio_device_impl",
+      "../../modules/audio_device:mock_audio_device",
+      "../../modules/audio_mixer:audio_mixer_impl",
+      "../../modules/audio_processing",
+      "../../modules/congestion_controller/goog_cc:test_goog_cc_printer",
+      "../../modules/rtp_rtcp",
+      "../../modules/rtp_rtcp:mock_rtp_rtcp",
+      "../../modules/rtp_rtcp:rtp_rtcp_format",
+      "../../modules/video_coding:video_codec_interface",
+      "../../modules/video_coding:video_coding_utility",
+      "../../modules/video_coding:webrtc_h264",
+      "../../modules/video_coding:webrtc_multiplex",
+      "../../modules/video_coding:webrtc_vp8",
+      "../../modules/video_coding:webrtc_vp9",
+      "../../modules/video_coding/svc:scalability_mode_util",
+      "../../rtc_base:checks",
+      "../../rtc_base:copy_on_write_buffer",
+      "../../rtc_base:net_helper",
+      "../../rtc_base:refcount",
+      "../../rtc_base:rtc_base_tests_utils",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:rtc_numerics",
+      "../../rtc_base:rtc_stats_counters",
+      "../../rtc_base:safe_minmax",
+      "../../rtc_base:socket_address",
+      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base:threading",
+      "../../rtc_base/synchronization:mutex",
+      "../../rtc_base/task_utils:repeating_task",
+      "../../system_wrappers",
+      "../../system_wrappers:field_trial",
+      "../../video/config:streams_config",
+      "../logging:log_writer",
+      "../network:emulated_network",
+      "../time_controller",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/flags:flag",
+      "//third_party/abseil-cpp/absl/flags:parse",
+      "//third_party/abseil-cpp/absl/functional:any_invocable",
+      "//third_party/abseil-cpp/absl/memory",
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+    if (is_android) {
+      deps += [ "../../modules/video_coding:android_codec_factory_helper" ]
+    } else if (is_ios || is_mac) {
+      deps += [ "../../modules/video_coding:objc_codec_factory_helper" ]
+    }
+    if (rtc_enable_protobuf) {
+      deps += [ "../../modules/audio_coding:ana_config_proto" ]
+    }
+    data = scenario_resources
+    if (is_ios) {
+      deps += [ ":scenario_resources_bundle_data" ]
+    }
+  }
+  rtc_library("scenario_unittests") {
+    testonly = true
+    sources = [
+      "performance_stats_unittest.cc",
+      "probing_test.cc",
+      "scenario_unittest.cc",
+      "stats_collection_unittest.cc",
+      "video_stream_unittest.cc",
+    ]
+    deps = [
+      ":scenario",
+      "../../api/test/network_emulation",
+      "../../api/test/network_emulation:create_cross_traffic",
+      "../../logging:mocks",
+      "../../rtc_base:checks",
+      "../../system_wrappers",
+      "../../system_wrappers:field_trial",
+      "../../test:field_trial",
+      "../../test:test_support",
+      "../logging:log_writer",
+      "//testing/gmock",
+    ]
+    data = scenario_unittest_resources
+    if (is_ios) {
+      deps += [ ":scenario_unittest_resources_bundle_data" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/test/scenario/OWNERS b/third_party/libwebrtc/test/scenario/OWNERS
new file mode 100644
index 0000000000..6698afbf02
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/OWNERS
@@ -0,0 +1,2 @@
+srte@webrtc.org
+perkj@webrtc.org
diff --git a/third_party/libwebrtc/test/scenario/audio_stream.cc b/third_party/libwebrtc/test/scenario/audio_stream.cc
new file mode 100644
index 0000000000..5f2eff12ff
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/audio_stream.cc
@@ -0,0 +1,241 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/audio_stream.h"
+
+#include "absl/memory/memory.h"
+#include "test/call_test.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#else
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+namespace test {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+  kAbsSendTimeExtensionId
+};
+
+absl::optional<std::string> CreateAdaptationString(
+    AudioStreamConfig::NetworkAdaptation config) {
+#if WEBRTC_ENABLE_PROTOBUF
+
+  audio_network_adaptor::config::ControllerManager cont_conf;
+  if (config.frame.max_rate_for_60_ms.IsFinite()) {
+    auto controller =
+        cont_conf.add_controllers()->mutable_frame_length_controller();
+    controller->set_fl_decreasing_packet_loss_fraction(
+        config.frame.min_packet_loss_for_decrease);
+    controller->set_fl_increasing_packet_loss_fraction(
+        config.frame.max_packet_loss_for_increase);
+
+    controller->set_fl_20ms_to_60ms_bandwidth_bps(
+        config.frame.min_rate_for_20_ms.bps<int32_t>());
+    controller->set_fl_60ms_to_20ms_bandwidth_bps(
+        config.frame.max_rate_for_60_ms.bps<int32_t>());
+
+    if (config.frame.max_rate_for_120_ms.IsFinite()) {
+      controller->set_fl_60ms_to_120ms_bandwidth_bps(
+          config.frame.min_rate_for_60_ms.bps<int32_t>());
+      controller->set_fl_120ms_to_60ms_bandwidth_bps(
+          config.frame.max_rate_for_120_ms.bps<int32_t>());
+    }
+  }
+  cont_conf.add_controllers()->mutable_bitrate_controller();
+  std::string config_string = cont_conf.SerializeAsString();
+  return config_string;
+#else
+  RTC_LOG(LS_ERROR) << "audio_network_adaptation is enabled"
+                       " but WEBRTC_ENABLE_PROTOBUF is false.\n"
+                       "Ignoring settings.";
+  return absl::nullopt;
+#endif  // WEBRTC_ENABLE_PROTOBUF
+}
+}  // namespace
+
+std::vector<RtpExtension> GetAudioRtpExtensions(
+    const AudioStreamConfig& config) {
+  std::vector<RtpExtension> extensions;
+  if (config.stream.in_bandwidth_estimation) {
+    extensions.push_back({RtpExtension::kTransportSequenceNumberUri,
+                          kTransportSequenceNumberExtensionId});
+  }
+  if (config.stream.abs_send_time) {
+    extensions.push_back(
+        {RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId});
+  }
+  return extensions;
+}
+
+SendAudioStream::SendAudioStream(
+    CallClient* sender,
+    AudioStreamConfig config,
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+    Transport* send_transport)
+    : sender_(sender), config_(config) {
+  AudioSendStream::Config send_config(send_transport);
+  ssrc_ = sender->GetNextAudioSsrc();
+  send_config.rtp.ssrc = ssrc_;
+  SdpAudioFormat::Parameters sdp_params;
+  if (config.source.channels == 2)
+    sdp_params["stereo"] = "1";
+  if (config.encoder.initial_frame_length != TimeDelta::Millis(20))
+    sdp_params["ptime"] =
+        std::to_string(config.encoder.initial_frame_length.ms());
+  if (config.encoder.enable_dtx)
+    sdp_params["usedtx"] = "1";
+
+  // SdpAudioFormat::num_channels indicates that the encoder is capable of
+  // stereo, but the actual channel count used is based on the "stereo"
+  // parameter.
+  send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+      CallTest::kAudioSendPayloadType, {"opus", 48000, 2, sdp_params});
+  RTC_DCHECK_LE(config.source.channels, 2);
+  send_config.encoder_factory = encoder_factory;
+
+  bool use_fixed_rate = !config.encoder.min_rate && !config.encoder.max_rate;
+  if (use_fixed_rate)
+    send_config.send_codec_spec->target_bitrate_bps =
+        config.encoder.fixed_rate.bps();
+  if (!config.adapt.binary_proto.empty()) {
+    send_config.audio_network_adaptor_config = config.adapt.binary_proto;
+  } else if (config.network_adaptation) {
+    send_config.audio_network_adaptor_config =
+        CreateAdaptationString(config.adapt);
+  }
+  if (config.encoder.allocate_bitrate ||
+      config.stream.in_bandwidth_estimation) {
+    DataRate min_rate = DataRate::Infinity();
+    DataRate max_rate = DataRate::Infinity();
+    if (use_fixed_rate) {
+      min_rate = config.encoder.fixed_rate;
+      max_rate = config.encoder.fixed_rate;
+    } else {
+      min_rate = *config.encoder.min_rate;
+      max_rate = *config.encoder.max_rate;
+    }
+    send_config.min_bitrate_bps = min_rate.bps();
+    send_config.max_bitrate_bps = max_rate.bps();
+  }
+
+  if (config.stream.in_bandwidth_estimation) {
+    send_config.send_codec_spec->transport_cc_enabled = true;
+  }
+  send_config.rtp.extensions = GetAudioRtpExtensions(config);
+
+  sender_->SendTask([&] {
+    send_stream_ = sender_->call_->CreateAudioSendStream(send_config);
+    sender->call_->OnAudioTransportOverheadChanged(
+        sender_->transport_->packet_overhead().bytes());
+  });
+}
+
+SendAudioStream::~SendAudioStream() {
+  sender_->SendTask(
+      [this] { sender_->call_->DestroyAudioSendStream(send_stream_); });
+}
+
+void SendAudioStream::Start() {
+  sender_->SendTask([this] {
+    send_stream_->Start();
+    sender_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+  });
+}
+
+void SendAudioStream::Stop() {
+  sender_->SendTask([this] { send_stream_->Stop(); });
+}
+
+void SendAudioStream::SetMuted(bool mute) {
+  sender_->SendTask([this, mute] { send_stream_->SetMuted(mute); });
+}
+
+ColumnPrinter SendAudioStream::StatsPrinter() {
+  return ColumnPrinter::Lambda(
+      "audio_target_rate",
+      [this](rtc::SimpleStringBuilder& sb) {
+        sender_->SendTask([this, &sb] {
+          AudioSendStream::Stats stats = send_stream_->GetStats();
+          sb.AppendFormat("%.0lf", stats.target_bitrate_bps / 8.0);
+        });
+      },
+      64);
+}
+
+ReceiveAudioStream::ReceiveAudioStream(
+    CallClient* receiver,
+    AudioStreamConfig config,
+    SendAudioStream* send_stream,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    Transport* feedback_transport)
+    : receiver_(receiver), config_(config) {
+  AudioReceiveStreamInterface::Config recv_config;
+  recv_config.rtp.local_ssrc = receiver_->GetNextAudioLocalSsrc();
+  recv_config.rtcp_send_transport = feedback_transport;
+  recv_config.rtp.remote_ssrc = send_stream->ssrc_;
+  receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;
+  recv_config.rtp.extensions = GetAudioRtpExtensions(config);
+  recv_config.decoder_factory = decoder_factory;
+  recv_config.decoder_map = {
+      {CallTest::kAudioSendPayloadType, {"opus", 48000, 2}}};
+  recv_config.sync_group = config.render.sync_group;
+  receiver_->SendTask([&] {
+    receive_stream_ = receiver_->call_->CreateAudioReceiveStream(recv_config);
+  });
+}
+ReceiveAudioStream::~ReceiveAudioStream() {
+  receiver_->SendTask(
+      [&] { receiver_->call_->DestroyAudioReceiveStream(receive_stream_); });
+}
+
+void ReceiveAudioStream::Start() {
+  receiver_->SendTask([&] {
+    receive_stream_->Start();
+    receiver_->call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
+  });
+}
+
+void ReceiveAudioStream::Stop() {
+  receiver_->SendTask([&] { receive_stream_->Stop(); });
+}
+
+AudioReceiveStreamInterface::Stats ReceiveAudioStream::GetStats() const {
+  AudioReceiveStreamInterface::Stats result;
+  receiver_->SendTask([&] {
+    result = receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true);
+  });
+  return result;
+}
+
+AudioStreamPair::~AudioStreamPair() = default;
+
+AudioStreamPair::AudioStreamPair(
+    CallClient* sender,
+    rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+    CallClient* receiver,
+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+    AudioStreamConfig config)
+    : config_(config),
+      send_stream_(sender, config, encoder_factory, sender->transport_.get()),
+      receive_stream_(receiver,
+                      config,
+                      &send_stream_,
+                      decoder_factory,
+                      receiver->transport_.get()) {}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/audio_stream.h b/third_party/libwebrtc/test/scenario/audio_stream.h
new file mode 100644
index 0000000000..cbaf9d29eb
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/audio_stream.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_AUDIO_STREAM_H_
+#define TEST_SCENARIO_AUDIO_STREAM_H_
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "test/scenario/call_client.h"
+#include "test/scenario/column_printer.h"
+#include "test/scenario/network_node.h"
+#include "test/scenario/scenario_config.h"
+
+namespace webrtc {
+namespace test {
+
+// SendAudioStream represents sending of audio. It can be used for starting the
+// stream if neccessary.
+class SendAudioStream {
+ public:
+  ~SendAudioStream();
+
+  SendAudioStream(const SendAudioStream&) = delete;
+  SendAudioStream& operator=(const SendAudioStream&) = delete;
+
+  void Start();
+  void Stop();
+  void SetMuted(bool mute);
+  ColumnPrinter StatsPrinter();
+
+ private:
+  friend class Scenario;
+  friend class AudioStreamPair;
+  friend class ReceiveAudioStream;
+  SendAudioStream(CallClient* sender,
+                  AudioStreamConfig config,
+                  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+                  Transport* send_transport);
+  AudioSendStream* send_stream_ = nullptr;
+  CallClient* const sender_;
+  const AudioStreamConfig config_;
+  uint32_t ssrc_;
+};
+
+// ReceiveAudioStream represents an audio receiver. It can't be used directly.
+class ReceiveAudioStream {
+ public:
+  ~ReceiveAudioStream();
+
+  ReceiveAudioStream(const ReceiveAudioStream&) = delete;
+  ReceiveAudioStream& operator=(const ReceiveAudioStream&) = delete;
+
+  void Start();
+  void Stop();
+  AudioReceiveStreamInterface::Stats GetStats() const;
+
+ private:
+  friend class Scenario;
+  friend class AudioStreamPair;
+  ReceiveAudioStream(CallClient* receiver,
+                     AudioStreamConfig config,
+                     SendAudioStream* send_stream,
+                     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+                     Transport* feedback_transport);
+  AudioReceiveStreamInterface* receive_stream_ = nullptr;
+  CallClient* const receiver_;
+  const AudioStreamConfig config_;
+};
+
+// AudioStreamPair represents an audio streaming session. It can be used to
+// access underlying send and receive classes. It can also be used in calls to
+// the Scenario class.
+class AudioStreamPair {
+ public:
+  ~AudioStreamPair();
+
+  AudioStreamPair(const AudioStreamPair&) = delete;
+  AudioStreamPair& operator=(const AudioStreamPair&) = delete;
+
+  SendAudioStream* send() { return &send_stream_; }
+  ReceiveAudioStream* receive() { return &receive_stream_; }
+
+ private:
+  friend class Scenario;
+  AudioStreamPair(CallClient* sender,
+                  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
+                  CallClient* receiver,
+                  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
+                  AudioStreamConfig config);
+
+ private:
+  const AudioStreamConfig config_;
+  SendAudioStream send_stream_;
+  ReceiveAudioStream receive_stream_;
+};
+
+std::vector<RtpExtension> GetAudioRtpExtensions(
+    const AudioStreamConfig& config);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCENARIO_AUDIO_STREAM_H_
diff --git a/third_party/libwebrtc/test/scenario/call_client.cc b/third_party/libwebrtc/test/scenario/call_client.cc
new file mode 100644
index 0000000000..c80f58eeeb
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/call_client.cc
@@ -0,0 +1,386 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/call_client.h"
+
+#include <iostream>
+#include <memory>
+#include <utility>
+
+#include "api/media_types.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/transport/network_types.h"
+#include "call/call.h"
+#include "call/rtp_transport_controller_send_factory.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_util.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+static constexpr size_t kNumSsrcs = 6;
+const uint32_t kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE, 0xBADCAFF,
+                                           0xBADCB00, 0xBADCB01, 0xBADCB02};
+const uint32_t kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF,
+                                             0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
+const uint32_t kVideoRecvLocalSsrcs[kNumSsrcs] = {0xDAB001, 0xDAB002, 0xDAB003,
+                                                  0xDAB004, 0xDAB005, 0xDAB006};
+const uint32_t kAudioSendSsrc = 0xDEADBEEF;
+const uint32_t kReceiverLocalAudioSsrc = 0x1234567;
+
+constexpr int kEventLogOutputIntervalMs = 5000;
+
+CallClientFakeAudio InitAudio(TimeController* time_controller) {
+  CallClientFakeAudio setup;
+  auto capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
+  auto renderer = TestAudioDeviceModule::CreateDiscardRenderer(48000);
+  setup.fake_audio_device = TestAudioDeviceModule::Create(
+      time_controller->GetTaskQueueFactory(), std::move(capturer),
+      std::move(renderer), 1.f);
+  setup.apm = AudioProcessingBuilder().Create();
+  setup.fake_audio_device->Init();
+  AudioState::Config audio_state_config;
+  audio_state_config.audio_mixer = AudioMixerImpl::Create();
+  audio_state_config.audio_processing = setup.apm;
+  audio_state_config.audio_device_module = setup.fake_audio_device;
+  setup.audio_state = AudioState::Create(audio_state_config);
+  setup.fake_audio_device->RegisterAudioCallback(
+      setup.audio_state->audio_transport());
+  return setup;
+}
+
+Call* CreateCall(TimeController* time_controller,
+                 RtcEventLog* event_log,
+                 CallClientConfig config,
+                 LoggingNetworkControllerFactory* network_controller_factory,
+                 rtc::scoped_refptr<AudioState> audio_state) {
+  CallConfig call_config(event_log);
+  call_config.bitrate_config.max_bitrate_bps =
+      config.transport.rates.max_rate.bps_or(-1);
+  call_config.bitrate_config.min_bitrate_bps =
+      config.transport.rates.min_rate.bps();
+  call_config.bitrate_config.start_bitrate_bps =
+      config.transport.rates.start_rate.bps();
+  call_config.task_queue_factory = time_controller->GetTaskQueueFactory();
+  call_config.network_controller_factory = network_controller_factory;
+  call_config.audio_state = audio_state;
+  call_config.pacer_burst_interval = config.pacer_burst_interval;
+  call_config.trials = config.field_trials;
+  Clock* clock = time_controller->GetClock();
+  return Call::Create(call_config, clock,
+                      RtpTransportControllerSendFactory().Create(
+                          call_config.ExtractTransportConfig(), clock));
+}
+
+std::unique_ptr<RtcEventLog> CreateEventLog(
+    TaskQueueFactory* task_queue_factory,
+    LogWriterFactoryInterface* log_writer_factory) {
+  if (!log_writer_factory) {
+    return std::make_unique<RtcEventLogNull>();
+  }
+  auto event_log = RtcEventLogFactory(task_queue_factory)
+                       .CreateRtcEventLog(RtcEventLog::EncodingType::NewFormat);
+  bool success = event_log->StartLogging(log_writer_factory->Create(".rtc.dat"),
+                                         kEventLogOutputIntervalMs);
+  RTC_CHECK(success);
+  return event_log;
+}
+}  // namespace
+NetworkControleUpdateCache::NetworkControleUpdateCache(
+    std::unique_ptr<NetworkControllerInterface> controller)
+    : controller_(std::move(controller)) {}
+NetworkControlUpdate NetworkControleUpdateCache::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  return Update(controller_->OnNetworkAvailability(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnNetworkRouteChange(
+    NetworkRouteChange msg) {
+  return Update(controller_->OnNetworkRouteChange(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnProcessInterval(
+    ProcessInterval msg) {
+  return Update(controller_->OnProcessInterval(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnRemoteBitrateReport(
+    RemoteBitrateReport msg) {
+  return Update(controller_->OnRemoteBitrateReport(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnRoundTripTimeUpdate(
+    RoundTripTimeUpdate msg) {
+  return Update(controller_->OnRoundTripTimeUpdate(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnSentPacket(SentPacket msg) {
+  return Update(controller_->OnSentPacket(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnReceivedPacket(
+    ReceivedPacket msg) {
+  return Update(controller_->OnReceivedPacket(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnStreamsConfig(
+    StreamsConfig msg) {
+  return Update(controller_->OnStreamsConfig(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnTargetRateConstraints(
+    TargetRateConstraints msg) {
+  return Update(controller_->OnTargetRateConstraints(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnTransportLossReport(
+    TransportLossReport msg) {
+  return Update(controller_->OnTransportLossReport(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnTransportPacketsFeedback(
+    TransportPacketsFeedback msg) {
+  return Update(controller_->OnTransportPacketsFeedback(msg));
+}
+NetworkControlUpdate NetworkControleUpdateCache::OnNetworkStateEstimate(
+    NetworkStateEstimate msg) {
+  return Update(controller_->OnNetworkStateEstimate(msg));
+}
+
+NetworkControlUpdate NetworkControleUpdateCache::update_state() const {
+  return update_state_;
+}
+NetworkControlUpdate NetworkControleUpdateCache::Update(
+    NetworkControlUpdate update) {
+  if (update.target_rate)
+    update_state_.target_rate = update.target_rate;
+  if (update.pacer_config)
+    update_state_.pacer_config = update.pacer_config;
+  if (update.congestion_window)
+    update_state_.congestion_window = update.congestion_window;
+  if (!update.probe_cluster_configs.empty())
+    update_state_.probe_cluster_configs = update.probe_cluster_configs;
+  return update;
+}
+
+LoggingNetworkControllerFactory::LoggingNetworkControllerFactory(
+    LogWriterFactoryInterface* log_writer_factory,
+    TransportControllerConfig config) {
+  if (config.cc_factory) {
+    cc_factory_ = config.cc_factory;
+    if (log_writer_factory)
+      RTC_LOG(LS_WARNING)
+          << "Can't log controller state for injected network controllers";
+  } else {
+    if (log_writer_factory) {
+      goog_cc_factory_.AttachWriter(
+          log_writer_factory->Create(".cc_state.txt"));
+      print_cc_state_ = true;
+    }
+    cc_factory_ = &goog_cc_factory_;
+  }
+}
+
+LoggingNetworkControllerFactory::~LoggingNetworkControllerFactory() {}
+
+void LoggingNetworkControllerFactory::LogCongestionControllerStats(
+    Timestamp at_time) {
+  if (print_cc_state_)
+    goog_cc_factory_.PrintState(at_time);
+}
+
+NetworkControlUpdate LoggingNetworkControllerFactory::GetUpdate() const {
+  if (last_controller_)
+    return last_controller_->update_state();
+  return NetworkControlUpdate();
+}
+
+std::unique_ptr<NetworkControllerInterface>
+LoggingNetworkControllerFactory::Create(NetworkControllerConfig config) {
+  auto controller =
+      std::make_unique<NetworkControleUpdateCache>(cc_factory_->Create(config));
+  last_controller_ = controller.get();
+  return controller;
+}
+
+TimeDelta LoggingNetworkControllerFactory::GetProcessInterval() const {
+  return cc_factory_->GetProcessInterval();
+}
+
+void LoggingNetworkControllerFactory::SetRemoteBitrateEstimate(
+    RemoteBitrateReport msg) {
+  if (last_controller_)
+    last_controller_->OnRemoteBitrateReport(msg);
+}
+
+CallClient::CallClient(
+    TimeController* time_controller,
+    std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
+    CallClientConfig config)
+    : time_controller_(time_controller),
+      clock_(time_controller->GetClock()),
+      log_writer_factory_(std::move(log_writer_factory)),
+      network_controller_factory_(log_writer_factory_.get(), config.transport),
+      task_queue_(time_controller->GetTaskQueueFactory()->CreateTaskQueue(
+          "CallClient",
+          TaskQueueFactory::Priority::NORMAL)) {
+  config.field_trials = &field_trials_;
+  SendTask([this, config] {
+    event_log_ = CreateEventLog(time_controller_->GetTaskQueueFactory(),
+                                log_writer_factory_.get());
+    fake_audio_setup_ = InitAudio(time_controller_);
+
+    call_.reset(CreateCall(time_controller_, event_log_.get(), config,
+                           &network_controller_factory_,
+                           fake_audio_setup_.audio_state));
+    transport_ = std::make_unique<NetworkNodeTransport>(clock_, call_.get());
+  });
+}
+
+CallClient::~CallClient() {
+  SendTask([&] {
+    call_.reset();
+    fake_audio_setup_ = {};
+    rtc::Event done;
+    event_log_->StopLogging([&done] { done.Set(); });
+    done.Wait(rtc::Event::kForever);
+    event_log_.reset();
+  });
+}
+
+ColumnPrinter CallClient::StatsPrinter() {
+  return ColumnPrinter::Lambda(
+      "pacer_delay call_send_bw",
+      [this](rtc::SimpleStringBuilder& sb) {
+        Call::Stats call_stats = call_->GetStats();
+        sb.AppendFormat("%.3lf %.0lf", call_stats.pacer_delay_ms / 1000.0,
+                        call_stats.send_bandwidth_bps / 8.0);
+      },
+      64);
+}
+
+Call::Stats CallClient::GetStats() {
+  // This call needs to be made on the thread that `call_` was constructed on.
+  Call::Stats stats;
+  SendTask([this, &stats] { stats = call_->GetStats(); });
+  return stats;
+}
+
+DataRate CallClient::target_rate() const {
+  return network_controller_factory_.GetUpdate().target_rate->target_rate;
+}
+
+DataRate CallClient::stable_target_rate() const {
+  return network_controller_factory_.GetUpdate()
+      .target_rate->stable_target_rate;
+}
+
+DataRate CallClient::padding_rate() const {
+  return network_controller_factory_.GetUpdate().pacer_config->pad_rate();
+}
+
+void CallClient::SetRemoteBitrate(DataRate bitrate) {
+  RemoteBitrateReport msg;
+  msg.bandwidth = bitrate;
+  msg.receive_time = clock_->CurrentTime();
+  network_controller_factory_.SetRemoteBitrateEstimate(msg);
+}
+
+void CallClient::UpdateBitrateConstraints(
+    const BitrateConstraints& constraints) {
+  SendTask([this, &constraints]() {
+    call_->GetTransportControllerSend()->SetSdpBitrateParameters(constraints);
+  });
+}
+
+void CallClient::SetAudioReceiveRtpHeaderExtensions(
+    rtc::ArrayView<RtpExtension> extensions) {
+  SendTask([this, &extensions]() {
+    audio_extensions_ = RtpHeaderExtensionMap(extensions);
+  });
+}
+
+void CallClient::SetVideoReceiveRtpHeaderExtensions(
+    rtc::ArrayView<RtpExtension> extensions) {
+  SendTask([this, &extensions]() {
+    video_extensions_ = RtpHeaderExtensionMap(extensions);
+  });
+}
+
+void CallClient::OnPacketReceived(EmulatedIpPacket packet) {
+  MediaType media_type = MediaType::ANY;
+  if (IsRtpPacket(packet.data)) {
+    media_type = ssrc_media_types_[ParseRtpSsrc(packet.data)];
+    task_queue_.PostTask([this, media_type,
+                          packet = std::move(packet)]() mutable {
+      RtpHeaderExtensionMap& extension_map = media_type == MediaType::AUDIO
+                                                 ? audio_extensions_
+                                                 : video_extensions_;
+      RtpPacketReceived received_packet(&extension_map, packet.arrival_time);
+      RTC_CHECK(received_packet.Parse(packet.data));
+      call_->Receiver()->DeliverRtpPacket(media_type, received_packet,
+                                          /*undemuxable_packet_handler=*/
+                                          [](const RtpPacketReceived& packet) {
+                                            RTC_CHECK_NOTREACHED();
+                                            return false;
+                                          });
+    });
+  } else {
+    task_queue_.PostTask(
+        [call = call_.get(), packet = std::move(packet)]() mutable {
+          call->Receiver()->DeliverRtcpPacket(packet.data);
+        });
+  }
+}
+
+std::unique_ptr<RtcEventLogOutput> CallClient::GetLogWriter(std::string name) {
+  if (!log_writer_factory_ || name.empty())
+    return nullptr;
+  return log_writer_factory_->Create(name);
+}
+
+uint32_t CallClient::GetNextVideoSsrc() {
+  RTC_CHECK_LT(next_video_ssrc_index_, kNumSsrcs);
+  return kVideoSendSsrcs[next_video_ssrc_index_++];
+}
+
+uint32_t CallClient::GetNextVideoLocalSsrc() {
+  RTC_CHECK_LT(next_video_local_ssrc_index_, kNumSsrcs);
+  return kVideoRecvLocalSsrcs[next_video_local_ssrc_index_++];
+}
+
+uint32_t CallClient::GetNextAudioSsrc() {
+  RTC_CHECK_LT(next_audio_ssrc_index_, 1);
+  next_audio_ssrc_index_++;
+  return kAudioSendSsrc;
+}
+
+uint32_t CallClient::GetNextAudioLocalSsrc() {
+  RTC_CHECK_LT(next_audio_local_ssrc_index_, 1);
+  next_audio_local_ssrc_index_++;
+  return kReceiverLocalAudioSsrc;
+}
+
+uint32_t CallClient::GetNextRtxSsrc() {
+  RTC_CHECK_LT(next_rtx_ssrc_index_, kNumSsrcs);
+  return kSendRtxSsrcs[next_rtx_ssrc_index_++];
+}
+
+void CallClient::SendTask(std::function<void()> task) {
+  task_queue_.SendTask(std::move(task));
+}
+
+int16_t CallClient::Bind(EmulatedEndpoint* endpoint) {
+  uint16_t port = endpoint->BindReceiver(0, this).value();
+  endpoints_.push_back({endpoint, port});
+  return port;
+}
+
+void CallClient::UnBind() {
+  for (auto ep_port : endpoints_)
+    ep_port.first->UnbindReceiver(ep_port.second);
+}
+
+CallClientPair::~CallClientPair() = default;
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/call_client.h b/third_party/libwebrtc/test/scenario/call_client.h
new file mode 100644
index 0000000000..5d62fc75e7
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/call_client.h
@@ -0,0 +1,204 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_CALL_CLIENT_H_
+#define TEST_SCENARIO_CALL_CLIENT_H_
+
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/rtp_parameters.h"
+#include "api/test/time_controller.h"
+#include "api/units/data_rate.h"
+#include "call/call.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "test/logging/log_writer.h"
+#include "test/network/network_emulation.h"
+#include "test/scenario/column_printer.h"
+#include "test/scenario/network_node.h"
+#include "test/scenario/scenario_config.h"
+
+namespace webrtc {
+
+namespace test {
+// Helper class to capture network controller state.
+class NetworkControleUpdateCache : public NetworkControllerInterface {
+ public:
+  explicit NetworkControleUpdateCache(
+      std::unique_ptr<NetworkControllerInterface> controller);
+
+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
+  NetworkControlUpdate OnReceivedPacket(ReceivedPacket msg) override;
+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
+  NetworkControlUpdate OnTargetRateConstraints(
+      TargetRateConstraints msg) override;
+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
+  NetworkControlUpdate OnTransportPacketsFeedback(
+      TransportPacketsFeedback msg) override;
+  NetworkControlUpdate OnNetworkStateEstimate(
+      NetworkStateEstimate msg) override;
+
+  NetworkControlUpdate update_state() const;
+
+ private:
+  NetworkControlUpdate Update(NetworkControlUpdate update);
+  const std::unique_ptr<NetworkControllerInterface> controller_;
+  NetworkControlUpdate update_state_;
+};
+
+class LoggingNetworkControllerFactory
+    : public NetworkControllerFactoryInterface {
+ public:
+  LoggingNetworkControllerFactory(LogWriterFactoryInterface* log_writer_factory,
+                                  TransportControllerConfig config);
+
+  ~LoggingNetworkControllerFactory();
+
+  LoggingNetworkControllerFactory(const LoggingNetworkControllerFactory&) =
+      delete;
+  LoggingNetworkControllerFactory& operator=(
+      const LoggingNetworkControllerFactory&) = delete;
+
+  std::unique_ptr<NetworkControllerInterface> Create(
+      NetworkControllerConfig config) override;
+  TimeDelta GetProcessInterval() const override;
+  // TODO(srte): Consider using the Columnprinter interface for this.
+  void LogCongestionControllerStats(Timestamp at_time);
+  void SetRemoteBitrateEstimate(RemoteBitrateReport msg);
+
+  NetworkControlUpdate GetUpdate() const;
+
+ private:
+  GoogCcDebugFactory goog_cc_factory_;
+  NetworkControllerFactoryInterface* cc_factory_ = nullptr;
+  bool print_cc_state_ = false;
+  NetworkControleUpdateCache* last_controller_ = nullptr;
+};
+
+struct CallClientFakeAudio {
+  rtc::scoped_refptr<AudioProcessing> apm;
+  rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device;
+  rtc::scoped_refptr<AudioState> audio_state;
+};
+// CallClient represents a participant in a call scenario. It is created by the
+// Scenario class and is used as sender and receiver when setting up a media
+// stream session.
+class CallClient : public EmulatedNetworkReceiverInterface {
+ public:
+  CallClient(TimeController* time_controller,
+             std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
+             CallClientConfig config);
+
+  ~CallClient();
+
+  CallClient(const CallClient&) = delete;
+  CallClient& operator=(const CallClient&) = delete;
+
+  ColumnPrinter StatsPrinter();
+  Call::Stats GetStats();
+  DataRate send_bandwidth() {
+    return DataRate::BitsPerSec(GetStats().send_bandwidth_bps);
+  }
+  DataRate target_rate() const;
+  DataRate stable_target_rate() const;
+  DataRate padding_rate() const;
+  void UpdateBitrateConstraints(const BitrateConstraints& constraints);
+  void SetRemoteBitrate(DataRate bitrate);
+
+  void SetAudioReceiveRtpHeaderExtensions(
+      rtc::ArrayView<RtpExtension> extensions);
+  void SetVideoReceiveRtpHeaderExtensions(
+      rtc::ArrayView<RtpExtension> extensions);
+
+  void OnPacketReceived(EmulatedIpPacket packet) override;
+  std::unique_ptr<RtcEventLogOutput> GetLogWriter(std::string name);
+
+  // Exposed publicly so that tests can execute tasks such as querying stats
+  // for media streams in the expected runtime environment (essentially what
+  // CallClient does internally for GetStats()).
+  void SendTask(std::function<void()> task);
+
+ private:
+  friend class Scenario;
+  friend class CallClientPair;
+  friend class SendVideoStream;
+  friend class VideoStreamPair;
+  friend class ReceiveVideoStream;
+  friend class SendAudioStream;
+  friend class ReceiveAudioStream;
+  friend class AudioStreamPair;
+  friend class NetworkNodeTransport;
+  uint32_t GetNextVideoSsrc();
+  uint32_t GetNextVideoLocalSsrc();
+  uint32_t GetNextAudioSsrc();
+  uint32_t GetNextAudioLocalSsrc();
+  uint32_t GetNextRtxSsrc();
+  int16_t Bind(EmulatedEndpoint* endpoint);
+  void UnBind();
+
+  TimeController* const time_controller_;
+  Clock* clock_;
+  const std::unique_ptr<LogWriterFactoryInterface> log_writer_factory_;
+  std::unique_ptr<RtcEventLog> event_log_;
+  LoggingNetworkControllerFactory network_controller_factory_;
+  CallClientFakeAudio fake_audio_setup_;
+  std::unique_ptr<Call> call_;
+  std::unique_ptr<NetworkNodeTransport> transport_;
+  std::vector<std::pair<EmulatedEndpoint*, uint16_t>> endpoints_;
+  RtpHeaderExtensionMap audio_extensions_;
+  RtpHeaderExtensionMap video_extensions_;
+
+  int next_video_ssrc_index_ = 0;
+  int next_video_local_ssrc_index_ = 0;
+  int next_rtx_ssrc_index_ = 0;
+  int next_audio_ssrc_index_ = 0;
+  int next_audio_local_ssrc_index_ = 0;
+  std::map<uint32_t, MediaType> ssrc_media_types_;
+  // Defined last so it's destroyed first.
+  TaskQueueForTest task_queue_;
+
+  const FieldTrialBasedConfig field_trials_;
+};
+
+class CallClientPair {
+ public:
+  ~CallClientPair();
+
+  CallClientPair(const CallClientPair&) = delete;
+  CallClientPair& operator=(const CallClientPair&) = delete;
+
+  CallClient* first() { return first_; }
+  CallClient* second() { return second_; }
+  std::pair<CallClient*, CallClient*> forward() { return {first(), second()}; }
+  std::pair<CallClient*, CallClient*> reverse() { return {second(), first()}; }
+
+ private:
+  friend class Scenario;
+  CallClientPair(CallClient* first, CallClient* second)
+      : first_(first), second_(second) {}
+  CallClient* const first_;
+  CallClient* const second_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCENARIO_CALL_CLIENT_H_
diff --git a/third_party/libwebrtc/test/scenario/column_printer.cc b/third_party/libwebrtc/test/scenario/column_printer.cc
new file mode 100644
index 0000000000..661c83bd0d
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/column_printer.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/column_printer.h"
+
+namespace webrtc {
+namespace test {
+
+ColumnPrinter::ColumnPrinter(const ColumnPrinter&) = default;
+ColumnPrinter::~ColumnPrinter() = default;
+
+ColumnPrinter::ColumnPrinter(
+    const char* headers,
+    std::function<void(rtc::SimpleStringBuilder&)> printer,
+    size_t max_length)
+    : headers_(headers), printer_(printer), max_length_(max_length) {}
+
+ColumnPrinter ColumnPrinter::Fixed(const char* headers, std::string fields) {
+  return ColumnPrinter(
+      headers, [fields](rtc::SimpleStringBuilder& sb) { sb << fields; },
+      fields.size());
+}
+
+ColumnPrinter ColumnPrinter::Lambda(
+    const char* headers,
+    std::function<void(rtc::SimpleStringBuilder&)> printer,
+    size_t max_length) {
+  return ColumnPrinter(headers, printer, max_length);
+}
+
+StatesPrinter::StatesPrinter(std::unique_ptr<RtcEventLogOutput> writer,
+                             std::vector<ColumnPrinter> printers)
+    : writer_(std::move(writer)), printers_(printers) {
+  RTC_CHECK(!printers_.empty());
+  for (auto& printer : printers_)
+    buffer_size_ += printer.max_length_ + 1;
+  buffer_.resize(buffer_size_);
+}
+
+StatesPrinter::~StatesPrinter() = default;
+
+void StatesPrinter::PrintHeaders() {
+  if (!writer_)
+    return;
+  writer_->Write(printers_[0].headers_);
+  for (size_t i = 1; i < printers_.size(); ++i) {
+    writer_->Write(" ");
+    writer_->Write(printers_[i].headers_);
+  }
+  writer_->Write("\n");
+}
+
+void StatesPrinter::PrintRow() {
+  // Note that this is run for null output to preserve side effects, this allows
+  // setting break points etc.
+  rtc::SimpleStringBuilder sb(buffer_);
+  printers_[0].printer_(sb);
+  for (size_t i = 1; i < printers_.size(); ++i) {
+    sb << ' ';
+    printers_[i].printer_(sb);
+  }
+  sb << "\n";
+  if (writer_)
+    writer_->Write(std::string(sb.str(), sb.size()));
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/column_printer.h b/third_party/libwebrtc/test/scenario/column_printer.h
new file mode 100644
index 0000000000..529f4597ec
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/column_printer.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_COLUMN_PRINTER_H_
+#define TEST_SCENARIO_COLUMN_PRINTER_H_
+#include <functional>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "rtc_base/strings/string_builder.h"
+#include "test/logging/log_writer.h"
+
+namespace webrtc {
+namespace test {
+class ColumnPrinter {
+ public:
+  ColumnPrinter(const ColumnPrinter&);
+  ~ColumnPrinter();
+  static ColumnPrinter Fixed(const char* headers, std::string fields);
+  static ColumnPrinter Lambda(
+      const char* headers,
+      std::function<void(rtc::SimpleStringBuilder&)> printer,
+      size_t max_length = 256);
+
+ protected:
+  friend class StatesPrinter;
+  const char* headers_;
+  std::function<void(rtc::SimpleStringBuilder&)> printer_;
+  size_t max_length_;
+
+ private:
+  ColumnPrinter(const char* headers,
+                std::function<void(rtc::SimpleStringBuilder&)> printer,
+                size_t max_length);
+};
+
+class StatesPrinter {
+ public:
+  StatesPrinter(std::unique_ptr<RtcEventLogOutput> writer,
+                std::vector<ColumnPrinter> printers);
+
+  ~StatesPrinter();
+
+  StatesPrinter(const StatesPrinter&) = delete;
+  StatesPrinter& operator=(const StatesPrinter&) = delete;
+
+  void PrintHeaders();
+  void PrintRow();
+
+ private:
+  const std::unique_ptr<RtcEventLogOutput> writer_;
+  const std::vector<ColumnPrinter> printers_;
+  size_t buffer_size_ = 0;
+  std::vector<char> buffer_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCENARIO_COLUMN_PRINTER_H_
diff --git a/third_party/libwebrtc/test/scenario/hardware_codecs.cc b/third_party/libwebrtc/test/scenario/hardware_codecs.cc
new file mode 100644
index 0000000000..cac0f10dc9
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/hardware_codecs.cc
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/hardware_codecs.h"
+
+#include "rtc_base/checks.h"
+
+#ifdef WEBRTC_ANDROID
+#include "modules/video_coding/codecs/test/android_codec_factory_helper.h"
+#endif
+#ifdef WEBRTC_MAC
+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
+#endif
+
+namespace webrtc {
+namespace test {
+std::unique_ptr<VideoEncoderFactory> CreateHardwareEncoderFactory() {
+#ifdef WEBRTC_ANDROID
+  InitializeAndroidObjects();
+  return CreateAndroidEncoderFactory();
+#else
+#ifdef WEBRTC_MAC
+  return CreateObjCEncoderFactory();
+#else
+  RTC_DCHECK_NOTREACHED()
+      << "Hardware encoder not implemented on this platform.";
+  return nullptr;
+#endif
+#endif
+}
+std::unique_ptr<VideoDecoderFactory> CreateHardwareDecoderFactory() {
+#ifdef WEBRTC_ANDROID
+  InitializeAndroidObjects();
+  return CreateAndroidDecoderFactory();
+#else
+#ifdef WEBRTC_MAC
+  return CreateObjCDecoderFactory();
+#else
+  RTC_DCHECK_NOTREACHED()
+      << "Hardware decoder not implemented on this platform.";
+  return nullptr;
+#endif
+#endif
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/hardware_codecs.h b/third_party/libwebrtc/test/scenario/hardware_codecs.h
new file mode 100644
index 0000000000..ae14a27d9e
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/hardware_codecs.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_HARDWARE_CODECS_H_
+#define TEST_SCENARIO_HARDWARE_CODECS_H_
+
+#include <memory>
+
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+
+namespace webrtc {
+namespace test {
+std::unique_ptr<VideoEncoderFactory> CreateHardwareEncoderFactory();
+std::unique_ptr<VideoDecoderFactory> CreateHardwareDecoderFactory();
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_SCENARIO_HARDWARE_CODECS_H_
diff --git a/third_party/libwebrtc/test/scenario/network_node.cc b/third_party/libwebrtc/test/scenario/network_node.cc
new file mode 100644
index 0000000000..e149bb11e0
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/network_node.cc
@@ -0,0 +1,144 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/network_node.h"
+
+#include <algorithm>
+#include <vector>
+
+#include <memory>
+#include "rtc_base/net_helper.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+constexpr char kDummyTransportName[] = "dummy";
+SimulatedNetwork::Config CreateSimulationConfig(
+    NetworkSimulationConfig config) {
+  SimulatedNetwork::Config sim_config;
+  sim_config.link_capacity_kbps = config.bandwidth.kbps_or(0);
+  sim_config.loss_percent = config.loss_rate * 100;
+  sim_config.queue_delay_ms = config.delay.ms();
+  sim_config.delay_standard_deviation_ms = config.delay_std_dev.ms();
+  sim_config.packet_overhead = config.packet_overhead.bytes<int>();
+  sim_config.queue_length_packets =
+      config.packet_queue_length_limit.value_or(0);
+  return sim_config;
+}
+}  // namespace
+
+SimulationNode::SimulationNode(NetworkSimulationConfig config,
+                               SimulatedNetwork* behavior,
+                               EmulatedNetworkNode* network_node)
+    : config_(config), simulation_(behavior), network_node_(network_node) {}
+
+std::unique_ptr<SimulatedNetwork> SimulationNode::CreateBehavior(
+    NetworkSimulationConfig config) {
+  SimulatedNetwork::Config sim_config = CreateSimulationConfig(config);
+  return std::make_unique<SimulatedNetwork>(sim_config);
+}
+
+void SimulationNode::UpdateConfig(
+    std::function<void(NetworkSimulationConfig*)> modifier) {
+  modifier(&config_);
+  SimulatedNetwork::Config sim_config = CreateSimulationConfig(config_);
+  simulation_->SetConfig(sim_config);
+}
+
+void SimulationNode::PauseTransmissionUntil(Timestamp until) {
+  simulation_->PauseTransmissionUntil(until.us());
+}
+
+ColumnPrinter SimulationNode::ConfigPrinter() const {
+  return ColumnPrinter::Lambda(
+      "propagation_delay capacity loss_rate",
+      [this](rtc::SimpleStringBuilder& sb) {
+        sb.AppendFormat("%.3lf %.0lf %.2lf", config_.delay.seconds<double>(),
+                        config_.bandwidth.bps() / 8.0, config_.loss_rate);
+      });
+}
+
+NetworkNodeTransport::NetworkNodeTransport(Clock* sender_clock,
+                                           Call* sender_call)
+    : sender_clock_(sender_clock), sender_call_(sender_call) {}
+
+NetworkNodeTransport::~NetworkNodeTransport() = default;
+
+bool NetworkNodeTransport::SendRtp(const uint8_t* packet,
+                                   size_t length,
+                                   const PacketOptions& options) {
+  int64_t send_time_ms = sender_clock_->TimeInMilliseconds();
+  rtc::SentPacket sent_packet;
+  sent_packet.packet_id = options.packet_id;
+  sent_packet.info.included_in_feedback = options.included_in_feedback;
+  sent_packet.info.included_in_allocation = options.included_in_allocation;
+  sent_packet.send_time_ms = send_time_ms;
+  sent_packet.info.packet_size_bytes = length;
+  sent_packet.info.packet_type = rtc::PacketType::kData;
+  sender_call_->OnSentPacket(sent_packet);
+
+  MutexLock lock(&mutex_);
+  if (!endpoint_)
+    return false;
+  rtc::CopyOnWriteBuffer buffer(packet, length);
+  endpoint_->SendPacket(local_address_, remote_address_, buffer,
+                        packet_overhead_.bytes());
+  return true;
+}
+
+bool NetworkNodeTransport::SendRtcp(const uint8_t* packet, size_t length) {
+  rtc::CopyOnWriteBuffer buffer(packet, length);
+  MutexLock lock(&mutex_);
+  if (!endpoint_)
+    return false;
+  endpoint_->SendPacket(local_address_, remote_address_, buffer,
+                        packet_overhead_.bytes());
+  return true;
+}
+
+void NetworkNodeTransport::Connect(EmulatedEndpoint* endpoint,
+                                   const rtc::SocketAddress& receiver_address,
+                                   DataSize packet_overhead) {
+  rtc::NetworkRoute route;
+  route.connected = true;
+  // We assume that the address will be unique in the lower bytes.
+  route.local = rtc::RouteEndpoint::CreateWithNetworkId(static_cast<uint16_t>(
+      receiver_address.ipaddr().v4AddressAsHostOrderInteger()));
+  route.remote = rtc::RouteEndpoint::CreateWithNetworkId(static_cast<uint16_t>(
+      receiver_address.ipaddr().v4AddressAsHostOrderInteger()));
+  route.packet_overhead = packet_overhead.bytes() +
+                          receiver_address.ipaddr().overhead() +
+                          cricket::kUdpHeaderSize;
+  {
+    // Only IPv4 address is supported.
+    RTC_CHECK_EQ(receiver_address.family(), AF_INET);
+    MutexLock lock(&mutex_);
+    endpoint_ = endpoint;
+    local_address_ = rtc::SocketAddress(endpoint_->GetPeerLocalAddress(), 0);
+    remote_address_ = receiver_address;
+    packet_overhead_ = packet_overhead;
+    current_network_route_ = route;
+  }
+
+  sender_call_->GetTransportControllerSend()->OnNetworkRouteChanged(
+      kDummyTransportName, route);
+}
+
+void NetworkNodeTransport::Disconnect() {
+  MutexLock lock(&mutex_);
+  current_network_route_.connected = false;
+  sender_call_->GetTransportControllerSend()->OnNetworkRouteChanged(
+      kDummyTransportName, current_network_route_);
+  current_network_route_ = {};
+  endpoint_ = nullptr;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/network_node.h b/third_party/libwebrtc/test/scenario/network_node.h
new file mode 100644
index 0000000000..fe87cefa26
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/network_node.h
@@ -0,0 +1,83 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_NETWORK_NODE_H_
+#define TEST_SCENARIO_NETWORK_NODE_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/call/transport.h"
+#include "api/units/timestamp.h"
+#include "call/call.h"
+#include "call/simulated_network.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/network/network_emulation.h"
+#include "test/scenario/column_printer.h"
+#include "test/scenario/scenario_config.h"
+
+namespace webrtc {
+namespace test {
+
+class SimulationNode {
+ public:
+  SimulationNode(NetworkSimulationConfig config,
+                 SimulatedNetwork* behavior,
+                 EmulatedNetworkNode* network_node);
+  static std::unique_ptr<SimulatedNetwork> CreateBehavior(
+      NetworkSimulationConfig config);
+
+  void UpdateConfig(std::function<void(NetworkSimulationConfig*)> modifier);
+  void PauseTransmissionUntil(Timestamp until);
+  ColumnPrinter ConfigPrinter() const;
+  EmulatedNetworkNode* node() { return network_node_; }
+
+ private:
+  NetworkSimulationConfig config_;
+  SimulatedNetwork* const simulation_;
+  EmulatedNetworkNode* const network_node_;
+};
+
+class NetworkNodeTransport : public Transport {
+ public:
+  NetworkNodeTransport(Clock* sender_clock, Call* sender_call);
+  ~NetworkNodeTransport() override;
+
+  bool SendRtp(const uint8_t* packet,
+               size_t length,
+               const PacketOptions& options) override;
+  bool SendRtcp(const uint8_t* packet, size_t length) override;
+
+  void Connect(EmulatedEndpoint* endpoint,
+               const rtc::SocketAddress& receiver_address,
+               DataSize packet_overhead);
+  void Disconnect();
+
+  DataSize packet_overhead() {
+    MutexLock lock(&mutex_);
+    return packet_overhead_;
+  }
+
+ private:
+  Mutex mutex_;
+  Clock* const sender_clock_;
+  Call* const sender_call_;
+  EmulatedEndpoint* endpoint_ RTC_GUARDED_BY(mutex_) = nullptr;
+  rtc::SocketAddress local_address_ RTC_GUARDED_BY(mutex_);
+  rtc::SocketAddress remote_address_ RTC_GUARDED_BY(mutex_);
+  DataSize packet_overhead_ RTC_GUARDED_BY(mutex_) = DataSize::Zero();
+  rtc::NetworkRoute current_network_route_ RTC_GUARDED_BY(mutex_);
+};
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_SCENARIO_NETWORK_NODE_H_
diff --git a/third_party/libwebrtc/test/scenario/performance_stats.cc b/third_party/libwebrtc/test/scenario/performance_stats.cc
new file mode 100644
index 0000000000..e12be8a003
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/performance_stats.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/performance_stats.h"
+
+#include <algorithm>
+
+namespace webrtc {
+namespace test {
+void VideoFramesStats::AddFrameInfo(const VideoFrameBuffer& frame,
+                                    Timestamp at_time) {
+  ++count;
+  RTC_DCHECK(at_time.IsFinite());
+  pixels.AddSample(frame.width() * frame.height());
+  resolution.AddSample(std::max(frame.width(), frame.height()));
+  frames.AddEvent(at_time);
+}
+
+void VideoFramesStats::AddStats(const VideoFramesStats& other) {
+  count += other.count;
+  pixels.AddSamples(other.pixels);
+  resolution.AddSamples(other.resolution);
+  frames.AddEvents(other.frames);
+}
+
+void VideoQualityStats::AddStats(const VideoQualityStats& other) {
+  capture.AddStats(other.capture);
+  render.AddStats(other.render);
+  lost_count += other.lost_count;
+  freeze_count += other.freeze_count;
+  capture_to_decoded_delay.AddSamples(other.capture_to_decoded_delay);
+  end_to_end_delay.AddSamples(other.end_to_end_delay);
+  psnr.AddSamples(other.psnr);
+  psnr_with_freeze.AddSamples(other.psnr_with_freeze);
+  skipped_between_rendered.AddSamples(other.skipped_between_rendered);
+  freeze_duration.AddSamples(other.freeze_duration);
+  time_between_freezes.AddSamples(other.time_between_freezes);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/performance_stats.h b/third_party/libwebrtc/test/scenario/performance_stats.h
new file mode 100644
index 0000000000..6974ab6d22
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/performance_stats.h
@@ -0,0 +1,108 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_PERFORMANCE_STATS_H_
+#define TEST_SCENARIO_PERFORMANCE_STATS_H_
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/video_frame_buffer.h"
+#include "rtc_base/numerics/event_rate_counter.h"
+#include "rtc_base/numerics/sample_stats.h"
+
+namespace webrtc {
+namespace test {
+
+struct VideoFramePair {
+  rtc::scoped_refptr<VideoFrameBuffer> captured;
+  rtc::scoped_refptr<VideoFrameBuffer> decoded;
+  Timestamp capture_time = Timestamp::MinusInfinity();
+  Timestamp decoded_time = Timestamp::PlusInfinity();
+  Timestamp render_time = Timestamp::PlusInfinity();
+  // A unique identifier for the spatial/temporal layer the decoded frame
+  // belongs to. Note that this does not reflect the id as defined by the
+  // underlying layer setup.
+  int layer_id = 0;
+  int capture_id = 0;
+  int decode_id = 0;
+  // Indicates the repeat count for the decoded frame. Meaning that the same
+  // decoded frame has matched differend captured frames.
+  int repeated = 0;
+};
+
+
+struct VideoFramesStats {
+  int count = 0;
+  SampleStats<double> pixels;
+  SampleStats<double> resolution;
+  EventRateCounter frames;
+  void AddFrameInfo(const VideoFrameBuffer& frame, Timestamp at_time);
+  void AddStats(const VideoFramesStats& other);
+};
+
+struct VideoQualityStats {
+  int lost_count = 0;
+  int freeze_count = 0;
+  VideoFramesStats capture;
+  VideoFramesStats render;
+  // Time from frame was captured on device to time frame was delivered from
+  // decoder.
+  SampleStats<TimeDelta> capture_to_decoded_delay;
+  // Time from frame was captured on device to time frame was displayed on
+  // device.
+  SampleStats<TimeDelta> end_to_end_delay;
+  // PSNR for delivered frames. Note that this might go up for a worse
+  // connection due to frame dropping.
+  SampleStats<double> psnr;
+  // PSNR for all frames, dropped or lost frames are compared to the last
+  // successfully delivered frame
+  SampleStats<double> psnr_with_freeze;
+  // Frames skipped between two nearest.
+  SampleStats<double> skipped_between_rendered;
+  // In the next 2 metrics freeze is a pause that is longer, than maximum:
+  //  1. 150ms
+  //  2. 3 * average time between two sequential frames.
+  // Item 1 will cover high fps video and is a duration, that is noticeable by
+  // human eye. Item 2 will cover low fps video like screen sharing.
+  SampleStats<TimeDelta> freeze_duration;
+  // Mean time between one freeze end and next freeze start.
+  SampleStats<TimeDelta> time_between_freezes;
+  void AddStats(const VideoQualityStats& other);
+};
+
+struct CollectedCallStats {
+  SampleStats<DataRate> target_rate;
+  SampleStats<TimeDelta> pacer_delay;
+  SampleStats<TimeDelta> round_trip_time;
+  SampleStats<double> memory_usage;
+};
+
+struct CollectedAudioReceiveStats {
+  SampleStats<double> expand_rate;
+  SampleStats<double> accelerate_rate;
+  SampleStats<TimeDelta> jitter_buffer;
+};
+struct CollectedVideoSendStats {
+  SampleStats<double> encode_frame_rate;
+  SampleStats<TimeDelta> encode_time;
+  SampleStats<double> encode_usage;
+  SampleStats<DataRate> media_bitrate;
+  SampleStats<DataRate> fec_bitrate;
+};
+struct CollectedVideoReceiveStats {
+  SampleStats<TimeDelta> decode_time;
+  SampleStats<TimeDelta> decode_time_max;
+  SampleStats<double> decode_pixels;
+  SampleStats<double> resolution;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_SCENARIO_PERFORMANCE_STATS_H_
diff --git a/third_party/libwebrtc/test/scenario/performance_stats_unittest.cc b/third_party/libwebrtc/test/scenario/performance_stats_unittest.cc
new file mode 100644
index 0000000000..8d87c87745
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/performance_stats_unittest.cc
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/performance_stats.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(EventRateCounter, ReturnsCorrectTotalDuration) {
+  EventRateCounter event_rate_counter;
+  EXPECT_EQ(event_rate_counter.TotalDuration(), TimeDelta::Zero());
+  event_rate_counter.AddEvent(Timestamp::Seconds(1));
+  EXPECT_EQ(event_rate_counter.TotalDuration(), TimeDelta::Zero());
+  event_rate_counter.AddEvent(Timestamp::Seconds(2));
+  EXPECT_EQ(event_rate_counter.TotalDuration(), TimeDelta::Seconds(1));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/probing_test.cc b/third_party/libwebrtc/test/scenario/probing_test.cc
new file mode 100644
index 0000000000..86653ced9b
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/probing_test.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(ProbingTest, InitialProbingRampsUpTargetRateWhenNetworkIsGood) {
+  Scenario s;
+  NetworkSimulationConfig good_network;
+  good_network.bandwidth = DataRate::KilobitsPerSec(2000);
+
+  VideoStreamConfig video_config;
+  video_config.encoder.codec =
+      VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+  CallClientConfig send_config;
+  auto* caller = s.CreateClient("caller", send_config);
+  auto* callee = s.CreateClient("callee", CallClientConfig());
+  auto route =
+      s.CreateRoutes(caller, {s.CreateSimulationNode(good_network)}, callee,
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), video_config);
+
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_GE(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            3 * send_config.transport.rates.start_rate);
+}
+
+TEST(ProbingTest, MidCallProbingRampupTriggeredByUpdatedBitrateConstraints) {
+  Scenario s;
+
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+  const DataRate kConstrainedRate = DataRate::KilobitsPerSec(100);
+  const DataRate kHighRate = DataRate::KilobitsPerSec(1500);
+
+  VideoStreamConfig video_config;
+  video_config.encoder.codec =
+      VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+  CallClientConfig send_call_config;
+  send_call_config.transport.rates.start_rate = kStartRate;
+  send_call_config.transport.rates.max_rate = kHighRate * 2;
+  auto* caller = s.CreateClient("caller", send_call_config);
+  auto* callee = s.CreateClient("callee", CallClientConfig());
+  auto route = s.CreateRoutes(
+      caller, {s.CreateSimulationNode(NetworkSimulationConfig())}, callee,
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), video_config);
+
+  // Wait until initial probing rampup is done and then set a low max bitrate.
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_GE(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            5 * send_call_config.transport.rates.start_rate);
+  BitrateConstraints bitrate_config;
+  bitrate_config.max_bitrate_bps = kConstrainedRate.bps();
+  caller->UpdateBitrateConstraints(bitrate_config);
+
+  // Wait until the low send bitrate has taken effect, and then set a much
+  // higher max bitrate.
+  s.RunFor(TimeDelta::Seconds(2));
+  EXPECT_LT(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            kConstrainedRate * 1.1);
+  bitrate_config.max_bitrate_bps = 2 * kHighRate.bps();
+  caller->UpdateBitrateConstraints(bitrate_config);
+
+  // Check that the max send bitrate is reached quicker than would be possible
+  // with simple AIMD rate control.
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_GE(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            kHighRate);
+}
+
+TEST(ProbingTest, ProbesRampsUpWhenVideoEncoderConfigChanges) {
+  Scenario s;
+  const DataRate kStartRate = DataRate::KilobitsPerSec(50);
+  const DataRate kHdRate = DataRate::KilobitsPerSec(3250);
+
+  // Set up 3-layer simulcast.
+  VideoStreamConfig video_config;
+  video_config.encoder.codec =
+      VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+  video_config.encoder.simulcast_streams = {webrtc::ScalabilityMode::kL1T3,
+                                            webrtc::ScalabilityMode::kL1T3,
+                                            webrtc::ScalabilityMode::kL1T3};
+  video_config.source.generator.width = 1280;
+  video_config.source.generator.height = 720;
+
+  CallClientConfig send_call_config;
+  send_call_config.transport.rates.start_rate = kStartRate;
+  send_call_config.transport.rates.max_rate = kHdRate * 2;
+  auto* caller = s.CreateClient("caller", send_call_config);
+  auto* callee = s.CreateClient("callee", CallClientConfig());
+  auto send_net =
+      s.CreateMutableSimulationNode([&](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(200);
+      });
+  auto route =
+      s.CreateRoutes(caller, {send_net->node()}, callee,
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  auto* video_stream = s.CreateVideoStream(route->forward(), video_config);
+
+  // Only QVGA enabled initially. Run until initial probing is done and BWE
+  // has settled.
+  video_stream->send()->UpdateActiveLayers({true, false, false});
+  s.RunFor(TimeDelta::Seconds(2));
+
+  // Remove network constraints and run for a while more, BWE should be much
+  // less than required HD rate.
+  send_net->UpdateConfig([&](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::PlusInfinity();
+  });
+  s.RunFor(TimeDelta::Seconds(2));
+
+  DataRate bandwidth =
+      DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps);
+  EXPECT_LT(bandwidth, kHdRate / 4);
+
+  // Enable all layers, triggering a probe.
+  video_stream->send()->UpdateActiveLayers({true, true, true});
+
+  // Run for a short while and verify BWE has ramped up fast.
+  s.RunFor(TimeDelta::Seconds(2));
+  EXPECT_GT(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            kHdRate);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/scenario.cc b/third_party/libwebrtc/test/scenario/scenario.cc
new file mode 100644
index 0000000000..93377120a1
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/scenario.cc
@@ -0,0 +1,355 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/scenario.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "rtc_base/socket_address.h"
+#include "test/logging/file_log_writer.h"
+#include "test/network/network_emulation.h"
+#include "test/scenario/video_stream.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(bool, scenario_logs, false, "Save logs from scenario framework.");
+ABSL_FLAG(std::string,
+          scenario_logs_root,
+          "",
+          "Output root path, based on project root if unset.");
+
+namespace webrtc {
+namespace test {
+namespace {
+
+std::unique_ptr<FileLogWriterFactory> GetScenarioLogManager(
+    absl::string_view file_name) {
+  if (absl::GetFlag(FLAGS_scenario_logs) && !file_name.empty()) {
+    std::string output_root = absl::GetFlag(FLAGS_scenario_logs_root);
+    if (output_root.empty())
+      output_root = OutputPath() + "output_data/";
+
+    auto base_filename = output_root + std::string(file_name) + ".";
+    RTC_LOG(LS_INFO) << "Saving scenario logs to: " << base_filename;
+    return std::make_unique<FileLogWriterFactory>(base_filename);
+  }
+  return nullptr;
+}
+}  // namespace
+
+Scenario::Scenario()
+    : Scenario(std::unique_ptr<LogWriterFactoryInterface>(),
+               /*real_time=*/false) {}
+
+Scenario::Scenario(const testing::TestInfo* test_info)
+    : Scenario(std::string(test_info->test_suite_name()) + "/" +
+               test_info->name()) {}
+
+Scenario::Scenario(absl::string_view file_name)
+    : Scenario(file_name, /*real_time=*/false) {}
+
+Scenario::Scenario(absl::string_view file_name, bool real_time)
+    : Scenario(GetScenarioLogManager(file_name), real_time) {}
+
+Scenario::Scenario(
+    std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
+    bool real_time)
+    : log_writer_factory_(std::move(log_writer_factory)),
+      network_manager_(real_time ? TimeMode::kRealTime : TimeMode::kSimulated,
+                       EmulatedNetworkStatsGatheringMode::kDefault),
+      clock_(network_manager_.time_controller()->GetClock()),
+      audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
+      audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
+      task_queue_(network_manager_.time_controller()
+                      ->GetTaskQueueFactory()
+                      ->CreateTaskQueue("Scenario",
+                                        TaskQueueFactory::Priority::NORMAL)) {}
+
+Scenario::~Scenario() {
+  if (start_time_.IsFinite())
+    Stop();
+  for (auto& call_client : clients_) {
+    call_client->transport_->Disconnect();
+    call_client->UnBind();
+  }
+}
+
+ColumnPrinter Scenario::TimePrinter() {
+  return ColumnPrinter::Lambda(
+      "time",
+      [this](rtc::SimpleStringBuilder& sb) {
+        sb.AppendFormat("%.3lf", Now().seconds<double>());
+      },
+      32);
+}
+
+StatesPrinter* Scenario::CreatePrinter(absl::string_view name,
+                                       TimeDelta interval,
+                                       std::vector<ColumnPrinter> printers) {
+  std::vector<ColumnPrinter> all_printers{TimePrinter()};
+  for (auto& printer : printers)
+    all_printers.push_back(printer);
+  StatesPrinter* printer = new StatesPrinter(GetLogWriter(name), all_printers);
+  printers_.emplace_back(printer);
+  printer->PrintHeaders();
+  if (interval.IsFinite())
+    Every(interval, [printer] { printer->PrintRow(); });
+  return printer;
+}
+
+CallClient* Scenario::CreateClient(absl::string_view name,
+                                   CallClientConfig config) {
+  CallClient* client = new CallClient(network_manager_.time_controller(),
+                                      GetLogWriterFactory(name), config);
+  if (config.transport.state_log_interval.IsFinite()) {
+    Every(config.transport.state_log_interval, [this, client]() {
+      client->network_controller_factory_.LogCongestionControllerStats(Now());
+    });
+  }
+  clients_.emplace_back(client);
+  return client;
+}
+
+CallClient* Scenario::CreateClient(
+    absl::string_view name,
+    std::function<void(CallClientConfig*)> config_modifier) {
+  CallClientConfig config;
+  config_modifier(&config);
+  return CreateClient(name, config);
+}
+
+CallClientPair* Scenario::CreateRoutes(
+    CallClient* first,
+    std::vector<EmulatedNetworkNode*> send_link,
+    CallClient* second,
+    std::vector<EmulatedNetworkNode*> return_link) {
+  return CreateRoutes(first, send_link,
+                      DataSize::Bytes(PacketOverhead::kDefault), second,
+                      return_link, DataSize::Bytes(PacketOverhead::kDefault));
+}
+
+CallClientPair* Scenario::CreateRoutes(
+    CallClient* first,
+    std::vector<EmulatedNetworkNode*> send_link,
+    DataSize first_overhead,
+    CallClient* second,
+    std::vector<EmulatedNetworkNode*> return_link,
+    DataSize second_overhead) {
+  CallClientPair* client_pair = new CallClientPair(first, second);
+  ChangeRoute(client_pair->forward(), send_link, first_overhead);
+  ChangeRoute(client_pair->reverse(), return_link, second_overhead);
+  client_pairs_.emplace_back(client_pair);
+  return client_pair;
+}
+
+void Scenario::ChangeRoute(std::pair<CallClient*, CallClient*> clients,
+                           std::vector<EmulatedNetworkNode*> over_nodes) {
+  ChangeRoute(clients, over_nodes, DataSize::Bytes(PacketOverhead::kDefault));
+}
+
+void Scenario::ChangeRoute(std::pair<CallClient*, CallClient*> clients,
+                           std::vector<EmulatedNetworkNode*> over_nodes,
+                           DataSize overhead) {
+  EmulatedRoute* route = network_manager_.CreateRoute(over_nodes);
+  uint16_t port = clients.second->Bind(route->to);
+  auto addr = rtc::SocketAddress(route->to->GetPeerLocalAddress(), port);
+  clients.first->transport_->Connect(route->from, addr, overhead);
+}
+
+EmulatedNetworkNode* Scenario::CreateSimulationNode(
+    std::function<void(NetworkSimulationConfig*)> config_modifier) {
+  NetworkSimulationConfig config;
+  config_modifier(&config);
+  return CreateSimulationNode(config);
+}
+
+EmulatedNetworkNode* Scenario::CreateSimulationNode(
+    NetworkSimulationConfig config) {
+  return network_manager_.CreateEmulatedNode(
+      SimulationNode::CreateBehavior(config));
+}
+
+SimulationNode* Scenario::CreateMutableSimulationNode(
+    std::function<void(NetworkSimulationConfig*)> config_modifier) {
+  NetworkSimulationConfig config;
+  config_modifier(&config);
+  return CreateMutableSimulationNode(config);
+}
+
+SimulationNode* Scenario::CreateMutableSimulationNode(
+    NetworkSimulationConfig config) {
+  std::unique_ptr<SimulatedNetwork> behavior =
+      SimulationNode::CreateBehavior(config);
+  SimulatedNetwork* behavior_ptr = behavior.get();
+  auto* emulated_node =
+      network_manager_.CreateEmulatedNode(std::move(behavior));
+  simulation_nodes_.emplace_back(
+      new SimulationNode(config, behavior_ptr, emulated_node));
+  return simulation_nodes_.back().get();
+}
+
+void Scenario::TriggerPacketBurst(std::vector<EmulatedNetworkNode*> over_nodes,
+                                  size_t num_packets,
+                                  size_t packet_size) {
+  network_manager_.CreateCrossTrafficRoute(over_nodes)
+      ->TriggerPacketBurst(num_packets, packet_size);
+}
+
+void Scenario::NetworkDelayedAction(
+    std::vector<EmulatedNetworkNode*> over_nodes,
+    size_t packet_size,
+    std::function<void()> action) {
+  network_manager_.CreateCrossTrafficRoute(over_nodes)
+      ->NetworkDelayedAction(packet_size, action);
+}
+
+VideoStreamPair* Scenario::CreateVideoStream(
+    std::pair<CallClient*, CallClient*> clients,
+    std::function<void(VideoStreamConfig*)> config_modifier) {
+  VideoStreamConfig config;
+  config_modifier(&config);
+  return CreateVideoStream(clients, config);
+}
+
+VideoStreamPair* Scenario::CreateVideoStream(
+    std::pair<CallClient*, CallClient*> clients,
+    VideoStreamConfig config) {
+  std::vector<RtpExtension> extensions = GetVideoRtpExtensions(config);
+  clients.first->SetVideoReceiveRtpHeaderExtensions(extensions);
+  clients.second->SetVideoReceiveRtpHeaderExtensions(extensions);
+  video_streams_.emplace_back(
+      new VideoStreamPair(clients.first, clients.second, config));
+  return video_streams_.back().get();
+}
+
+AudioStreamPair* Scenario::CreateAudioStream(
+    std::pair<CallClient*, CallClient*> clients,
+    std::function<void(AudioStreamConfig*)> config_modifier) {
+  AudioStreamConfig config;
+  config_modifier(&config);
+  return CreateAudioStream(clients, config);
+}
+
+AudioStreamPair* Scenario::CreateAudioStream(
+    std::pair<CallClient*, CallClient*> clients,
+    AudioStreamConfig config) {
+  std::vector<RtpExtension> extensions = GetAudioRtpExtensions(config);
+  clients.first->SetAudioReceiveRtpHeaderExtensions(extensions);
+  clients.second->SetAudioReceiveRtpHeaderExtensions(extensions);
+  audio_streams_.emplace_back(
+      new AudioStreamPair(clients.first, audio_encoder_factory_, clients.second,
+                          audio_decoder_factory_, config));
+  return audio_streams_.back().get();
+}
+
+void Scenario::Every(TimeDelta interval,
+                     absl::AnyInvocable<void(TimeDelta)> function) {
+  RepeatingTaskHandle::DelayedStart(
+      task_queue_.get(), interval,
+      [interval, function = std::move(function)]() mutable {
+        function(interval);
+        return interval;
+      });
+}
+
+void Scenario::Every(TimeDelta interval, absl::AnyInvocable<void()> function) {
+  RepeatingTaskHandle::DelayedStart(
+      task_queue_.get(), interval,
+      [interval, function = std::move(function)]() mutable {
+        function();
+        return interval;
+      });
+}
+
+void Scenario::Post(absl::AnyInvocable<void() &&> function) {
+  task_queue_->PostTask(std::move(function));
+}
+
+void Scenario::At(TimeDelta offset, absl::AnyInvocable<void() &&> function) {
+  RTC_DCHECK_GT(offset, TimeSinceStart());
+  task_queue_->PostDelayedTask(std::move(function), TimeUntilTarget(offset));
+}
+
+void Scenario::RunFor(TimeDelta duration) {
+  if (start_time_.IsInfinite())
+    Start();
+  network_manager_.time_controller()->AdvanceTime(duration);
+}
+
+void Scenario::RunUntil(TimeDelta target_time_since_start) {
+  RunFor(TimeUntilTarget(target_time_since_start));
+}
+
+void Scenario::RunUntil(TimeDelta target_time_since_start,
+                        TimeDelta check_interval,
+                        std::function<bool()> exit_function) {
+  if (start_time_.IsInfinite())
+    Start();
+  while (check_interval >= TimeUntilTarget(target_time_since_start)) {
+    network_manager_.time_controller()->AdvanceTime(check_interval);
+    if (exit_function())
+      return;
+  }
+  network_manager_.time_controller()->AdvanceTime(
+      TimeUntilTarget(target_time_since_start));
+}
+
+void Scenario::Start() {
+  start_time_ = clock_->CurrentTime();
+  for (auto& stream_pair : video_streams_)
+    stream_pair->receive()->Start();
+  for (auto& stream_pair : audio_streams_)
+    stream_pair->receive()->Start();
+  for (auto& stream_pair : video_streams_) {
+    if (stream_pair->config_.autostart) {
+      stream_pair->send()->Start();
+    }
+  }
+  for (auto& stream_pair : audio_streams_) {
+    if (stream_pair->config_.autostart) {
+      stream_pair->send()->Start();
+    }
+  }
+}
+
+void Scenario::Stop() {
+  RTC_DCHECK(start_time_.IsFinite());
+  for (auto& stream_pair : video_streams_) {
+    stream_pair->send()->Stop();
+  }
+  for (auto& stream_pair : audio_streams_)
+    stream_pair->send()->Stop();
+  for (auto& stream_pair : video_streams_)
+    stream_pair->receive()->Stop();
+  for (auto& stream_pair : audio_streams_)
+    stream_pair->receive()->Stop();
+  start_time_ = Timestamp::PlusInfinity();
+}
+
+Timestamp Scenario::Now() {
+  return clock_->CurrentTime();
+}
+
+TimeDelta Scenario::TimeSinceStart() {
+  if (start_time_.IsInfinite())
+    return TimeDelta::Zero();
+  return Now() - start_time_;
+}
+
+TimeDelta Scenario::TimeUntilTarget(TimeDelta target_time_offset) {
+  return target_time_offset - TimeSinceStart();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/scenario.h b/third_party/libwebrtc/test/scenario/scenario.h
new file mode 100644
index 0000000000..cad9210002
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/scenario.h
@@ -0,0 +1,189 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_SCENARIO_H_
+#define TEST_SCENARIO_SCENARIO_H_
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/test/time_controller.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "test/gtest.h"
+#include "test/logging/log_writer.h"
+#include "test/network/network_emulation_manager.h"
+#include "test/scenario/audio_stream.h"
+#include "test/scenario/call_client.h"
+#include "test/scenario/column_printer.h"
+#include "test/scenario/network_node.h"
+#include "test/scenario/scenario_config.h"
+#include "test/scenario/video_stream.h"
+
+namespace webrtc {
+namespace test {
+// Scenario is a class owning everything for a test scenario. It creates and
+// holds network nodes, call clients and media streams. It also provides methods
+// for changing behavior at runtime. Since it always keeps ownership of the
+// created components, it generally returns non-owning pointers. It maintains
+// the life of its objects until it is destroyed.
+// For methods accepting configuration structs, a modifier function interface is
+// generally provided. This allows simple partial overriding of the default
+// configuration.
+class Scenario {
+ public:
+  Scenario();
+  explicit Scenario(const testing::TestInfo* test_info);
+  explicit Scenario(absl::string_view file_name);
+  Scenario(absl::string_view file_name, bool real_time);
+  Scenario(std::unique_ptr<LogWriterFactoryInterface> log_writer_manager,
+           bool real_time);
+
+  ~Scenario();
+
+  Scenario(const Scenario&) = delete;
+  Scenario& operator=(const Scenario&) = delete;
+
+  NetworkEmulationManagerImpl* net() { return &network_manager_; }
+
+  EmulatedNetworkNode* CreateSimulationNode(NetworkSimulationConfig config);
+  EmulatedNetworkNode* CreateSimulationNode(
+      std::function<void(NetworkSimulationConfig*)> config_modifier);
+
+  SimulationNode* CreateMutableSimulationNode(NetworkSimulationConfig config);
+  SimulationNode* CreateMutableSimulationNode(
+      std::function<void(NetworkSimulationConfig*)> config_modifier);
+
+  CallClient* CreateClient(absl::string_view name, CallClientConfig config);
+  CallClient* CreateClient(
+      absl::string_view name,
+      std::function<void(CallClientConfig*)> config_modifier);
+
+  CallClientPair* CreateRoutes(CallClient* first,
+                               std::vector<EmulatedNetworkNode*> send_link,
+                               CallClient* second,
+                               std::vector<EmulatedNetworkNode*> return_link);
+
+  CallClientPair* CreateRoutes(CallClient* first,
+                               std::vector<EmulatedNetworkNode*> send_link,
+                               DataSize first_overhead,
+                               CallClient* second,
+                               std::vector<EmulatedNetworkNode*> return_link,
+                               DataSize second_overhead);
+
+  void ChangeRoute(std::pair<CallClient*, CallClient*> clients,
+                   std::vector<EmulatedNetworkNode*> over_nodes);
+
+  void ChangeRoute(std::pair<CallClient*, CallClient*> clients,
+                   std::vector<EmulatedNetworkNode*> over_nodes,
+                   DataSize overhead);
+
+  VideoStreamPair* CreateVideoStream(
+      std::pair<CallClient*, CallClient*> clients,
+      std::function<void(VideoStreamConfig*)> config_modifier);
+  VideoStreamPair* CreateVideoStream(
+      std::pair<CallClient*, CallClient*> clients,
+      VideoStreamConfig config);
+
+  AudioStreamPair* CreateAudioStream(
+      std::pair<CallClient*, CallClient*> clients,
+      std::function<void(AudioStreamConfig*)> config_modifier);
+  AudioStreamPair* CreateAudioStream(
+      std::pair<CallClient*, CallClient*> clients,
+      AudioStreamConfig config);
+
+  // Runs the provided function with a fixed interval. For real time tests,
+  // `function` starts being called after `interval` from the call to Every().
+  void Every(TimeDelta interval, absl::AnyInvocable<void(TimeDelta)> function);
+  void Every(TimeDelta interval, absl::AnyInvocable<void()> function);
+
+  // Runs the provided function on the internal task queue. This ensure that
+  // it's run on the main thread for simulated time tests.
+  void Post(absl::AnyInvocable<void() &&> function);
+
+  // Runs the provided function after given duration has passed. For real time
+  // tests, `function` is called after `target_time_since_start` from the call
+  // to Every().
+  void At(TimeDelta offset, absl::AnyInvocable<void() &&> function);
+
+  // Sends a packet over the nodes and runs `action` when it has been delivered.
+  void NetworkDelayedAction(std::vector<EmulatedNetworkNode*> over_nodes,
+                            size_t packet_size,
+                            std::function<void()> action);
+
+  // Runs the scenario for the given time.
+  void RunFor(TimeDelta duration);
+  // Runs the scenario until `target_time_since_start`.
+  void RunUntil(TimeDelta target_time_since_start);
+  // Runs the scenario until `target_time_since_start` or `exit_function`
+  // returns true. `exit_function` is polled after each `check_interval` has
+  // passed.
+  void RunUntil(TimeDelta target_time_since_start,
+                TimeDelta check_interval,
+                std::function<bool()> exit_function);
+  void Start();
+  void Stop();
+
+  // Triggers sending of dummy packets over the given nodes.
+  void TriggerPacketBurst(std::vector<EmulatedNetworkNode*> over_nodes,
+                          size_t num_packets,
+                          size_t packet_size);
+
+  ColumnPrinter TimePrinter();
+  StatesPrinter* CreatePrinter(absl::string_view name,
+                               TimeDelta interval,
+                               std::vector<ColumnPrinter> printers);
+
+  // Returns the current time.
+  Timestamp Now();
+  // Return the duration of the current session so far.
+  TimeDelta TimeSinceStart();
+
+  std::unique_ptr<RtcEventLogOutput> GetLogWriter(absl::string_view name) {
+    if (!log_writer_factory_ || name.empty())
+      return nullptr;
+    return log_writer_factory_->Create(name);
+  }
+  std::unique_ptr<LogWriterFactoryInterface> GetLogWriterFactory(
+      absl::string_view name) {
+    if (!log_writer_factory_ || name.empty())
+      return nullptr;
+    return std::make_unique<LogWriterFactoryAddPrefix>(
+        log_writer_factory_.get(), name);
+  }
+
+ private:
+  TimeDelta TimeUntilTarget(TimeDelta target_time_offset);
+
+  const std::unique_ptr<LogWriterFactoryInterface> log_writer_factory_;
+  NetworkEmulationManagerImpl network_manager_;
+  Clock* clock_;
+
+  std::vector<std::unique_ptr<CallClient>> clients_;
+  std::vector<std::unique_ptr<CallClientPair>> client_pairs_;
+  std::vector<std::unique_ptr<VideoStreamPair>> video_streams_;
+  std::vector<std::unique_ptr<AudioStreamPair>> audio_streams_;
+  std::vector<std::unique_ptr<SimulationNode>> simulation_nodes_;
+  std::vector<std::unique_ptr<StatesPrinter>> printers_;
+
+  rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
+  rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
+
+  Timestamp start_time_ = Timestamp::PlusInfinity();
+  // Defined last so it's destroyed first.
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCENARIO_SCENARIO_H_
diff --git a/third_party/libwebrtc/test/scenario/scenario_config.cc b/third_party/libwebrtc/test/scenario/scenario_config.cc
new file mode 100644
index 0000000000..3f8a70a162
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/scenario_config.cc
@@ -0,0 +1,47 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/scenario_config.h"
+
+namespace webrtc {
+namespace test {
+
+TransportControllerConfig::Rates::Rates() = default;
+TransportControllerConfig::Rates::Rates(
+    const TransportControllerConfig::Rates&) = default;
+TransportControllerConfig::Rates::~Rates() = default;
+
+PacketStreamConfig::PacketStreamConfig() = default;
+PacketStreamConfig::PacketStreamConfig(const PacketStreamConfig&) = default;
+PacketStreamConfig::~PacketStreamConfig() = default;
+
+VideoStreamConfig::Encoder::Encoder() = default;
+VideoStreamConfig::Encoder::Encoder(const VideoStreamConfig::Encoder&) =
+    default;
+VideoStreamConfig::Encoder::~Encoder() = default;
+
+VideoStreamConfig::Stream::Stream() = default;
+VideoStreamConfig::Stream::Stream(const VideoStreamConfig::Stream&) = default;
+VideoStreamConfig::Stream::~Stream() = default;
+
+AudioStreamConfig::AudioStreamConfig() = default;
+AudioStreamConfig::AudioStreamConfig(const AudioStreamConfig&) = default;
+AudioStreamConfig::~AudioStreamConfig() = default;
+
+AudioStreamConfig::Encoder::Encoder() = default;
+AudioStreamConfig::Encoder::Encoder(const AudioStreamConfig::Encoder&) =
+    default;
+AudioStreamConfig::Encoder::~Encoder() = default;
+
+AudioStreamConfig::Stream::Stream() = default;
+AudioStreamConfig::Stream::Stream(const AudioStreamConfig::Stream&) = default;
+AudioStreamConfig::Stream::~Stream() = default;
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/scenario_config.h b/third_party/libwebrtc/test/scenario/scenario_config.h
new file mode 100644
index 0000000000..9ce99401d7
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/scenario_config.h
@@ -0,0 +1,231 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_SCENARIO_CONFIG_H_
+#define TEST_SCENARIO_SCENARIO_CONFIG_H_
+
+#include <stddef.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/fec_controller.h"
+#include "api/rtp_parameters.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/transport/network_control.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/video/video_codec_type.h"
+#include "api/video_codecs/scalability_mode.h"
+#include "test/scenario/performance_stats.h"
+
+namespace webrtc {
+namespace test {
+struct PacketOverhead {
+  static constexpr size_t kSrtp = 10;
+  static constexpr size_t kStun = 4;
+  // TURN messages can be sent either with or without an establieshed channel.
+  // In the latter case, a TURN Send/Data Indication is sent which has
+  // significantly more overhead.
+  static constexpr size_t kTurnChannelMessage = 4;
+  static constexpr size_t kTurnIndicationMessage = 36;
+  static constexpr size_t kDefault = kSrtp;
+};
+struct TransportControllerConfig {
+  struct Rates {
+    Rates();
+    Rates(const Rates&);
+    ~Rates();
+    DataRate min_rate = DataRate::KilobitsPerSec(30);
+    DataRate max_rate = DataRate::KilobitsPerSec(3000);
+    DataRate start_rate = DataRate::KilobitsPerSec(300);
+  } rates;
+  NetworkControllerFactoryInterface* cc_factory = nullptr;
+  TimeDelta state_log_interval = TimeDelta::Millis(100);
+};
+
+struct CallClientConfig {
+  TransportControllerConfig transport;
+  // Allows the pacer to send out multiple packets in a burst.
+  // The number of bites that can be sent in one burst is pacer_burst_interval *
+  // current bwe. 40ms is the default Chrome setting.
+  TimeDelta pacer_burst_interval = TimeDelta::Millis(40);
+  const FieldTrialsView* field_trials = nullptr;
+};
+
+struct PacketStreamConfig {
+  PacketStreamConfig();
+  PacketStreamConfig(const PacketStreamConfig&);
+  ~PacketStreamConfig();
+  int frame_rate = 30;
+  DataRate max_data_rate = DataRate::Infinity();
+  DataSize max_packet_size = DataSize::Bytes(1400);
+  DataSize min_frame_size = DataSize::Bytes(100);
+  double keyframe_multiplier = 1;
+  DataSize packet_overhead = DataSize::Bytes(PacketOverhead::kDefault);
+};
+
+struct VideoStreamConfig {
+  bool autostart = true;
+  struct Source {
+    enum Capture {
+      kGenerator,
+      kVideoFile,
+      kGenerateSlides,
+      kImageSlides,
+      // Support for explicit frame triggers should be added here if needed.
+    } capture = Capture::kGenerator;
+    struct Slides {
+      TimeDelta change_interval = TimeDelta::Seconds(10);
+      struct Generator {
+        int width = 1600;
+        int height = 1200;
+      } generator;
+      struct Images {
+        struct Crop {
+          TimeDelta scroll_duration = TimeDelta::Seconds(0);
+          absl::optional<int> width;
+          absl::optional<int> height;
+        } crop;
+        int width = 1850;
+        int height = 1110;
+        std::vector<std::string> paths = {
+            "web_screenshot_1850_1110",
+            "presentation_1850_1110",
+            "photo_1850_1110",
+            "difficult_photo_1850_1110",
+        };
+      } images;
+    } slides;
+    struct Generator {
+      using PixelFormat = FrameGeneratorInterface::OutputType;
+      PixelFormat pixel_format = PixelFormat::kI420;
+      int width = 320;
+      int height = 180;
+    } generator;
+    struct VideoFile {
+      std::string name;
+      // Must be set to width and height of the source video file.
+      int width = 0;
+      int height = 0;
+    } video_file;
+    int framerate = 30;
+  } source;
+  struct Encoder {
+    Encoder();
+    Encoder(const Encoder&);
+    ~Encoder();
+    enum class ContentType {
+      kVideo,
+      kScreen,
+    } content_type = ContentType::kVideo;
+    enum Implementation { kFake, kSoftware, kHardware } implementation = kFake;
+    struct Fake {
+      DataRate max_rate = DataRate::Infinity();
+    } fake;
+
+    using Codec = VideoCodecType;
+    Codec codec = Codec::kVideoCodecGeneric;
+    absl::optional<DataRate> max_data_rate;
+    absl::optional<DataRate> min_data_rate;
+    absl::optional<int> max_framerate;
+    // Counted in frame count.
+    absl::optional<int> key_frame_interval = 3000;
+    bool frame_dropping = true;
+    struct SingleLayer {
+      bool denoising = true;
+      bool automatic_scaling = true;
+    } single;
+    std::vector<webrtc::ScalabilityMode> simulcast_streams = {
+        webrtc::ScalabilityMode::kL1T1};
+
+    DegradationPreference degradation_preference =
+        DegradationPreference::MAINTAIN_FRAMERATE;
+    bool suspend_below_min_bitrate = false;
+  } encoder;
+  struct Stream {
+    Stream();
+    Stream(const Stream&);
+    ~Stream();
+    bool abs_send_time = false;
+    bool packet_feedback = true;
+    bool use_rtx = true;
+    DataRate pad_to_rate = DataRate::Zero();
+    TimeDelta nack_history_time = TimeDelta::Millis(1000);
+    bool use_flexfec = false;
+    bool use_ulpfec = false;
+    FecControllerFactoryInterface* fec_controller_factory = nullptr;
+  } stream;
+  struct Rendering {
+    enum Type { kFake } type = kFake;
+    std::string sync_group;
+  } render;
+  struct Hooks {
+    std::vector<std::function<void(const VideoFramePair&)>> frame_pair_handlers;
+  } hooks;
+};
+
+struct AudioStreamConfig {
+  AudioStreamConfig();
+  AudioStreamConfig(const AudioStreamConfig&);
+  ~AudioStreamConfig();
+  bool autostart = true;
+  struct Source {
+    int channels = 1;
+  } source;
+  bool network_adaptation = false;
+  struct NetworkAdaptation {
+    struct FrameLength {
+      double min_packet_loss_for_decrease = 0;
+      double max_packet_loss_for_increase = 1;
+      DataRate min_rate_for_20_ms = DataRate::Zero();
+      DataRate max_rate_for_60_ms = DataRate::Infinity();
+      DataRate min_rate_for_60_ms = DataRate::Zero();
+      DataRate max_rate_for_120_ms = DataRate::Infinity();
+    } frame;
+    std::string binary_proto;
+  } adapt;
+  struct Encoder {
+    Encoder();
+    Encoder(const Encoder&);
+    ~Encoder();
+    bool allocate_bitrate = false;
+    bool enable_dtx = false;
+    DataRate fixed_rate = DataRate::KilobitsPerSec(32);
+    // Overrides fixed rate.
+    absl::optional<DataRate> min_rate;
+    absl::optional<DataRate> max_rate;
+    TimeDelta initial_frame_length = TimeDelta::Millis(20);
+  } encoder;
+  struct Stream {
+    Stream();
+    Stream(const Stream&);
+    ~Stream();
+    bool abs_send_time = true;
+    bool in_bandwidth_estimation = true;
+  } stream;
+  struct Rendering {
+    std::string sync_group;
+  } render;
+};
+
+// TODO(srte): Merge this with BuiltInNetworkBehaviorConfig.
+struct NetworkSimulationConfig {
+  DataRate bandwidth = DataRate::Infinity();
+  TimeDelta delay = TimeDelta::Zero();
+  TimeDelta delay_std_dev = TimeDelta::Zero();
+  double loss_rate = 0;
+  absl::optional<int> packet_queue_length_limit;
+  DataSize packet_overhead = DataSize::Zero();
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCENARIO_SCENARIO_CONFIG_H_
diff --git a/third_party/libwebrtc/test/scenario/scenario_unittest.cc b/third_party/libwebrtc/test/scenario/scenario_unittest.cc
new file mode 100644
index 0000000000..6861151a2d
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/scenario_unittest.cc
@@ -0,0 +1,196 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/scenario.h"
+
+#include <atomic>
+
+#include "api/test/network_emulation/create_cross_traffic.h"
+#include "api/test/network_emulation/cross_traffic.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/logging/memory_log_writer.h"
+#include "test/scenario/stats_collection.h"
+
+namespace webrtc {
+namespace test {
+TEST(ScenarioTest, StartsAndStopsWithoutErrors) {
+  std::atomic<bool> packet_received(false);
+  std::atomic<bool> bitrate_changed(false);
+  Scenario s;
+  CallClientConfig call_client_config;
+  call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto* alice = s.CreateClient("alice", call_client_config);
+  auto* bob = s.CreateClient("bob", call_client_config);
+  NetworkSimulationConfig network_config;
+  auto alice_net = s.CreateSimulationNode(network_config);
+  auto bob_net = s.CreateSimulationNode(network_config);
+  auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
+
+  VideoStreamConfig video_stream_config;
+  s.CreateVideoStream(route->forward(), video_stream_config);
+  s.CreateVideoStream(route->reverse(), video_stream_config);
+
+  AudioStreamConfig audio_stream_config;
+  audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6);
+  audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64);
+  audio_stream_config.encoder.allocate_bitrate = true;
+  audio_stream_config.stream.in_bandwidth_estimation = false;
+  s.CreateAudioStream(route->forward(), audio_stream_config);
+  s.CreateAudioStream(route->reverse(), audio_stream_config);
+
+  RandomWalkConfig cross_traffic_config;
+  s.net()->StartCrossTraffic(CreateRandomWalkCrossTraffic(
+      s.net()->CreateCrossTrafficRoute({alice_net}), cross_traffic_config));
+
+  s.NetworkDelayedAction({alice_net, bob_net}, 100,
+                         [&packet_received] { packet_received = true; });
+  s.Every(TimeDelta::Millis(10), [alice, bob, &bitrate_changed] {
+    if (alice->GetStats().send_bandwidth_bps != 300000 &&
+        bob->GetStats().send_bandwidth_bps != 300000)
+      bitrate_changed = true;
+  });
+  s.RunUntil(TimeDelta::Seconds(2), TimeDelta::Millis(5),
+             [&bitrate_changed, &packet_received] {
+               return packet_received && bitrate_changed;
+             });
+  EXPECT_TRUE(packet_received);
+  EXPECT_TRUE(bitrate_changed);
+}
+namespace {
+void SetupVideoCall(Scenario& s, VideoQualityAnalyzer* analyzer) {
+  CallClientConfig call_config;
+  auto* alice = s.CreateClient("alice", call_config);
+  auto* bob = s.CreateClient("bob", call_config);
+  NetworkSimulationConfig network_config;
+  network_config.bandwidth = DataRate::KilobitsPerSec(1000);
+  network_config.delay = TimeDelta::Millis(50);
+  auto alice_net = s.CreateSimulationNode(network_config);
+  auto bob_net = s.CreateSimulationNode(network_config);
+  auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
+  VideoStreamConfig video;
+  if (analyzer) {
+    video.source.capture = VideoStreamConfig::Source::Capture::kVideoFile;
+    video.source.video_file.name = "foreman_cif";
+    video.source.video_file.width = 352;
+    video.source.video_file.height = 288;
+    video.source.framerate = 30;
+    video.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+    video.encoder.implementation =
+        VideoStreamConfig::Encoder::Implementation::kSoftware;
+    video.hooks.frame_pair_handlers = {analyzer->Handler()};
+  }
+  s.CreateVideoStream(route->forward(), video);
+  s.CreateAudioStream(route->forward(), AudioStreamConfig());
+}
+}  // namespace
+
+TEST(ScenarioTest, SimTimeEncoding) {
+  VideoQualityAnalyzerConfig analyzer_config;
+  analyzer_config.psnr_coverage = 0.1;
+  VideoQualityAnalyzer analyzer(analyzer_config);
+  {
+    Scenario s("scenario/encode_sim", false);
+    SetupVideoCall(s, &analyzer);
+    s.RunFor(TimeDelta::Seconds(2));
+  }
+  // Regression tests based on previous runs.
+  EXPECT_EQ(analyzer.stats().lost_count, 0);
+  EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 38, 5);
+}
+
+// TODO(bugs.webrtc.org/10515): Remove this when performance has been improved.
+#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
+#define MAYBE_RealTimeEncoding DISABLED_RealTimeEncoding
+#else
+#define MAYBE_RealTimeEncoding RealTimeEncoding
+#endif
+TEST(ScenarioTest, MAYBE_RealTimeEncoding) {
+  VideoQualityAnalyzerConfig analyzer_config;
+  analyzer_config.psnr_coverage = 0.1;
+  VideoQualityAnalyzer analyzer(analyzer_config);
+  {
+    Scenario s("scenario/encode_real", true);
+    SetupVideoCall(s, &analyzer);
+    s.RunFor(TimeDelta::Seconds(2));
+  }
+  // Regression tests based on previous runs.
+  EXPECT_LT(analyzer.stats().lost_count, 2);
+  // This far below expected but ensures that we get something.
+  EXPECT_GT(analyzer.stats().psnr_with_freeze.Mean(), 10);
+}
+
+TEST(ScenarioTest, SimTimeFakeing) {
+  Scenario s("scenario/encode_sim", false);
+  SetupVideoCall(s, nullptr);
+  s.RunFor(TimeDelta::Seconds(2));
+}
+
+TEST(ScenarioTest, WritesToRtcEventLog) {
+  MemoryLogStorage storage;
+  {
+    Scenario s(storage.CreateFactory(), false);
+    SetupVideoCall(s, nullptr);
+    s.RunFor(TimeDelta::Seconds(1));
+  }
+  auto logs = storage.logs();
+  // We expect that a rtc event log has been created and that it has some data.
+  EXPECT_GE(storage.logs().at("alice.rtc.dat").size(), 1u);
+}
+
+TEST(ScenarioTest,
+     RetransmitsVideoPacketsInAudioAndVideoCallWithSendSideBweAndLoss) {
+  // Make sure audio packets are included in transport feedback.
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-ABWENoTWCC/Disabled/");
+
+  Scenario s;
+  CallClientConfig call_client_config;
+  call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto* alice = s.CreateClient("alice", call_client_config);
+  auto* bob = s.CreateClient("bob", call_client_config);
+  NetworkSimulationConfig network_config;
+  // Add some loss and delay.
+  network_config.delay = TimeDelta::Millis(200);
+  network_config.loss_rate = 0.05;
+  auto alice_net = s.CreateSimulationNode(network_config);
+  auto bob_net = s.CreateSimulationNode(network_config);
+  auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
+
+  // First add an audio stream, then a video stream.
+  // Needed to make sure audio RTP module is selected first when sending
+  // transport feedback message.
+  AudioStreamConfig audio_stream_config;
+  audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6);
+  audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64);
+  audio_stream_config.encoder.allocate_bitrate = true;
+  audio_stream_config.stream.in_bandwidth_estimation = true;
+  s.CreateAudioStream(route->forward(), audio_stream_config);
+  s.CreateAudioStream(route->reverse(), audio_stream_config);
+
+  VideoStreamConfig video_stream_config;
+  auto video = s.CreateVideoStream(route->forward(), video_stream_config);
+  s.CreateVideoStream(route->reverse(), video_stream_config);
+
+  // Run for 10 seconds.
+  s.RunFor(TimeDelta::Seconds(10));
+  // Make sure retransmissions have happened.
+  int retransmit_packets = 0;
+
+  VideoSendStream::Stats stats;
+  alice->SendTask([&]() { stats = video->send()->GetStats(); });
+
+  for (const auto& substream : stats.substreams) {
+    retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
+  }
+  EXPECT_GT(retransmit_packets, 0);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/stats_collection.cc b/third_party/libwebrtc/test/scenario/stats_collection.cc
new file mode 100644
index 0000000000..e32696de71
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/stats_collection.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/scenario/stats_collection.h"
+
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/memory_usage.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace test {
+
+VideoQualityAnalyzer::VideoQualityAnalyzer(
+    VideoQualityAnalyzerConfig config,
+    std::unique_ptr<RtcEventLogOutput> writer)
+    : config_(config), writer_(std::move(writer)) {
+  if (writer_) {
+    PrintHeaders();
+  }
+}
+
+VideoQualityAnalyzer::~VideoQualityAnalyzer() = default;
+
+void VideoQualityAnalyzer::PrintHeaders() {
+  writer_->Write(
+      "capture_time render_time capture_width capture_height render_width "
+      "render_height psnr\n");
+}
+
+std::function<void(const VideoFramePair&)> VideoQualityAnalyzer::Handler() {
+  return [this](VideoFramePair pair) { HandleFramePair(pair); };
+}
+
+void VideoQualityAnalyzer::HandleFramePair(VideoFramePair sample, double psnr) {
+  layer_analyzers_[sample.layer_id].HandleFramePair(sample, psnr,
+                                                    writer_.get());
+  cached_.reset();
+}
+
+void VideoQualityAnalyzer::HandleFramePair(VideoFramePair sample) {
+  double psnr = NAN;
+  if (sample.decoded)
+    psnr = I420PSNR(*sample.captured->ToI420(), *sample.decoded->ToI420());
+
+  if (config_.thread) {
+    config_.thread->PostTask(
+        [this, sample, psnr] { HandleFramePair(std::move(sample), psnr); });
+  } else {
+    HandleFramePair(std::move(sample), psnr);
+  }
+}
+
+std::vector<VideoQualityStats> VideoQualityAnalyzer::layer_stats() const {
+  std::vector<VideoQualityStats> res;
+  for (auto& layer : layer_analyzers_)
+    res.push_back(layer.second.stats_);
+  return res;
+}
+
+VideoQualityStats& VideoQualityAnalyzer::stats() {
+  if (!cached_) {
+    cached_ = VideoQualityStats();
+    for (auto& layer : layer_analyzers_)
+      cached_->AddStats(layer.second.stats_);
+  }
+  return *cached_;
+}
+
+void VideoLayerAnalyzer::HandleFramePair(VideoFramePair sample,
+                                         double psnr,
+                                         RtcEventLogOutput* writer) {
+  RTC_CHECK(sample.captured);
+  HandleCapturedFrame(sample);
+  if (!sample.decoded) {
+    // Can only happen in the beginning of a call or if the resolution is
+    // reduced. Otherwise we will detect a freeze.
+    ++stats_.lost_count;
+    ++skip_count_;
+  } else {
+    stats_.psnr_with_freeze.AddSample(psnr);
+    if (sample.repeated) {
+      ++stats_.freeze_count;
+      ++skip_count_;
+    } else {
+      stats_.psnr.AddSample(psnr);
+      HandleRenderedFrame(sample);
+    }
+  }
+  if (writer) {
+    LogWriteFormat(writer, "%.3f %.3f %.3f %i %i %i %i %.3f\n",
+                   sample.capture_time.seconds<double>(),
+                   sample.render_time.seconds<double>(),
+                   sample.captured->width(), sample.captured->height(),
+                   sample.decoded ? sample.decoded->width() : 0,
+                   sample.decoded ? sample.decoded->height() : 0, psnr);
+  }
+}
+
+void VideoLayerAnalyzer::HandleCapturedFrame(const VideoFramePair& sample) {
+  stats_.capture.AddFrameInfo(*sample.captured, sample.capture_time);
+  if (last_freeze_time_.IsInfinite())
+    last_freeze_time_ = sample.capture_time;
+}
+
+void VideoLayerAnalyzer::HandleRenderedFrame(const VideoFramePair& sample) {
+  stats_.capture_to_decoded_delay.AddSample(sample.decoded_time -
+                                            sample.capture_time);
+  stats_.end_to_end_delay.AddSample(sample.render_time - sample.capture_time);
+  stats_.render.AddFrameInfo(*sample.decoded, sample.render_time);
+  stats_.skipped_between_rendered.AddSample(skip_count_);
+  skip_count_ = 0;
+
+  if (last_render_time_.IsFinite()) {
+    RTC_DCHECK(sample.render_time.IsFinite());
+    TimeDelta render_interval = sample.render_time - last_render_time_;
+    TimeDelta mean_interval = stats_.render.frames.interval().Mean();
+    if (render_interval > TimeDelta::Millis(150) + mean_interval ||
+        render_interval > 3 * mean_interval) {
+      stats_.freeze_duration.AddSample(render_interval);
+      stats_.time_between_freezes.AddSample(last_render_time_ -
+                                            last_freeze_time_);
+      last_freeze_time_ = sample.render_time;
+    }
+  }
+  last_render_time_ = sample.render_time;
+}
+
+void CallStatsCollector::AddStats(Call::Stats sample) {
+  if (sample.send_bandwidth_bps > 0)
+    stats_.target_rate.AddSampleBps(sample.send_bandwidth_bps);
+  if (sample.pacer_delay_ms > 0)
+    stats_.pacer_delay.AddSample(TimeDelta::Millis(sample.pacer_delay_ms));
+  if (sample.rtt_ms > 0)
+    stats_.round_trip_time.AddSample(TimeDelta::Millis(sample.rtt_ms));
+  stats_.memory_usage.AddSample(rtc::GetProcessResidentSizeBytes());
+}
+
+void AudioReceiveStatsCollector::AddStats(
+    AudioReceiveStreamInterface::Stats sample) {
+  stats_.expand_rate.AddSample(sample.expand_rate);
+  stats_.accelerate_rate.AddSample(sample.accelerate_rate);
+  stats_.jitter_buffer.AddSampleMs(sample.jitter_buffer_ms);
+}
+
+void VideoSendStatsCollector::AddStats(VideoSendStream::Stats sample,
+                                       Timestamp at_time) {
+  // It's not certain that we yet have estimates for any of these stats.
+  // Check that they are positive before mixing them in.
+  if (sample.encode_frame_rate <= 0)
+    return;
+
+  stats_.encode_frame_rate.AddSample(sample.encode_frame_rate);
+  stats_.encode_time.AddSampleMs(sample.avg_encode_time_ms);
+  stats_.encode_usage.AddSample(sample.encode_usage_percent / 100.0);
+  stats_.media_bitrate.AddSampleBps(sample.media_bitrate_bps);
+
+  size_t fec_bytes = 0;
+  for (const auto& kv : sample.substreams) {
+    fec_bytes += kv.second.rtp_stats.fec.payload_bytes +
+                 kv.second.rtp_stats.fec.padding_bytes;
+  }
+  if (last_update_.IsFinite()) {
+    auto fec_delta = DataSize::Bytes(fec_bytes - last_fec_bytes_);
+    auto time_delta = at_time - last_update_;
+    stats_.fec_bitrate.AddSample(fec_delta / time_delta);
+  }
+  last_fec_bytes_ = fec_bytes;
+  last_update_ = at_time;
+}
+
+void VideoReceiveStatsCollector::AddStats(
+    VideoReceiveStreamInterface::Stats sample) {
+  if (sample.decode_ms > 0)
+    stats_.decode_time.AddSampleMs(sample.decode_ms);
+  if (sample.max_decode_ms > 0)
+    stats_.decode_time_max.AddSampleMs(sample.max_decode_ms);
+  if (sample.width > 0 && sample.height > 0) {
+    stats_.decode_pixels.AddSample(sample.width * sample.height);
+    stats_.resolution.AddSample(sample.height);
+  }
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/stats_collection.h b/third_party/libwebrtc/test/scenario/stats_collection.h
new file mode 100644
index 0000000000..1f5d8daea7
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/stats_collection.h
@@ -0,0 +1,110 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_STATS_COLLECTION_H_
+#define TEST_SCENARIO_STATS_COLLECTION_H_
+
+#include <map>
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "call/call.h"
+#include "rtc_base/thread.h"
+#include "test/logging/log_writer.h"
+#include "test/scenario/performance_stats.h"
+
+namespace webrtc {
+namespace test {
+
+struct VideoQualityAnalyzerConfig {
+  double psnr_coverage = 1;
+  rtc::Thread* thread = nullptr;
+};
+
+class VideoLayerAnalyzer {
+ public:
+  void HandleCapturedFrame(const VideoFramePair& sample);
+  void HandleRenderedFrame(const VideoFramePair& sample);
+  void HandleFramePair(VideoFramePair sample,
+                       double psnr,
+                       RtcEventLogOutput* writer);
+  VideoQualityStats stats_;
+  Timestamp last_capture_time_ = Timestamp::MinusInfinity();
+  Timestamp last_render_time_ = Timestamp::MinusInfinity();
+  Timestamp last_freeze_time_ = Timestamp::MinusInfinity();
+  int skip_count_ = 0;
+};
+
+class VideoQualityAnalyzer {
+ public:
+  explicit VideoQualityAnalyzer(
+      VideoQualityAnalyzerConfig config = VideoQualityAnalyzerConfig(),
+      std::unique_ptr<RtcEventLogOutput> writer = nullptr);
+  ~VideoQualityAnalyzer();
+  void HandleFramePair(VideoFramePair sample);
+  std::vector<VideoQualityStats> layer_stats() const;
+  VideoQualityStats& stats();
+  void PrintHeaders();
+  void PrintFrameInfo(const VideoFramePair& sample);
+  std::function<void(const VideoFramePair&)> Handler();
+
+ private:
+  void HandleFramePair(VideoFramePair sample, double psnr);
+  const VideoQualityAnalyzerConfig config_;
+  std::map<int, VideoLayerAnalyzer> layer_analyzers_;
+  const std::unique_ptr<RtcEventLogOutput> writer_;
+  absl::optional<VideoQualityStats> cached_;
+};
+
+class CallStatsCollector {
+ public:
+  void AddStats(Call::Stats sample);
+  CollectedCallStats& stats() { return stats_; }
+
+ private:
+  CollectedCallStats stats_;
+};
+class AudioReceiveStatsCollector {
+ public:
+  void AddStats(AudioReceiveStreamInterface::Stats sample);
+  CollectedAudioReceiveStats& stats() { return stats_; }
+
+ private:
+  CollectedAudioReceiveStats stats_;
+};
+class VideoSendStatsCollector {
+ public:
+  void AddStats(VideoSendStream::Stats sample, Timestamp at_time);
+  CollectedVideoSendStats& stats() { return stats_; }
+
+ private:
+  CollectedVideoSendStats stats_;
+  Timestamp last_update_ = Timestamp::MinusInfinity();
+  size_t last_fec_bytes_ = 0;
+};
+class VideoReceiveStatsCollector {
+ public:
+  void AddStats(VideoReceiveStreamInterface::Stats sample);
+  CollectedVideoReceiveStats& stats() { return stats_; }
+
+ private:
+  CollectedVideoReceiveStats stats_;
+};
+
+struct CallStatsCollectors {
+  CallStatsCollector call;
+  AudioReceiveStatsCollector audio_receive;
+  VideoSendStatsCollector video_send;
+  VideoReceiveStatsCollector video_receive;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCENARIO_STATS_COLLECTION_H_
diff --git a/third_party/libwebrtc/test/scenario/stats_collection_unittest.cc b/third_party/libwebrtc/test/scenario/stats_collection_unittest.cc
new file mode 100644
index 0000000000..9f46f10073
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/stats_collection_unittest.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/stats_collection.h"
+
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+void CreateAnalyzedStream(Scenario* s,
+                          NetworkSimulationConfig network_config,
+                          VideoQualityAnalyzer* analyzer,
+                          CallStatsCollectors* collectors) {
+  VideoStreamConfig config;
+  config.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+  config.encoder.implementation =
+      VideoStreamConfig::Encoder::Implementation::kSoftware;
+  config.hooks.frame_pair_handlers = {analyzer->Handler()};
+  auto* caller = s->CreateClient("caller", CallClientConfig());
+  auto* callee = s->CreateClient("callee", CallClientConfig());
+  auto route =
+      s->CreateRoutes(caller, {s->CreateSimulationNode(network_config)}, callee,
+                      {s->CreateSimulationNode(NetworkSimulationConfig())});
+  VideoStreamPair* video = s->CreateVideoStream(route->forward(), config);
+  auto* audio = s->CreateAudioStream(route->forward(), AudioStreamConfig());
+  s->Every(TimeDelta::Seconds(1), [=] {
+    collectors->call.AddStats(caller->GetStats());
+
+    VideoSendStream::Stats send_stats;
+    caller->SendTask([&]() { send_stats = video->send()->GetStats(); });
+    collectors->video_send.AddStats(send_stats, s->Now());
+
+    AudioReceiveStreamInterface::Stats receive_stats;
+    caller->SendTask([&]() { receive_stats = audio->receive()->GetStats(); });
+    collectors->audio_receive.AddStats(receive_stats);
+
+    // Querying the video stats from within the expected runtime environment
+    // (i.e. the TQ that belongs to the CallClient, not the Scenario TQ that
+    // we're currently on).
+    VideoReceiveStreamInterface::Stats video_receive_stats;
+    auto* video_stream = video->receive();
+    callee->SendTask([&video_stream, &video_receive_stats]() {
+      video_receive_stats = video_stream->GetStats();
+    });
+    collectors->video_receive.AddStats(video_receive_stats);
+  });
+}
+}  // namespace
+
+TEST(ScenarioAnalyzerTest, PsnrIsHighWhenNetworkIsGood) {
+  VideoQualityAnalyzer analyzer;
+  CallStatsCollectors stats;
+  {
+    Scenario s;
+    NetworkSimulationConfig good_network;
+    good_network.bandwidth = DataRate::KilobitsPerSec(1000);
+    CreateAnalyzedStream(&s, good_network, &analyzer, &stats);
+    s.RunFor(TimeDelta::Seconds(3));
+  }
+  // This is a change detecting test, the targets are based on previous runs and
+  // might change due to changes in configuration and encoder etc. The main
+  // purpose is to show how the stats can be used. To avoid being overly
+  // sensistive to change, the ranges are chosen to be quite large.
+  EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 43, 10);
+  EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 700, 300);
+  EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 500, 200);
+  EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
+  EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 40, 20);
+}
+
+TEST(ScenarioAnalyzerTest, PsnrIsLowWhenNetworkIsBad) {
+  VideoQualityAnalyzer analyzer;
+  CallStatsCollectors stats;
+  {
+    Scenario s;
+    NetworkSimulationConfig bad_network;
+    bad_network.bandwidth = DataRate::KilobitsPerSec(100);
+    bad_network.loss_rate = 0.02;
+    CreateAnalyzedStream(&s, bad_network, &analyzer, &stats);
+    s.RunFor(TimeDelta::Seconds(3));
+  }
+  // This is a change detecting test, the targets are based on previous runs and
+  // might change due to changes in configuration and encoder etc.
+  EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 20, 10);
+  EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 75, 50);
+  EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 70, 30);
+  EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
+  EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 250, 200);
+}
+
+TEST(ScenarioAnalyzerTest, CountsCapturedButNotRendered) {
+  VideoQualityAnalyzer analyzer;
+  CallStatsCollectors stats;
+  {
+    Scenario s;
+    NetworkSimulationConfig long_delays;
+    long_delays.delay = TimeDelta::Seconds(5);
+    CreateAnalyzedStream(&s, long_delays, &analyzer, &stats);
+    // Enough time to send frames but not enough to deliver.
+    s.RunFor(TimeDelta::Millis(100));
+  }
+  EXPECT_GE(analyzer.stats().capture.count, 1);
+  EXPECT_EQ(analyzer.stats().render.count, 0);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/video_frame_matcher.cc b/third_party/libwebrtc/test/scenario/video_frame_matcher.cc
new file mode 100644
index 0000000000..dc8cd59756
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/video_frame_matcher.cc
@@ -0,0 +1,188 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/video_frame_matcher.h"
+
+#include <utility>
+
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+constexpr int kThumbWidth = 96;
+constexpr int kThumbHeight = 96;
+}  // namespace
+
+VideoFrameMatcher::VideoFrameMatcher(
+    std::vector<std::function<void(const VideoFramePair&)> >
+        frame_pair_handlers)
+    : frame_pair_handlers_(std::move(frame_pair_handlers)),
+      task_queue_("VideoAnalyzer") {}
+
+VideoFrameMatcher::~VideoFrameMatcher() {
+  task_queue_.SendTask([this] { Finalize(); });
+}
+
+void VideoFrameMatcher::RegisterLayer(int layer_id) {
+  task_queue_.PostTask([this, layer_id] { layers_[layer_id] = VideoLayer(); });
+}
+
+void VideoFrameMatcher::OnCapturedFrame(const VideoFrame& frame,
+                                        Timestamp at_time) {
+  CapturedFrame captured;
+  captured.id = next_capture_id_++;
+  captured.capture_time = at_time;
+  captured.frame = frame.video_frame_buffer();
+  captured.thumb = ScaleVideoFrameBuffer(*frame.video_frame_buffer()->ToI420(),
+                                         kThumbWidth, kThumbHeight),
+  task_queue_.PostTask([this, captured]() {
+    for (auto& layer : layers_) {
+      CapturedFrame copy = captured;
+      if (layer.second.last_decode &&
+          layer.second.last_decode->frame->width() <= captured.frame->width()) {
+        copy.best_score = I420SSE(*captured.thumb->GetI420(),
+                                  *layer.second.last_decode->thumb->GetI420());
+        copy.best_decode = layer.second.last_decode;
+      }
+      layer.second.captured_frames.push_back(std::move(copy));
+    }
+  });
+}
+
+void VideoFrameMatcher::OnDecodedFrame(const VideoFrame& frame,
+                                       int layer_id,
+                                       Timestamp render_time,
+                                       Timestamp at_time) {
+  rtc::scoped_refptr<DecodedFrame> decoded(new DecodedFrame{});
+  decoded->decoded_time = at_time;
+  decoded->render_time = render_time;
+  decoded->frame = frame.video_frame_buffer();
+  decoded->thumb = ScaleVideoFrameBuffer(*frame.video_frame_buffer()->ToI420(),
+                                         kThumbWidth, kThumbHeight);
+
+  task_queue_.PostTask([this, decoded, layer_id] {
+    auto& layer = layers_[layer_id];
+    decoded->id = layer.next_decoded_id++;
+    layer.last_decode = decoded;
+    for (auto& captured : layer.captured_frames) {
+      // We can't match with a smaller capture.
+      if (captured.frame->width() < decoded->frame->width()) {
+        captured.matched = true;
+        continue;
+      }
+      double score =
+          I420SSE(*captured.thumb->GetI420(), *decoded->thumb->GetI420());
+      if (score < captured.best_score) {
+        captured.best_score = score;
+        captured.best_decode = decoded;
+        captured.matched = false;
+      } else {
+        captured.matched = true;
+      }
+    }
+    while (!layer.captured_frames.empty() &&
+           layer.captured_frames.front().matched) {
+      HandleMatch(std::move(layer.captured_frames.front()), layer_id);
+      layer.captured_frames.pop_front();
+    }
+  });
+}
+
+bool VideoFrameMatcher::Active() const {
+  return !frame_pair_handlers_.empty();
+}
+
+void VideoFrameMatcher::HandleMatch(VideoFrameMatcher::CapturedFrame captured,
+                                    int layer_id) {
+  VideoFramePair frame_pair;
+  frame_pair.layer_id = layer_id;
+  frame_pair.captured = captured.frame;
+  frame_pair.capture_id = captured.id;
+  frame_pair.capture_time = captured.capture_time;
+  if (captured.best_decode) {
+    frame_pair.decode_id = captured.best_decode->id;
+    frame_pair.decoded = captured.best_decode->frame;
+    frame_pair.decoded_time = captured.best_decode->decoded_time;
+    // We can't render frames before they have been decoded.
+    frame_pair.render_time = std::max(captured.best_decode->render_time,
+                                      captured.best_decode->decoded_time);
+    frame_pair.repeated = captured.best_decode->repeat_count++;
+  }
+  for (auto& handler : frame_pair_handlers_)
+    handler(frame_pair);
+}
+
+void VideoFrameMatcher::Finalize() {
+  for (auto& layer : layers_) {
+    while (!layer.second.captured_frames.empty()) {
+      HandleMatch(std::move(layer.second.captured_frames.front()), layer.first);
+      layer.second.captured_frames.pop_front();
+    }
+  }
+}
+
+CapturedFrameTap::CapturedFrameTap(Clock* clock, VideoFrameMatcher* matcher)
+    : clock_(clock), matcher_(matcher) {}
+
+void CapturedFrameTap::OnFrame(const VideoFrame& frame) {
+  matcher_->OnCapturedFrame(frame, clock_->CurrentTime());
+}
+void CapturedFrameTap::OnDiscardedFrame() {
+  discarded_count_++;
+}
+
+ForwardingCapturedFrameTap::ForwardingCapturedFrameTap(
+    Clock* clock,
+    VideoFrameMatcher* matcher,
+    rtc::VideoSourceInterface<VideoFrame>* source)
+    : clock_(clock), matcher_(matcher), source_(source) {}
+
+void ForwardingCapturedFrameTap::OnFrame(const VideoFrame& frame) {
+  RTC_CHECK(sink_);
+  matcher_->OnCapturedFrame(frame, clock_->CurrentTime());
+  sink_->OnFrame(frame);
+}
+void ForwardingCapturedFrameTap::OnDiscardedFrame() {
+  RTC_CHECK(sink_);
+  discarded_count_++;
+  sink_->OnDiscardedFrame();
+}
+
+void ForwardingCapturedFrameTap::AddOrUpdateSink(
+    VideoSinkInterface<VideoFrame>* sink,
+    const rtc::VideoSinkWants& wants) {
+  if (!sink_)
+    sink_ = sink;
+  RTC_DCHECK_EQ(sink_, sink);
+  source_->AddOrUpdateSink(this, wants);
+}
+void ForwardingCapturedFrameTap::RemoveSink(
+    VideoSinkInterface<VideoFrame>* sink) {
+  source_->RemoveSink(this);
+  sink_ = nullptr;
+}
+
+DecodedFrameTap::DecodedFrameTap(Clock* clock,
+                                 VideoFrameMatcher* matcher,
+                                 int layer_id)
+    : clock_(clock), matcher_(matcher), layer_id_(layer_id) {
+  matcher_->RegisterLayer(layer_id_);
+}
+
+void DecodedFrameTap::OnFrame(const VideoFrame& frame) {
+  matcher_->OnDecodedFrame(frame, layer_id_,
+                           Timestamp::Millis(frame.render_time_ms()),
+                           clock_->CurrentTime());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/video_frame_matcher.h b/third_party/libwebrtc/test/scenario/video_frame_matcher.h
new file mode 100644
index 0000000000..a3aa85447d
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/video_frame_matcher.h
@@ -0,0 +1,134 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_VIDEO_FRAME_MATCHER_H_
+#define TEST_SCENARIO_VIDEO_FRAME_MATCHER_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+#include <vector>
+
+#include "api/units/timestamp.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "system_wrappers/include/clock.h"
+#include "test/scenario/performance_stats.h"
+
+namespace webrtc {
+namespace test {
+
+class VideoFrameMatcher {
+ public:
+  explicit VideoFrameMatcher(
+      std::vector<std::function<void(const VideoFramePair&)>>
+          frame_pair_handlers);
+  ~VideoFrameMatcher();
+  void RegisterLayer(int layer_id);
+  void OnCapturedFrame(const VideoFrame& frame, Timestamp at_time);
+  void OnDecodedFrame(const VideoFrame& frame,
+                      int layer_id,
+                      Timestamp render_time,
+                      Timestamp at_time);
+  bool Active() const;
+
+ private:
+  struct DecodedFrameBase {
+    int id;
+    Timestamp decoded_time = Timestamp::PlusInfinity();
+    Timestamp render_time = Timestamp::PlusInfinity();
+    rtc::scoped_refptr<VideoFrameBuffer> frame;
+    rtc::scoped_refptr<VideoFrameBuffer> thumb;
+    int repeat_count = 0;
+  };
+  using DecodedFrame = rtc::FinalRefCountedObject<DecodedFrameBase>;
+  struct CapturedFrame {
+    int id;
+    Timestamp capture_time = Timestamp::PlusInfinity();
+    rtc::scoped_refptr<VideoFrameBuffer> frame;
+    rtc::scoped_refptr<VideoFrameBuffer> thumb;
+    double best_score = INFINITY;
+    rtc::scoped_refptr<DecodedFrame> best_decode;
+    bool matched = false;
+  };
+  struct VideoLayer {
+    int layer_id;
+    std::deque<CapturedFrame> captured_frames;
+    rtc::scoped_refptr<DecodedFrame> last_decode;
+    int next_decoded_id = 1;
+  };
+  void HandleMatch(CapturedFrame captured, int layer_id);
+  void Finalize();
+  int next_capture_id_ = 1;
+  std::vector<std::function<void(const VideoFramePair&)>> frame_pair_handlers_;
+  std::map<int, VideoLayer> layers_;
+  TaskQueueForTest task_queue_;
+};
+
+class CapturedFrameTap : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  CapturedFrameTap(Clock* clock, VideoFrameMatcher* matcher);
+  CapturedFrameTap(CapturedFrameTap&) = delete;
+  CapturedFrameTap& operator=(CapturedFrameTap&) = delete;
+
+  void OnFrame(const VideoFrame& frame) override;
+  void OnDiscardedFrame() override;
+
+ private:
+  Clock* const clock_;
+  VideoFrameMatcher* const matcher_;
+  int discarded_count_ = 0;
+};
+
+class ForwardingCapturedFrameTap
+    : public rtc::VideoSinkInterface<VideoFrame>,
+      public rtc::VideoSourceInterface<VideoFrame> {
+ public:
+  ForwardingCapturedFrameTap(Clock* clock,
+                             VideoFrameMatcher* matcher,
+                             rtc::VideoSourceInterface<VideoFrame>* source);
+  ForwardingCapturedFrameTap(ForwardingCapturedFrameTap&) = delete;
+  ForwardingCapturedFrameTap& operator=(ForwardingCapturedFrameTap&) = delete;
+
+  // VideoSinkInterface interface
+  void OnFrame(const VideoFrame& frame) override;
+  void OnDiscardedFrame() override;
+
+  // VideoSourceInterface interface
+  void AddOrUpdateSink(VideoSinkInterface<VideoFrame>* sink,
+                       const rtc::VideoSinkWants& wants) override;
+  void RemoveSink(VideoSinkInterface<VideoFrame>* sink) override;
+
+ private:
+  Clock* const clock_;
+  VideoFrameMatcher* const matcher_;
+  rtc::VideoSourceInterface<VideoFrame>* const source_;
+  VideoSinkInterface<VideoFrame>* sink_ = nullptr;
+  int discarded_count_ = 0;
+};
+
+class DecodedFrameTap : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  DecodedFrameTap(Clock* clock, VideoFrameMatcher* matcher, int layer_id);
+  // VideoSinkInterface interface
+  void OnFrame(const VideoFrame& frame) override;
+
+ private:
+  Clock* const clock_;
+  VideoFrameMatcher* const matcher_;
+  int layer_id_;
+};
+}  // namespace test
+}  // namespace webrtc
+#endif  // TEST_SCENARIO_VIDEO_FRAME_MATCHER_H_
diff --git a/third_party/libwebrtc/test/scenario/video_stream.cc b/third_party/libwebrtc/test/scenario/video_stream.cc
new file mode 100644
index 0000000000..8d627d8893
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/video_stream.cc
@@ -0,0 +1,636 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/scenario/video_stream.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/test/create_frame_generator.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/video/function_video_encoder_factory.h"
+#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "media/base/media_constants.h"
+#include "media/engine/internal_decoder_factory.h"
+#include "media/engine/internal_encoder_factory.h"
+#include "media/engine/webrtc_video_engine.h"
+#include "modules/video_coding/svc/scalability_mode_util.h"
+#include "test/call_test.h"
+#include "test/fake_encoder.h"
+#include "test/scenario/hardware_codecs.h"
+#include "test/testsupport/file_utils.h"
+#include "video/config/encoder_stream_factory.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+enum : int {  // The first valid value is 1.
+  kTransportSequenceNumberExtensionId = 1,
+  kAbsSendTimeExtensionId,
+  kVideoContentTypeExtensionId,
+  kVideoRotationRtpExtensionId,
+};
+
+constexpr int kDefaultMaxQp = cricket::WebRtcVideoChannel::kDefaultQpMax;
+uint8_t CodecTypeToPayloadType(VideoCodecType codec_type) {
+  switch (codec_type) {
+    case VideoCodecType::kVideoCodecGeneric:
+      return CallTest::kFakeVideoSendPayloadType;
+    case VideoCodecType::kVideoCodecVP8:
+      return CallTest::kPayloadTypeVP8;
+    case VideoCodecType::kVideoCodecVP9:
+      return CallTest::kPayloadTypeVP9;
+    case VideoCodecType::kVideoCodecH264:
+      return CallTest::kPayloadTypeH264;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  return {};
+}
+std::string CodecTypeToCodecName(VideoCodecType codec_type) {
+  switch (codec_type) {
+    case VideoCodecType::kVideoCodecGeneric:
+      return "";
+    case VideoCodecType::kVideoCodecVP8:
+      return cricket::kVp8CodecName;
+    case VideoCodecType::kVideoCodecVP9:
+      return cricket::kVp9CodecName;
+    case VideoCodecType::kVideoCodecH264:
+      return cricket::kH264CodecName;
+    default:
+      RTC_DCHECK_NOTREACHED();
+  }
+  return {};
+}
+VideoEncoderConfig::ContentType ConvertContentType(
+    VideoStreamConfig::Encoder::ContentType content_type) {
+  switch (content_type) {
+    case VideoStreamConfig::Encoder::ContentType::kVideo:
+      return VideoEncoderConfig::ContentType::kRealtimeVideo;
+    case VideoStreamConfig::Encoder::ContentType::kScreen:
+      return VideoEncoderConfig::ContentType::kScreen;
+  }
+}
+
+std::string TransformFilePath(std::string path) {
+  static const std::string resource_prefix = "res://";
+  int ext_pos = path.rfind('.');
+  if (ext_pos < 0) {
+    return test::ResourcePath(path, "yuv");
+  } else if (absl::StartsWith(path, resource_prefix)) {
+    std::string name = path.substr(resource_prefix.length(), ext_pos);
+    std::string ext = path.substr(ext_pos, path.size());
+    return test::ResourcePath(name, ext);
+  }
+  return path;
+}
+
+VideoSendStream::Config CreateVideoSendStreamConfig(
+    VideoStreamConfig config,
+    std::vector<uint32_t> ssrcs,
+    std::vector<uint32_t> rtx_ssrcs,
+    Transport* send_transport) {
+  VideoSendStream::Config send_config(send_transport);
+  send_config.rtp.payload_name = CodecTypeToPayloadString(config.encoder.codec);
+  send_config.rtp.payload_type = CodecTypeToPayloadType(config.encoder.codec);
+  send_config.rtp.nack.rtp_history_ms =
+      config.stream.nack_history_time.ms<int>();
+
+  send_config.rtp.ssrcs = ssrcs;
+  send_config.rtp.extensions = GetVideoRtpExtensions(config);
+
+  if (config.stream.use_rtx) {
+    send_config.rtp.rtx.payload_type = CallTest::kSendRtxPayloadType;
+    send_config.rtp.rtx.ssrcs = rtx_ssrcs;
+  }
+  if (config.stream.use_flexfec) {
+    send_config.rtp.flexfec.payload_type = CallTest::kFlexfecPayloadType;
+    send_config.rtp.flexfec.ssrc = CallTest::kFlexfecSendSsrc;
+    send_config.rtp.flexfec.protected_media_ssrcs = ssrcs;
+  }
+  if (config.stream.use_ulpfec) {
+    send_config.rtp.ulpfec.red_payload_type = CallTest::kRedPayloadType;
+    send_config.rtp.ulpfec.ulpfec_payload_type = CallTest::kUlpfecPayloadType;
+    send_config.rtp.ulpfec.red_rtx_payload_type = CallTest::kRtxRedPayloadType;
+  }
+  return send_config;
+}
+rtc::scoped_refptr<VideoEncoderConfig::EncoderSpecificSettings>
+CreateVp9SpecificSettings(VideoStreamConfig video_config) {
+  constexpr auto kScreen = VideoStreamConfig::Encoder::ContentType::kScreen;
+  VideoStreamConfig::Encoder conf = video_config.encoder;
+  VideoCodecVP9 vp9 = VideoEncoder::GetDefaultVp9Settings();
+  // TODO(bugs.webrtc.org/11607): Support separate scalability mode per
+  // simulcast stream.
+  ScalabilityMode scalability_mode = conf.simulcast_streams[0];
+  vp9.keyFrameInterval = conf.key_frame_interval.value_or(0);
+  vp9.numberOfTemporalLayers =
+      ScalabilityModeToNumTemporalLayers(scalability_mode);
+  vp9.numberOfSpatialLayers =
+      ScalabilityModeToNumSpatialLayers(scalability_mode);
+  vp9.interLayerPred = ScalabilityModeToInterLayerPredMode(scalability_mode);
+
+  if (conf.content_type == kScreen &&
+      (video_config.source.framerate > 5 || vp9.numberOfSpatialLayers >= 3)) {
+    vp9.flexibleMode = true;
+  }
+
+  if (conf.content_type == kScreen || vp9.numberOfTemporalLayers > 1 ||
+      vp9.numberOfSpatialLayers > 1) {
+    vp9.automaticResizeOn = false;
+    vp9.denoisingOn = false;
+  } else {
+    vp9.automaticResizeOn = conf.single.automatic_scaling;
+    vp9.denoisingOn = conf.single.denoising;
+  }
+  return rtc::make_ref_counted<VideoEncoderConfig::Vp9EncoderSpecificSettings>(
+      vp9);
+}
+
+rtc::scoped_refptr<VideoEncoderConfig::EncoderSpecificSettings>
+CreateVp8SpecificSettings(VideoStreamConfig config) {
+  VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
+  vp8_settings.keyFrameInterval = config.encoder.key_frame_interval.value_or(0);
+  // TODO(bugs.webrtc.org/11607): Support separate scalability mode per
+  // simulcast stream.
+  ScalabilityMode scalability_mode = config.encoder.simulcast_streams[0];
+  vp8_settings.numberOfTemporalLayers =
+      ScalabilityModeToNumTemporalLayers(scalability_mode);
+  if (vp8_settings.numberOfTemporalLayers > 1 ||
+      config.encoder.simulcast_streams.size() > 1) {
+    vp8_settings.automaticResizeOn = false;
+    vp8_settings.denoisingOn = false;
+  } else {
+    vp8_settings.automaticResizeOn = config.encoder.single.automatic_scaling;
+    vp8_settings.denoisingOn = config.encoder.single.denoising;
+  }
+  return rtc::make_ref_counted<VideoEncoderConfig::Vp8EncoderSpecificSettings>(
+      vp8_settings);
+}
+
+rtc::scoped_refptr<VideoEncoderConfig::EncoderSpecificSettings>
+CreateH264SpecificSettings(VideoStreamConfig config) {
+  RTC_DCHECK_EQ(config.encoder.simulcast_streams.size(), 1);
+  RTC_DCHECK(config.encoder.simulcast_streams[0] == ScalabilityMode::kL1T1);
+  // TODO(bugs.webrtc.org/6883): Set a key frame interval as a setting that
+  // isn't codec specific.
+  RTC_CHECK_EQ(0, config.encoder.key_frame_interval.value_or(0));
+  return nullptr;
+}
+
+rtc::scoped_refptr<VideoEncoderConfig::EncoderSpecificSettings>
+CreateEncoderSpecificSettings(VideoStreamConfig config) {
+  using Codec = VideoStreamConfig::Encoder::Codec;
+  switch (config.encoder.codec) {
+    case Codec::kVideoCodecH264:
+      return CreateH264SpecificSettings(config);
+    case Codec::kVideoCodecVP8:
+      return CreateVp8SpecificSettings(config);
+    case Codec::kVideoCodecVP9:
+      return CreateVp9SpecificSettings(config);
+    case Codec::kVideoCodecGeneric:
+    case Codec::kVideoCodecAV1:
+      return nullptr;
+    case Codec::kVideoCodecMultiplex:
+      RTC_DCHECK_NOTREACHED();
+      return nullptr;
+  }
+}
+
+VideoEncoderConfig CreateVideoEncoderConfig(VideoStreamConfig config) {
+  webrtc::VideoEncoder::EncoderInfo encoder_info;
+  VideoEncoderConfig encoder_config;
+  encoder_config.codec_type = config.encoder.codec;
+  encoder_config.content_type = ConvertContentType(config.encoder.content_type);
+  encoder_config.video_format =
+      SdpVideoFormat(CodecTypeToPayloadString(config.encoder.codec), {});
+
+  encoder_config.number_of_streams = config.encoder.simulcast_streams.size();
+  encoder_config.simulcast_layers =
+      std::vector<VideoStream>(encoder_config.number_of_streams);
+  encoder_config.min_transmit_bitrate_bps = config.stream.pad_to_rate.bps();
+
+  std::string cricket_codec = CodecTypeToCodecName(config.encoder.codec);
+  if (!cricket_codec.empty()) {
+    bool screenshare = config.encoder.content_type ==
+                       VideoStreamConfig::Encoder::ContentType::kScreen;
+    encoder_config.video_stream_factory =
+        rtc::make_ref_counted<cricket::EncoderStreamFactory>(
+            cricket_codec, kDefaultMaxQp, screenshare, screenshare,
+            encoder_info);
+  } else {
+    encoder_config.video_stream_factory =
+        rtc::make_ref_counted<DefaultVideoStreamFactory>();
+  }
+
+  // TODO(srte): Base this on encoder capabilities.
+  encoder_config.max_bitrate_bps =
+      config.encoder.max_data_rate.value_or(DataRate::KilobitsPerSec(10000))
+          .bps();
+
+  encoder_config.frame_drop_enabled = config.encoder.frame_dropping;
+  encoder_config.encoder_specific_settings =
+      CreateEncoderSpecificSettings(config);
+
+  for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
+    auto& layer = encoder_config.simulcast_layers[i];
+    if (config.encoder.max_framerate) {
+      layer.max_framerate = *config.encoder.max_framerate;
+      layer.min_bitrate_bps = config.encoder.min_data_rate->bps_or(-1);
+    }
+    layer.scalability_mode = config.encoder.simulcast_streams[i];
+  }
+
+  return encoder_config;
+}
+
+std::unique_ptr<FrameGeneratorInterface> CreateImageSlideGenerator(
+    Clock* clock,
+    VideoStreamConfig::Source::Slides slides,
+    int framerate) {
+  std::vector<std::string> paths = slides.images.paths;
+  for (std::string& path : paths)
+    path = TransformFilePath(path);
+  if (slides.images.crop.width || slides.images.crop.height) {
+    TimeDelta pause_duration =
+        slides.change_interval - slides.images.crop.scroll_duration;
+    RTC_CHECK_GE(pause_duration, TimeDelta::Zero());
+    int crop_width = slides.images.crop.width.value_or(slides.images.width);
+    int crop_height = slides.images.crop.height.value_or(slides.images.height);
+    RTC_CHECK_LE(crop_width, slides.images.width);
+    RTC_CHECK_LE(crop_height, slides.images.height);
+    return CreateScrollingInputFromYuvFilesFrameGenerator(
+        clock, paths, slides.images.width, slides.images.height, crop_width,
+        crop_height, slides.images.crop.scroll_duration.ms(),
+        pause_duration.ms());
+  } else {
+    return CreateFromYuvFileFrameGenerator(
+        paths, slides.images.width, slides.images.height,
+        slides.change_interval.seconds<double>() * framerate);
+  }
+}
+
+std::unique_ptr<FrameGeneratorInterface> CreateFrameGenerator(
+    Clock* clock,
+    VideoStreamConfig::Source source) {
+  using Capture = VideoStreamConfig::Source::Capture;
+  switch (source.capture) {
+    case Capture::kGenerator:
+      return CreateSquareFrameGenerator(
+          source.generator.width, source.generator.height,
+          source.generator.pixel_format, /*num_squares*/ absl::nullopt);
+    case Capture::kVideoFile:
+      RTC_CHECK(source.video_file.width && source.video_file.height);
+      return CreateFromYuvFileFrameGenerator(
+          {TransformFilePath(source.video_file.name)}, source.video_file.width,
+          source.video_file.height, /*frame_repeat_count*/ 1);
+    case Capture::kGenerateSlides:
+      return CreateSlideFrameGenerator(
+          source.slides.generator.width, source.slides.generator.height,
+          source.slides.change_interval.seconds<double>() * source.framerate);
+    case Capture::kImageSlides:
+      return CreateImageSlideGenerator(clock, source.slides, source.framerate);
+  }
+}
+
+VideoReceiveStreamInterface::Config CreateVideoReceiveStreamConfig(
+    VideoStreamConfig config,
+    Transport* feedback_transport,
+    VideoDecoderFactory* decoder_factory,
+    VideoReceiveStreamInterface::Decoder decoder,
+    rtc::VideoSinkInterface<VideoFrame>* renderer,
+    uint32_t local_ssrc,
+    uint32_t ssrc,
+    uint32_t rtx_ssrc) {
+  VideoReceiveStreamInterface::Config recv(feedback_transport);
+  recv.rtp.local_ssrc = local_ssrc;
+  recv.rtp.extensions = GetVideoRtpExtensions(config);
+
+  RTC_DCHECK(!config.stream.use_rtx ||
+             config.stream.nack_history_time > TimeDelta::Zero());
+  recv.rtp.nack.rtp_history_ms = config.stream.nack_history_time.ms();
+  recv.rtp.protected_by_flexfec = config.stream.use_flexfec;
+  recv.rtp.remote_ssrc = ssrc;
+  recv.decoder_factory = decoder_factory;
+  recv.decoders.push_back(decoder);
+  recv.renderer = renderer;
+  if (config.stream.use_rtx) {
+    recv.rtp.rtx_ssrc = rtx_ssrc;
+    recv.rtp.rtx_associated_payload_types[CallTest::kSendRtxPayloadType] =
+        CodecTypeToPayloadType(config.encoder.codec);
+  }
+  if (config.stream.use_ulpfec) {
+    recv.rtp.red_payload_type = CallTest::kRedPayloadType;
+    recv.rtp.ulpfec_payload_type = CallTest::kUlpfecPayloadType;
+    recv.rtp.rtx_associated_payload_types[CallTest::kRtxRedPayloadType] =
+        CallTest::kRedPayloadType;
+  }
+  recv.sync_group = config.render.sync_group;
+  return recv;
+}
+}  // namespace
+
+std::vector<RtpExtension> GetVideoRtpExtensions(
+    const VideoStreamConfig config) {
+  std::vector<RtpExtension> res = {
+      RtpExtension(RtpExtension::kVideoContentTypeUri,
+                   kVideoContentTypeExtensionId),
+      RtpExtension(RtpExtension::kVideoRotationUri,
+                   kVideoRotationRtpExtensionId)};
+  if (config.stream.packet_feedback) {
+    res.push_back(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                               kTransportSequenceNumberExtensionId));
+  }
+  if (config.stream.abs_send_time) {
+    res.push_back(
+        RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
+  }
+  return res;
+}
+
+SendVideoStream::SendVideoStream(CallClient* sender,
+                                 VideoStreamConfig config,
+                                 Transport* send_transport,
+                                 VideoFrameMatcher* matcher)
+    : sender_(sender), config_(config) {
+  video_capturer_ = std::make_unique<FrameGeneratorCapturer>(
+      sender_->clock_, CreateFrameGenerator(sender_->clock_, config.source),
+      config.source.framerate,
+      *sender->time_controller_->GetTaskQueueFactory());
+  video_capturer_->Init();
+
+  using Encoder = VideoStreamConfig::Encoder;
+  using Codec = VideoStreamConfig::Encoder::Codec;
+  switch (config.encoder.implementation) {
+    case Encoder::Implementation::kFake:
+      encoder_factory_ =
+          std::make_unique<FunctionVideoEncoderFactory>([this]() {
+            MutexLock lock(&mutex_);
+            std::unique_ptr<FakeEncoder> encoder;
+            if (config_.encoder.codec == Codec::kVideoCodecVP8) {
+              encoder = std::make_unique<test::FakeVp8Encoder>(sender_->clock_);
+            } else if (config_.encoder.codec == Codec::kVideoCodecGeneric) {
+              encoder = std::make_unique<test::FakeEncoder>(sender_->clock_);
+            } else {
+              RTC_DCHECK_NOTREACHED();
+            }
+            fake_encoders_.push_back(encoder.get());
+            if (config_.encoder.fake.max_rate.IsFinite())
+              encoder->SetMaxBitrate(config_.encoder.fake.max_rate.kbps());
+            return encoder;
+          });
+      break;
+    case VideoStreamConfig::Encoder::Implementation::kSoftware:
+      encoder_factory_.reset(new InternalEncoderFactory());
+      break;
+    case VideoStreamConfig::Encoder::Implementation::kHardware:
+      encoder_factory_ = CreateHardwareEncoderFactory();
+      break;
+  }
+  RTC_CHECK(encoder_factory_);
+
+  bitrate_allocator_factory_ = CreateBuiltinVideoBitrateAllocatorFactory();
+  RTC_CHECK(bitrate_allocator_factory_);
+
+  VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(config);
+  for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
+    ssrcs_.push_back(sender->GetNextVideoSsrc());
+    rtx_ssrcs_.push_back(sender->GetNextRtxSsrc());
+  }
+  VideoSendStream::Config send_config =
+      CreateVideoSendStreamConfig(config, ssrcs_, rtx_ssrcs_, send_transport);
+  send_config.encoder_settings.encoder_factory = encoder_factory_.get();
+  send_config.encoder_settings.bitrate_allocator_factory =
+      bitrate_allocator_factory_.get();
+  send_config.suspend_below_min_bitrate =
+      config.encoder.suspend_below_min_bitrate;
+
+  sender_->SendTask([&] {
+    if (config.stream.fec_controller_factory) {
+      send_stream_ = sender_->call_->CreateVideoSendStream(
+          std::move(send_config), std::move(encoder_config),
+          config.stream.fec_controller_factory->CreateFecController());
+    } else {
+      send_stream_ = sender_->call_->CreateVideoSendStream(
+          std::move(send_config), std::move(encoder_config));
+    }
+
+    if (matcher->Active()) {
+      frame_tap_ = std::make_unique<ForwardingCapturedFrameTap>(
+          sender_->clock_, matcher, video_capturer_.get());
+      send_stream_->SetSource(frame_tap_.get(),
+                              config.encoder.degradation_preference);
+    } else {
+      send_stream_->SetSource(video_capturer_.get(),
+                              config.encoder.degradation_preference);
+    }
+  });
+}
+
+SendVideoStream::~SendVideoStream() {
+  sender_->SendTask(
+      [this] { sender_->call_->DestroyVideoSendStream(send_stream_); });
+}
+
+void SendVideoStream::Start() {
+  sender_->SendTask([this] {
+    send_stream_->Start();
+    sender_->call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
+  });
+}
+
+void SendVideoStream::Stop() {
+  sender_->SendTask([this] { send_stream_->Stop(); });
+}
+
+void SendVideoStream::UpdateConfig(
+    std::function<void(VideoStreamConfig*)> modifier) {
+  sender_->SendTask([&] {
+    MutexLock lock(&mutex_);
+    VideoStreamConfig prior_config = config_;
+    modifier(&config_);
+    if (prior_config.encoder.fake.max_rate != config_.encoder.fake.max_rate) {
+      for (auto* encoder : fake_encoders_) {
+        encoder->SetMaxBitrate(config_.encoder.fake.max_rate.kbps());
+      }
+    }
+    // TODO(srte): Add more conditions that should cause reconfiguration.
+    if (prior_config.encoder.max_framerate != config_.encoder.max_framerate ||
+        prior_config.encoder.max_data_rate != config_.encoder.max_data_rate) {
+      VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(config_);
+      send_stream_->ReconfigureVideoEncoder(std::move(encoder_config));
+    }
+    if (prior_config.source.framerate != config_.source.framerate) {
+      SetCaptureFramerate(config_.source.framerate);
+    }
+  });
+}
+
+void SendVideoStream::UpdateActiveLayers(std::vector<bool> active_layers) {
+  sender_->task_queue_.PostTask([=] {
+    MutexLock lock(&mutex_);
+    if (config_.encoder.codec ==
+        VideoStreamConfig::Encoder::Codec::kVideoCodecVP8) {
+      send_stream_->StartPerRtpStream(active_layers);
+    }
+    VideoEncoderConfig encoder_config = CreateVideoEncoderConfig(config_);
+    RTC_CHECK_EQ(encoder_config.simulcast_layers.size(), active_layers.size());
+    for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i)
+      encoder_config.simulcast_layers[i].active = active_layers[i];
+    send_stream_->ReconfigureVideoEncoder(std::move(encoder_config));
+  });
+}
+
+bool SendVideoStream::UsingSsrc(uint32_t ssrc) const {
+  for (uint32_t owned : ssrcs_) {
+    if (owned == ssrc)
+      return true;
+  }
+  return false;
+}
+
+bool SendVideoStream::UsingRtxSsrc(uint32_t ssrc) const {
+  for (uint32_t owned : rtx_ssrcs_) {
+    if (owned == ssrc)
+      return true;
+  }
+  return false;
+}
+
+void SendVideoStream::SetCaptureFramerate(int framerate) {
+  sender_->SendTask([&] { video_capturer_->ChangeFramerate(framerate); });
+}
+
+VideoSendStream::Stats SendVideoStream::GetStats() const {
+  return send_stream_->GetStats();
+}
+
+ColumnPrinter SendVideoStream::StatsPrinter() {
+  return ColumnPrinter::Lambda(
+      "video_target_rate video_sent_rate width height",
+      [this](rtc::SimpleStringBuilder& sb) {
+        VideoSendStream::Stats video_stats = send_stream_->GetStats();
+        int width = 0;
+        int height = 0;
+        for (const auto& stream_stat : video_stats.substreams) {
+          width = std::max(width, stream_stat.second.width);
+          height = std::max(height, stream_stat.second.height);
+        }
+        sb.AppendFormat("%.0lf %.0lf %i %i",
+                        video_stats.target_media_bitrate_bps / 8.0,
+                        video_stats.media_bitrate_bps / 8.0, width, height);
+      },
+      64);
+}
+
+ReceiveVideoStream::ReceiveVideoStream(CallClient* receiver,
+                                       VideoStreamConfig config,
+                                       SendVideoStream* send_stream,
+                                       size_t chosen_stream,
+                                       Transport* feedback_transport,
+                                       VideoFrameMatcher* matcher)
+    : receiver_(receiver), config_(config) {
+  if (config.encoder.codec ==
+          VideoStreamConfig::Encoder::Codec::kVideoCodecGeneric ||
+      config.encoder.implementation == VideoStreamConfig::Encoder::kFake) {
+    decoder_factory_ = std::make_unique<FunctionVideoDecoderFactory>(
+        []() { return std::make_unique<FakeDecoder>(); });
+  } else {
+    decoder_factory_ = std::make_unique<InternalDecoderFactory>();
+  }
+
+  VideoReceiveStreamInterface::Decoder decoder =
+      CreateMatchingDecoder(CodecTypeToPayloadType(config.encoder.codec),
+                            CodecTypeToPayloadString(config.encoder.codec));
+  size_t num_streams = config.encoder.simulcast_streams.size();
+  for (size_t i = 0; i < num_streams; ++i) {
+    rtc::VideoSinkInterface<VideoFrame>* renderer = &fake_renderer_;
+    if (matcher->Active()) {
+      render_taps_.emplace_back(
+          std::make_unique<DecodedFrameTap>(receiver_->clock_, matcher, i));
+      renderer = render_taps_.back().get();
+    }
+    auto recv_config = CreateVideoReceiveStreamConfig(
+        config, feedback_transport, decoder_factory_.get(), decoder, renderer,
+        receiver_->GetNextVideoLocalSsrc(), send_stream->ssrcs_[i],
+        send_stream->rtx_ssrcs_[i]);
+    if (config.stream.use_flexfec) {
+      RTC_DCHECK(num_streams == 1);
+      FlexfecReceiveStream::Config flexfec(feedback_transport);
+      flexfec.payload_type = CallTest::kFlexfecPayloadType;
+      flexfec.rtp.remote_ssrc = CallTest::kFlexfecSendSsrc;
+      flexfec.protected_media_ssrcs = send_stream->rtx_ssrcs_;
+      flexfec.rtp.local_ssrc = recv_config.rtp.local_ssrc;
+      receiver_->ssrc_media_types_[flexfec.rtp.remote_ssrc] = MediaType::VIDEO;
+
+      receiver_->SendTask([this, &flexfec] {
+        flecfec_stream_ = receiver_->call_->CreateFlexfecReceiveStream(flexfec);
+      });
+    }
+    receiver_->ssrc_media_types_[recv_config.rtp.remote_ssrc] =
+        MediaType::VIDEO;
+    if (config.stream.use_rtx)
+      receiver_->ssrc_media_types_[recv_config.rtp.rtx_ssrc] = MediaType::VIDEO;
+    receiver_->SendTask([this, &recv_config] {
+      receive_streams_.push_back(
+          receiver_->call_->CreateVideoReceiveStream(std::move(recv_config)));
+    });
+  }
+}
+
+ReceiveVideoStream::~ReceiveVideoStream() {
+  receiver_->SendTask([this] {
+    for (auto* recv_stream : receive_streams_)
+      receiver_->call_->DestroyVideoReceiveStream(recv_stream);
+    if (flecfec_stream_)
+      receiver_->call_->DestroyFlexfecReceiveStream(flecfec_stream_);
+  });
+}
+
+void ReceiveVideoStream::Start() {
+  receiver_->SendTask([this] {
+    for (auto* recv_stream : receive_streams_)
+      recv_stream->Start();
+    receiver_->call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
+  });
+}
+
+void ReceiveVideoStream::Stop() {
+  receiver_->SendTask([this] {
+    for (auto* recv_stream : receive_streams_)
+      recv_stream->Stop();
+  });
+}
+
+VideoReceiveStreamInterface::Stats ReceiveVideoStream::GetStats() const {
+  if (receive_streams_.empty())
+    return VideoReceiveStreamInterface::Stats();
+  // TODO(srte): Handle multiple receive streams.
+  return receive_streams_.back()->GetStats();
+}
+
+VideoStreamPair::~VideoStreamPair() = default;
+
+VideoStreamPair::VideoStreamPair(CallClient* sender,
+                                 CallClient* receiver,
+                                 VideoStreamConfig config)
+    : config_(config),
+      matcher_(config.hooks.frame_pair_handlers),
+      send_stream_(sender, config, sender->transport_.get(), &matcher_),
+      receive_stream_(receiver,
+                      config,
+                      &send_stream_,
+                      /*chosen_stream=*/0,
+                      receiver->transport_.get(),
+                      &matcher_) {}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scenario/video_stream.h b/third_party/libwebrtc/test/scenario/video_stream.h
new file mode 100644
index 0000000000..43c51eab73
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/video_stream.h
@@ -0,0 +1,138 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_SCENARIO_VIDEO_STREAM_H_
+#define TEST_SCENARIO_VIDEO_STREAM_H_
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "rtc_base/synchronization/mutex.h"
+#include "test/fake_encoder.h"
+#include "test/fake_videorenderer.h"
+#include "test/frame_generator_capturer.h"
+#include "test/logging/log_writer.h"
+#include "test/scenario/call_client.h"
+#include "test/scenario/column_printer.h"
+#include "test/scenario/network_node.h"
+#include "test/scenario/scenario_config.h"
+#include "test/scenario/video_frame_matcher.h"
+#include "test/test_video_capturer.h"
+
+namespace webrtc {
+namespace test {
+// SendVideoStream provides an interface for changing parameters and retrieving
+// states at run time.
+class SendVideoStream {
+ public:
+  ~SendVideoStream();
+
+  SendVideoStream(const SendVideoStream&) = delete;
+  SendVideoStream& operator=(const SendVideoStream&) = delete;
+
+  void SetCaptureFramerate(int framerate);
+  VideoSendStream::Stats GetStats() const;
+  ColumnPrinter StatsPrinter();
+  void Start();
+  void Stop();
+  void UpdateConfig(std::function<void(VideoStreamConfig*)> modifier);
+  void UpdateActiveLayers(std::vector<bool> active_layers);
+  bool UsingSsrc(uint32_t ssrc) const;
+  bool UsingRtxSsrc(uint32_t ssrc) const;
+
+ private:
+  friend class Scenario;
+  friend class VideoStreamPair;
+  friend class ReceiveVideoStream;
+  // Handles RTCP feedback for this stream.
+  SendVideoStream(CallClient* sender,
+                  VideoStreamConfig config,
+                  Transport* send_transport,
+                  VideoFrameMatcher* matcher);
+
+  Mutex mutex_;
+  std::vector<uint32_t> ssrcs_;
+  std::vector<uint32_t> rtx_ssrcs_;
+  VideoSendStream* send_stream_ = nullptr;
+  CallClient* const sender_;
+  VideoStreamConfig config_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<VideoEncoderFactory> encoder_factory_;
+  std::vector<test::FakeEncoder*> fake_encoders_ RTC_GUARDED_BY(mutex_);
+  std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
+  std::unique_ptr<FrameGeneratorCapturer> video_capturer_;
+  std::unique_ptr<ForwardingCapturedFrameTap> frame_tap_;
+  int next_local_network_id_ = 0;
+  int next_remote_network_id_ = 0;
+};
+
+// ReceiveVideoStream represents a video receiver. It can't be used directly.
+class ReceiveVideoStream {
+ public:
+  ~ReceiveVideoStream();
+
+  ReceiveVideoStream(const ReceiveVideoStream&) = delete;
+  ReceiveVideoStream& operator=(const ReceiveVideoStream&) = delete;
+
+  void Start();
+  void Stop();
+  VideoReceiveStreamInterface::Stats GetStats() const;
+
+ private:
+  friend class Scenario;
+  friend class VideoStreamPair;
+  ReceiveVideoStream(CallClient* receiver,
+                     VideoStreamConfig config,
+                     SendVideoStream* send_stream,
+                     size_t chosen_stream,
+                     Transport* feedback_transport,
+                     VideoFrameMatcher* matcher);
+
+  std::vector<VideoReceiveStreamInterface*> receive_streams_;
+  FlexfecReceiveStream* flecfec_stream_ = nullptr;
+  FakeVideoRenderer fake_renderer_;
+  std::vector<std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>>>
+      render_taps_;
+  CallClient* const receiver_;
+  const VideoStreamConfig config_;
+  std::unique_ptr<VideoDecoderFactory> decoder_factory_;
+};
+
+// VideoStreamPair represents a video streaming session. It can be used to
+// access underlying send and receive classes. It can also be used in calls to
+// the Scenario class.
+class VideoStreamPair {
+ public:
+  ~VideoStreamPair();
+
+  VideoStreamPair(const VideoStreamPair&) = delete;
+  VideoStreamPair& operator=(const VideoStreamPair&) = delete;
+
+  SendVideoStream* send() { return &send_stream_; }
+  ReceiveVideoStream* receive() { return &receive_stream_; }
+  VideoFrameMatcher* matcher() { return &matcher_; }
+
+ private:
+  friend class Scenario;
+  VideoStreamPair(CallClient* sender,
+                  CallClient* receiver,
+                  VideoStreamConfig config);
+
+  const VideoStreamConfig config_;
+
+  VideoFrameMatcher matcher_;
+  SendVideoStream send_stream_;
+  ReceiveVideoStream receive_stream_;
+};
+
+std::vector<RtpExtension> GetVideoRtpExtensions(const VideoStreamConfig config);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCENARIO_VIDEO_STREAM_H_
diff --git a/third_party/libwebrtc/test/scenario/video_stream_unittest.cc b/third_party/libwebrtc/test/scenario/video_stream_unittest.cc
new file mode 100644
index 0000000000..e53af4ef2b
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/video_stream_unittest.cc
@@ -0,0 +1,322 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <atomic>
+
+#include "api/test/network_emulation/create_cross_traffic.h"
+#include "api/test/network_emulation/cross_traffic.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+using Capture = VideoStreamConfig::Source::Capture;
+using ContentType = VideoStreamConfig::Encoder::ContentType;
+using Codec = VideoStreamConfig::Encoder::Codec;
+using CodecImpl = VideoStreamConfig::Encoder::Implementation;
+}  // namespace
+
+TEST(VideoStreamTest, ReceivesFramesFromFileBasedStreams) {
+  TimeDelta kRunTime = TimeDelta::Millis(500);
+  std::vector<int> kFrameRates = {15, 30};
+  std::deque<std::atomic<int>> frame_counts(2);
+  frame_counts[0] = 0;
+  frame_counts[1] = 0;
+  {
+    Scenario s;
+    auto route =
+        s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+                       {s.CreateSimulationNode(NetworkSimulationConfig())},
+                       s.CreateClient("callee", CallClientConfig()),
+                       {s.CreateSimulationNode(NetworkSimulationConfig())});
+
+    s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+      c->hooks.frame_pair_handlers = {
+          [&](const VideoFramePair&) { frame_counts[0]++; }};
+      c->source.capture = Capture::kVideoFile;
+      c->source.video_file.name = "foreman_cif";
+      c->source.video_file.width = 352;
+      c->source.video_file.height = 288;
+      c->source.framerate = kFrameRates[0];
+      c->encoder.implementation = CodecImpl::kSoftware;
+      c->encoder.codec = Codec::kVideoCodecVP8;
+    });
+    s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+      c->hooks.frame_pair_handlers = {
+          [&](const VideoFramePair&) { frame_counts[1]++; }};
+      c->source.capture = Capture::kImageSlides;
+      c->source.slides.images.crop.width = 320;
+      c->source.slides.images.crop.height = 240;
+      c->source.framerate = kFrameRates[1];
+      c->encoder.implementation = CodecImpl::kSoftware;
+      c->encoder.codec = Codec::kVideoCodecVP9;
+    });
+    s.RunFor(kRunTime);
+  }
+  std::vector<int> expected_counts;
+  for (int fps : kFrameRates)
+    expected_counts.push_back(
+        static_cast<int>(kRunTime.seconds<double>() * fps * 0.8));
+
+  EXPECT_GE(frame_counts[0], expected_counts[0]);
+  EXPECT_GE(frame_counts[1], expected_counts[1]);
+}
+
+TEST(VideoStreamTest, ReceivesVp8SimulcastFrames) {
+  TimeDelta kRunTime = TimeDelta::Millis(500);
+  int kFrameRate = 30;
+
+  std::deque<std::atomic<int>> frame_counts(3);
+  frame_counts[0] = 0;
+  frame_counts[1] = 0;
+  frame_counts[2] = 0;
+  {
+    Scenario s;
+    auto route =
+        s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+                       {s.CreateSimulationNode(NetworkSimulationConfig())},
+                       s.CreateClient("callee", CallClientConfig()),
+                       {s.CreateSimulationNode(NetworkSimulationConfig())});
+    s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+      // TODO(srte): Replace with code checking for all simulcast streams when
+      // there's a hook available for that.
+      c->hooks.frame_pair_handlers = {[&](const VideoFramePair& info) {
+        frame_counts[info.layer_id]++;
+        RTC_DCHECK(info.decoded);
+        printf("%i: [%3i->%3i, %i], %i->%i, \n", info.layer_id, info.capture_id,
+               info.decode_id, info.repeated, info.captured->width(),
+               info.decoded->width());
+      }};
+      c->source.framerate = kFrameRate;
+      // The resolution must be high enough to allow smaller layers to be
+      // created.
+      c->source.generator.width = 1024;
+      c->source.generator.height = 768;
+      c->encoder.implementation = CodecImpl::kSoftware;
+      c->encoder.codec = Codec::kVideoCodecVP8;
+      // Enable simulcast.
+      c->encoder.simulcast_streams = {webrtc::ScalabilityMode::kL1T1,
+                                      webrtc::ScalabilityMode::kL1T1,
+                                      webrtc::ScalabilityMode::kL1T1};
+
+    });
+    s.RunFor(kRunTime);
+  }
+
+  // Using high error margin to avoid flakyness.
+  const int kExpectedCount =
+      static_cast<int>(kRunTime.seconds<double>() * kFrameRate * 0.5);
+
+  EXPECT_GE(frame_counts[0], kExpectedCount);
+  EXPECT_GE(frame_counts[1], kExpectedCount);
+  EXPECT_GE(frame_counts[2], kExpectedCount);
+}
+
+TEST(VideoStreamTest, SendsNacksOnLoss) {
+  Scenario s;
+  auto route =
+      s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+                     {s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+                       c->loss_rate = 0.2;
+                     })},
+                     s.CreateClient("callee", CallClientConfig()),
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  // NACK retransmissions are enabled by default.
+  auto video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  s.RunFor(TimeDelta::Seconds(1));
+  int retransmit_packets = 0;
+  VideoSendStream::Stats stats;
+  route->first()->SendTask([&]() { stats = video->send()->GetStats(); });
+  for (const auto& substream : stats.substreams) {
+    retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
+  }
+  EXPECT_GT(retransmit_packets, 0);
+}
+
+TEST(VideoStreamTest, SendsFecWithUlpFec) {
+  Scenario s;
+  auto route =
+      s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+                     {s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+                       c->loss_rate = 0.1;
+                       c->delay = TimeDelta::Millis(100);
+                     })},
+                     s.CreateClient("callee", CallClientConfig()),
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+    // We do not allow NACK+ULPFEC for generic codec, using VP8.
+    c->encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+    c->stream.use_ulpfec = true;
+  });
+  s.RunFor(TimeDelta::Seconds(5));
+  VideoSendStream::Stats video_stats;
+  route->first()->SendTask([&]() { video_stats = video->send()->GetStats(); });
+  EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
+}
+TEST(VideoStreamTest, SendsFecWithFlexFec) {
+  Scenario s;
+  auto route =
+      s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+                     {s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+                       c->loss_rate = 0.1;
+                       c->delay = TimeDelta::Millis(100);
+                     })},
+                     s.CreateClient("callee", CallClientConfig()),
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+    c->stream.use_flexfec = true;
+  });
+  s.RunFor(TimeDelta::Seconds(5));
+  VideoSendStream::Stats video_stats;
+  route->first()->SendTask([&]() { video_stats = video->send()->GetStats(); });
+  EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
+}
+
+TEST(VideoStreamTest, ResolutionAdaptsToAvailableBandwidth) {
+  // Declared before scenario to avoid use after free.
+  std::atomic<size_t> num_qvga_frames_(0);
+  std::atomic<size_t> num_vga_frames_(0);
+
+  Scenario s;
+  // Link has enough capacity for VGA.
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(800);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(800);
+  });
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+  auto* route = s.CreateRoutes(
+      client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+
+  s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+    c->hooks.frame_pair_handlers = {[&](const VideoFramePair& info) {
+      if (info.decoded->width() == 640) {
+        ++num_vga_frames_;
+      } else if (info.decoded->width() == 320) {
+        ++num_qvga_frames_;
+      } else {
+        ADD_FAILURE() << "Unexpected resolution: " << info.decoded->width();
+      }
+    }};
+    c->source.framerate = 30;
+    // The resolution must be high enough to allow smaller layers to be
+    // created.
+    c->source.generator.width = 640;
+    c->source.generator.height = 480;
+    c->encoder.implementation = CodecImpl::kSoftware;
+    c->encoder.codec = Codec::kVideoCodecVP9;
+    // Enable SVC.
+    c->encoder.simulcast_streams = {webrtc::ScalabilityMode::kL2T1};
+  });
+
+  // Run for a few seconds, until streams have stabilized,
+  // check that we are sending VGA.
+  s.RunFor(TimeDelta::Seconds(5));
+  EXPECT_GT(num_vga_frames_, 0u);
+
+  // Trigger cross traffic, run until we have seen 3 consecutive
+  // seconds with no VGA frames due to reduced available bandwidth.
+  auto cross_traffic = s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+      s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+      FakeTcpConfig()));
+
+  int num_seconds_without_vga = 0;
+  int num_iterations = 0;
+  do {
+    ASSERT_LE(++num_iterations, 100);
+    num_qvga_frames_ = 0;
+    num_vga_frames_ = 0;
+    s.RunFor(TimeDelta::Seconds(1));
+    if (num_qvga_frames_ > 0 && num_vga_frames_ == 0) {
+      ++num_seconds_without_vga;
+    } else {
+      num_seconds_without_vga = 0;
+    }
+  } while (num_seconds_without_vga < 3);
+
+  // Stop cross traffic, make sure we recover and get VGA frames agian.
+  s.net()->StopCrossTraffic(cross_traffic);
+  num_qvga_frames_ = 0;
+  num_vga_frames_ = 0;
+
+  s.RunFor(TimeDelta::Seconds(40));
+  EXPECT_GT(num_qvga_frames_, 0u);
+  EXPECT_GT(num_vga_frames_, 0u);
+}
+
+TEST(VideoStreamTest, SuspendsBelowMinBitrate) {
+  const DataRate kMinVideoBitrate = DataRate::KilobitsPerSec(30);
+
+  // Declared before scenario to avoid use after free.
+  std::atomic<Timestamp> last_frame_timestamp(Timestamp::MinusInfinity());
+
+  Scenario s;
+  NetworkSimulationConfig net_config;
+  net_config.bandwidth = kMinVideoBitrate * 4;
+  net_config.delay = TimeDelta::Millis(10);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    // Min transmit rate needs to be lower than kMinVideoBitrate for this test
+    // to make sense.
+    c->transport.rates.min_rate = kMinVideoBitrate / 2;
+    c->transport.rates.start_rate = kMinVideoBitrate;
+    c->transport.rates.max_rate = kMinVideoBitrate * 2;
+  });
+  auto send_net = s.CreateMutableSimulationNode(
+      [&](NetworkSimulationConfig* c) { *c = net_config; });
+  auto ret_net = {s.CreateSimulationNode(net_config)};
+  auto* route =
+      s.CreateRoutes(client, {send_net->node()},
+                     s.CreateClient("return", CallClientConfig()), ret_net);
+
+  s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+    c->hooks.frame_pair_handlers = {[&](const VideoFramePair& pair) {
+      if (pair.repeated == 0) {
+        last_frame_timestamp = pair.capture_time;
+      }
+    }};
+    c->source.framerate = 30;
+    c->source.generator.width = 320;
+    c->source.generator.height = 180;
+    c->encoder.implementation = CodecImpl::kFake;
+    c->encoder.codec = Codec::kVideoCodecVP8;
+    c->encoder.min_data_rate = kMinVideoBitrate;
+    c->encoder.suspend_below_min_bitrate = true;
+    c->stream.pad_to_rate = kMinVideoBitrate;
+  });
+
+  // Run for a few seconds, check we have received at least one frame.
+  s.RunFor(TimeDelta::Seconds(2));
+  EXPECT_TRUE(last_frame_timestamp.load().IsFinite());
+
+  // Degrade network to below min bitrate.
+  send_net->UpdateConfig([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kMinVideoBitrate * 0.9;
+  });
+
+  // Run for 20s, verify that no frames arrive that were captured after the
+  // first five seconds, allowing some margin for BWE backoff to trigger and
+  // packets already in the pipeline to potentially arrive.
+  s.RunFor(TimeDelta::Seconds(20));
+  EXPECT_GT(s.Now() - last_frame_timestamp, TimeDelta::Seconds(15));
+
+  // Relax the network constraints and run for a while more, verify that we
+  // start receiving frames again.
+  send_net->UpdateConfig(
+      [&](NetworkSimulationConfig* c) { c->bandwidth = kMinVideoBitrate * 4; });
+  last_frame_timestamp = Timestamp::MinusInfinity();
+  s.RunFor(TimeDelta::Seconds(15));
+  EXPECT_TRUE(last_frame_timestamp.load().IsFinite());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scoped_key_value_config.cc b/third_party/libwebrtc/test/scoped_key_value_config.cc
new file mode 100644
index 0000000000..df84462637
--- /dev/null
+++ b/third_party/libwebrtc/test/scoped_key_value_config.cc
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/scoped_key_value_config.h"
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+
+namespace {
+
+// This part is copied from system_wrappers/field_trial.cc.
+void InsertIntoMap(
+    std::map<std::string, std::string, std::less<>>& key_value_map,
+    absl::string_view s) {
+  std::string::size_type field_start = 0;
+  while (field_start < s.size()) {
+    std::string::size_type separator_pos = s.find('/', field_start);
+    RTC_CHECK_NE(separator_pos, std::string::npos)
+        << "Missing separator '/' after field trial key.";
+    RTC_CHECK_GT(separator_pos, field_start)
+        << "Field trial key cannot be empty.";
+    std::string key(s.substr(field_start, separator_pos - field_start));
+    field_start = separator_pos + 1;
+
+    RTC_CHECK_LT(field_start, s.size())
+        << "Missing value after field trial key. String ended.";
+    separator_pos = s.find('/', field_start);
+    RTC_CHECK_NE(separator_pos, std::string::npos)
+        << "Missing terminating '/' in field trial string.";
+    RTC_CHECK_GT(separator_pos, field_start)
+        << "Field trial value cannot be empty.";
+    std::string value(s.substr(field_start, separator_pos - field_start));
+    field_start = separator_pos + 1;
+
+    key_value_map[key] = value;
+  }
+  // This check is technically redundant due to earlier checks.
+  // We nevertheless keep the check to make it clear that the entire
+  // string has been processed, and without indexing past the end.
+  RTC_CHECK_EQ(field_start, s.size());
+}
+
+}  // namespace
+
+namespace webrtc {
+namespace test {
+
+ScopedKeyValueConfig::ScopedKeyValueConfig()
+    : ScopedKeyValueConfig(nullptr, "") {}
+
+ScopedKeyValueConfig::ScopedKeyValueConfig(absl::string_view s)
+    : ScopedKeyValueConfig(nullptr, s) {}
+
+ScopedKeyValueConfig::ScopedKeyValueConfig(ScopedKeyValueConfig& parent,
+                                           absl::string_view s)
+    : ScopedKeyValueConfig(&parent, s) {}
+
+ScopedKeyValueConfig::ScopedKeyValueConfig(ScopedKeyValueConfig* parent,
+                                           absl::string_view s)
+    : parent_(parent), leaf_(nullptr) {
+  InsertIntoMap(key_value_map_, s);
+
+  if (!s.empty()) {
+    // Also store field trials in global string (until we get rid of it).
+    scoped_field_trials_ = std::make_unique<ScopedFieldTrials>(s);
+  }
+
+  if (parent == nullptr) {
+    // We are root, set leaf_.
+    leaf_ = this;
+  } else {
+    // Link root to new leaf.
+    GetRoot(parent)->leaf_ = this;
+    RTC_DCHECK(leaf_ == nullptr);
+  }
+}
+
+ScopedKeyValueConfig::~ScopedKeyValueConfig() {
+  if (parent_) {
+    GetRoot(parent_)->leaf_ = parent_;
+  }
+}
+
+ScopedKeyValueConfig* ScopedKeyValueConfig::GetRoot(ScopedKeyValueConfig* n) {
+  while (n->parent_ != nullptr) {
+    n = n->parent_;
+  }
+  return n;
+}
+
+std::string ScopedKeyValueConfig::GetValue(absl::string_view key) const {
+  if (parent_ == nullptr) {
+    return leaf_->LookupRecurse(key);
+  } else {
+    return LookupRecurse(key);
+  }
+}
+
+std::string ScopedKeyValueConfig::LookupRecurse(absl::string_view key) const {
+  auto it = key_value_map_.find(key);
+  if (it != key_value_map_.end())
+    return it->second;
+
+  if (parent_) {
+    return parent_->LookupRecurse(key);
+  }
+
+  // When at the root, check the global string so that test programs using
+  // a mix between ScopedKeyValueConfig and the global string continue to work
+  return webrtc::field_trial::FindFullName(std::string(key));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/scoped_key_value_config.h b/third_party/libwebrtc/test/scoped_key_value_config.h
new file mode 100644
index 0000000000..c0023f8228
--- /dev/null
+++ b/third_party/libwebrtc/test/scoped_key_value_config.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_SCOPED_KEY_VALUE_CONFIG_H_
+#define TEST_SCOPED_KEY_VALUE_CONFIG_H_
+
+#include <functional>
+#include <map>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/field_trials_registry.h"
+#include "test/field_trial.h"
+
+namespace webrtc {
+namespace test {
+
+class ScopedKeyValueConfig : public FieldTrialsRegistry {
+ public:
+  virtual ~ScopedKeyValueConfig();
+  ScopedKeyValueConfig();
+  explicit ScopedKeyValueConfig(absl::string_view s);
+  ScopedKeyValueConfig(ScopedKeyValueConfig& parent, absl::string_view s);
+
+ private:
+  ScopedKeyValueConfig(ScopedKeyValueConfig* parent, absl::string_view s);
+  ScopedKeyValueConfig* GetRoot(ScopedKeyValueConfig* n);
+  std::string GetValue(absl::string_view key) const override;
+  std::string LookupRecurse(absl::string_view key) const;
+
+  ScopedKeyValueConfig* const parent_;
+
+  // The leaf in a list of stacked ScopedKeyValueConfig.
+  // Only set on root (e.g with parent_ == nullptr).
+  const ScopedKeyValueConfig* leaf_;
+
+  // Unlike std::less<std::string>, std::less<> is transparent and allows
+  // heterogeneous lookup directly with absl::string_view.
+  std::map<std::string, std::string, std::less<>> key_value_map_;
+  std::unique_ptr<ScopedFieldTrials> scoped_field_trials_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_SCOPED_KEY_VALUE_CONFIG_H_
diff --git a/third_party/libwebrtc/test/test_flags.cc b/third_party/libwebrtc/test/test_flags.cc
new file mode 100644
index 0000000000..a0fff747fe
--- /dev/null
+++ b/third_party/libwebrtc/test/test_flags.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/test_flags.h"
+
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+
+ABSL_FLAG(std::string,
+          force_fieldtrials,
+          "",
+          "Field trials control experimental feature code which can be forced. "
+          "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
+          " will assign the group Enable to field trial WebRTC-FooFeature.");
+
+ABSL_FLAG(std::vector<std::string>,
+          plot,
+          {},
+          "List of metrics that should be exported for plotting (if they are "
+          "available). Example: psnr,ssim,encode_time. To plot all available "
+          " metrics pass 'all' as flag value");
+
+ABSL_FLAG(
+    std::string,
+    isolated_script_test_perf_output,
+    "",
+    "Path where the perf results should be stored in proto format described "
+    "described by histogram.proto in "
+    "https://chromium.googlesource.com/catapult/.");
+
+ABSL_FLAG(std::string,
+          webrtc_test_metrics_output_path,
+          "",
+          "Path where the test perf metrics should be stored using "
+          "api/test/metrics/metric.proto proto format. File will contain "
+          "MetricsSet as a root proto. On iOS, this MUST be a file name "
+          "and the file will be stored under NSDocumentDirectory.");
+
+ABSL_FLAG(bool,
+          export_perf_results_new_api,
+          false,
+          "Tells to initialize new API for exporting performance metrics");
diff --git a/third_party/libwebrtc/test/test_flags.h b/third_party/libwebrtc/test/test_flags.h
new file mode 100644
index 0000000000..30f918fc7d
--- /dev/null
+++ b/third_party/libwebrtc/test/test_flags.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TEST_FLAGS_H_
+#define TEST_TEST_FLAGS_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/flags/declare.h"
+
+ABSL_DECLARE_FLAG(std::string, force_fieldtrials);
+ABSL_DECLARE_FLAG(std::vector<std::string>, plot);
+ABSL_DECLARE_FLAG(std::string, isolated_script_test_perf_output);
+ABSL_DECLARE_FLAG(std::string, webrtc_test_metrics_output_path);
+ABSL_DECLARE_FLAG(bool, export_perf_results_new_api);
+
+#endif  // TEST_TEST_FLAGS_H_
diff --git a/third_party/libwebrtc/test/test_main.cc b/third_party/libwebrtc/test/test_main.cc
new file mode 100644
index 0000000000..d811fd0e6d
--- /dev/null
+++ b/third_party/libwebrtc/test/test_main.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <regex>
+#include <string>
+#include <vector>
+
+#include "absl/debugging/failure_signal_handler.h"
+#include "absl/debugging/symbolize.h"
+#include "absl/flags/parse.h"
+#include "test/gmock.h"
+#include "test/test_main_lib.h"
+
+namespace {
+
+std::vector<std::string> ReplaceDashesWithUnderscores(int argc, char* argv[]) {
+  std::vector<std::string> args(argv, argv + argc);
+  for (std::string& arg : args) {
+    // Only replace arguments that starts with a dash.
+    if (!arg.empty() && arg[0] == '-') {
+      // Don't replace the 2 first characters.
+      auto begin = arg.begin() + 2;
+      // Replace dashes on the left of '=' or on all the arg if no '=' is found.
+      auto end = std::find(arg.begin(), arg.end(), '=');
+      std::replace(begin, end, '-', '_');
+    }
+  }
+  return args;
+}
+
+std::vector<char*> VectorOfStringsToVectorOfPointers(
+    std::vector<std::string>& input) {
+  std::vector<char*> output(input.size());
+  for (size_t i = 0; i < input.size(); ++i) {
+    output[i] = &(input[i][0]);
+  }
+  return output;
+}
+
+}  // namespace
+
+int main(int argc, char* argv[]) {
+  // Initialize the symbolizer to get a human-readable stack trace
+  absl::InitializeSymbolizer(argv[0]);
+  testing::InitGoogleMock(&argc, argv);
+  // Before parsing the arguments with the absl flag library, any internal '-'
+  // characters will be converted to '_' characters to make sure the string is a
+  // valid attribute name.
+  std::vector<std::string> new_argv = ReplaceDashesWithUnderscores(argc, argv);
+  std::vector<char*> raw_new_argv = VectorOfStringsToVectorOfPointers(new_argv);
+  absl::ParseCommandLine(argc, &raw_new_argv[0]);
+
+// This absl handler use unsupported features/instructions on Fuchsia
+#if !defined(WEBRTC_FUCHSIA)
+  absl::FailureSignalHandlerOptions options;
+  absl::InstallFailureSignalHandler(options);
+#endif
+
+  std::unique_ptr<webrtc::TestMain> main = webrtc::TestMain::Create();
+  int err_code = main->Init();
+  if (err_code != 0) {
+    return err_code;
+  }
+  return main->Run(argc, argv);
+}
diff --git a/third_party/libwebrtc/test/test_main_lib.cc b/third_party/libwebrtc/test/test_main_lib.cc
new file mode 100644
index 0000000000..4c80315ac5
--- /dev/null
+++ b/third_party/libwebrtc/test/test_main_lib.cc
@@ -0,0 +1,267 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/test_main_lib.h"
+
+#include <fstream>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/memory/memory.h"
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/test/metrics/chrome_perf_dashboard_metrics_exporter.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metrics_exporter.h"
+#include "api/test/metrics/metrics_set_proto_file_exporter.h"
+#include "api/test/metrics/print_result_proxy_metrics_exporter.h"
+#include "api/test/metrics/stdout_metrics_exporter.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event_tracer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/ssl_adapter.h"
+#include "rtc_base/ssl_stream_adapter.h"
+#include "rtc_base/thread.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/test_flags.h"
+#include "test/testsupport/perf_test.h"
+#include "test/testsupport/resources_dir_flag.h"
+
+#if defined(WEBRTC_WIN)
+#include "rtc_base/win32_socket_init.h"
+#endif
+
+#if defined(WEBRTC_IOS)
+#include "test/ios/test_support.h"
+
+ABSL_FLAG(std::string,
+          NSTreatUnknownArgumentsAsOpen,
+          "",
+          "Intentionally ignored flag intended for iOS test runner.");
+ABSL_FLAG(std::string,
+          ApplePersistenceIgnoreState,
+          "",
+          "Intentionally ignored flag intended for iOS test runner.");
+ABSL_FLAG(bool,
+          enable_run_ios_unittests_with_xctest,
+          false,
+          "Intentionally ignored flag intended for iOS test runner.");
+ABSL_FLAG(bool,
+          write_compiled_tests_json_to_writable_path,
+          false,
+          "Intentionally ignored flag intended for iOS test runner.");
+
+// This is the cousin of isolated_script_test_perf_output, but we can't dictate
+// where to write on iOS so the semantics of this flag are a bit different.
+ABSL_FLAG(
+    bool,
+    write_perf_output_on_ios,
+    false,
+    "Store the perf results in Documents/perftest_result.pb in the format "
+    "described by histogram.proto in "
+    "https://chromium.googlesource.com/catapult/.");
+
+#elif defined(WEBRTC_FUCHSIA)
+ABSL_FLAG(std::string, use_vulkan, "", "Intentionally ignored flag.");
+#else
+// TODO(bugs.webrtc.org/8115): Remove workaround when fixed.
+ABSL_FLAG(bool, no_sandbox, false, "Intentionally ignored flag.");
+ABSL_FLAG(bool, test_launcher_bot_mode, false, "Intentionally ignored flag.");
+#endif
+
+ABSL_FLAG(std::string,
+          isolated_script_test_output,
+          "",
+          "Path to output an empty JSON file which Chromium infra requires.");
+
+ABSL_FLAG(bool, logs, true, "print logs to stderr");
+ABSL_FLAG(bool, verbose, false, "verbose logs to stderr");
+
+ABSL_FLAG(std::string,
+          trace_event,
+          "",
+          "Path to collect trace events (json file) for chrome://tracing. "
+          "If not set, events aren't captured.");
+
+ABSL_FLAG(std::string,
+          test_launcher_shard_index,
+          "",
+          "Index of the test shard to run, from 0 to "
+          "the value specified with --test_launcher_total_shards.");
+
+ABSL_FLAG(std::string,
+          test_launcher_total_shards,
+          "",
+          "Total number of shards.");
+
+namespace webrtc {
+
+namespace {
+
+constexpr char kPlotAllMetrics[] = "all";
+
+class TestMainImpl : public TestMain {
+ public:
+  int Init(int* argc, char* argv[]) override { return Init(); }
+
+  int Init() override {
+    // Make sure we always pull in the --resources_dir flag, even if the test
+    // binary doesn't link with fileutils (downstream expects all test mains to
+    // have this flag).
+    (void)absl::GetFlag(FLAGS_resources_dir);
+
+    // Default to LS_INFO, even for release builds to provide better test
+    // logging.
+    if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO)
+      rtc::LogMessage::LogToDebug(rtc::LS_INFO);
+
+    if (absl::GetFlag(FLAGS_verbose))
+      rtc::LogMessage::LogToDebug(rtc::LS_VERBOSE);
+
+    rtc::LogMessage::SetLogToStderr(absl::GetFlag(FLAGS_logs) ||
+                                    absl::GetFlag(FLAGS_verbose));
+
+    // The sharding arguments take precedence over the sharding environment
+    // variables.
+    if (!absl::GetFlag(FLAGS_test_launcher_shard_index).empty() &&
+        !absl::GetFlag(FLAGS_test_launcher_total_shards).empty()) {
+      std::string shard_index =
+          "GTEST_SHARD_INDEX=" + absl::GetFlag(FLAGS_test_launcher_shard_index);
+      std::string total_shards =
+          "GTEST_TOTAL_SHARDS=" +
+          absl::GetFlag(FLAGS_test_launcher_total_shards);
+      putenv(shard_index.data());
+      putenv(total_shards.data());
+    }
+
+    // InitFieldTrialsFromString stores the char*, so the char array must
+    // outlive the application.
+    field_trials_ = absl::GetFlag(FLAGS_force_fieldtrials);
+    webrtc::field_trial::InitFieldTrialsFromString(field_trials_.c_str());
+    webrtc::metrics::Enable();
+
+#if defined(WEBRTC_WIN)
+    winsock_init_ = std::make_unique<rtc::WinsockInitializer>();
+#endif
+
+    // Initialize SSL which are used by several tests.
+    rtc::InitializeSSL();
+    rtc::SSLStreamAdapter::EnableTimeCallbackForTesting();
+
+    return 0;
+  }
+
+  int Run(int argc, char* argv[]) override {
+    std::string trace_event_path = absl::GetFlag(FLAGS_trace_event);
+    const bool capture_events = !trace_event_path.empty();
+    if (capture_events) {
+      rtc::tracing::SetupInternalTracer();
+      rtc::tracing::StartInternalCapture(trace_event_path);
+    }
+
+    absl::optional<std::vector<std::string>> metrics_to_plot =
+        absl::GetFlag(FLAGS_plot);
+
+    if (metrics_to_plot->empty()) {
+      metrics_to_plot = absl::nullopt;
+    } else {
+      if (metrics_to_plot->size() == 1 &&
+          (*metrics_to_plot)[0] == kPlotAllMetrics) {
+        metrics_to_plot->clear();
+      }
+    }
+
+#if defined(WEBRTC_IOS)
+    rtc::test::InitTestSuite(
+        RUN_ALL_TESTS, argc, argv,
+        absl::GetFlag(FLAGS_write_perf_output_on_ios),
+        absl::GetFlag(FLAGS_export_perf_results_new_api),
+        absl::GetFlag(FLAGS_webrtc_test_metrics_output_path), metrics_to_plot);
+    rtc::test::RunTestsFromIOSApp();
+    int exit_code = 0;
+#else
+    int exit_code = RUN_ALL_TESTS();
+
+    std::vector<std::unique_ptr<test::MetricsExporter>> exporters;
+    if (absl::GetFlag(FLAGS_export_perf_results_new_api)) {
+      exporters.push_back(std::make_unique<test::StdoutMetricsExporter>());
+      if (!absl::GetFlag(FLAGS_webrtc_test_metrics_output_path).empty()) {
+        exporters.push_back(
+            std::make_unique<webrtc::test::MetricsSetProtoFileExporter>(
+                webrtc::test::MetricsSetProtoFileExporter::Options(
+                    absl::GetFlag(FLAGS_webrtc_test_metrics_output_path))));
+      }
+      if (!absl::GetFlag(FLAGS_isolated_script_test_perf_output).empty()) {
+        exporters.push_back(
+            std::make_unique<test::ChromePerfDashboardMetricsExporter>(
+                absl::GetFlag(FLAGS_isolated_script_test_perf_output)));
+      }
+    } else {
+      exporters.push_back(
+          std::make_unique<test::PrintResultProxyMetricsExporter>());
+    }
+    test::ExportPerfMetric(*test::GetGlobalMetricsLogger(),
+                           std::move(exporters));
+    if (!absl::GetFlag(FLAGS_export_perf_results_new_api)) {
+      std::string perf_output_file =
+          absl::GetFlag(FLAGS_isolated_script_test_perf_output);
+      if (!perf_output_file.empty()) {
+        if (!webrtc::test::WritePerfResults(perf_output_file)) {
+          return 1;
+        }
+      }
+      if (metrics_to_plot) {
+        webrtc::test::PrintPlottableResults(*metrics_to_plot);
+      }
+    }
+
+    std::string result_filename =
+        absl::GetFlag(FLAGS_isolated_script_test_output);
+    if (!result_filename.empty()) {
+      std::ofstream result_file(result_filename);
+      result_file << "{\"version\": 3}";
+    }
+#endif
+
+    if (capture_events) {
+      rtc::tracing::StopInternalCapture();
+    }
+
+#if defined(ADDRESS_SANITIZER) || defined(LEAK_SANITIZER) ||  \
+    defined(MEMORY_SANITIZER) || defined(THREAD_SANITIZER) || \
+    defined(UNDEFINED_SANITIZER)
+    // We want the test flagged as failed only for sanitizer defects,
+    // in which case the sanitizer will override exit code with 66.
+    exit_code = 0;
+#endif
+
+    return exit_code;
+  }
+
+  ~TestMainImpl() override = default;
+
+ private:
+#if defined(WEBRTC_WIN)
+  std::unique_ptr<rtc::WinsockInitializer> winsock_init_;
+#endif
+};
+
+}  // namespace
+
+std::unique_ptr<TestMain> TestMain::Create() {
+  return std::make_unique<TestMainImpl>();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/test_main_lib.h b/third_party/libwebrtc/test/test_main_lib.h
new file mode 100644
index 0000000000..2233171c60
--- /dev/null
+++ b/third_party/libwebrtc/test/test_main_lib.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TEST_MAIN_LIB_H_
+#define TEST_TEST_MAIN_LIB_H_
+
+#include <memory>
+#include <string>
+
+namespace webrtc {
+
+// Class to initialize test environment and run tests.
+class TestMain {
+ public:
+  virtual ~TestMain() {}
+
+  static std::unique_ptr<TestMain> Create();
+
+  // Initializes test environment. Clients can add their own initialization
+  // steps after call to this method and before running tests.
+  // Returns 0 if initialization was successful and non 0 otherwise.
+  virtual int Init() = 0;
+  // Temporary for backward compatibility
+  virtual int Init(int* argc, char* argv[]) = 0;
+
+  // Runs test end return result error code. 0 - no errors.
+  virtual int Run(int argc, char* argv[]) = 0;
+
+ protected:
+  TestMain() = default;
+
+  std::string field_trials_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_TEST_MAIN_LIB_H_
diff --git a/third_party/libwebrtc/test/test_video_capturer.cc b/third_party/libwebrtc/test/test_video_capturer.cc
new file mode 100644
index 0000000000..4a4adc61d7
--- /dev/null
+++ b/third_party/libwebrtc/test/test_video_capturer.cc
@@ -0,0 +1,107 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/test_video_capturer.h"
+
+#include <algorithm>
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "api/video/video_rotation.h"
+
+namespace webrtc {
+namespace test {
+TestVideoCapturer::~TestVideoCapturer() = default;
+
+void TestVideoCapturer::OnOutputFormatRequest(
+    int width,
+    int height,
+    const absl::optional<int>& max_fps) {
+  absl::optional<std::pair<int, int>> target_aspect_ratio =
+      std::make_pair(width, height);
+  absl::optional<int> max_pixel_count = width * height;
+  video_adapter_.OnOutputFormatRequest(target_aspect_ratio, max_pixel_count,
+                                       max_fps);
+}
+
+void TestVideoCapturer::OnFrame(const VideoFrame& original_frame) {
+  int cropped_width = 0;
+  int cropped_height = 0;
+  int out_width = 0;
+  int out_height = 0;
+
+  VideoFrame frame = MaybePreprocess(original_frame);
+
+  if (!video_adapter_.AdaptFrameResolution(
+          frame.width(), frame.height(), frame.timestamp_us() * 1000,
+          &cropped_width, &cropped_height, &out_width, &out_height)) {
+    // Drop frame in order to respect frame rate constraint.
+    return;
+  }
+
+  if (out_height != frame.height() || out_width != frame.width()) {
+    // Video adapter has requested a down-scale. Allocate a new buffer and
+    // return scaled version.
+    // For simplicity, only scale here without cropping.
+    rtc::scoped_refptr<I420Buffer> scaled_buffer =
+        I420Buffer::Create(out_width, out_height);
+    scaled_buffer->ScaleFrom(*frame.video_frame_buffer()->ToI420());
+    VideoFrame::Builder new_frame_builder =
+        VideoFrame::Builder()
+            .set_video_frame_buffer(scaled_buffer)
+            .set_rotation(kVideoRotation_0)
+            .set_timestamp_us(frame.timestamp_us())
+            .set_id(frame.id());
+    if (frame.has_update_rect()) {
+      VideoFrame::UpdateRect new_rect = frame.update_rect().ScaleWithFrame(
+          frame.width(), frame.height(), 0, 0, frame.width(), frame.height(),
+          out_width, out_height);
+      new_frame_builder.set_update_rect(new_rect);
+    }
+    broadcaster_.OnFrame(new_frame_builder.build());
+
+  } else {
+    // No adaptations needed, just return the frame as is.
+    broadcaster_.OnFrame(frame);
+  }
+}
+
+rtc::VideoSinkWants TestVideoCapturer::GetSinkWants() {
+  return broadcaster_.wants();
+}
+
+void TestVideoCapturer::AddOrUpdateSink(
+    rtc::VideoSinkInterface<VideoFrame>* sink,
+    const rtc::VideoSinkWants& wants) {
+  broadcaster_.AddOrUpdateSink(sink, wants);
+  UpdateVideoAdapter();
+}
+
+void TestVideoCapturer::RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
+  broadcaster_.RemoveSink(sink);
+  UpdateVideoAdapter();
+}
+
+void TestVideoCapturer::UpdateVideoAdapter() {
+  video_adapter_.OnSinkWants(broadcaster_.wants());
+}
+
+VideoFrame TestVideoCapturer::MaybePreprocess(const VideoFrame& frame) {
+  MutexLock lock(&lock_);
+  if (preprocessor_ != nullptr) {
+    return preprocessor_->Preprocess(frame);
+  } else {
+    return frame;
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/test_video_capturer.h b/third_party/libwebrtc/test/test_video_capturer.h
new file mode 100644
index 0000000000..6fafd96efb
--- /dev/null
+++ b/third_party/libwebrtc/test/test_video_capturer.h
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TEST_VIDEO_CAPTURER_H_
+#define TEST_TEST_VIDEO_CAPTURER_H_
+
+#include <stddef.h>
+
+#include <memory>
+
+#include "api/video/video_frame.h"
+#include "api/video/video_source_interface.h"
+#include "media/base/video_adapter.h"
+#include "media/base/video_broadcaster.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace test {
+
+class TestVideoCapturer : public rtc::VideoSourceInterface<VideoFrame> {
+ public:
+  class FramePreprocessor {
+   public:
+    virtual ~FramePreprocessor() = default;
+
+    virtual VideoFrame Preprocess(const VideoFrame& frame) = 0;
+  };
+
+  ~TestVideoCapturer() override;
+
+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
+                       const rtc::VideoSinkWants& wants) override;
+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override;
+  void SetFramePreprocessor(std::unique_ptr<FramePreprocessor> preprocessor) {
+    MutexLock lock(&lock_);
+    preprocessor_ = std::move(preprocessor);
+  }
+  void OnOutputFormatRequest(int width,
+                             int height,
+                             const absl::optional<int>& max_fps);
+
+ protected:
+  void OnFrame(const VideoFrame& frame);
+  rtc::VideoSinkWants GetSinkWants();
+
+ private:
+  void UpdateVideoAdapter();
+  VideoFrame MaybePreprocess(const VideoFrame& frame);
+
+  Mutex lock_;
+  std::unique_ptr<FramePreprocessor> preprocessor_ RTC_GUARDED_BY(lock_);
+  rtc::VideoBroadcaster broadcaster_;
+  cricket::VideoAdapter video_adapter_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TEST_VIDEO_CAPTURER_H_
diff --git a/third_party/libwebrtc/test/testsupport/DEPS b/third_party/libwebrtc/test/testsupport/DEPS
new file mode 100644
index 0000000000..6f6150ad30
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/DEPS
@@ -0,0 +1,4 @@
+include_rules = [
+  # Histogram C++ API, used by perf tests.
+  "+third_party/catapult/tracing/tracing/value"
+]
diff --git a/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.cc b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.cc
new file mode 100644
index 0000000000..6de8e7fd99
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.cc
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/copy_to_file_audio_capturer.h"
+
+#include <memory>
+#include <utility>
+
+namespace webrtc {
+namespace test {
+
+CopyToFileAudioCapturer::CopyToFileAudioCapturer(
+    std::unique_ptr<TestAudioDeviceModule::Capturer> delegate,
+    std::string stream_dump_file_name)
+    : delegate_(std::move(delegate)),
+      wav_writer_(std::make_unique<WavWriter>(std::move(stream_dump_file_name),
+                                              delegate_->SamplingFrequency(),
+                                              delegate_->NumChannels())) {}
+CopyToFileAudioCapturer::~CopyToFileAudioCapturer() = default;
+
+int CopyToFileAudioCapturer::SamplingFrequency() const {
+  return delegate_->SamplingFrequency();
+}
+
+int CopyToFileAudioCapturer::NumChannels() const {
+  return delegate_->NumChannels();
+}
+
+bool CopyToFileAudioCapturer::Capture(rtc::BufferT<int16_t>* buffer) {
+  bool result = delegate_->Capture(buffer);
+  if (result) {
+    wav_writer_->WriteSamples(buffer->data(), buffer->size());
+  }
+  return result;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h
new file mode 100644
index 0000000000..a410beeea8
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_COPY_TO_FILE_AUDIO_CAPTURER_H_
+#define TEST_TESTSUPPORT_COPY_TO_FILE_AUDIO_CAPTURER_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_device/include/test_audio_device.h"
+#include "rtc_base/buffer.h"
+
+namespace webrtc {
+namespace test {
+
+// TestAudioDeviceModule::Capturer that will store audio data, captured by
+// delegate to the specified output file. Can be used to create a copy of
+// generated audio data to be able then to compare it as a reference with
+// audio on the TestAudioDeviceModule::Renderer side.
+class CopyToFileAudioCapturer : public TestAudioDeviceModule::Capturer {
+ public:
+  CopyToFileAudioCapturer(
+      std::unique_ptr<TestAudioDeviceModule::Capturer> delegate,
+      std::string stream_dump_file_name);
+  ~CopyToFileAudioCapturer() override;
+
+  int SamplingFrequency() const override;
+  int NumChannels() const override;
+  bool Capture(rtc::BufferT<int16_t>* buffer) override;
+
+ private:
+  std::unique_ptr<TestAudioDeviceModule::Capturer> delegate_;
+  std::unique_ptr<WavWriter> wav_writer_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_COPY_TO_FILE_AUDIO_CAPTURER_H_
diff --git a/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer_unittest.cc b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer_unittest.cc
new file mode 100644
index 0000000000..3831c28580
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/copy_to_file_audio_capturer_unittest.cc
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/copy_to_file_audio_capturer.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/audio_device/include/test_audio_device.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+class CopyToFileAudioCapturerTest : public ::testing::Test {
+ protected:
+  void SetUp() override {
+    temp_filename_ = webrtc::test::TempFilename(
+        webrtc::test::OutputPath(), "copy_to_file_audio_capturer_unittest");
+    std::unique_ptr<TestAudioDeviceModule::Capturer> delegate =
+        TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, 48000);
+    capturer_ = std::make_unique<CopyToFileAudioCapturer>(std::move(delegate),
+                                                          temp_filename_);
+  }
+
+  void TearDown() override { ASSERT_EQ(remove(temp_filename_.c_str()), 0); }
+
+  std::unique_ptr<CopyToFileAudioCapturer> capturer_;
+  std::string temp_filename_;
+};
+
+TEST_F(CopyToFileAudioCapturerTest, Capture) {
+  rtc::BufferT<int16_t> expected_buffer;
+  ASSERT_TRUE(capturer_->Capture(&expected_buffer));
+  ASSERT_TRUE(!expected_buffer.empty());
+  // Destruct capturer to close wav file.
+  capturer_.reset(nullptr);
+
+  // Read resulted file content with `wav_file_capture` and compare with
+  // what was captured.
+  std::unique_ptr<TestAudioDeviceModule::Capturer> wav_file_capturer =
+      TestAudioDeviceModule::CreateWavFileReader(temp_filename_, 48000);
+  rtc::BufferT<int16_t> actual_buffer;
+  wav_file_capturer->Capture(&actual_buffer);
+  ASSERT_EQ(actual_buffer, expected_buffer);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/file_utils.cc b/third_party/libwebrtc/test/testsupport/file_utils.cc
new file mode 100644
index 0000000000..ff0d5a854c
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/file_utils.cc
@@ -0,0 +1,250 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/file_utils.h"
+
+
+#if defined(WEBRTC_POSIX)
+#include <unistd.h>
+#endif
+
+#if defined(WEBRTC_WIN)
+#include <direct.h>
+#include <tchar.h>
+#include <windows.h>
+
+#include <algorithm>
+#include <codecvt>
+#include <locale>
+
+#include "Shlwapi.h"
+#include "WinDef.h"
+#include "rtc_base/win32.h"
+
+#define GET_CURRENT_DIR _getcwd
+#else
+#include <dirent.h>
+
+#define GET_CURRENT_DIR getcwd
+#endif
+
+#include <sys/stat.h>  // To check for directory existence.
+#ifndef S_ISDIR        // Not defined in stat.h on Windows.
+#define S_ISDIR(mode) (((mode)&S_IFMT) == S_IFDIR)
+#endif
+
+#include <stdio.h>
+#include <stdlib.h>
+
+#include <memory>
+#include <type_traits>
+#include <utility>
+
+#if defined(WEBRTC_IOS)
+#include "test/testsupport/ios_file_utils.h"
+#elif defined(WEBRTC_MAC)
+#include "test/testsupport/mac_file_utils.h"
+#endif
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/testsupport/file_utils_override.h"
+
+namespace webrtc {
+namespace test {
+
+#if defined(WEBRTC_WIN)
+ABSL_CONST_INIT const absl::string_view kPathDelimiter = "\\";
+#else
+ABSL_CONST_INIT const absl::string_view kPathDelimiter = "/";
+#endif
+
+std::string DirName(absl::string_view path) {
+  if (path.empty())
+    return "";
+  if (path == kPathDelimiter)
+    return std::string(path);
+
+  if (path.back() == kPathDelimiter[0])
+    path.remove_suffix(1);  // Remove trailing separator.
+
+  return std::string(path.substr(0, path.find_last_of(kPathDelimiter)));
+}
+
+bool FileExists(absl::string_view file_name) {
+  struct stat file_info = {0};
+  return stat(std::string(file_name).c_str(), &file_info) == 0;
+}
+
+bool DirExists(absl::string_view directory_name) {
+  struct stat directory_info = {0};
+  return stat(std::string(directory_name).c_str(), &directory_info) == 0 &&
+         S_ISDIR(directory_info.st_mode);
+}
+
+std::string OutputPath() {
+  return webrtc::test::internal::OutputPath();
+}
+
+std::string WorkingDir() {
+  return webrtc::test::internal::WorkingDir();
+}
+
+// Generate a temporary filename in a safe way.
+// Largely copied from talk/base/{unixfilesystem,win32filesystem}.cc.
+std::string TempFilename(absl::string_view dir, absl::string_view prefix) {
+#ifdef WIN32
+  wchar_t filename[MAX_PATH];
+  if (::GetTempFileNameW(rtc::ToUtf16(dir).c_str(),
+                         rtc::ToUtf16(prefix).c_str(), 0, filename) != 0)
+    return rtc::ToUtf8(filename);
+  RTC_DCHECK_NOTREACHED();
+  return "";
+#else
+  rtc::StringBuilder os;
+  os << dir << "/" << prefix << "XXXXXX";
+  std::string tempname = os.Release();
+
+  int fd = ::mkstemp(tempname.data());
+  if (fd == -1) {
+    RTC_DCHECK_NOTREACHED();
+    return "";
+  } else {
+    ::close(fd);
+  }
+  return tempname;
+#endif
+}
+
+std::string GenerateTempFilename(absl::string_view dir,
+                                 absl::string_view prefix) {
+  std::string filename = TempFilename(dir, prefix);
+  RemoveFile(filename);
+  return filename;
+}
+
+absl::optional<std::vector<std::string>> ReadDirectory(absl::string_view path) {
+  if (path.length() == 0)
+    return absl::optional<std::vector<std::string>>();
+
+  std::string path_str(path);
+
+#if defined(WEBRTC_WIN)
+  // Append separator character if needed.
+  if (path_str.back() != '\\')
+    path_str += '\\';
+
+  // Init.
+  WIN32_FIND_DATAW data;
+  HANDLE handle = ::FindFirstFileW(rtc::ToUtf16(path_str + '*').c_str(), &data);
+  if (handle == INVALID_HANDLE_VALUE)
+    return absl::optional<std::vector<std::string>>();
+
+  // Populate output.
+  std::vector<std::string> found_entries;
+  do {
+    const std::string name = rtc::ToUtf8(data.cFileName);
+    if (name != "." && name != "..")
+      found_entries.emplace_back(path_str + name);
+  } while (::FindNextFileW(handle, &data) == TRUE);
+
+  // Release resources.
+  if (handle != INVALID_HANDLE_VALUE)
+    ::FindClose(handle);
+#else
+  // Append separator character if needed.
+  if (path_str.back() != '/')
+    path_str += '/';
+
+  // Init.
+  DIR* dir = ::opendir(path_str.c_str());
+  if (dir == nullptr)
+    return absl::optional<std::vector<std::string>>();
+
+  // Populate output.
+  std::vector<std::string> found_entries;
+  while (dirent* dirent = readdir(dir)) {
+    const std::string& name = dirent->d_name;
+    if (name != "." && name != "..")
+      found_entries.emplace_back(path_str + name);
+  }
+
+  // Release resources.
+  closedir(dir);
+#endif
+
+  return absl::optional<std::vector<std::string>>(std::move(found_entries));
+}
+
+bool CreateDir(absl::string_view directory_name) {
+  std::string directory_name_str(directory_name);
+  struct stat path_info = {0};
+  // Check if the path exists already:
+  if (stat(directory_name_str.c_str(), &path_info) == 0) {
+    if (!S_ISDIR(path_info.st_mode)) {
+      fprintf(stderr,
+              "Path %s exists but is not a directory! Remove this "
+              "file and re-run to create the directory.\n",
+              directory_name_str.c_str());
+      return false;
+    }
+  } else {
+#ifdef WIN32
+    return _mkdir(directory_name_str.c_str()) == 0;
+#else
+    return mkdir(directory_name_str.c_str(), S_IRWXU | S_IRWXG | S_IRWXO) == 0;
+#endif
+  }
+  return true;
+}
+
+bool RemoveDir(absl::string_view directory_name) {
+#ifdef WIN32
+  return RemoveDirectoryA(std::string(directory_name).c_str()) != FALSE;
+#else
+  return rmdir(std::string(directory_name).c_str()) == 0;
+#endif
+}
+
+bool RemoveFile(absl::string_view file_name) {
+#ifdef WIN32
+  return DeleteFileA(std::string(file_name).c_str()) != FALSE;
+#else
+  return unlink(std::string(file_name).c_str()) == 0;
+#endif
+}
+
+std::string ResourcePath(absl::string_view name, absl::string_view extension) {
+  return webrtc::test::internal::ResourcePath(name, extension);
+}
+
+std::string JoinFilename(absl::string_view dir, absl::string_view name) {
+  RTC_CHECK(!dir.empty()) << "Special cases not implemented.";
+  rtc::StringBuilder os;
+  os << dir << kPathDelimiter << name;
+  return os.Release();
+}
+
+size_t GetFileSize(absl::string_view filename) {
+  FILE* f = fopen(std::string(filename).c_str(), "rb");
+  size_t size = 0;
+  if (f != NULL) {
+    if (fseek(f, 0, SEEK_END) == 0) {
+      size = ftell(f);
+    }
+    fclose(f);
+  }
+  return size;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/file_utils.h b/third_party/libwebrtc/test/testsupport/file_utils.h
new file mode 100644
index 0000000000..ab80ca4454
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/file_utils.h
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#ifndef TEST_TESTSUPPORT_FILE_UTILS_H_
+#define TEST_TESTSUPPORT_FILE_UTILS_H_
+
+#include <string>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+
+namespace webrtc {
+namespace test {
+
+// Slash or backslash, depending on platform.
+ABSL_CONST_INIT extern const absl::string_view kPathDelimiter;
+
+// Returns the absolute path to the output directory where log files and other
+// test artifacts should be put. The output directory is generally a directory
+// named "out" at the project root. This root is assumed to be two levels above
+// where the test binary is located; this is because tests execute in a dir
+// out/Whatever relative to the project root. This convention is also followed
+// in Chromium.
+//
+// The exception is Android where we use /sdcard/ instead.
+//
+// If symbolic links occur in the path they will be resolved and the actual
+// directory will be returned.
+//
+// Returns the path WITH a trailing path delimiter. If the project root is not
+// found, the current working directory ("./") is returned as a fallback.
+std::string OutputPath();
+
+// Generates an empty file with a unique name in the specified directory and
+// returns the file name and path.
+// TODO(titovartem) rename to TempFile and next method to TempFilename
+std::string TempFilename(absl::string_view dir, absl::string_view prefix);
+
+// Generates a unique file name that can be used for file creation. Doesn't
+// create any files.
+std::string GenerateTempFilename(absl::string_view dir,
+                                 absl::string_view prefix);
+
+// Returns a path to a resource file in [project-root]/resources/ dir.
+// Returns an absolute path
+//
+// Arguments:
+//    name - Name of the resource file. If a plain filename (no directory path)
+//           is supplied, the file is assumed to be located in resources/
+//           If a directory path is prepended to the filename, a subdirectory
+//           hierarchy reflecting that path is assumed to be present.
+//    extension - File extension, without the dot, i.e. "bmp" or "yuv".
+std::string ResourcePath(absl::string_view name, absl::string_view extension);
+
+// Joins directory name and file name, separated by the path delimiter.
+std::string JoinFilename(absl::string_view dir, absl::string_view name);
+
+// Gets the current working directory for the executing program.
+// Returns "./" if for some reason it is not possible to find the working
+// directory.
+std::string WorkingDir();
+
+// Reads the content of a directory and, in case of success, returns a vector
+// of strings with one element for each found file or directory. Each element is
+// a path created by prepending `dir` to the file/directory name. "." and ".."
+// are never added in the returned vector.
+absl::optional<std::vector<std::string>> ReadDirectory(absl::string_view path);
+
+// Creates a directory if it not already exists.
+// Returns true if successful. Will print an error message to stderr and return
+// false if a file with the same name already exists.
+bool CreateDir(absl::string_view directory_name);
+
+// Removes a directory, which must already be empty.
+bool RemoveDir(absl::string_view directory_name);
+
+// Removes a file.
+bool RemoveFile(absl::string_view file_name);
+
+// Checks if a file exists.
+// TOOD(alito): Merge these once absl::string_view adoption is complete for this
+// file.
+bool FileExists(absl::string_view file_name);
+
+// Checks if a directory exists.
+bool DirExists(absl::string_view directory_name);
+
+// Strips the rightmost path segment from a path.
+std::string DirName(absl::string_view path);
+
+// File size of the supplied file in bytes. Will return 0 if the file is
+// empty or if the file does not exist/is readable.
+size_t GetFileSize(absl::string_view filename);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_FILE_UTILS_H_
diff --git a/third_party/libwebrtc/test/testsupport/file_utils_override.cc b/third_party/libwebrtc/test/testsupport/file_utils_override.cc
new file mode 100644
index 0000000000..7d0a3e3312
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/file_utils_override.cc
@@ -0,0 +1,170 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/file_utils_override.h"
+
+#include <limits.h>
+#include <stdio.h>
+
+#if defined(WEBRTC_WIN)
+#include <direct.h>
+#include <tchar.h>
+#include <windows.h>
+
+#include <algorithm>
+#include <codecvt>
+#include <locale>
+
+#include "Shlwapi.h"
+#include "WinDef.h"
+#include "rtc_base/win32.h"
+
+#define GET_CURRENT_DIR _getcwd
+#else
+#include <unistd.h>
+
+#define GET_CURRENT_DIR getcwd
+#endif
+
+#if defined(WEBRTC_IOS)
+#include "test/testsupport/ios_file_utils.h"
+#endif
+
+#if defined(WEBRTC_MAC)
+#include "test/testsupport/mac_file_utils.h"
+#endif
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace test {
+
+std::string DirName(absl::string_view path);
+bool CreateDir(absl::string_view directory_name);
+
+namespace internal {
+
+namespace {
+#if defined(WEBRTC_WIN)
+const absl::string_view kPathDelimiter = "\\";
+#elif !defined(WEBRTC_IOS)
+const absl::string_view kPathDelimiter = "/";
+#endif
+
+#if defined(WEBRTC_ANDROID)
+// This is a special case in Chrome infrastructure. See
+// base/test/test_support_android.cc.
+const absl::string_view kAndroidChromiumTestsRoot =
+    "/sdcard/chromium_tests_root/";
+#endif
+#if defined(WEBRTC_FUCHSIA)
+const absl::string_view kFuchsiaTestRoot = "/pkg/";
+const absl::string_view kFuchsiaTempWritableDir = "/tmp/";
+#endif
+#if !defined(WEBRTC_IOS)
+const absl::string_view kResourcesDirName = "resources";
+#endif
+
+}  // namespace
+
+// Finds the WebRTC src dir.
+// The returned path always ends with a path separator.
+absl::optional<std::string> ProjectRootPath() {
+#if defined(WEBRTC_ANDROID)
+  return std::string(kAndroidChromiumTestsRoot);
+#elif defined WEBRTC_IOS
+  return IOSRootPath();
+#elif defined(WEBRTC_MAC)
+  std::string path;
+  GetNSExecutablePath(&path);
+  std::string exe_dir = DirName(path);
+  // On Mac, tests execute in out/Whatever, so src is two levels up except if
+  // the test is bundled (which our tests are not), in which case it's 5 levels.
+  return DirName(DirName(exe_dir)) + std::string(kPathDelimiter);
+#elif defined(WEBRTC_POSIX)
+// Fuchsia uses POSIX defines as well but does not have full POSIX
+// functionality.
+#if defined(WEBRTC_FUCHSIA)
+  return std::string(kFuchsiaTestRoot);
+#else
+  char buf[PATH_MAX];
+  ssize_t count = ::readlink("/proc/self/exe", buf, arraysize(buf));
+  if (count <= 0) {
+    RTC_DCHECK_NOTREACHED() << "Unable to resolve /proc/self/exe.";
+    return absl::nullopt;
+  }
+  // On POSIX, tests execute in out/Whatever, so src is two levels up.
+  std::string exe_dir = DirName(absl::string_view(buf, count));
+  return DirName(DirName(exe_dir)) + std::string(kPathDelimiter);
+#endif
+#elif defined(WEBRTC_WIN)
+  wchar_t buf[MAX_PATH];
+  buf[0] = 0;
+  if (GetModuleFileNameW(NULL, buf, MAX_PATH) == 0)
+    return absl::nullopt;
+
+  std::string exe_path = rtc::ToUtf8(std::wstring(buf));
+  std::string exe_dir = DirName(exe_path);
+  return DirName(DirName(exe_dir)) + std::string(kPathDelimiter);
+#endif
+}
+
+std::string OutputPath() {
+#if defined(WEBRTC_IOS)
+  return IOSOutputPath();
+#elif defined(WEBRTC_ANDROID)
+  return std::string(kAndroidChromiumTestsRoot);
+#elif defined(WEBRTC_FUCHSIA)
+  return std::string(kFuchsiaTempWritableDir);
+#else
+  absl::optional<std::string> path_opt = ProjectRootPath();
+  RTC_DCHECK(path_opt);
+  std::string path = *path_opt + "out";
+  if (!CreateDir(path)) {
+    return "./";
+  }
+  return path + std::string(kPathDelimiter);
+#endif
+}
+
+std::string WorkingDir() {
+#if defined(WEBRTC_ANDROID)
+  return std::string(kAndroidChromiumTestsRoot);
+#else
+  char path_buffer[FILENAME_MAX];
+  if (!GET_CURRENT_DIR(path_buffer, sizeof(path_buffer))) {
+    fprintf(stderr, "Cannot get current directory!\n");
+    return "./";
+  } else {
+    return std::string(path_buffer);
+  }
+#endif
+}
+
+std::string ResourcePath(absl::string_view name, absl::string_view extension) {
+#if defined(WEBRTC_IOS)
+  return IOSResourcePath(name, extension);
+#else
+  absl::optional<std::string> path_opt = ProjectRootPath();
+  RTC_DCHECK(path_opt);
+  rtc::StringBuilder os(*path_opt);
+  os << kResourcesDirName << kPathDelimiter << name << "." << extension;
+  return os.Release();
+#endif
+}
+
+}  // namespace internal
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/file_utils_override.h b/third_party/libwebrtc/test/testsupport/file_utils_override.h
new file mode 100644
index 0000000000..9f119e6d4e
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/file_utils_override.h
@@ -0,0 +1,57 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_FILE_UTILS_OVERRIDE_H_
+#define TEST_TESTSUPPORT_FILE_UTILS_OVERRIDE_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+namespace internal {
+
+// Returns the absolute path to the output directory where log files and other
+// test artifacts should be put. The output directory is generally a directory
+// named "out" at the project root. This root is assumed to be two levels above
+// where the test binary is located; this is because tests execute in a dir
+// out/Whatever relative to the project root. This convention is also followed
+// in Chromium.
+//
+// The exception is Android where we use /sdcard/ instead.
+//
+// If symbolic links occur in the path they will be resolved and the actual
+// directory will be returned.
+//
+// Returns the path WITH a trailing path delimiter. If the project root is not
+// found, the current working directory ("./") is returned as a fallback.
+std::string OutputPath();
+
+// Gets the current working directory for the executing program.
+// Returns "./" if for some reason it is not possible to find the working
+// directory.
+std::string WorkingDir();
+
+// Returns a full path to a resource file in the resources_dir dir.
+//
+// Arguments:
+//    name - Name of the resource file. If a plain filename (no directory path)
+//           is supplied, the file is assumed to be located in resources/
+//           If a directory path is prepended to the filename, a subdirectory
+//           hierarchy reflecting that path is assumed to be present.
+//    extension - File extension, without the dot, i.e. "bmp" or "yuv".
+std::string ResourcePath(absl::string_view name, absl::string_view extension);
+
+}  // namespace internal
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_FILE_UTILS_OVERRIDE_H_
diff --git a/third_party/libwebrtc/test/testsupport/file_utils_unittest.cc b/third_party/libwebrtc/test/testsupport/file_utils_unittest.cc
new file mode 100644
index 0000000000..b9de01d09d
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/file_utils_unittest.cc
@@ -0,0 +1,277 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/file_utils.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <fstream>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+#ifdef WIN32
+#define chdir _chdir
+#endif
+
+using ::testing::EndsWith;
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+std::string Path(absl::string_view path) {
+  std::string result(path);
+  std::replace(result.begin(), result.end(), '/', kPathDelimiter[0]);
+  return result;
+}
+
+// Remove files and directories in a directory non-recursively and writes the
+// number of deleted items in `num_deleted_entries`.
+void CleanDir(absl::string_view dir, size_t* num_deleted_entries) {
+  RTC_DCHECK(num_deleted_entries);
+  *num_deleted_entries = 0;
+  absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
+  EXPECT_TRUE(dir_content);
+  for (const auto& entry : *dir_content) {
+    if (DirExists(entry)) {
+      EXPECT_TRUE(RemoveDir(entry));
+      (*num_deleted_entries)++;
+    } else if (FileExists(entry)) {
+      EXPECT_TRUE(RemoveFile(entry));
+      (*num_deleted_entries)++;
+    } else {
+      FAIL();
+    }
+  }
+}
+
+void WriteStringInFile(absl::string_view what, absl::string_view file_path) {
+  std::ofstream out(std::string{file_path});
+  out << what;
+  out.close();
+}
+
+}  // namespace
+
+// Test fixture to restore the working directory between each test, since some
+// of them change it with chdir during execution (not restored by the
+// gtest framework).
+class FileUtilsTest : public ::testing::Test {
+ protected:
+  FileUtilsTest() {}
+  ~FileUtilsTest() override {}
+  // Runs before the first test
+  static void SetUpTestSuite() {
+    original_working_dir_ = webrtc::test::WorkingDir();
+  }
+  void SetUp() override { ASSERT_EQ(chdir(original_working_dir_.c_str()), 0); }
+  void TearDown() override {
+    ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
+  }
+
+ private:
+  static std::string original_working_dir_;
+};
+
+std::string FileUtilsTest::original_working_dir_ = "";
+
+// The location will vary depending on where the webrtc checkout is on the
+// system, but it should end as described above and be an absolute path.
+std::string ExpectedRootDirByPlatform() {
+#if defined(WEBRTC_ANDROID)
+  return Path("chromium_tests_root/");
+#elif defined(WEBRTC_IOS)
+  return Path("tmp/");
+#else
+  return Path("out/");
+#endif
+}
+
+TEST_F(FileUtilsTest, OutputPathFromUnchangedWorkingDir) {
+  std::string expected_end = ExpectedRootDirByPlatform();
+  std::string result = webrtc::test::OutputPath();
+
+  ASSERT_THAT(result, EndsWith(expected_end));
+}
+
+// Tests with current working directory set to a directory higher up in the
+// directory tree than the project root dir.
+TEST_F(FileUtilsTest, OutputPathFromRootWorkingDir) {
+  ASSERT_EQ(0, chdir(kPathDelimiter.data()));
+
+  std::string expected_end = ExpectedRootDirByPlatform();
+  std::string result = webrtc::test::OutputPath();
+
+  ASSERT_THAT(result, EndsWith(expected_end));
+}
+
+TEST_F(FileUtilsTest, TempFilename) {
+  std::string temp_filename = webrtc::test::TempFilename(
+      webrtc::test::OutputPath(), "TempFilenameTest");
+  ASSERT_TRUE(webrtc::test::FileExists(temp_filename))
+      << "Couldn't find file: " << temp_filename;
+  remove(temp_filename.c_str());
+}
+
+TEST_F(FileUtilsTest, GenerateTempFilename) {
+  std::string temp_filename = webrtc::test::GenerateTempFilename(
+      webrtc::test::OutputPath(), "TempFilenameTest");
+  ASSERT_FALSE(webrtc::test::FileExists(temp_filename))
+      << "File exists: " << temp_filename;
+  FILE* file = fopen(temp_filename.c_str(), "wb");
+  ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename;
+  ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0)
+      << "Failed to write to file: " << temp_filename;
+  fclose(file);
+  remove(temp_filename.c_str());
+}
+
+// Only tests that the code executes
+#if defined(WEBRTC_IOS)
+#define MAYBE_CreateDir DISABLED_CreateDir
+#else
+#define MAYBE_CreateDir CreateDir
+#endif
+TEST_F(FileUtilsTest, MAYBE_CreateDir) {
+  std::string directory = "fileutils-unittest-empty-dir";
+  // Make sure it's removed if a previous test has failed:
+  remove(directory.c_str());
+  ASSERT_TRUE(webrtc::test::CreateDir(directory));
+  remove(directory.c_str());
+}
+
+TEST_F(FileUtilsTest, WorkingDirReturnsValue) {
+  // This will obviously be different depending on where the webrtc checkout is,
+  // so just check something is returned.
+  std::string working_dir = webrtc::test::WorkingDir();
+  ASSERT_GT(working_dir.length(), 0u);
+}
+
+TEST_F(FileUtilsTest, ResourcePathReturnsCorrectPath) {
+  std::string result = webrtc::test::ResourcePath(
+      Path("video_coding/frame-ethernet-ii"), "pcap");
+#if defined(WEBRTC_IOS)
+  // iOS bundles resources straight into the bundle root.
+  std::string expected_end = Path("/frame-ethernet-ii.pcap");
+#else
+  // Other platforms: it's a separate dir.
+  std::string expected_end =
+      Path("resources/video_coding/frame-ethernet-ii.pcap");
+#endif
+
+  ASSERT_THAT(result, EndsWith(expected_end));
+  ASSERT_TRUE(FileExists(result)) << "Expected " << result
+                                  << " to exist; did "
+                                     "ResourcePath return an incorrect path?";
+}
+
+TEST_F(FileUtilsTest, ResourcePathFromRootWorkingDir) {
+  ASSERT_EQ(0, chdir(kPathDelimiter.data()));
+  std::string resource = webrtc::test::ResourcePath("whatever", "ext");
+#if !defined(WEBRTC_IOS)
+  ASSERT_NE(resource.find("resources"), std::string::npos);
+#endif
+  ASSERT_GT(resource.find("whatever"), 0u);
+  ASSERT_GT(resource.find("ext"), 0u);
+}
+
+TEST_F(FileUtilsTest, GetFileSizeExistingFile) {
+  // Create a file with some dummy data in.
+  std::string temp_filename = webrtc::test::TempFilename(
+      webrtc::test::OutputPath(), "fileutils_unittest");
+  FILE* file = fopen(temp_filename.c_str(), "wb");
+  ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename;
+  ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0)
+      << "Failed to write to file: " << temp_filename;
+  fclose(file);
+  ASSERT_GT(webrtc::test::GetFileSize(temp_filename), 0u);
+  remove(temp_filename.c_str());
+}
+
+TEST_F(FileUtilsTest, GetFileSizeNonExistingFile) {
+  ASSERT_EQ(0u, webrtc::test::GetFileSize("non-existing-file.tmp"));
+}
+
+TEST_F(FileUtilsTest, DirExists) {
+  // Check that an existing directory is recognized as such.
+  ASSERT_TRUE(webrtc::test::DirExists(webrtc::test::OutputPath()))
+      << "Existing directory not found";
+
+  // Check that a non-existing directory is recognized as such.
+  std::string directory = "direxists-unittest-non_existing-dir";
+  ASSERT_FALSE(webrtc::test::DirExists(directory))
+      << "Non-existing directory found";
+
+  // Check that an existing file is not recognized as an existing directory.
+  std::string temp_filename = webrtc::test::TempFilename(
+      webrtc::test::OutputPath(), "TempFilenameTest");
+  ASSERT_TRUE(webrtc::test::FileExists(temp_filename))
+      << "Couldn't find file: " << temp_filename;
+  ASSERT_FALSE(webrtc::test::DirExists(temp_filename))
+      << "Existing file recognized as existing directory";
+  remove(temp_filename.c_str());
+}
+
+TEST_F(FileUtilsTest, WriteReadDeleteFilesAndDirs) {
+  size_t num_deleted_entries;
+
+  // Create an empty temporary directory for this test.
+  const std::string temp_directory =
+      OutputPath() + Path("TempFileUtilsTestReadDirectory/");
+  CreateDir(temp_directory);
+  EXPECT_NO_FATAL_FAILURE(CleanDir(temp_directory, &num_deleted_entries));
+  EXPECT_TRUE(DirExists(temp_directory));
+
+  // Add a file.
+  const std::string temp_filename = temp_directory + "TempFilenameTest";
+  WriteStringInFile("test\n", temp_filename);
+  EXPECT_TRUE(FileExists(temp_filename));
+
+  // Add an empty directory.
+  const std::string temp_subdir = temp_directory + Path("subdir/");
+  EXPECT_TRUE(CreateDir(temp_subdir));
+  EXPECT_TRUE(DirExists(temp_subdir));
+
+  // Checks.
+  absl::optional<std::vector<std::string>> dir_content =
+      ReadDirectory(temp_directory);
+  EXPECT_TRUE(dir_content);
+  EXPECT_EQ(2u, dir_content->size());
+  EXPECT_NO_FATAL_FAILURE(CleanDir(temp_directory, &num_deleted_entries));
+  EXPECT_EQ(2u, num_deleted_entries);
+  EXPECT_TRUE(RemoveDir(temp_directory));
+  EXPECT_FALSE(DirExists(temp_directory));
+}
+
+TEST_F(FileUtilsTest, DirNameStripsFilename) {
+  EXPECT_EQ(Path("/some/path"), DirName(Path("/some/path/file.txt")));
+}
+
+TEST_F(FileUtilsTest, DirNameKeepsStrippingRightmostPathComponent) {
+  EXPECT_EQ(Path("/some"), DirName(DirName(Path("/some/path/file.txt"))));
+}
+
+TEST_F(FileUtilsTest, DirNameDoesntCareIfAPathEndsInPathSeparator) {
+  EXPECT_EQ(Path("/some"), DirName(Path("/some/path/")));
+}
+
+TEST_F(FileUtilsTest, DirNameStopsAtRoot) {
+  EXPECT_EQ(Path("/"), DirName(Path("/")));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter.cc b/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter.cc
new file mode 100644
index 0000000000..531dade0e8
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter.cc
@@ -0,0 +1,114 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/fixed_fps_video_frame_writer_adapter.h"
+
+#include <cmath>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "api/video/video_sink_interface.h"
+#include "rtc_base/checks.h"
+#include "test/testsupport/video_frame_writer.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr TimeDelta kOneSecond = TimeDelta::Seconds(1);
+
+}  // namespace
+
+FixedFpsVideoFrameWriterAdapter::FixedFpsVideoFrameWriterAdapter(
+    int fps,
+    Clock* clock,
+    std::unique_ptr<VideoFrameWriter> delegate)
+    : inter_frame_interval_(kOneSecond / fps),
+      clock_(clock),
+      delegate_(std::move(delegate)) {}
+
+FixedFpsVideoFrameWriterAdapter::~FixedFpsVideoFrameWriterAdapter() {
+  Close();
+}
+
+void FixedFpsVideoFrameWriterAdapter::Close() {
+  if (is_closed_) {
+    return;
+  }
+  is_closed_ = true;
+  if (!last_frame_.has_value()) {
+    return;
+  }
+  Timestamp now = Now();
+  RTC_CHECK(WriteMissedSlotsExceptLast(now));
+  RTC_CHECK(delegate_->WriteFrame(*last_frame_));
+  delegate_->Close();
+}
+
+bool FixedFpsVideoFrameWriterAdapter::WriteFrame(const VideoFrame& frame) {
+  RTC_CHECK(!is_closed_);
+  Timestamp now = Now();
+  if (!last_frame_.has_value()) {
+    RTC_CHECK(!last_frame_time_.IsFinite());
+    last_frame_ = frame;
+    last_frame_time_ = now;
+    return true;
+  }
+
+  RTC_CHECK(last_frame_time_.IsFinite());
+
+  if (last_frame_time_ > now) {
+    // New frame was recevied before expected time "slot" for current
+    // `last_frame_` came => just replace current `last_frame_` with
+    // received `frame`.
+    RTC_CHECK_LE(last_frame_time_ - now, inter_frame_interval_ / 2);
+    last_frame_ = frame;
+    return true;
+  }
+
+  if (!WriteMissedSlotsExceptLast(now)) {
+    return false;
+  }
+
+  if (now - last_frame_time_ < inter_frame_interval_ / 2) {
+    // New frame was received closer to the expected time "slot" for current
+    // `last_frame_` than to the next "slot" => just replace current
+    // `last_frame_` with received `frame`.
+    last_frame_ = frame;
+    return true;
+  }
+
+  if (!delegate_->WriteFrame(*last_frame_)) {
+    return false;
+  }
+  last_frame_ = frame;
+  last_frame_time_ = last_frame_time_ + inter_frame_interval_;
+  return true;
+}
+
+bool FixedFpsVideoFrameWriterAdapter::WriteMissedSlotsExceptLast(
+    Timestamp now) {
+  RTC_CHECK(last_frame_time_.IsFinite());
+  while (now - last_frame_time_ > inter_frame_interval_) {
+    if (!delegate_->WriteFrame(*last_frame_)) {
+      return false;
+    }
+    last_frame_time_ = last_frame_time_ + inter_frame_interval_;
+  }
+  return true;
+}
+
+Timestamp FixedFpsVideoFrameWriterAdapter::Now() const {
+  return clock_->CurrentTime();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter.h b/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter.h
new file mode 100644
index 0000000000..d4d95e9f82
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter.h
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_FIXED_FPS_VIDEO_FRAME_WRITER_ADAPTER_H_
+#define TEST_TESTSUPPORT_FIXED_FPS_VIDEO_FRAME_WRITER_ADAPTER_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/test/video/video_frame_writer.h"
+#include "api/video/video_sink_interface.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/video_frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+// Writes video to the specified video writer with specified fixed frame rate.
+// If at the point in time X no new frames are passed to the writer, the
+// previous frame is used to fill the gap and preserve frame rate.
+//
+// This adaptor uses next algorithm:
+// There are output "slots" at a fixed frame rate (starting at the time of the
+// first received frame). Each incoming frame is assigned to the closest output
+// slot. Then empty slots are filled by repeating the closest filled slot before
+// empty one. If there are multiple frames closest to the same slot, the latest
+// received one is used.
+//
+// The frames are outputted for the whole duration of the class life after the
+// first frame was written or until it will be closed.
+//
+// For example if frames from A to F were received, then next output sequence
+// will be generated:
+// Received frames:  A            B       C      D  EF        Destructor called
+//                   |            |       |      |  ||          |
+//                   v            v       v      v  vv          v
+//                   X----X----X----X----X----X----X----X----X----+----+--
+//                   |    |    |    |    |    |    |    |    |
+// Produced frames:  A    A    A    B    C    C    F    F    F
+//
+// This class is not thread safe.
+class FixedFpsVideoFrameWriterAdapter : public VideoFrameWriter {
+ public:
+  FixedFpsVideoFrameWriterAdapter(int fps,
+                                  Clock* clock,
+                                  std::unique_ptr<VideoFrameWriter> delegate);
+  ~FixedFpsVideoFrameWriterAdapter() override;
+
+  bool WriteFrame(const webrtc::VideoFrame& frame) override;
+
+  // Closes adapter and underlying delegate. User mustn't call to the WriteFrame
+  // after calling this method.
+  void Close() override;
+
+ private:
+  // Writes `last_frame_` for each "slot" from `last_frame_time_` up to now
+  // excluding the last one.
+  // Updates `last_frame_time_` to the position of the last NOT WRITTEN frame.
+  // Returns true if all writes were successful, otherwise retuns false. In such
+  // case it is not guranteed how many frames were actually written.
+  bool WriteMissedSlotsExceptLast(Timestamp now);
+  Timestamp Now() const;
+
+  // Because `TimeDelta` stores time with microseconds precision
+  // `last_frame_time_` may have a small drift and for very long streams it
+  // must be updated to use double for time.
+  const TimeDelta inter_frame_interval_;
+  Clock* const clock_;
+  std::unique_ptr<VideoFrameWriter> delegate_;
+  bool is_closed_ = false;
+
+  // Expected time slot for the last frame.
+  Timestamp last_frame_time_ = Timestamp::MinusInfinity();
+  absl::optional<VideoFrame> last_frame_ = absl::nullopt;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_FIXED_FPS_VIDEO_FRAME_WRITER_ADAPTER_H_
diff --git a/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter_test.cc b/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter_test.cc
new file mode 100644
index 0000000000..5ee4701cc9
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/fixed_fps_video_frame_writer_adapter_test.cc
@@ -0,0 +1,320 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/fixed_fps_video_frame_writer_adapter.h"
+
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/video_frame_writer.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr TimeDelta kOneSecond = TimeDelta::Seconds(1);
+
+using ::testing::ElementsAre;
+
+class InMemoryVideoWriter : public VideoFrameWriter {
+ public:
+  ~InMemoryVideoWriter() override = default;
+
+  bool WriteFrame(const webrtc::VideoFrame& frame) override {
+    MutexLock lock(&mutex_);
+    received_frames_.push_back(frame);
+    return true;
+  }
+
+  void Close() override {}
+
+  std::vector<VideoFrame> received_frames() const {
+    MutexLock lock(&mutex_);
+    return received_frames_;
+  }
+
+ private:
+  mutable Mutex mutex_;
+  std::vector<VideoFrame> received_frames_ RTC_GUARDED_BY(mutex_);
+};
+
+VideoFrame EmptyFrameWithId(uint16_t frame_id) {
+  return VideoFrame::Builder()
+      .set_video_frame_buffer(I420Buffer::Create(1, 1))
+      .set_id(frame_id)
+      .build();
+}
+
+std::vector<uint16_t> FrameIds(const std::vector<VideoFrame>& frames) {
+  std::vector<uint16_t> out;
+  for (const VideoFrame& frame : frames) {
+    out.push_back(frame.id());
+  }
+  return out;
+}
+
+std::unique_ptr<TimeController> CreateSimulatedTimeController() {
+  // Using an offset of 100000 to get nice fixed width and readable
+  // timestamps in typical test scenarios.
+  const Timestamp kSimulatedStartTime = Timestamp::Seconds(100000);
+  return std::make_unique<GlobalSimulatedTimeController>(kSimulatedStartTime);
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     WhenWrittenWithSameFpsVideoIsCorrect) {
+  auto time_controller = CreateSimulatedTimeController();
+  int fps = 25;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(fps, time_controller->GetClock(),
+                                               std::move(inmemory_writer));
+
+  for (int i = 1; i <= 30; ++i) {
+    video_writer.WriteFrame(EmptyFrameWithId(i));
+    time_controller->AdvanceTime(kOneSecond / fps);
+  }
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(
+      FrameIds(received_frames),
+      ElementsAre(1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18,
+                  19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest, FrameIsRepeatedWhenThereIsAFreeze) {
+  auto time_controller = CreateSimulatedTimeController();
+  int fps = 25;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(fps, time_controller->GetClock(),
+                                               std::move(inmemory_writer));
+
+  // Write 10 frames
+  for (int i = 1; i <= 10; ++i) {
+    video_writer.WriteFrame(EmptyFrameWithId(i));
+    time_controller->AdvanceTime(kOneSecond / fps);
+  }
+
+  // Freeze for 4 frames
+  time_controller->AdvanceTime(4 * kOneSecond / fps);
+
+  // Write 10 more frames
+  for (int i = 11; i <= 20; ++i) {
+    video_writer.WriteFrame(EmptyFrameWithId(i));
+    time_controller->AdvanceTime(kOneSecond / fps);
+  }
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames),
+              ElementsAre(1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 10, 10, 10, 10, 11, 12,
+                          13, 14, 15, 16, 17, 18, 19, 20));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest, NoFramesWritten) {
+  auto time_controller = CreateSimulatedTimeController();
+  int fps = 25;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(fps, time_controller->GetClock(),
+                                               std::move(inmemory_writer));
+  time_controller->AdvanceTime(TimeDelta::Millis(100));
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  ASSERT_TRUE(received_frames.empty());
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     FreezeInTheMiddleAndNewFrameReceivedBeforeMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(2.3 * kInterval);
+  video_writer.WriteFrame(EmptyFrameWithId(2));
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(1, 1, 2));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     FreezeInTheMiddleAndNewFrameReceivedAfterMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(2.5 * kInterval);
+  video_writer.WriteFrame(EmptyFrameWithId(2));
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(1, 1, 1, 2));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     NewFrameReceivedBeforeMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(0.3 * kInterval);
+  video_writer.WriteFrame(EmptyFrameWithId(2));
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(2));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     NewFrameReceivedAfterMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(0.5 * kInterval);
+  video_writer.WriteFrame(EmptyFrameWithId(2));
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(1, 2));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     FreeezeAtTheEndAndDestroyBeforeMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(2.3 * kInterval);
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(1, 1, 1));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     FreeezeAtTheEndAndDestroyAfterMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(2.5 * kInterval);
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(1, 1, 1));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     DestroyBeforeMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(0.3 * kInterval);
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(1));
+}
+
+TEST(FixedFpsVideoFrameWriterAdapterTest,
+     DestroyAfterMiddleOfExpectedInterval) {
+  auto time_controller = CreateSimulatedTimeController();
+  constexpr int kFps = 10;
+  constexpr TimeDelta kInterval = kOneSecond / kFps;
+
+  auto inmemory_writer = std::make_unique<InMemoryVideoWriter>();
+  InMemoryVideoWriter* inmemory_writer_ref = inmemory_writer.get();
+
+  FixedFpsVideoFrameWriterAdapter video_writer(
+      kFps, time_controller->GetClock(), std::move(inmemory_writer));
+  video_writer.WriteFrame(EmptyFrameWithId(1));
+  time_controller->AdvanceTime(0.5 * kInterval);
+  video_writer.Close();
+
+  std::vector<VideoFrame> received_frames =
+      inmemory_writer_ref->received_frames();
+  EXPECT_THAT(FrameIds(received_frames), ElementsAre(1));
+}
+
+}  // namespace
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/frame_reader.h b/third_party/libwebrtc/test/testsupport/frame_reader.h
new file mode 100644
index 0000000000..7856476ca0
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/frame_reader.h
@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_FRAME_READER_H_
+#define TEST_TESTSUPPORT_FRAME_READER_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/scoped_refptr.h"
+#include "api/video/resolution.h"
+
+namespace webrtc {
+class I420Buffer;
+namespace test {
+
+// Handles reading of I420 frames from video files.
+class FrameReader {
+ public:
+  struct Ratio {
+    int num = 1;
+    int den = 1;
+  };
+
+  static constexpr Ratio kNoScale = Ratio({.num = 1, .den = 1});
+
+  virtual ~FrameReader() {}
+
+  // Reads and returns next frame. Returns `nullptr` if reading failed or end of
+  // stream is reached.
+  virtual rtc::scoped_refptr<I420Buffer> PullFrame() = 0;
+
+  // Reads and returns next frame. `frame_num` stores unwrapped frame number
+  // which can be passed to `ReadFrame` to re-read this frame later. Returns
+  // `nullptr` if reading failed or end of stream is reached.
+  virtual rtc::scoped_refptr<I420Buffer> PullFrame(int* frame_num) = 0;
+
+  // Reads and returns frame specified by `frame_num`. Returns `nullptr` if
+  // reading failed.
+  virtual rtc::scoped_refptr<I420Buffer> ReadFrame(int frame_num) = 0;
+
+  // Reads next frame, resizes and returns it. `frame_num` stores unwrapped
+  // frame number which can be passed to `ReadFrame` to re-read this frame
+  // later. `resolution` specifies resolution of the returned frame.
+  // `framerate_scale` specifies frame rate scale factor. Frame rate scaling is
+  // done by skipping or repeating frames.
+  virtual rtc::scoped_refptr<I420Buffer> PullFrame(int* frame_num,
+                                                   Resolution resolution,
+                                                   Ratio framerate_scale) = 0;
+
+  // Reads frame specified by `frame_num`, resizes and returns it. Returns
+  // `nullptr` if reading failed.
+  virtual rtc::scoped_refptr<I420Buffer> ReadFrame(int frame_num,
+                                                   Resolution resolution) = 0;
+
+  // Total number of retrievable frames.
+  virtual int num_frames() const = 0;
+};
+
+class YuvFrameReaderImpl : public FrameReader {
+ public:
+  enum class RepeatMode { kSingle, kRepeat, kPingPong };
+
+  // Creates the frame reader for a YUV file specified by `filepath`.
+  // `resolution` specifies width and height of frames in pixels. `repeat_mode`
+  // specifies behaviour of the reader at reaching the end of file (stop, read
+  // it over from the beginning or read in reverse order). The file is assumed
+  // to exist, be readable and to contain at least 1 frame.
+  YuvFrameReaderImpl(std::string filepath,
+                     Resolution resolution,
+                     RepeatMode repeat_mode);
+
+  ~YuvFrameReaderImpl() override;
+
+  virtual void Init();
+
+  rtc::scoped_refptr<I420Buffer> PullFrame() override;
+
+  rtc::scoped_refptr<I420Buffer> PullFrame(int* frame_num) override;
+
+  rtc::scoped_refptr<I420Buffer> PullFrame(int* frame_num,
+                                           Resolution resolution,
+                                           Ratio framerate_scale) override;
+
+  rtc::scoped_refptr<I420Buffer> ReadFrame(int frame_num) override;
+
+  rtc::scoped_refptr<I420Buffer> ReadFrame(int frame_num,
+                                           Resolution resolution) override;
+
+  int num_frames() const override { return num_frames_; }
+
+ protected:
+  class RateScaler {
+   public:
+    int Skip(Ratio framerate_scale);
+
+   private:
+    absl::optional<int> ticks_;
+  };
+
+  const std::string filepath_;
+  Resolution resolution_;
+  const RepeatMode repeat_mode_;
+  int num_frames_;
+  int frame_num_;
+  int frame_size_bytes_;
+  int header_size_bytes_;
+  FILE* file_;
+  RateScaler framerate_scaler_;
+};
+
+class Y4mFrameReaderImpl : public YuvFrameReaderImpl {
+ public:
+  // Creates the frame reader for a Y4M file specified by `filepath`.
+  // `repeat_mode` specifies behaviour of the reader at reaching the end of file
+  // (stop, read it over from the beginning or read in reverse order). The file
+  // is assumed to exist, be readable and to contain at least 1 frame.
+  Y4mFrameReaderImpl(std::string filepath, RepeatMode repeat_mode);
+
+  void Init() override;
+};
+
+std::unique_ptr<FrameReader> CreateYuvFrameReader(std::string filepath,
+                                                  Resolution resolution);
+
+std::unique_ptr<FrameReader> CreateYuvFrameReader(
+    std::string filepath,
+    Resolution resolution,
+    YuvFrameReaderImpl::RepeatMode repeat_mode);
+
+std::unique_ptr<FrameReader> CreateY4mFrameReader(std::string filepath);
+
+std::unique_ptr<FrameReader> CreateY4mFrameReader(
+    std::string filepath,
+    YuvFrameReaderImpl::RepeatMode repeat_mode);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_FRAME_READER_H_
diff --git a/third_party/libwebrtc/test/testsupport/frame_writer.h b/third_party/libwebrtc/test/testsupport/frame_writer.h
new file mode 100644
index 0000000000..5f85d8bcd4
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/frame_writer.h
@@ -0,0 +1,104 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_FRAME_WRITER_H_
+#define TEST_TESTSUPPORT_FRAME_WRITER_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "api/video/video_frame.h"
+
+namespace webrtc {
+namespace test {
+
+// Handles writing of video files.
+class FrameWriter {
+ public:
+  virtual ~FrameWriter() {}
+
+  // Initializes the file handler, i.e. opens the input and output files etc.
+  // This must be called before reading or writing frames has started.
+  // Returns false if an error has occurred, in addition to printing to stderr.
+  virtual bool Init() = 0;
+
+  // Writes a frame of the configured frame length to the output file.
+  // Returns true if the write was successful, false otherwise.
+  virtual bool WriteFrame(const uint8_t* frame_buffer) = 0;
+
+  // Closes the output file if open. Essentially makes this class impossible
+  // to use anymore. Will also be invoked by the destructor.
+  virtual void Close() = 0;
+
+  // Frame length in bytes of a single frame image.
+  virtual size_t FrameLength() = 0;
+};
+
+// Writes raw I420 frames in sequence.
+class YuvFrameWriterImpl : public FrameWriter {
+ public:
+  // Creates a file handler. The input file is assumed to exist and be readable
+  // and the output file must be writable.
+  // Parameters:
+  //   output_filename         The file to write. Will be overwritten if already
+  //                           existing.
+  //   width, height           Size of each frame to read.
+  YuvFrameWriterImpl(std::string output_filename, int width, int height);
+  ~YuvFrameWriterImpl() override;
+  bool Init() override;
+  bool WriteFrame(const uint8_t* frame_buffer) override;
+  void Close() override;
+  size_t FrameLength() override;
+
+ protected:
+  const std::string output_filename_;
+  size_t frame_length_in_bytes_;
+  const int width_;
+  const int height_;
+  FILE* output_file_;
+};
+
+// Writes raw I420 frames in sequence, but with Y4M file and frame headers for
+// more convenient playback in external media players.
+class Y4mFrameWriterImpl : public YuvFrameWriterImpl {
+ public:
+  Y4mFrameWriterImpl(std::string output_filename,
+                     int width,
+                     int height,
+                     int frame_rate);
+  ~Y4mFrameWriterImpl() override;
+  bool Init() override;
+  bool WriteFrame(const uint8_t* frame_buffer) override;
+
+ private:
+  const int frame_rate_;
+};
+
+// LibJpeg is not available on iOS. This class will do nothing on iOS.
+class JpegFrameWriter {
+ public:
+  JpegFrameWriter(const std::string& output_filename);
+  // Quality can be from 0 (worst) to 100 (best). Best quality is still lossy.
+  // WriteFrame can be called only once. Subsequent calls will fail.
+  bool WriteFrame(const VideoFrame& input_frame, int quality);
+
+#if !defined(WEBRTC_IOS)
+ private:
+  bool frame_written_;
+  const std::string output_filename_;
+  FILE* output_file_;
+#endif
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_FRAME_WRITER_H_
diff --git a/third_party/libwebrtc/test/testsupport/ios_file_utils.h b/third_party/libwebrtc/test/testsupport/ios_file_utils.h
new file mode 100644
index 0000000000..49df3b9010
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/ios_file_utils.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_IOS_FILE_UTILS_H_
+#define TEST_TESTSUPPORT_IOS_FILE_UTILS_H_
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+namespace webrtc {
+namespace test {
+
+std::string IOSOutputPath();
+std::string IOSRootPath();
+std::string IOSResourcePath(absl::string_view name,
+                            absl::string_view extension);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_IOS_FILE_UTILS_H_
diff --git a/third_party/libwebrtc/test/testsupport/ios_file_utils.mm b/third_party/libwebrtc/test/testsupport/ios_file_utils.mm
new file mode 100644
index 0000000000..ef36937e6a
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/ios_file_utils.mm
@@ -0,0 +1,61 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#if defined(WEBRTC_IOS)
+
+#import <Foundation/Foundation.h>
+#include <string.h>
+
+#import "sdk/objc/helpers/NSString+StdString.h"
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+// For iOS, resource files are added to the application bundle in the root
+// and not in separate folders as is the case for other platforms. This method
+// therefore removes any prepended folders and uses only the actual file name.
+std::string IOSResourcePath(absl::string_view name, absl::string_view extension) {
+  @autoreleasepool {
+    NSString* path = [NSString stringForAbslStringView:name];
+    NSString* fileName = path.lastPathComponent;
+    NSString* fileType = [NSString stringForAbslStringView:extension];
+    // Get full pathname for the resource identified by the name and extension.
+    NSString* pathString = [[NSBundle mainBundle] pathForResource:fileName
+                                                           ofType:fileType];
+    return [NSString stdStringForString:pathString];
+  }
+}
+
+std::string IOSRootPath() {
+  @autoreleasepool {
+    NSBundle* mainBundle = [NSBundle mainBundle];
+    return [NSString stdStringForString:mainBundle.bundlePath] + "/";
+  }
+}
+
+// For iOS, we don't have access to the output directory. Return the path to the
+// temporary directory instead. This is mostly used by tests that need to write
+// output files to disk.
+std::string IOSOutputPath()  {
+  @autoreleasepool {
+    NSString* tempDir = NSTemporaryDirectory();
+    if (tempDir == nil)
+        tempDir = @"/tmp";
+    return [NSString stdStringForString:tempDir];
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // defined(WEBRTC_IOS)
diff --git a/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator.cc b/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator.cc
new file mode 100644
index 0000000000..92700e192f
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator.cc
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/ivf_video_frame_generator.h"
+
+#include <limits>
+
+#include "api/video/encoded_image.h"
+#include "api/video/i420_buffer.h"
+#include "api/video_codecs/video_codec.h"
+#include "media/base/media_constants.h"
+#include "modules/video_coding/codecs/h264/include/h264.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/system/file_wrapper.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr TimeDelta kMaxNextFrameWaitTimeout = TimeDelta::Seconds(1);
+
+}  // namespace
+
+IvfVideoFrameGenerator::IvfVideoFrameGenerator(const std::string& file_name)
+    : callback_(this),
+      file_reader_(IvfFileReader::Create(FileWrapper::OpenReadOnly(file_name))),
+      video_decoder_(CreateVideoDecoder(file_reader_->GetVideoCodecType())),
+      width_(file_reader_->GetFrameWidth()),
+      height_(file_reader_->GetFrameHeight()) {
+  RTC_CHECK(video_decoder_) << "No decoder found for file's video codec type";
+  VideoDecoder::Settings decoder_settings;
+  decoder_settings.set_codec_type(file_reader_->GetVideoCodecType());
+  decoder_settings.set_max_render_resolution(
+      {file_reader_->GetFrameWidth(), file_reader_->GetFrameHeight()});
+  // Set buffer pool size to max value to ensure that if users of generator,
+  // ex. test frameworks, will retain frames for quite a long time, decoder
+  // won't crash with buffers pool overflow error.
+  decoder_settings.set_buffer_pool_size(std::numeric_limits<int>::max());
+  RTC_CHECK_EQ(video_decoder_->RegisterDecodeCompleteCallback(&callback_),
+               WEBRTC_VIDEO_CODEC_OK);
+  RTC_CHECK(video_decoder_->Configure(decoder_settings));
+}
+IvfVideoFrameGenerator::~IvfVideoFrameGenerator() {
+  MutexLock lock(&lock_);
+  if (!file_reader_) {
+    return;
+  }
+  file_reader_->Close();
+  file_reader_.reset();
+  // Reset decoder to prevent it from async access to `this`.
+  video_decoder_.reset();
+  {
+    MutexLock frame_lock(&frame_decode_lock_);
+    next_frame_ = absl::nullopt;
+    // Set event in case another thread is waiting on it.
+    next_frame_decoded_.Set();
+  }
+}
+
+FrameGeneratorInterface::VideoFrameData IvfVideoFrameGenerator::NextFrame() {
+  MutexLock lock(&lock_);
+  next_frame_decoded_.Reset();
+  RTC_CHECK(file_reader_);
+  if (!file_reader_->HasMoreFrames()) {
+    file_reader_->Reset();
+  }
+  absl::optional<EncodedImage> image = file_reader_->NextFrame();
+  RTC_CHECK(image);
+  // Last parameter is undocumented and there is no usage of it found.
+  RTC_CHECK_EQ(WEBRTC_VIDEO_CODEC_OK,
+               video_decoder_->Decode(*image, /*missing_frames=*/false,
+                                      /*render_time_ms=*/0));
+  bool decoded = next_frame_decoded_.Wait(kMaxNextFrameWaitTimeout);
+  RTC_CHECK(decoded) << "Failed to decode next frame in "
+                     << kMaxNextFrameWaitTimeout << ". Can't continue";
+
+  MutexLock frame_lock(&frame_decode_lock_);
+  rtc::scoped_refptr<VideoFrameBuffer> buffer =
+      next_frame_->video_frame_buffer();
+  if (width_ != static_cast<size_t>(buffer->width()) ||
+      height_ != static_cast<size_t>(buffer->height())) {
+    // Video adapter has requested a down-scale. Allocate a new buffer and
+    // return scaled version.
+    rtc::scoped_refptr<I420Buffer> scaled_buffer =
+        I420Buffer::Create(width_, height_);
+    scaled_buffer->ScaleFrom(*buffer->ToI420());
+    buffer = scaled_buffer;
+  }
+  return VideoFrameData(buffer, next_frame_->update_rect());
+}
+
+void IvfVideoFrameGenerator::ChangeResolution(size_t width, size_t height) {
+  MutexLock lock(&lock_);
+  width_ = width;
+  height_ = height;
+}
+
+int32_t IvfVideoFrameGenerator::DecodedCallback::Decoded(
+    VideoFrame& decoded_image) {
+  Decoded(decoded_image, 0, 0);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+int32_t IvfVideoFrameGenerator::DecodedCallback::Decoded(
+    VideoFrame& decoded_image,
+    int64_t decode_time_ms) {
+  Decoded(decoded_image, decode_time_ms, 0);
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+void IvfVideoFrameGenerator::DecodedCallback::Decoded(
+    VideoFrame& decoded_image,
+    absl::optional<int32_t> decode_time_ms,
+    absl::optional<uint8_t> qp) {
+  reader_->OnFrameDecoded(decoded_image);
+}
+
+void IvfVideoFrameGenerator::OnFrameDecoded(const VideoFrame& decoded_frame) {
+  MutexLock lock(&frame_decode_lock_);
+  next_frame_ = decoded_frame;
+  next_frame_decoded_.Set();
+}
+
+std::unique_ptr<VideoDecoder> IvfVideoFrameGenerator::CreateVideoDecoder(
+    VideoCodecType codec_type) {
+  if (codec_type == VideoCodecType::kVideoCodecVP8) {
+    return VP8Decoder::Create();
+  }
+  if (codec_type == VideoCodecType::kVideoCodecVP9) {
+    return VP9Decoder::Create();
+  }
+  if (codec_type == VideoCodecType::kVideoCodecH264) {
+    return H264Decoder::Create();
+  }
+  return nullptr;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator.h b/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator.h
new file mode 100644
index 0000000000..4b6d116383
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator.h
@@ -0,0 +1,87 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_IVF_VIDEO_FRAME_GENERATOR_H_
+#define TEST_TESTSUPPORT_IVF_VIDEO_FRAME_GENERATOR_H_
+
+#include <memory>
+#include <string>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/video/video_codec_type.h"
+#include "api/video/video_frame.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/utility/ivf_file_reader.h"
+#include "rtc_base/event.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+namespace test {
+
+// All methods except constructor must be used from the same thread.
+class IvfVideoFrameGenerator : public FrameGeneratorInterface {
+ public:
+  explicit IvfVideoFrameGenerator(const std::string& file_name);
+  ~IvfVideoFrameGenerator() override;
+
+  VideoFrameData NextFrame() override;
+  void ChangeResolution(size_t width, size_t height) override;
+
+ private:
+  class DecodedCallback : public DecodedImageCallback {
+   public:
+    explicit DecodedCallback(IvfVideoFrameGenerator* reader)
+        : reader_(reader) {}
+
+    int32_t Decoded(VideoFrame& decoded_image) override;
+    int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override;
+    void Decoded(VideoFrame& decoded_image,
+                 absl::optional<int32_t> decode_time_ms,
+                 absl::optional<uint8_t> qp) override;
+
+   private:
+    IvfVideoFrameGenerator* const reader_;
+  };
+
+  void OnFrameDecoded(const VideoFrame& decoded_frame);
+  static std::unique_ptr<VideoDecoder> CreateVideoDecoder(
+      VideoCodecType codec_type);
+
+  DecodedCallback callback_;
+  std::unique_ptr<IvfFileReader> file_reader_;
+  std::unique_ptr<VideoDecoder> video_decoder_;
+
+  size_t width_;
+  size_t height_;
+
+  // This lock is used to ensure that all API method will be called
+  // sequentially. It is required because we need to ensure that generator
+  // won't be destroyed while it is reading the next frame on another thread,
+  // because it will cause SIGSEGV when decoder callback will be invoked.
+  //
+  // FrameGenerator is injected into PeerConnection via some scoped_ref object
+  // and it can happen that the last pointer will be destroyed on the different
+  // thread comparing to the one from which frames were read.
+  Mutex lock_;
+  // This lock is used to sync between sending and receiving frame from decoder.
+  // We can't reuse `lock_` because then generator can be destroyed between
+  // frame was sent to decoder and decoder callback was invoked.
+  Mutex frame_decode_lock_;
+
+  rtc::Event next_frame_decoded_;
+  absl::optional<VideoFrame> next_frame_ RTC_GUARDED_BY(frame_decode_lock_);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_IVF_VIDEO_FRAME_GENERATOR_H_
diff --git a/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator_unittest.cc b/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator_unittest.cc
new file mode 100644
index 0000000000..dd60f3d3fc
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/ivf_video_frame_generator_unittest.cc
@@ -0,0 +1,212 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/ivf_video_frame_generator.h"
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/test/create_frame_generator.h"
+#include "api/units/time_delta.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_codec_type.h"
+#include "api/video_codecs/video_codec.h"
+#include "api/video_codecs/video_encoder.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "media/base/codec.h"
+#include "media/base/media_constants.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "modules/video_coding/utility/ivf_file_writer.h"
+#include "rtc_base/event.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/video_codec_settings.h"
+
+#if defined(WEBRTC_USE_H264)
+#include "modules/video_coding/codecs/h264/include/h264.h"
+#include "rtc_base/synchronization/mutex.h"
+
+#endif
+
+namespace webrtc {
+namespace test {
+namespace {
+
+constexpr int kWidth = 320;
+constexpr int kHeight = 240;
+constexpr int kVideoFramesCount = 30;
+constexpr int kMaxFramerate = 30;
+constexpr TimeDelta kMaxFrameEncodeWaitTimeout = TimeDelta::Seconds(2);
+static const VideoEncoder::Capabilities kCapabilities(false);
+
+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM64)
+constexpr double kExpectedMinPsnr = 35;
+#else
+constexpr double kExpectedMinPsnr = 39;
+#endif
+
+class IvfFileWriterEncodedCallback : public EncodedImageCallback {
+ public:
+  IvfFileWriterEncodedCallback(const std::string& file_name,
+                               VideoCodecType video_codec_type,
+                               int expected_frames_count)
+      : file_writer_(
+            IvfFileWriter::Wrap(FileWrapper::OpenWriteOnly(file_name), 0)),
+        video_codec_type_(video_codec_type),
+        expected_frames_count_(expected_frames_count) {
+    EXPECT_TRUE(file_writer_.get());
+  }
+  ~IvfFileWriterEncodedCallback() { EXPECT_TRUE(file_writer_->Close()); }
+
+  Result OnEncodedImage(const EncodedImage& encoded_image,
+                        const CodecSpecificInfo* codec_specific_info) override {
+    EXPECT_TRUE(file_writer_->WriteFrame(encoded_image, video_codec_type_));
+
+    MutexLock lock(&lock_);
+    received_frames_count_++;
+    RTC_CHECK_LE(received_frames_count_, expected_frames_count_);
+    if (received_frames_count_ == expected_frames_count_) {
+      expected_frames_count_received_.Set();
+    }
+    return Result(Result::Error::OK);
+  }
+
+  bool WaitForExpectedFramesReceived(TimeDelta timeout) {
+    return expected_frames_count_received_.Wait(timeout);
+  }
+
+ private:
+  std::unique_ptr<IvfFileWriter> file_writer_;
+  const VideoCodecType video_codec_type_;
+  const int expected_frames_count_;
+
+  Mutex lock_;
+  int received_frames_count_ RTC_GUARDED_BY(lock_) = 0;
+  rtc::Event expected_frames_count_received_;
+};
+
+class IvfVideoFrameGeneratorTest : public ::testing::Test {
+ protected:
+  void SetUp() override {
+    file_name_ =
+        webrtc::test::TempFilename(webrtc::test::OutputPath(), "test_file.ivf");
+  }
+  void TearDown() override { webrtc::test::RemoveFile(file_name_); }
+
+  VideoFrame BuildFrame(FrameGeneratorInterface::VideoFrameData frame_data) {
+    return VideoFrame::Builder()
+        .set_video_frame_buffer(frame_data.buffer)
+        .set_update_rect(frame_data.update_rect)
+        .build();
+  }
+
+  void CreateTestVideoFile(VideoCodecType video_codec_type,
+                           std::unique_ptr<VideoEncoder> video_encoder) {
+    std::unique_ptr<test::FrameGeneratorInterface> frame_generator =
+        test::CreateSquareFrameGenerator(
+            kWidth, kHeight, test::FrameGeneratorInterface::OutputType::kI420,
+            absl::nullopt);
+
+    VideoCodec codec_settings;
+    webrtc::test::CodecSettings(video_codec_type, &codec_settings);
+    codec_settings.width = kWidth;
+    codec_settings.height = kHeight;
+    codec_settings.maxFramerate = kMaxFramerate;
+    const uint32_t kBitrateBps = 500000;
+    VideoBitrateAllocation bitrate_allocation;
+    bitrate_allocation.SetBitrate(0, 0, kBitrateBps);
+
+    IvfFileWriterEncodedCallback ivf_writer_callback(
+        file_name_, video_codec_type, kVideoFramesCount);
+
+    video_encoder->RegisterEncodeCompleteCallback(&ivf_writer_callback);
+    video_encoder->SetRates(VideoEncoder::RateControlParameters(
+        bitrate_allocation, static_cast<double>(codec_settings.maxFramerate)));
+    ASSERT_EQ(WEBRTC_VIDEO_CODEC_OK,
+              video_encoder->InitEncode(
+                  &codec_settings,
+                  VideoEncoder::Settings(kCapabilities, /*number_of_cores=*/1,
+                                         /*max_payload_size=*/0)));
+
+    uint32_t last_frame_timestamp = 0;
+
+    for (int i = 0; i < kVideoFramesCount; ++i) {
+      VideoFrame frame = BuildFrame(frame_generator->NextFrame());
+      const uint32_t timestamp =
+          last_frame_timestamp +
+          kVideoPayloadTypeFrequency / codec_settings.maxFramerate;
+      frame.set_timestamp(timestamp);
+
+      last_frame_timestamp = timestamp;
+
+      ASSERT_EQ(WEBRTC_VIDEO_CODEC_OK, video_encoder->Encode(frame, nullptr));
+      video_frames_.push_back(frame);
+    }
+
+    ASSERT_TRUE(ivf_writer_callback.WaitForExpectedFramesReceived(
+        kMaxFrameEncodeWaitTimeout));
+  }
+
+  std::string file_name_;
+  std::vector<VideoFrame> video_frames_;
+};
+
+}  // namespace
+
+TEST_F(IvfVideoFrameGeneratorTest, Vp8) {
+  CreateTestVideoFile(VideoCodecType::kVideoCodecVP8, VP8Encoder::Create());
+  IvfVideoFrameGenerator generator(file_name_);
+  for (size_t i = 0; i < video_frames_.size(); ++i) {
+    auto& expected_frame = video_frames_[i];
+    VideoFrame actual_frame = BuildFrame(generator.NextFrame());
+    EXPECT_GT(I420PSNR(&expected_frame, &actual_frame), kExpectedMinPsnr);
+  }
+}
+
+TEST_F(IvfVideoFrameGeneratorTest, Vp8DoubleRead) {
+  CreateTestVideoFile(VideoCodecType::kVideoCodecVP8, VP8Encoder::Create());
+  IvfVideoFrameGenerator generator(file_name_);
+  for (size_t i = 0; i < video_frames_.size() * 2; ++i) {
+    auto& expected_frame = video_frames_[i % video_frames_.size()];
+    VideoFrame actual_frame = BuildFrame(generator.NextFrame());
+    EXPECT_GT(I420PSNR(&expected_frame, &actual_frame), kExpectedMinPsnr);
+  }
+}
+
+TEST_F(IvfVideoFrameGeneratorTest, Vp9) {
+  CreateTestVideoFile(VideoCodecType::kVideoCodecVP9, VP9Encoder::Create());
+  IvfVideoFrameGenerator generator(file_name_);
+  for (size_t i = 0; i < video_frames_.size(); ++i) {
+    auto& expected_frame = video_frames_[i];
+    VideoFrame actual_frame = BuildFrame(generator.NextFrame());
+    EXPECT_GT(I420PSNR(&expected_frame, &actual_frame), kExpectedMinPsnr);
+  }
+}
+
+#if defined(WEBRTC_USE_H264)
+TEST_F(IvfVideoFrameGeneratorTest, H264) {
+  CreateTestVideoFile(
+      VideoCodecType::kVideoCodecH264,
+      H264Encoder::Create(cricket::VideoCodec(cricket::kH264CodecName)));
+  IvfVideoFrameGenerator generator(file_name_);
+  for (size_t i = 0; i < video_frames_.size(); ++i) {
+    auto& expected_frame = video_frames_[i];
+    VideoFrame actual_frame = BuildFrame(generator.NextFrame());
+    EXPECT_GT(I420PSNR(&expected_frame, &actual_frame), kExpectedMinPsnr);
+  }
+}
+#endif
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/jpeg_frame_writer.cc b/third_party/libwebrtc/test/testsupport/jpeg_frame_writer.cc
new file mode 100644
index 0000000000..8bf1ee4630
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/jpeg_frame_writer.cc
@@ -0,0 +1,88 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/testsupport/frame_writer.h"
+
+extern "C" {
+#if defined(USE_SYSTEM_LIBJPEG)
+#include <jpeglib.h>
+#else
+// Include directory supplied by gn
+#include "jpeglib.h"  // NOLINT
+#endif
+}
+
+namespace webrtc {
+namespace test {
+
+JpegFrameWriter::JpegFrameWriter(const std::string& output_filename)
+    : frame_written_(false),
+      output_filename_(output_filename),
+      output_file_(nullptr) {}
+
+bool JpegFrameWriter::WriteFrame(const VideoFrame& input_frame, int quality) {
+  if (frame_written_) {
+    RTC_LOG(LS_ERROR) << "Only a single frame can be saved to Jpeg.";
+    return false;
+  }
+  const int kColorPlanes = 3;  // R, G and B.
+  size_t rgb_len = input_frame.height() * input_frame.width() * kColorPlanes;
+  std::unique_ptr<uint8_t[]> rgb_buf(new uint8_t[rgb_len]);
+
+  // kRGB24 actually corresponds to FourCC 24BG which is 24-bit BGR.
+  if (ConvertFromI420(input_frame, VideoType::kRGB24, 0, rgb_buf.get()) < 0) {
+    RTC_LOG(LS_ERROR) << "Could not convert input frame to RGB.";
+    return false;
+  }
+  output_file_ = fopen(output_filename_.c_str(), "wb");
+  if (!output_file_) {
+    RTC_LOG(LS_ERROR) << "Couldn't open file to write jpeg frame to:"
+                      << output_filename_;
+    return false;
+  }
+
+  // Invoking LIBJPEG
+  struct jpeg_compress_struct cinfo;
+  struct jpeg_error_mgr jerr;
+  JSAMPROW row_pointer[1];
+  cinfo.err = jpeg_std_error(&jerr);
+  jpeg_create_compress(&cinfo);
+
+  jpeg_stdio_dest(&cinfo, output_file_);
+
+  cinfo.image_width = input_frame.width();
+  cinfo.image_height = input_frame.height();
+  cinfo.input_components = kColorPlanes;
+  cinfo.in_color_space = JCS_EXT_BGR;
+  jpeg_set_defaults(&cinfo);
+  jpeg_set_quality(&cinfo, quality, TRUE);
+
+  jpeg_start_compress(&cinfo, TRUE);
+  int row_stride = input_frame.width() * kColorPlanes;
+  while (cinfo.next_scanline < cinfo.image_height) {
+    row_pointer[0] = &rgb_buf.get()[cinfo.next_scanline * row_stride];
+    jpeg_write_scanlines(&cinfo, row_pointer, 1);
+  }
+
+  jpeg_finish_compress(&cinfo);
+  jpeg_destroy_compress(&cinfo);
+  fclose(output_file_);
+
+  frame_written_ = true;
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/jpeg_frame_writer_ios.cc b/third_party/libwebrtc/test/testsupport/jpeg_frame_writer_ios.cc
new file mode 100644
index 0000000000..e72fea102f
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/jpeg_frame_writer_ios.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+JpegFrameWriter::JpegFrameWriter(const std::string& /*output_filename*/) {}
+
+bool JpegFrameWriter::WriteFrame(const VideoFrame& /*input_frame*/,
+                                 int /*quality*/) {
+  RTC_LOG(LS_WARNING)
+      << "Libjpeg isn't available on IOS. Jpeg frame writer is not "
+         "supported. No frame will be saved.";
+  // Don't fail.
+  return true;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/mac_file_utils.h b/third_party/libwebrtc/test/testsupport/mac_file_utils.h
new file mode 100644
index 0000000000..c6cbdc580d
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/mac_file_utils.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_MAC_FILE_UTILS_H_
+#define TEST_TESTSUPPORT_MAC_FILE_UTILS_H_
+
+#include <string>
+
+namespace webrtc {
+namespace test {
+
+void GetNSExecutablePath(std::string* path);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_MAC_FILE_UTILS_H_
diff --git a/third_party/libwebrtc/test/testsupport/mac_file_utils.mm b/third_party/libwebrtc/test/testsupport/mac_file_utils.mm
new file mode 100644
index 0000000000..270ecbc9a5
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/mac_file_utils.mm
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#include <dlfcn.h>
+#include <mach-o/dyld.h>
+#include <stdint.h>
+#include <stdlib.h>
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+void GetNSExecutablePath(std::string* path) {
+  RTC_DCHECK(path);
+  // Executable path can have relative references ("..") depending on
+  // how the app was launched.
+  uint32_t executable_length = 0;
+  _NSGetExecutablePath(NULL, &executable_length);
+  RTC_DCHECK_GT(executable_length, 1u);
+  char executable_path[PATH_MAX + 1];
+  int rv = _NSGetExecutablePath(executable_path, &executable_length);
+  RTC_DCHECK_EQ(rv, 0);
+
+  char full_path[PATH_MAX];
+  if (realpath(executable_path, full_path) == nullptr) {
+    *path = "";
+    return;
+  }
+
+  *path = full_path;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/mock/mock_frame_reader.h b/third_party/libwebrtc/test/testsupport/mock/mock_frame_reader.h
new file mode 100644
index 0000000000..f68bbf8368
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/mock/mock_frame_reader.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
+#define TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
+
+#include "api/video/i420_buffer.h"
+#include "test/gmock.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace test {
+
+class MockFrameReader : public FrameReader {
+ public:
+  MOCK_METHOD(rtc::scoped_refptr<I420Buffer>, PullFrame, (), (override));
+  MOCK_METHOD(rtc::scoped_refptr<I420Buffer>, PullFrame, (int*), (override));
+  MOCK_METHOD(rtc::scoped_refptr<I420Buffer>,
+              PullFrame,
+              (int*, Resolution, Ratio),
+              (override));
+  MOCK_METHOD(rtc::scoped_refptr<I420Buffer>, ReadFrame, (int), (override));
+  MOCK_METHOD(rtc::scoped_refptr<I420Buffer>,
+              ReadFrame,
+              (int, Resolution),
+              (override));
+  MOCK_METHOD(int, num_frames, (), (const override));
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
diff --git a/third_party/libwebrtc/test/testsupport/perf_test.cc b/third_party/libwebrtc/test/testsupport/perf_test.cc
new file mode 100644
index 0000000000..bbea5f841a
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test.cc
@@ -0,0 +1,355 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/perf_test.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <fstream>
+#include <set>
+#include <sstream>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/perf_test_histogram_writer.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+std::string UnitWithDirection(
+    absl::string_view units,
+    webrtc::test::ImproveDirection improve_direction) {
+  switch (improve_direction) {
+    case webrtc::test::ImproveDirection::kNone:
+      return std::string(units);
+    case webrtc::test::ImproveDirection::kSmallerIsBetter:
+      return std::string(units) + "_smallerIsBetter";
+    case webrtc::test::ImproveDirection::kBiggerIsBetter:
+      return std::string(units) + "_biggerIsBetter";
+  }
+}
+
+std::vector<SamplesStatsCounter::StatsSample> GetSortedSamples(
+    const SamplesStatsCounter& counter) {
+  rtc::ArrayView<const SamplesStatsCounter::StatsSample> view =
+      counter.GetTimedSamples();
+  std::vector<SamplesStatsCounter::StatsSample> out(view.begin(), view.end());
+  std::stable_sort(out.begin(), out.end(),
+                   [](const SamplesStatsCounter::StatsSample& a,
+                      const SamplesStatsCounter::StatsSample& b) {
+                     return a.time < b.time;
+                   });
+  return out;
+}
+
+template <typename Container>
+void OutputListToStream(std::ostream* ostream, const Container& values) {
+  const char* sep = "";
+  for (const auto& v : values) {
+    (*ostream) << sep << v;
+    sep = ",";
+  }
+}
+
+struct PlottableCounter {
+  std::string graph_name;
+  std::string trace_name;
+  webrtc::SamplesStatsCounter counter;
+  std::string units;
+};
+
+class PlottableCounterPrinter {
+ public:
+  PlottableCounterPrinter() : output_(stdout) {}
+
+  void SetOutput(FILE* output) {
+    MutexLock lock(&mutex_);
+    output_ = output;
+  }
+
+  void AddCounter(absl::string_view graph_name,
+                  absl::string_view trace_name,
+                  const webrtc::SamplesStatsCounter& counter,
+                  absl::string_view units) {
+    MutexLock lock(&mutex_);
+    plottable_counters_.push_back({std::string(graph_name),
+                                   std::string(trace_name), counter,
+                                   std::string(units)});
+  }
+
+  void Print(const std::vector<std::string>& desired_graphs_raw) const {
+    std::set<std::string> desired_graphs(desired_graphs_raw.begin(),
+                                         desired_graphs_raw.end());
+    MutexLock lock(&mutex_);
+    for (auto& counter : plottable_counters_) {
+      if (!desired_graphs.empty()) {
+        auto it = desired_graphs.find(counter.graph_name);
+        if (it == desired_graphs.end()) {
+          continue;
+        }
+      }
+
+      std::ostringstream value_stream;
+      value_stream.precision(8);
+      value_stream << R"({"graph_name":")" << counter.graph_name << R"(",)";
+      value_stream << R"("trace_name":")" << counter.trace_name << R"(",)";
+      value_stream << R"("units":")" << counter.units << R"(",)";
+      if (!counter.counter.IsEmpty()) {
+        value_stream << R"("mean":)" << counter.counter.GetAverage() << ',';
+        value_stream << R"("std":)" << counter.counter.GetStandardDeviation()
+                     << ',';
+      }
+      value_stream << R"("samples":[)";
+      const char* sep = "";
+      for (const auto& sample : counter.counter.GetTimedSamples()) {
+        value_stream << sep << R"({"time":)" << sample.time.us() << ','
+                     << R"("value":)" << sample.value << '}';
+        sep = ",";
+      }
+      value_stream << "]}";
+
+      fprintf(output_, "PLOTTABLE_DATA: %s\n", value_stream.str().c_str());
+    }
+  }
+
+ private:
+  mutable Mutex mutex_;
+  std::vector<PlottableCounter> plottable_counters_ RTC_GUARDED_BY(&mutex_);
+  FILE* output_ RTC_GUARDED_BY(&mutex_);
+};
+
+PlottableCounterPrinter& GetPlottableCounterPrinter() {
+  static PlottableCounterPrinter* printer_ = new PlottableCounterPrinter();
+  return *printer_;
+}
+
+class ResultsLinePrinter {
+ public:
+  ResultsLinePrinter() : output_(stdout) {}
+
+  void SetOutput(FILE* output) {
+    MutexLock lock(&mutex_);
+    output_ = output;
+  }
+
+  void PrintResult(absl::string_view graph_name,
+                   absl::string_view trace_name,
+                   const double value,
+                   absl::string_view units,
+                   bool important,
+                   ImproveDirection improve_direction) {
+    std::ostringstream value_stream;
+    value_stream.precision(8);
+    value_stream << value;
+
+    PrintResultImpl(graph_name, trace_name, value_stream.str(), std::string(),
+                    std::string(), UnitWithDirection(units, improve_direction),
+                    important);
+  }
+
+  void PrintResultMeanAndError(absl::string_view graph_name,
+                               absl::string_view trace_name,
+                               const double mean,
+                               const double error,
+                               absl::string_view units,
+                               bool important,
+                               ImproveDirection improve_direction) {
+    std::ostringstream value_stream;
+    value_stream.precision(8);
+    value_stream << mean << ',' << error;
+    PrintResultImpl(graph_name, trace_name, value_stream.str(), "{", "}",
+                    UnitWithDirection(units, improve_direction), important);
+  }
+
+  void PrintResultList(absl::string_view graph_name,
+                       absl::string_view trace_name,
+                       const rtc::ArrayView<const double> values,
+                       absl::string_view units,
+                       const bool important,
+                       webrtc::test::ImproveDirection improve_direction) {
+    std::ostringstream value_stream;
+    value_stream.precision(8);
+    OutputListToStream(&value_stream, values);
+    PrintResultImpl(graph_name, trace_name, value_stream.str(), "[", "]", units,
+                    important);
+  }
+
+ private:
+  void PrintResultImpl(absl::string_view graph_name,
+                       absl::string_view trace_name,
+                       absl::string_view values,
+                       absl::string_view prefix,
+                       absl::string_view suffix,
+                       absl::string_view units,
+                       bool important) {
+    MutexLock lock(&mutex_);
+    rtc::StringBuilder message;
+    message << (important ? "*" : "") << "RESULT " << graph_name << ": "
+            << trace_name << "= " << prefix << values << suffix << " " << units;
+    // <*>RESULT <graph_name>: <trace_name>= <value> <units>
+    // <*>RESULT <graph_name>: <trace_name>= {<mean>, <std deviation>} <units>
+    // <*>RESULT <graph_name>: <trace_name>= [<value>,value,value,...,] <units>
+    fprintf(output_, "%s\n", message.str().c_str());
+  }
+
+  Mutex mutex_;
+  FILE* output_ RTC_GUARDED_BY(&mutex_);
+};
+
+ResultsLinePrinter& GetResultsLinePrinter() {
+  static ResultsLinePrinter* const printer_ = new ResultsLinePrinter();
+  return *printer_;
+}
+
+PerfTestResultWriter& GetPerfWriter() {
+  static PerfTestResultWriter* writer = CreateHistogramWriter();
+  return *writer;
+}
+
+}  // namespace
+
+void ClearPerfResults() {
+  GetPerfWriter().ClearResults();
+}
+
+void SetPerfResultsOutput(FILE* output) {
+  GetPlottableCounterPrinter().SetOutput(output);
+  GetResultsLinePrinter().SetOutput(output);
+}
+
+std::string GetPerfResults() {
+  return GetPerfWriter().Serialize();
+}
+
+void PrintPlottableResults(const std::vector<std::string>& desired_graphs) {
+  GetPlottableCounterPrinter().Print(desired_graphs);
+}
+
+bool WritePerfResults(const std::string& output_path) {
+  std::string results = GetPerfResults();
+  CreateDir(DirName(output_path));
+  FILE* output = fopen(output_path.c_str(), "wb");
+  if (output == NULL) {
+    printf("Failed to write to %s.\n", output_path.c_str());
+    return false;
+  }
+  size_t written =
+      fwrite(results.c_str(), sizeof(char), results.size(), output);
+  fclose(output);
+
+  if (written != results.size()) {
+    long expected = results.size();
+    printf("Wrote %zu, tried to write %lu\n", written, expected);
+    return false;
+  }
+
+  return true;
+}
+
+void PrintResult(absl::string_view measurement,
+                 absl::string_view modifier,
+                 absl::string_view trace,
+                 const double value,
+                 absl::string_view units,
+                 bool important,
+                 ImproveDirection improve_direction) {
+  rtc::StringBuilder graph_name;
+  graph_name << measurement << modifier;
+  RTC_CHECK(std::isfinite(value))
+      << "Expected finite value for graph " << graph_name.str()
+      << ", trace name " << trace << ", units " << units << ", got " << value;
+  GetPerfWriter().LogResult(graph_name.str(), trace, value, units, important,
+                            improve_direction);
+  GetResultsLinePrinter().PrintResult(graph_name.str(), trace, value, units,
+                                      important, improve_direction);
+}
+
+void PrintResult(absl::string_view measurement,
+                 absl::string_view modifier,
+                 absl::string_view trace,
+                 const SamplesStatsCounter& counter,
+                 absl::string_view units,
+                 const bool important,
+                 ImproveDirection improve_direction) {
+  rtc::StringBuilder graph_name;
+  graph_name << measurement << modifier;
+  GetPlottableCounterPrinter().AddCounter(graph_name.str(), trace, counter,
+                                          units);
+
+  double mean = counter.IsEmpty() ? 0 : counter.GetAverage();
+  double error = counter.IsEmpty() ? 0 : counter.GetStandardDeviation();
+
+  std::vector<SamplesStatsCounter::StatsSample> timed_samples =
+      GetSortedSamples(counter);
+  std::vector<double> samples(timed_samples.size());
+  for (size_t i = 0; i < timed_samples.size(); ++i) {
+    samples[i] = timed_samples[i].value;
+  }
+  // If we have an empty counter, default it to 0.
+  if (samples.empty()) {
+    samples.push_back(0);
+  }
+
+  GetPerfWriter().LogResultList(graph_name.str(), trace, samples, units,
+                                important, improve_direction);
+  GetResultsLinePrinter().PrintResultMeanAndError(graph_name.str(), trace, mean,
+                                                  error, units, important,
+                                                  improve_direction);
+}
+
+void PrintResultMeanAndError(absl::string_view measurement,
+                             absl::string_view modifier,
+                             absl::string_view trace,
+                             const double mean,
+                             const double error,
+                             absl::string_view units,
+                             bool important,
+                             ImproveDirection improve_direction) {
+  RTC_CHECK(std::isfinite(mean));
+  RTC_CHECK(std::isfinite(error));
+
+  rtc::StringBuilder graph_name;
+  graph_name << measurement << modifier;
+  GetPerfWriter().LogResultMeanAndError(graph_name.str(), trace, mean, error,
+                                        units, important, improve_direction);
+  GetResultsLinePrinter().PrintResultMeanAndError(graph_name.str(), trace, mean,
+                                                  error, units, important,
+                                                  improve_direction);
+}
+
+void PrintResultList(absl::string_view measurement,
+                     absl::string_view modifier,
+                     absl::string_view trace,
+                     const rtc::ArrayView<const double> values,
+                     absl::string_view units,
+                     bool important,
+                     ImproveDirection improve_direction) {
+  for (double v : values) {
+    RTC_CHECK(std::isfinite(v));
+  }
+
+  rtc::StringBuilder graph_name;
+  graph_name << measurement << modifier;
+  GetPerfWriter().LogResultList(graph_name.str(), trace, values, units,
+                                important, improve_direction);
+  GetResultsLinePrinter().PrintResultList(graph_name.str(), trace, values,
+                                          units, important, improve_direction);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/perf_test.h b/third_party/libwebrtc/test/testsupport/perf_test.h
new file mode 100644
index 0000000000..732fff7d14
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test.h
@@ -0,0 +1,124 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_PERF_TEST_H_
+#define TEST_TESTSUPPORT_PERF_TEST_H_
+
+#include <sstream>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/numerics/samples_stats_counter.h"
+
+namespace webrtc {
+namespace test {
+
+enum class ImproveDirection {
+  // Direction is undefined.
+  kNone,
+  // Smaller value is better.
+  kSmallerIsBetter,
+  // Bigger value is better.
+  kBiggerIsBetter,
+};
+
+// Prints a performance test result.
+//
+// For example,
+// PrintResult("ramp_up_time_", "turn_over_tcp",
+//             "bwe_15s", 1234.2, "ms", false);
+//
+// will show up in the http://chromeperf.appspot.com under
+//
+// (test binary name) > (bot) > ramp_up_time_turn_over_tcp > bwe_15s.
+//
+// The `measurement` + `modifier` is what we're measuring. `user_story` is the
+// scenario we're testing under.
+//
+// The binary this runs in must be hooked up as a perf test in the WebRTC
+// recipes for this to actually be uploaded to chromeperf.appspot.com.
+void PrintResult(absl::string_view measurement,
+                 absl::string_view modifier,
+                 absl::string_view user_story,
+                 double value,
+                 absl::string_view units,
+                 bool important,
+                 ImproveDirection improve_direction = ImproveDirection::kNone);
+
+// Like PrintResult(), but prints a (mean, standard deviation) result pair.
+// The |<values>| should be two comma-separated numbers, the mean and
+// standard deviation (or other error metric) of the measurement.
+// DEPRECATED: soon unsupported.
+void PrintResultMeanAndError(
+    absl::string_view measurement,
+    absl::string_view modifier,
+    absl::string_view user_story,
+    double mean,
+    double error,
+    absl::string_view units,
+    bool important,
+    ImproveDirection improve_direction = ImproveDirection::kNone);
+
+// Like PrintResult(), but prints an entire list of results. The `values`
+// will generally be a list of comma-separated numbers. A typical
+// post-processing step might produce plots of their mean and standard
+// deviation.
+void PrintResultList(
+    absl::string_view measurement,
+    absl::string_view modifier,
+    absl::string_view user_story,
+    rtc::ArrayView<const double> values,
+    absl::string_view units,
+    bool important,
+    ImproveDirection improve_direction = ImproveDirection::kNone);
+
+// Like PrintResult(), but prints a (mean, standard deviation) from stats
+// counter. Also add specified metric to the plotable metrics output.
+void PrintResult(absl::string_view measurement,
+                 absl::string_view modifier,
+                 absl::string_view user_story,
+                 const SamplesStatsCounter& counter,
+                 absl::string_view units,
+                 bool important,
+                 ImproveDirection improve_direction = ImproveDirection::kNone);
+
+// Returns a string-encoded proto as described in
+// tracing/tracing/proto/histogram.proto in
+// https://github.com/catapult-project/catapult/blob/master/.
+// If you want to print the proto in human readable format, use
+// tracing/bin/proto2json from third_party/catapult in your WebRTC checkout.
+std::string GetPerfResults();
+
+// Print into stdout plottable metrics for further post processing.
+// `desired_graphs` - list of metrics, that should be plotted. If empty - all
+// available metrics will be plotted. If some of `desired_graphs` are missing
+// they will be skipped.
+void PrintPlottableResults(const std::vector<std::string>& desired_graphs);
+
+// Call GetPerfResults() and write its output to a file. Returns false if we
+// failed to write to the file. If you want to print the proto in human readable
+// format, use tracing/bin/proto2json from third_party/catapult in your WebRTC
+// checkout.
+bool WritePerfResults(const std::string& output_path);
+
+// By default, human-readable perf results are printed to stdout. Set the FILE*
+// to where they should be printing instead. These results are not used to
+// upload to the dashboard, however - this is only through WritePerfResults.
+void SetPerfResultsOutput(FILE* output);
+
+// Only for use by tests.
+void ClearPerfResults();
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_PERF_TEST_H_
diff --git a/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer.cc b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer.cc
new file mode 100644
index 0000000000..93924ba16c
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer.cc
@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/perf_test_histogram_writer.h"
+
+#include <stdlib.h>
+
+#include <map>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "third_party/catapult/tracing/tracing/value/diagnostics/reserved_infos.h"
+#include "third_party/catapult/tracing/tracing/value/histogram.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+namespace proto = catapult::tracing::tracing::proto;
+
+std::string AsJsonString(const std::string string) {
+  return "\"" + string + "\"";
+}
+
+class PerfTestHistogramWriter : public PerfTestResultWriter {
+ public:
+  PerfTestHistogramWriter() : mutex_() {}
+  void ClearResults() override {
+    MutexLock lock(&mutex_);
+    histograms_.clear();
+  }
+
+  void LogResult(absl::string_view graph_name,
+                 absl::string_view trace_name,
+                 const double value,
+                 absl::string_view units,
+                 const bool important,
+                 ImproveDirection improve_direction) override {
+    (void)important;
+    AddSample(graph_name, trace_name, value, units, improve_direction);
+  }
+  void LogResultMeanAndError(absl::string_view graph_name,
+                             absl::string_view trace_name,
+                             const double mean,
+                             const double error,
+                             absl::string_view units,
+                             const bool important,
+                             ImproveDirection improve_direction) override {
+    RTC_LOG(LS_WARNING) << "Discarding stddev, not supported by histograms";
+    (void)error;
+    (void)important;
+
+    AddSample(graph_name, trace_name, mean, units, improve_direction);
+  }
+  void LogResultList(absl::string_view graph_name,
+                     absl::string_view trace_name,
+                     const rtc::ArrayView<const double> values,
+                     absl::string_view units,
+                     const bool important,
+                     ImproveDirection improve_direction) override {
+    (void)important;
+    for (double value : values) {
+      AddSample(graph_name, trace_name, value, units, improve_direction);
+    }
+  }
+  std::string Serialize() const override {
+    proto::HistogramSet histogram_set;
+
+    MutexLock lock(&mutex_);
+    for (const auto& histogram : histograms_) {
+      std::unique_ptr<proto::Histogram> proto = histogram.second->toProto();
+      histogram_set.mutable_histograms()->AddAllocated(proto.release());
+    }
+
+    std::string output;
+    bool ok = histogram_set.SerializeToString(&output);
+    RTC_DCHECK(ok) << "Failed to serialize histogram set to string";
+    return output;
+  }
+
+ private:
+  void AddSample(absl::string_view original_graph_name,
+                 absl::string_view trace_name,
+                 const double value,
+                 absl::string_view units,
+                 ImproveDirection improve_direction) {
+    // WebRTC annotates the units into the metric name when they are not
+    // supported by the Histogram API.
+    std::string graph_name(original_graph_name);
+    if (units == "dB") {
+      graph_name += "_dB";
+    } else if (units == "fps") {
+      graph_name += "_fps";
+    } else if (units == "%") {
+      graph_name += "_%";
+    }
+
+    // Lookup on graph name + trace name (or measurement + story in catapult
+    // parlance). There should be several histograms with the same measurement
+    // if they're for different stories.
+    rtc::StringBuilder measurement_and_story;
+    measurement_and_story << graph_name << trace_name;
+    MutexLock lock(&mutex_);
+    if (histograms_.count(measurement_and_story.str()) == 0) {
+      proto::UnitAndDirection unit = ParseUnit(units, improve_direction);
+      std::unique_ptr<catapult::HistogramBuilder> builder =
+          std::make_unique<catapult::HistogramBuilder>(graph_name, unit);
+
+      // Set all summary options as false - we don't want to generate
+      // metric_std, metric_count, and so on for all metrics.
+      builder->SetSummaryOptions(proto::SummaryOptions());
+      histograms_[measurement_and_story.str()] = std::move(builder);
+
+      proto::Diagnostic stories;
+      proto::GenericSet* generic_set = stories.mutable_generic_set();
+      generic_set->add_values(AsJsonString(std::string(trace_name)));
+      histograms_[measurement_and_story.str()]->AddDiagnostic(
+          catapult::kStoriesDiagnostic, stories);
+    }
+
+    if (units == "bps") {
+      // Bps has been interpreted as bits per second in WebRTC tests.
+      histograms_[measurement_and_story.str()]->AddSample(value / 8);
+    } else {
+      histograms_[measurement_and_story.str()]->AddSample(value);
+    }
+  }
+
+  proto::UnitAndDirection ParseUnit(absl::string_view units,
+                                    ImproveDirection improve_direction) {
+    RTC_DCHECK(units.find('_') == std::string::npos)
+        << "The unit_bigger|smallerIsBetter syntax isn't supported in WebRTC, "
+           "use the enum instead.";
+
+    proto::UnitAndDirection result;
+    result.set_improvement_direction(ParseDirection(improve_direction));
+    if (units == "bps") {
+      result.set_unit(proto::BYTES_PER_SECOND);
+    } else if (units == "dB") {
+      result.set_unit(proto::UNITLESS);
+    } else if (units == "fps") {
+      result.set_unit(proto::HERTZ);
+    } else if (units == "frames") {
+      result.set_unit(proto::COUNT);
+    } else if (units == "ms") {
+      result.set_unit(proto::MS_BEST_FIT_FORMAT);
+    } else if (units == "%") {
+      result.set_unit(proto::UNITLESS);
+    } else {
+      proto::Unit unit = catapult::UnitFromJsonUnit(std::string(units));
+
+      // UnitFromJsonUnit returns UNITLESS if it doesn't recognize the unit.
+      if (unit == proto::UNITLESS && units != "unitless") {
+        RTC_LOG(LS_WARNING) << "Unit " << units << " is unsupported.";
+      }
+
+      result.set_unit(unit);
+    }
+    return result;
+  }
+
+  proto::ImprovementDirection ParseDirection(
+      ImproveDirection improve_direction) {
+    switch (improve_direction) {
+      case ImproveDirection::kNone:
+        return proto::NOT_SPECIFIED;
+      case ImproveDirection::kSmallerIsBetter:
+        return proto::SMALLER_IS_BETTER;
+      case ImproveDirection::kBiggerIsBetter:
+        return proto::BIGGER_IS_BETTER;
+      default:
+        RTC_DCHECK_NOTREACHED() << "Invalid enum value " << improve_direction;
+    }
+  }
+
+ private:
+  mutable Mutex mutex_;
+  std::map<std::string, std::unique_ptr<catapult::HistogramBuilder>> histograms_
+      RTC_GUARDED_BY(&mutex_);
+};
+
+}  // namespace
+
+PerfTestResultWriter* CreateHistogramWriter() {
+  return new PerfTestHistogramWriter();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer.h b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer.h
new file mode 100644
index 0000000000..244e69fc45
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_PERF_TEST_HISTOGRAM_WRITER_H_
+#define TEST_TESTSUPPORT_PERF_TEST_HISTOGRAM_WRITER_H_
+
+#include "test/testsupport/perf_test_result_writer.h"
+
+namespace webrtc {
+namespace test {
+
+PerfTestResultWriter* CreateHistogramWriter();
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_PERF_TEST_HISTOGRAM_WRITER_H_
diff --git a/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer_no_protobuf.cc b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer_no_protobuf.cc
new file mode 100644
index 0000000000..6bc810b94d
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer_no_protobuf.cc
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/perf_test_histogram_writer.h"
+
+namespace webrtc {
+namespace test {
+
+PerfTestResultWriter* CreateHistogramWriter() {
+  RTC_DCHECK_NOTREACHED()
+      << "Cannot run perf tests with rtc_enable_protobuf = false. "
+         "Perf write results as protobufs.";
+  return nullptr;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer_unittest.cc b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer_unittest.cc
new file mode 100644
index 0000000000..83025a7447
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test_histogram_writer_unittest.cc
@@ -0,0 +1,216 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/perf_test_histogram_writer.h"
+
+#include <memory>
+#include <string>
+
+#include "test/gtest.h"
+#include "third_party/catapult/tracing/tracing/value/histogram.h"
+
+namespace webrtc {
+namespace test {
+
+namespace proto = catapult::tracing::tracing::proto;
+
+TEST(PerfHistogramWriterUnittest, TestSimpleHistogram) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("-", "-", 0, "ms", false, ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  EXPECT_TRUE(histogram_set.ParseFromString(writer->Serialize()))
+      << "Expected valid histogram set";
+
+  ASSERT_EQ(histogram_set.histograms_size(), 1);
+}
+
+TEST(PerfHistogramWriterUnittest, TestListOfValuesHistogram) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  std::vector<double> samples{0, 1, 2};
+  writer->LogResultList("-", "-", samples, "ms", false,
+                        ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  EXPECT_TRUE(histogram_set.ParseFromString(writer->Serialize()))
+      << "Expected valid histogram set";
+
+  ASSERT_EQ(histogram_set.histograms_size(), 1);
+  ASSERT_EQ(histogram_set.histograms(0).sample_values_size(), 3);
+  EXPECT_EQ(histogram_set.histograms(0).sample_values(0), 0);
+  EXPECT_EQ(histogram_set.histograms(0).sample_values(1), 1);
+  EXPECT_EQ(histogram_set.histograms(0).sample_values(2), 2);
+}
+
+TEST(PerfHistogramWriterUnittest, WritesSamplesAndUserStory) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("measurement", "user_story", 15e7, "Hz", false,
+                    ImproveDirection::kBiggerIsBetter);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist1.name(), "measurement");
+
+  EXPECT_EQ(hist1.unit().unit(), proto::HERTZ);
+  EXPECT_EQ(hist1.unit().improvement_direction(), proto::BIGGER_IS_BETTER);
+
+  EXPECT_EQ(hist1.sample_values_size(), 1);
+  EXPECT_EQ(hist1.sample_values(0), 15e7);
+
+  EXPECT_EQ(hist1.diagnostics().diagnostic_map().count("stories"), 1u);
+  const proto::Diagnostic& stories =
+      hist1.diagnostics().diagnostic_map().at("stories");
+  ASSERT_EQ(stories.generic_set().values_size(), 1);
+  EXPECT_EQ(stories.generic_set().values(0), "\"user_story\"");
+}
+
+TEST(PerfHistogramWriterUnittest, WritesOneHistogramPerMeasurementAndStory) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("measurement", "story1", 1, "ms", false,
+                    ImproveDirection::kNone);
+  writer->LogResult("measurement", "story1", 2, "ms", false,
+                    ImproveDirection::kNone);
+  writer->LogResult("measurement", "story2", 2, "ms", false,
+                    ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  ASSERT_EQ(histogram_set.histograms_size(), 2);
+
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+  const proto::Histogram& hist2 = histogram_set.histograms(1);
+
+  EXPECT_EQ(hist1.name(), "measurement");
+  EXPECT_EQ(hist2.name(), "measurement");
+
+  const proto::Diagnostic& stories1 =
+      hist1.diagnostics().diagnostic_map().at("stories");
+  EXPECT_EQ(stories1.generic_set().values(0), "\"story1\"");
+  EXPECT_EQ(hist1.sample_values_size(), 2);
+
+  const proto::Diagnostic& stories2 =
+      hist2.diagnostics().diagnostic_map().at("stories");
+  EXPECT_EQ(stories2.generic_set().values(0), "\"story2\"");
+  EXPECT_EQ(hist2.sample_values_size(), 1);
+}
+
+TEST(PerfHistogramWriterUnittest, IgnoresError) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResultMeanAndError("-", "-", 17, 12345, "ms", false,
+                                ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist1.running().mean(), 17);
+  EXPECT_EQ(hist1.running().variance(), 0) << "The error should be ignored.";
+}
+
+TEST(PerfHistogramWriterUnittest, WritesDecibelIntoMeasurementName) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("measurement", "-", 0, "dB", false,
+                    ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist1.unit().unit(), proto::UNITLESS)
+      << "dB should map to unitless";
+  EXPECT_EQ(hist1.name(), "measurement_dB") << "measurement should be renamed";
+}
+
+TEST(PerfHistogramWriterUnittest, WritesFpsIntoMeasurementName) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("measurement", "-", 0, "fps", false,
+                    ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist1.unit().unit(), proto::HERTZ) << "fps should map to hertz";
+  EXPECT_EQ(hist1.name(), "measurement_fps") << "measurement should be renamed";
+}
+
+TEST(PerfHistogramWriterUnittest, WritesPercentIntoMeasurementName) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("measurement", "-", 0, "%", false, ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist1.unit().unit(), proto::UNITLESS)
+      << "percent should map to hertz";
+  EXPECT_EQ(hist1.name(), "measurement_%") << "measurement should be renamed";
+}
+
+TEST(PerfHistogramWriterUnittest, BitsPerSecondIsConvertedToBytes) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("-", "-", 1024, "bps", false, ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist1.sample_values(0), 128) << "1024 bits = 128 bytes";
+}
+
+TEST(PerfHistogramWriterUnittest, ParsesDirection) {
+  std::unique_ptr<PerfTestResultWriter> writer =
+      std::unique_ptr<PerfTestResultWriter>(CreateHistogramWriter());
+
+  writer->LogResult("measurement1", "-", 0, "bps", false,
+                    ImproveDirection::kBiggerIsBetter);
+  writer->LogResult("measurement2", "-", 0, "frames", false,
+                    ImproveDirection::kSmallerIsBetter);
+  writer->LogResult("measurement3", "-", 0, "sigma", false,
+                    ImproveDirection::kNone);
+
+  proto::HistogramSet histogram_set;
+  histogram_set.ParseFromString(writer->Serialize());
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+  const proto::Histogram& hist2 = histogram_set.histograms(1);
+  const proto::Histogram& hist3 = histogram_set.histograms(2);
+
+  EXPECT_EQ(hist1.unit().unit(), proto::BYTES_PER_SECOND);
+  EXPECT_EQ(hist1.unit().improvement_direction(), proto::BIGGER_IS_BETTER);
+
+  EXPECT_EQ(hist2.unit().unit(), proto::COUNT);
+  EXPECT_EQ(hist2.unit().improvement_direction(), proto::SMALLER_IS_BETTER);
+
+  EXPECT_EQ(hist3.unit().unit(), proto::SIGMA);
+  EXPECT_EQ(hist3.unit().improvement_direction(), proto::NOT_SPECIFIED);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/perf_test_result_writer.h b/third_party/libwebrtc/test/testsupport/perf_test_result_writer.h
new file mode 100644
index 0000000000..1b93bc9583
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test_result_writer.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_PERF_TEST_RESULT_WRITER_H_
+#define TEST_TESTSUPPORT_PERF_TEST_RESULT_WRITER_H_
+
+#include <stdio.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "test/testsupport/perf_test.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface for classes that write perf results to some kind of JSON format.
+class PerfTestResultWriter {
+ public:
+  virtual ~PerfTestResultWriter() = default;
+
+  virtual void ClearResults() = 0;
+  virtual void LogResult(absl::string_view graph_name,
+                         absl::string_view trace_name,
+                         double value,
+                         absl::string_view units,
+                         bool important,
+                         webrtc::test::ImproveDirection improve_direction) = 0;
+  virtual void LogResultMeanAndError(
+      absl::string_view graph_name,
+      absl::string_view trace_name,
+      double mean,
+      double error,
+      absl::string_view units,
+      bool important,
+      webrtc::test::ImproveDirection improve_direction) = 0;
+  virtual void LogResultList(
+      absl::string_view graph_name,
+      absl::string_view trace_name,
+      rtc::ArrayView<const double> values,
+      absl::string_view units,
+      bool important,
+      webrtc::test::ImproveDirection improve_direction) = 0;
+
+  virtual std::string Serialize() const = 0;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_PERF_TEST_RESULT_WRITER_H_
diff --git a/third_party/libwebrtc/test/testsupport/perf_test_unittest.cc b/third_party/libwebrtc/test/testsupport/perf_test_unittest.cc
new file mode 100644
index 0000000000..509882db08
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/perf_test_unittest.cc
@@ -0,0 +1,205 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/perf_test.h"
+
+#include <algorithm>
+#include <limits>
+#include <string>
+
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/rtc_expect_death.h"
+
+#if WEBRTC_ENABLE_PROTOBUF
+#include "third_party/catapult/tracing/tracing/value/histogram.h"
+namespace proto = catapult::tracing::tracing::proto;
+#endif
+
+namespace webrtc {
+namespace test {
+
+class PerfTest : public ::testing::Test {
+ protected:
+  void TearDown() override { ClearPerfResults(); }
+};
+
+#if defined(WEBRTC_IOS)
+#define MAYBE_TestPrintResult DISABLED_TestPrintResult
+#else
+#define MAYBE_TestPrintResult TestPrintResult
+#endif
+TEST_F(PerfTest, MAYBE_TestPrintResult) {
+  ::testing::internal::CaptureStdout();
+  std::string expected;
+
+  expected += "RESULT measurementmodifier: trace= 42 units\n";
+  PrintResult("measurement", "modifier", "trace", 42, "units", false);
+
+  expected += "*RESULT foobar: baz_v= 1423730 widgets\n";
+  PrintResult("foo", "bar", "baz_v", 1423730, "widgets", true);
+
+  expected += "RESULT foobar: baz_me= {1,2} lemurs\n";
+  PrintResultMeanAndError("foo", "bar", "baz_me", 1, 2, "lemurs", false);
+
+  const double kListOfScalars[] = {1, 2, 3};
+  expected += "RESULT foobar: baz_vl= [1,2,3] units\n";
+  PrintResultList("foo", "bar", "baz_vl", kListOfScalars, "units", false);
+
+  EXPECT_EQ(expected, ::testing::internal::GetCapturedStdout());
+}
+
+TEST_F(PerfTest, TestClearPerfResults) {
+  PrintResult("measurement", "modifier", "trace", 42, "units", false);
+  ClearPerfResults();
+  EXPECT_EQ("", GetPerfResults());
+}
+
+#if WEBRTC_ENABLE_PROTOBUF
+
+TEST_F(PerfTest, TestGetPerfResultsHistograms) {
+  ClearPerfResults();
+  PrintResult("measurement", "_modifier", "story_1", 42, "ms", false);
+  PrintResult("foo", "bar", "story_1", 7, "sigma", true);
+  // Note: the error will be ignored, not supported by histograms.
+  PrintResultMeanAndError("foo", "bar", "story_1", 1, 2000, "sigma", false);
+  const double kListOfScalars[] = {1, 2, 3};
+  PrintResultList("foo", "bar", "story_1", kListOfScalars, "sigma", false);
+
+  proto::HistogramSet histogram_set;
+  EXPECT_TRUE(histogram_set.ParseFromString(GetPerfResults()))
+      << "Expected valid histogram set";
+
+  ASSERT_EQ(histogram_set.histograms_size(), 2)
+      << "Should be two histograms: foobar and measurement_modifier";
+  const proto::Histogram& hist1 = histogram_set.histograms(0);
+  const proto::Histogram& hist2 = histogram_set.histograms(1);
+
+  EXPECT_EQ(hist1.name(), "foobar");
+
+  // Spot check some things in here (there's a more thorough test on the
+  // histogram writer itself).
+  EXPECT_EQ(hist1.unit().unit(), proto::SIGMA);
+  EXPECT_EQ(hist1.sample_values_size(), 5);
+  EXPECT_EQ(hist1.sample_values(0), 7);
+  EXPECT_EQ(hist1.sample_values(1), 1);
+  EXPECT_EQ(hist1.sample_values(2), 1);
+  EXPECT_EQ(hist1.sample_values(3), 2);
+  EXPECT_EQ(hist1.sample_values(4), 3);
+
+  EXPECT_EQ(hist1.diagnostics().diagnostic_map().count("stories"), 1u);
+  const proto::Diagnostic& stories =
+      hist1.diagnostics().diagnostic_map().at("stories");
+  ASSERT_EQ(stories.generic_set().values_size(), 1);
+  EXPECT_EQ(stories.generic_set().values(0), "\"story_1\"");
+
+  EXPECT_EQ(hist2.name(), "measurement_modifier");
+  EXPECT_EQ(hist2.unit().unit(), proto::MS_BEST_FIT_FORMAT);
+}
+
+TEST_F(PerfTest, TestGetPerfResultsHistogramsWithEmptyCounter) {
+  ClearPerfResults();
+  ::testing::internal::CaptureStdout();
+
+  SamplesStatsCounter empty_counter;
+  PrintResult("measurement", "_modifier", "story", empty_counter, "ms", false);
+
+  proto::HistogramSet histogram_set;
+  EXPECT_TRUE(histogram_set.ParseFromString(GetPerfResults()))
+      << "Expected valid histogram set";
+
+  ASSERT_EQ(histogram_set.histograms_size(), 1)
+      << "Should be one histogram: measurement_modifier";
+  const proto::Histogram& hist = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist.name(), "measurement_modifier");
+
+  // Spot check some things in here (there's a more thorough test on the
+  // histogram writer itself).
+  EXPECT_EQ(hist.unit().unit(), proto::MS_BEST_FIT_FORMAT);
+  EXPECT_EQ(hist.sample_values_size(), 1);
+  EXPECT_EQ(hist.sample_values(0), 0);
+
+  EXPECT_EQ(hist.diagnostics().diagnostic_map().count("stories"), 1u);
+  const proto::Diagnostic& stories =
+      hist.diagnostics().diagnostic_map().at("stories");
+  ASSERT_EQ(stories.generic_set().values_size(), 1);
+  EXPECT_EQ(stories.generic_set().values(0), "\"story\"");
+
+  std::string expected = "RESULT measurement_modifier: story= {0,0} ms\n";
+  EXPECT_EQ(expected, ::testing::internal::GetCapturedStdout());
+}
+
+TEST_F(PerfTest, TestGetPerfResultsHistogramsWithStatsCounter) {
+  ClearPerfResults();
+  ::testing::internal::CaptureStdout();
+
+  SamplesStatsCounter counter;
+  counter.AddSample(1);
+  counter.AddSample(2);
+  counter.AddSample(3);
+  counter.AddSample(4);
+  counter.AddSample(5);
+  PrintResult("measurement", "_modifier", "story", counter, "ms", false);
+
+  proto::HistogramSet histogram_set;
+  EXPECT_TRUE(histogram_set.ParseFromString(GetPerfResults()))
+      << "Expected valid histogram set";
+
+  ASSERT_EQ(histogram_set.histograms_size(), 1)
+      << "Should be one histogram: measurement_modifier";
+  const proto::Histogram& hist = histogram_set.histograms(0);
+
+  EXPECT_EQ(hist.name(), "measurement_modifier");
+
+  // Spot check some things in here (there's a more thorough test on the
+  // histogram writer itself).
+  EXPECT_EQ(hist.unit().unit(), proto::MS_BEST_FIT_FORMAT);
+  EXPECT_EQ(hist.sample_values_size(), 5);
+  EXPECT_THAT(hist.sample_values(), testing::ElementsAre(1, 2, 3, 4, 5));
+
+  EXPECT_EQ(hist.diagnostics().diagnostic_map().count("stories"), 1u);
+  const proto::Diagnostic& stories =
+      hist.diagnostics().diagnostic_map().at("stories");
+  ASSERT_EQ(stories.generic_set().values_size(), 1);
+  EXPECT_EQ(stories.generic_set().values(0), "\"story\"");
+
+  // mean = 3; std = sqrt(2)
+  std::string expected =
+      "RESULT measurement_modifier: story= {3,1.4142136} ms\n";
+  EXPECT_EQ(expected, ::testing::internal::GetCapturedStdout());
+}
+
+#endif  // WEBRTC_ENABLE_PROTOBUF
+
+#if GTEST_HAS_DEATH_TEST
+using PerfDeathTest = PerfTest;
+
+TEST_F(PerfDeathTest, TestFiniteResultError) {
+  const double kNan = std::numeric_limits<double>::quiet_NaN();
+  const double kInf = std::numeric_limits<double>::infinity();
+
+  RTC_EXPECT_DEATH(PrintResult("a", "b", "c", kNan, "d", false), "finit");
+  RTC_EXPECT_DEATH(PrintResult("a", "b", "c", kInf, "d", false), "finit");
+
+  RTC_EXPECT_DEATH(PrintResultMeanAndError("a", "b", "c", kNan, 1, "d", false),
+                   "");
+  RTC_EXPECT_DEATH(PrintResultMeanAndError("a", "b", "c", 1, kInf, "d", false),
+                   "");
+
+  const double kNanList[] = {kNan, kNan};
+  RTC_EXPECT_DEATH(PrintResultList("a", "b", "c", kNanList, "d", false), "");
+  const double kInfList[] = {0, kInf};
+  RTC_EXPECT_DEATH(PrintResultList("a", "b", "c", kInfList, "d", false), "");
+}
+#endif
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/resources_dir_flag.cc b/third_party/libwebrtc/test/testsupport/resources_dir_flag.cc
new file mode 100644
index 0000000000..87a449a401
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/resources_dir_flag.cc
@@ -0,0 +1,21 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/resources_dir_flag.h"
+
+#include "absl/flags/flag.h"
+
+ABSL_FLAG(std::string,
+          resources_dir,
+          "",
+          "Where to look for the runtime dependencies. If not specified, we "
+          "will use a reasonable default depending on where we are running. "
+          "This flag is useful if we copy over test resources to a phone and "
+          "need to tell the tests where their resources are.");
diff --git a/third_party/libwebrtc/test/testsupport/resources_dir_flag.h b/third_party/libwebrtc/test/testsupport/resources_dir_flag.h
new file mode 100644
index 0000000000..7d6f192d9b
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/resources_dir_flag.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <string>
+
+#ifndef TEST_TESTSUPPORT_RESOURCES_DIR_FLAG_H__
+#define TEST_TESTSUPPORT_RESOURCES_DIR_FLAG_H__
+
+#include "absl/flags/declare.h"
+
+ABSL_DECLARE_FLAG(std::string, resources_dir);
+
+#endif  // TEST_TESTSUPPORT_RESOURCES_DIR_FLAG_H__
diff --git a/third_party/libwebrtc/test/testsupport/rtc_expect_death.h b/third_party/libwebrtc/test/testsupport/rtc_expect_death.h
new file mode 100644
index 0000000000..5941e12bd2
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/rtc_expect_death.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_RTC_EXPECT_DEATH_H_
+#define TEST_TESTSUPPORT_RTC_EXPECT_DEATH_H_
+
+#include "test/gtest.h"
+
+#if RTC_CHECK_MSG_ENABLED
+#define RTC_EXPECT_DEATH(statement, regex) EXPECT_DEATH(statement, regex)
+#else
+// If RTC_CHECKs messages are disabled we can't validate failure message
+#define RTC_EXPECT_DEATH(statement, regex) EXPECT_DEATH(statement, "")
+#endif
+
+#endif  // TEST_TESTSUPPORT_RTC_EXPECT_DEATH_H_
diff --git a/third_party/libwebrtc/test/testsupport/test_artifacts.cc b/third_party/libwebrtc/test/testsupport/test_artifacts.cc
new file mode 100644
index 0000000000..6f062e5fe4
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/test_artifacts.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/test_artifacts.h"
+
+#include <string.h>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "test/testsupport/file_utils.h"
+
+namespace {
+const std::string& DefaultArtifactPath() {
+  static const std::string path = webrtc::test::OutputPath();
+  return path;
+}
+}  // namespace
+
+ABSL_FLAG(std::string,
+          test_artifacts_dir,
+          DefaultArtifactPath().c_str(),
+          "The output folder where test output should be saved.");
+
+namespace webrtc {
+namespace test {
+
+bool GetTestArtifactsDir(std::string* out_dir) {
+  if (absl::GetFlag(FLAGS_test_artifacts_dir).empty()) {
+    RTC_LOG(LS_WARNING) << "No test_out_dir defined.";
+    return false;
+  }
+  *out_dir = absl::GetFlag(FLAGS_test_artifacts_dir);
+  return true;
+}
+
+bool WriteToTestArtifactsDir(const char* filename,
+                             const uint8_t* buffer,
+                             size_t length) {
+  if (absl::GetFlag(FLAGS_test_artifacts_dir).empty()) {
+    RTC_LOG(LS_WARNING) << "No test_out_dir defined.";
+    return false;
+  }
+
+  if (filename == nullptr || strlen(filename) == 0) {
+    RTC_LOG(LS_WARNING) << "filename must be provided.";
+    return false;
+  }
+
+  FileWrapper output = FileWrapper::OpenWriteOnly(
+      JoinFilename(absl::GetFlag(FLAGS_test_artifacts_dir), filename));
+
+  return output.is_open() && output.Write(buffer, length);
+}
+
+bool WriteToTestArtifactsDir(const char* filename, const std::string& content) {
+  return WriteToTestArtifactsDir(
+      filename, reinterpret_cast<const uint8_t*>(content.c_str()),
+      content.length());
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/test_artifacts.h b/third_party/libwebrtc/test/testsupport/test_artifacts.h
new file mode 100644
index 0000000000..ba0d4d39cb
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/test_artifacts.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_TEST_ARTIFACTS_H_
+#define TEST_TESTSUPPORT_TEST_ARTIFACTS_H_
+
+#include <stdint.h>
+#include <stdlib.h>
+
+#include <string>
+
+namespace webrtc {
+namespace test {
+
+// If the test_artifacts_dir flag is set, returns true and copies the location
+// of the dir to `out_dir`. Otherwise, return false.
+bool GetTestArtifactsDir(std::string* out_dir);
+
+// Writes a `length` bytes array `buffer` to `filename` in isolated output
+// directory defined by swarming. If the file is existing, content will be
+// appended. Otherwise a new file will be created. This function returns false
+// if isolated output directory has not been defined, or `filename` indicates an
+// invalid or non-writable file, or underlying file system errors.
+bool WriteToTestArtifactsDir(const char* filename,
+                             const uint8_t* buffer,
+                             size_t length);
+
+bool WriteToTestArtifactsDir(const char* filename, const std::string& content);
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_TEST_ARTIFACTS_H_
diff --git a/third_party/libwebrtc/test/testsupport/test_artifacts_unittest.cc b/third_party/libwebrtc/test/testsupport/test_artifacts_unittest.cc
new file mode 100644
index 0000000000..fb577610fb
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/test_artifacts_unittest.cc
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/test_artifacts.h"
+
+#include <string.h>
+
+#include <string>
+
+#include "absl/flags/declare.h"
+#include "absl/flags/flag.h"
+#include "rtc_base/system/file_wrapper.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_DECLARE_FLAG(std::string, test_artifacts_dir);
+
+namespace webrtc {
+namespace test {
+
+TEST(IsolatedOutputTest, ShouldRejectInvalidIsolatedOutDir) {
+  const std::string backup = absl::GetFlag(FLAGS_test_artifacts_dir);
+  absl::SetFlag(&FLAGS_test_artifacts_dir, "");
+  ASSERT_FALSE(WriteToTestArtifactsDir("a-file", "some-contents"));
+  absl::SetFlag(&FLAGS_test_artifacts_dir, backup);
+}
+
+TEST(IsolatedOutputTest, ShouldRejectInvalidFileName) {
+  ASSERT_FALSE(WriteToTestArtifactsDir(nullptr, "some-contents"));
+  ASSERT_FALSE(WriteToTestArtifactsDir("", "some-contents"));
+}
+
+// Sets isolated_out_dir=<a-writable-path> to execute this test.
+TEST(IsolatedOutputTest, ShouldBeAbleToWriteContent) {
+  const char* filename = "a-file";
+  const char* content = "some-contents";
+  if (WriteToTestArtifactsDir(filename, content)) {
+    std::string out_file =
+        JoinFilename(absl::GetFlag(FLAGS_test_artifacts_dir), filename);
+    FileWrapper input = FileWrapper::OpenReadOnly(out_file);
+    EXPECT_TRUE(input.is_open());
+    EXPECT_TRUE(input.Rewind());
+    uint8_t buffer[32];
+    EXPECT_EQ(input.Read(buffer, strlen(content)), strlen(content));
+    buffer[strlen(content)] = 0;
+    EXPECT_EQ(std::string(content),
+              std::string(reinterpret_cast<char*>(buffer)));
+    input.Close();
+
+    EXPECT_TRUE(RemoveFile(out_file));
+  }
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/video_frame_writer.cc b/third_party/libwebrtc/test/testsupport/video_frame_writer.cc
new file mode 100644
index 0000000000..c36ebdeed7
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/video_frame_writer.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/video_frame_writer.h"
+
+#include <cmath>
+#include <cstdlib>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+rtc::Buffer ExtractI420BufferWithSize(const VideoFrame& frame,
+                                      int width,
+                                      int height) {
+  if (frame.width() != width || frame.height() != height) {
+    RTC_CHECK_LE(std::abs(static_cast<double>(width) / height -
+                          static_cast<double>(frame.width()) / frame.height()),
+                 2 * std::numeric_limits<double>::epsilon());
+    // Same aspect ratio, no cropping needed.
+    rtc::scoped_refptr<I420Buffer> scaled(I420Buffer::Create(width, height));
+    scaled->ScaleFrom(*frame.video_frame_buffer()->ToI420());
+
+    size_t length =
+        CalcBufferSize(VideoType::kI420, scaled->width(), scaled->height());
+    rtc::Buffer buffer(length);
+    RTC_CHECK_NE(ExtractBuffer(scaled, length, buffer.data()), -1);
+    return buffer;
+  }
+
+  // No resize.
+  size_t length =
+      CalcBufferSize(VideoType::kI420, frame.width(), frame.height());
+  rtc::Buffer buffer(length);
+  RTC_CHECK_NE(ExtractBuffer(frame, length, buffer.data()), -1);
+  return buffer;
+}
+
+}  // namespace
+
+Y4mVideoFrameWriterImpl::Y4mVideoFrameWriterImpl(std::string output_file_name,
+                                                 int width,
+                                                 int height,
+                                                 int fps)
+    // We will move string here to prevent extra copy. We won't use const ref
+    // to not corrupt caller variable with move and don't assume that caller's
+    // variable won't be destructed before writer.
+    : width_(width),
+      height_(height),
+      frame_writer_(
+          std::make_unique<Y4mFrameWriterImpl>(std::move(output_file_name),
+                                               width_,
+                                               height_,
+                                               fps)) {
+  // Init underlying frame writer and ensure that it is operational.
+  RTC_CHECK(frame_writer_->Init());
+}
+
+bool Y4mVideoFrameWriterImpl::WriteFrame(const webrtc::VideoFrame& frame) {
+  rtc::Buffer frame_buffer = ExtractI420BufferWithSize(frame, width_, height_);
+  RTC_CHECK_EQ(frame_buffer.size(), frame_writer_->FrameLength());
+  return frame_writer_->WriteFrame(frame_buffer.data());
+}
+
+void Y4mVideoFrameWriterImpl::Close() {
+  frame_writer_->Close();
+}
+
+YuvVideoFrameWriterImpl::YuvVideoFrameWriterImpl(std::string output_file_name,
+                                                 int width,
+                                                 int height)
+    // We will move string here to prevent extra copy. We won't use const ref
+    // to not corrupt caller variable with move and don't assume that caller's
+    // variable won't be destructed before writer.
+    : width_(width),
+      height_(height),
+      frame_writer_(
+          std::make_unique<YuvFrameWriterImpl>(std::move(output_file_name),
+                                               width_,
+                                               height_)) {
+  // Init underlying frame writer and ensure that it is operational.
+  RTC_CHECK(frame_writer_->Init());
+}
+
+bool YuvVideoFrameWriterImpl::WriteFrame(const webrtc::VideoFrame& frame) {
+  rtc::Buffer frame_buffer = ExtractI420BufferWithSize(frame, width_, height_);
+  RTC_CHECK_EQ(frame_buffer.size(), frame_writer_->FrameLength());
+  return frame_writer_->WriteFrame(frame_buffer.data());
+}
+
+void YuvVideoFrameWriterImpl::Close() {
+  frame_writer_->Close();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/video_frame_writer.h b/third_party/libwebrtc/test/testsupport/video_frame_writer.h
new file mode 100644
index 0000000000..1cc4e56284
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/video_frame_writer.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_TESTSUPPORT_VIDEO_FRAME_WRITER_H_
+#define TEST_TESTSUPPORT_VIDEO_FRAME_WRITER_H_
+
+#include <memory>
+#include <string>
+
+#include "api/test/video/video_frame_writer.h"
+#include "api/video/video_frame.h"
+#include "rtc_base/buffer.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+// Writes webrtc::VideoFrame to specified file with y4m frame writer
+class Y4mVideoFrameWriterImpl : public VideoFrameWriter {
+ public:
+  Y4mVideoFrameWriterImpl(std::string output_file_name,
+                          int width,
+                          int height,
+                          int fps);
+  ~Y4mVideoFrameWriterImpl() override = default;
+
+  bool WriteFrame(const webrtc::VideoFrame& frame) override;
+  void Close() override;
+
+ private:
+  const int width_;
+  const int height_;
+
+  std::unique_ptr<FrameWriter> frame_writer_;
+};
+
+// Writes webrtc::VideoFrame to specified file with yuv frame writer
+class YuvVideoFrameWriterImpl : public VideoFrameWriter {
+ public:
+  YuvVideoFrameWriterImpl(std::string output_file_name, int width, int height);
+  ~YuvVideoFrameWriterImpl() override = default;
+
+  bool WriteFrame(const webrtc::VideoFrame& frame) override;
+  void Close() override;
+
+ private:
+  const int width_;
+  const int height_;
+
+  std::unique_ptr<FrameWriter> frame_writer_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_TESTSUPPORT_VIDEO_FRAME_WRITER_H_
diff --git a/third_party/libwebrtc/test/testsupport/video_frame_writer_unittest.cc b/third_party/libwebrtc/test/testsupport/video_frame_writer_unittest.cc
new file mode 100644
index 0000000000..9d59627c0f
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/video_frame_writer_unittest.cc
@@ -0,0 +1,173 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/testsupport/video_frame_writer.h"
+
+#include <stdint.h>
+#include <stdio.h>
+#include <string.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/test/video/video_frame_writer.h"
+#include "api/video/i420_buffer.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+const size_t kFrameWidth = 50;
+const size_t kFrameHeight = 20;
+const size_t kFrameLength = 3 * kFrameWidth * kFrameHeight / 2;  // I420.
+const size_t kFrameRate = 30;
+
+// Size of header: "YUV4MPEG2 W50 H20 F30:1 C420\n"
+const size_t kFileHeaderSize = 29;
+// Size of header: "FRAME\n"
+const size_t kFrameHeaderSize = 6;
+
+rtc::scoped_refptr<I420Buffer> CreateI420Buffer(int width, int height) {
+  rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(width, height));
+  for (int x = 0; x < width; x++) {
+    for (int y = 0; y < height; y++) {
+      buffer->MutableDataY()[x + y * width] = 128;
+    }
+  }
+  int chroma_width = buffer->ChromaWidth();
+  int chroma_height = buffer->ChromaHeight();
+  for (int x = 0; x < chroma_width; x++) {
+    for (int y = 0; y < chroma_height; y++) {
+      buffer->MutableDataU()[x + y * chroma_width] = 1;
+      buffer->MutableDataV()[x + y * chroma_width] = 255;
+    }
+  }
+  return buffer;
+}
+
+void AssertI420BuffersEq(
+    rtc::scoped_refptr<webrtc::I420BufferInterface> actual,
+    rtc::scoped_refptr<webrtc::I420BufferInterface> expected) {
+  ASSERT_TRUE(actual);
+
+  ASSERT_EQ(actual->width(), expected->width());
+  ASSERT_EQ(actual->height(), expected->height());
+  const int width = expected->width();
+  const int height = expected->height();
+  for (int x = 0; x < width; x++) {
+    for (int y = 0; y < height; y++) {
+      ASSERT_EQ(actual->DataY()[x + y * width],
+                expected->DataY()[x + y * width]);
+    }
+  }
+
+  ASSERT_EQ(actual->ChromaWidth(), expected->ChromaWidth());
+  ASSERT_EQ(actual->ChromaHeight(), expected->ChromaHeight());
+  int chroma_width = expected->ChromaWidth();
+  int chroma_height = expected->ChromaHeight();
+  for (int x = 0; x < chroma_width; x++) {
+    for (int y = 0; y < chroma_height; y++) {
+      ASSERT_EQ(actual->DataU()[x + y * chroma_width],
+                expected->DataU()[x + y * chroma_width]);
+      ASSERT_EQ(actual->DataV()[x + y * chroma_width],
+                expected->DataV()[x + y * chroma_width]);
+    }
+  }
+}
+
+}  // namespace
+
+class VideoFrameWriterTest : public ::testing::Test {
+ protected:
+  VideoFrameWriterTest() = default;
+  ~VideoFrameWriterTest() override = default;
+
+  void SetUp() override {
+    temp_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                "video_frame_writer_unittest");
+    frame_writer_ = CreateFrameWriter();
+  }
+
+  virtual std::unique_ptr<VideoFrameWriter> CreateFrameWriter() = 0;
+
+  void TearDown() override { remove(temp_filename_.c_str()); }
+
+  std::unique_ptr<VideoFrameWriter> frame_writer_;
+  std::string temp_filename_;
+};
+
+class Y4mVideoFrameWriterTest : public VideoFrameWriterTest {
+ protected:
+  std::unique_ptr<VideoFrameWriter> CreateFrameWriter() override {
+    return std::make_unique<Y4mVideoFrameWriterImpl>(
+        temp_filename_, kFrameWidth, kFrameHeight, kFrameRate);
+  }
+};
+
+class YuvVideoFrameWriterTest : public VideoFrameWriterTest {
+ protected:
+  std::unique_ptr<VideoFrameWriter> CreateFrameWriter() override {
+    return std::make_unique<YuvVideoFrameWriterImpl>(temp_filename_,
+                                                     kFrameWidth, kFrameHeight);
+  }
+};
+
+TEST_F(Y4mVideoFrameWriterTest, InitSuccess) {}
+
+TEST_F(Y4mVideoFrameWriterTest, WriteFrame) {
+  rtc::scoped_refptr<I420Buffer> expected_buffer =
+      CreateI420Buffer(kFrameWidth, kFrameHeight);
+
+  VideoFrame frame =
+      VideoFrame::Builder().set_video_frame_buffer(expected_buffer).build();
+
+  ASSERT_TRUE(frame_writer_->WriteFrame(frame));
+  ASSERT_TRUE(frame_writer_->WriteFrame(frame));
+
+  frame_writer_->Close();
+  EXPECT_EQ(kFileHeaderSize + 2 * kFrameHeaderSize + 2 * kFrameLength,
+            GetFileSize(temp_filename_));
+
+  std::unique_ptr<FrameReader> frame_reader =
+      CreateY4mFrameReader(temp_filename_);
+  AssertI420BuffersEq(frame_reader->PullFrame(), expected_buffer);
+  AssertI420BuffersEq(frame_reader->PullFrame(), expected_buffer);
+  EXPECT_FALSE(frame_reader->PullFrame());  // End of file.
+}
+
+TEST_F(YuvVideoFrameWriterTest, InitSuccess) {}
+
+TEST_F(YuvVideoFrameWriterTest, WriteFrame) {
+  rtc::scoped_refptr<I420Buffer> expected_buffer =
+      CreateI420Buffer(kFrameWidth, kFrameHeight);
+
+  VideoFrame frame =
+      VideoFrame::Builder().set_video_frame_buffer(expected_buffer).build();
+
+  ASSERT_TRUE(frame_writer_->WriteFrame(frame));
+  ASSERT_TRUE(frame_writer_->WriteFrame(frame));
+
+  frame_writer_->Close();
+  EXPECT_EQ(2 * kFrameLength, GetFileSize(temp_filename_));
+
+  std::unique_ptr<FrameReader> frame_reader = CreateYuvFrameReader(
+      temp_filename_,
+      Resolution({.width = kFrameWidth, .height = kFrameHeight}));
+  AssertI420BuffersEq(frame_reader->PullFrame(), expected_buffer);
+  AssertI420BuffersEq(frame_reader->PullFrame(), expected_buffer);
+  EXPECT_FALSE(frame_reader->PullFrame());  // End of file.
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/y4m_frame_reader.cc b/third_party/libwebrtc/test/testsupport/y4m_frame_reader.cc
new file mode 100644
index 0000000000..72fb9b5188
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/y4m_frame_reader.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+constexpr int kFrameHeaderSize = 6;  // "FRAME\n"
+}  // namespace
+
+void ParseY4mHeader(std::string filepath,
+                    Resolution* resolution,
+                    int* header_size) {
+  FILE* file = fopen(filepath.c_str(), "r");
+  RTC_CHECK(file != NULL) << "Cannot open " << filepath;
+
+  // Length of Y4M header is technically unlimited due to the comment tag 'X'.
+  char h[1024];
+  RTC_CHECK(fgets(h, sizeof(h), file) != NULL)
+      << "File " << filepath << " is too small";
+  fclose(file);
+
+  RTC_CHECK(sscanf(h, "YUV4MPEG2 W%d H%d", &resolution->width,
+                   &resolution->height) == 2)
+      << filepath << " is not a valid Y4M file";
+
+  RTC_CHECK_GT(resolution->width, 0) << "Width must be positive";
+  RTC_CHECK_GT(resolution->height, 0) << "Height must be positive";
+
+  *header_size = strcspn(h, "\n") + 1;
+  RTC_CHECK(static_cast<unsigned>(*header_size) < sizeof(h))
+      << filepath << " has unexpectedly large header";
+}
+
+Y4mFrameReaderImpl::Y4mFrameReaderImpl(std::string filepath,
+                                       RepeatMode repeat_mode)
+    : YuvFrameReaderImpl(filepath, Resolution(), repeat_mode) {}
+
+void Y4mFrameReaderImpl::Init() {
+  file_ = fopen(filepath_.c_str(), "rb");
+  RTC_CHECK(file_ != nullptr) << "Cannot open " << filepath_;
+
+  ParseY4mHeader(filepath_, &resolution_, &header_size_bytes_);
+  frame_size_bytes_ =
+      CalcBufferSize(VideoType::kI420, resolution_.width, resolution_.height);
+  frame_size_bytes_ += kFrameHeaderSize;
+
+  size_t file_size_bytes = GetFileSize(filepath_);
+  RTC_CHECK_GT(file_size_bytes, 0u) << "File " << filepath_ << " is empty";
+  RTC_CHECK_GT(file_size_bytes, header_size_bytes_)
+      << "File " << filepath_ << " is too small";
+
+  num_frames_ = static_cast<int>((file_size_bytes - header_size_bytes_) /
+                                 frame_size_bytes_);
+  RTC_CHECK_GT(num_frames_, 0u) << "File " << filepath_ << " is too small";
+  header_size_bytes_ += kFrameHeaderSize;
+}
+
+std::unique_ptr<FrameReader> CreateY4mFrameReader(std::string filepath) {
+  return CreateY4mFrameReader(filepath,
+                              YuvFrameReaderImpl::RepeatMode::kSingle);
+}
+
+std::unique_ptr<FrameReader> CreateY4mFrameReader(
+    std::string filepath,
+    YuvFrameReaderImpl::RepeatMode repeat_mode) {
+  Y4mFrameReaderImpl* frame_reader =
+      new Y4mFrameReaderImpl(filepath, repeat_mode);
+  frame_reader->Init();
+  return std::unique_ptr<FrameReader>(frame_reader);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/y4m_frame_reader_unittest.cc b/third_party/libwebrtc/test/testsupport/y4m_frame_reader_unittest.cc
new file mode 100644
index 0000000000..df81a8135b
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/y4m_frame_reader_unittest.cc
@@ -0,0 +1,158 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdint.h>
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+using Ratio = FrameReader::Ratio;
+using RepeatMode = YuvFrameReaderImpl::RepeatMode;
+
+constexpr Resolution kResolution({.width = 1, .height = 1});
+constexpr char kFileHeader[] = "YUV4MPEG2 W1 H1 F30:1 C420\n";
+constexpr char kFrameHeader[] = "FRAME\n";
+constexpr char kFrameContent[3][3] = {{0, 1, 2}, {1, 2, 3}, {2, 3, 4}};
+constexpr int kNumFrames = sizeof(kFrameContent) / sizeof(kFrameContent[0]);
+}  // namespace
+
+class Y4mFrameReaderTest : public ::testing::Test {
+ protected:
+  Y4mFrameReaderTest() = default;
+  ~Y4mFrameReaderTest() override = default;
+
+  void SetUp() override {
+    filepath_ = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                           "y4m_frame_reader_unittest");
+    FILE* file = fopen(filepath_.c_str(), "wb");
+    fwrite(kFileHeader, 1, sizeof(kFileHeader) - 1, file);
+    for (int n = 0; n < kNumFrames; ++n) {
+      fwrite(kFrameHeader, 1, sizeof(kFrameHeader) - 1, file);
+      fwrite(kFrameContent[n], 1, sizeof(kFrameContent[n]), file);
+    }
+    fclose(file);
+
+    reader_ = CreateY4mFrameReader(filepath_);
+  }
+
+  void TearDown() override { remove(filepath_.c_str()); }
+
+  std::string filepath_;
+  std::unique_ptr<FrameReader> reader_;
+};
+
+TEST_F(Y4mFrameReaderTest, num_frames) {
+  EXPECT_EQ(kNumFrames, reader_->num_frames());
+}
+
+TEST_F(Y4mFrameReaderTest, PullFrame_frameResolution) {
+  rtc::scoped_refptr<I420BufferInterface> buffer = reader_->PullFrame();
+  EXPECT_EQ(kResolution.width, buffer->width());
+  EXPECT_EQ(kResolution.height, buffer->height());
+}
+
+TEST_F(Y4mFrameReaderTest, PullFrame_frameContent) {
+  rtc::scoped_refptr<I420BufferInterface> buffer = reader_->PullFrame();
+  EXPECT_EQ(kFrameContent[0][0], *buffer->DataY());
+  EXPECT_EQ(kFrameContent[0][1], *buffer->DataU());
+  EXPECT_EQ(kFrameContent[0][2], *buffer->DataV());
+}
+
+TEST_F(Y4mFrameReaderTest, ReadFrame_randomOrder) {
+  std::vector<int> expected_frames = {2, 0, 1};
+  std::vector<int> actual_frames;
+  for (int frame_num : expected_frames) {
+    rtc::scoped_refptr<I420BufferInterface> buffer =
+        reader_->ReadFrame(frame_num);
+    actual_frames.push_back(*buffer->DataY());
+  }
+  EXPECT_EQ(expected_frames, actual_frames);
+}
+
+TEST_F(Y4mFrameReaderTest, PullFrame_scale) {
+  rtc::scoped_refptr<I420BufferInterface> buffer = reader_->PullFrame(
+      /*pulled_frame_num=*/nullptr, Resolution({.width = 2, .height = 2}),
+      FrameReader::kNoScale);
+  EXPECT_EQ(2, buffer->width());
+  EXPECT_EQ(2, buffer->height());
+}
+
+class Y4mFrameReaderRepeatModeTest
+    : public Y4mFrameReaderTest,
+      public ::testing::WithParamInterface<
+          std::tuple<RepeatMode, std::vector<int>>> {};
+
+TEST_P(Y4mFrameReaderRepeatModeTest, PullFrame) {
+  RepeatMode mode = std::get<0>(GetParam());
+  std::vector<int> expected_frames = std::get<1>(GetParam());
+
+  reader_ = CreateY4mFrameReader(filepath_, mode);
+  std::vector<int> read_frames;
+  for (size_t i = 0; i < expected_frames.size(); ++i) {
+    rtc::scoped_refptr<I420BufferInterface> buffer = reader_->PullFrame();
+    read_frames.push_back(*buffer->DataY());
+  }
+  EXPECT_EQ(expected_frames, read_frames);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    Y4mFrameReaderTest,
+    Y4mFrameReaderRepeatModeTest,
+    ::testing::ValuesIn(
+        {std::make_tuple(RepeatMode::kSingle, std::vector<int>{0, 1, 2}),
+         std::make_tuple(RepeatMode::kRepeat,
+                         std::vector<int>{0, 1, 2, 0, 1, 2}),
+         std::make_tuple(RepeatMode::kPingPong,
+                         std::vector<int>{0, 1, 2, 1, 0, 1, 2})}));
+
+class Y4mFrameReaderFramerateScaleTest
+    : public Y4mFrameReaderTest,
+      public ::testing::WithParamInterface<
+          std::tuple<Ratio, std::vector<int>>> {};
+
+TEST_P(Y4mFrameReaderFramerateScaleTest, PullFrame) {
+  Ratio framerate_scale = std::get<0>(GetParam());
+  std::vector<int> expected_frames = std::get<1>(GetParam());
+
+  std::vector<int> actual_frames;
+  for (size_t i = 0; i < expected_frames.size(); ++i) {
+    int pulled_frame;
+    rtc::scoped_refptr<I420BufferInterface> buffer =
+        reader_->PullFrame(&pulled_frame, kResolution, framerate_scale);
+    actual_frames.push_back(pulled_frame);
+  }
+  EXPECT_EQ(expected_frames, actual_frames);
+}
+
+INSTANTIATE_TEST_SUITE_P(Y4mFrameReaderTest,
+                         Y4mFrameReaderFramerateScaleTest,
+                         ::testing::ValuesIn({
+                             std::make_tuple(Ratio({.num = 1, .den = 2}),
+                                             std::vector<int>{0, 2, 4}),
+                             std::make_tuple(Ratio({.num = 2, .den = 3}),
+                                             std::vector<int>{0, 1, 3, 4, 6}),
+                             std::make_tuple(Ratio({.num = 2, .den = 1}),
+                                             std::vector<int>{0, 0, 1, 1}),
+                         }));
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/y4m_frame_writer.cc b/third_party/libwebrtc/test/testsupport/y4m_frame_writer.cc
new file mode 100644
index 0000000000..1bb4543963
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/y4m_frame_writer.cc
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <string>
+
+#include "rtc_base/logging.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+Y4mFrameWriterImpl::Y4mFrameWriterImpl(std::string output_filename,
+                                       int width,
+                                       int height,
+                                       int frame_rate)
+    : YuvFrameWriterImpl(output_filename, width, height),
+      frame_rate_(frame_rate) {}
+
+Y4mFrameWriterImpl::~Y4mFrameWriterImpl() = default;
+
+bool Y4mFrameWriterImpl::Init() {
+  if (!YuvFrameWriterImpl::Init()) {
+    return false;
+  }
+  int bytes_written = fprintf(output_file_, "YUV4MPEG2 W%d H%d F%d:1 C420\n",
+                              width_, height_, frame_rate_);
+  if (bytes_written < 0) {
+    RTC_LOG(LS_ERROR) << "Failed to write Y4M file header to file: "
+                      << output_filename_.c_str();
+    return false;
+  }
+  return true;
+}
+
+bool Y4mFrameWriterImpl::WriteFrame(const uint8_t* frame_buffer) {
+  if (output_file_ == nullptr) {
+    RTC_LOG(LS_ERROR) << "Y4mFrameWriterImpl is not initialized.";
+    return false;
+  }
+  int bytes_written = fprintf(output_file_, "FRAME\n");
+  if (bytes_written < 0) {
+    RTC_LOG(LS_ERROR) << "Couldn't write Y4M frame header to file: "
+                      << output_filename_.c_str();
+    return false;
+  }
+  return YuvFrameWriterImpl::WriteFrame(frame_buffer);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/y4m_frame_writer_unittest.cc b/third_party/libwebrtc/test/testsupport/y4m_frame_writer_unittest.cc
new file mode 100644
index 0000000000..f12a4b8e4f
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/y4m_frame_writer_unittest.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdint.h>
+#include <stdio.h>
+#include <string.h>
+
+#include <memory>
+#include <string>
+
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const size_t kFrameWidth = 50;
+const size_t kFrameHeight = 20;
+const size_t kFrameLength = 3 * kFrameWidth * kFrameHeight / 2;  // I420.
+const size_t kFrameRate = 30;
+
+const std::string kFileHeader = "YUV4MPEG2 W50 H20 F30:1 C420\n";
+const std::string kFrameHeader = "FRAME\n";
+}  // namespace
+
+class Y4mFrameWriterTest : public ::testing::Test {
+ protected:
+  Y4mFrameWriterTest() = default;
+  ~Y4mFrameWriterTest() override = default;
+
+  void SetUp() override {
+    temp_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                "y4m_frame_writer_unittest");
+    frame_writer_.reset(new Y4mFrameWriterImpl(temp_filename_, kFrameWidth,
+                                               kFrameHeight, kFrameRate));
+    ASSERT_TRUE(frame_writer_->Init());
+  }
+
+  void TearDown() override { remove(temp_filename_.c_str()); }
+
+  std::unique_ptr<FrameWriter> frame_writer_;
+  std::string temp_filename_;
+};
+
+TEST_F(Y4mFrameWriterTest, InitSuccess) {}
+
+TEST_F(Y4mFrameWriterTest, FrameLength) {
+  EXPECT_EQ(kFrameLength, frame_writer_->FrameLength());
+}
+
+TEST_F(Y4mFrameWriterTest, WriteFrame) {
+  uint8_t buffer[kFrameLength];
+  memset(buffer, 9, kFrameLength);  // Write lots of 9s to the buffer.
+  bool result = frame_writer_->WriteFrame(buffer);
+  ASSERT_TRUE(result);
+  result = frame_writer_->WriteFrame(buffer);
+  ASSERT_TRUE(result);
+
+  frame_writer_->Close();
+  EXPECT_EQ(kFileHeader.size() + 2 * kFrameHeader.size() + 2 * kFrameLength,
+            GetFileSize(temp_filename_));
+}
+
+TEST_F(Y4mFrameWriterTest, WriteFrameUninitialized) {
+  uint8_t buffer[kFrameLength];
+  Y4mFrameWriterImpl frame_writer(temp_filename_, kFrameWidth, kFrameHeight,
+                                  kFrameRate);
+  EXPECT_FALSE(frame_writer.WriteFrame(buffer));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/yuv_frame_reader.cc b/third_party/libwebrtc/test/testsupport/yuv_frame_reader.cc
new file mode 100644
index 0000000000..02c1a68008
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/yuv_frame_reader.cc
@@ -0,0 +1,162 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/logging.h"
+#include "test/frame_utils.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+using RepeatMode = YuvFrameReaderImpl::RepeatMode;
+
+int WrapFrameNum(int frame_num, int num_frames, RepeatMode mode) {
+  RTC_CHECK_GE(frame_num, 0) << "frame_num cannot be negative";
+  RTC_CHECK_GT(num_frames, 0) << "num_frames must be greater than 0";
+  if (mode == RepeatMode::kSingle) {
+    return frame_num;
+  }
+  if (mode == RepeatMode::kRepeat) {
+    return frame_num % num_frames;
+  }
+
+  RTC_CHECK_EQ(RepeatMode::kPingPong, mode);
+  int cycle_len = 2 * (num_frames - 1);
+  int wrapped_num = frame_num % cycle_len;
+  if (wrapped_num >= num_frames) {
+    return cycle_len - wrapped_num;
+  }
+  return wrapped_num;
+}
+
+rtc::scoped_refptr<I420Buffer> Scale(rtc::scoped_refptr<I420Buffer> buffer,
+                                     Resolution resolution) {
+  if (buffer->width() == resolution.width &&
+      buffer->height() == resolution.height) {
+    return buffer;
+  }
+  rtc::scoped_refptr<I420Buffer> scaled(
+      I420Buffer::Create(resolution.width, resolution.height));
+  scaled->ScaleFrom(*buffer.get());
+  return scaled;
+}
+}  // namespace
+
+int YuvFrameReaderImpl::RateScaler::Skip(Ratio framerate_scale) {
+  ticks_ = ticks_.value_or(framerate_scale.num);
+  int skip = 0;
+  while (ticks_ <= 0) {
+    *ticks_ += framerate_scale.num;
+    ++skip;
+  }
+  *ticks_ -= framerate_scale.den;
+  return skip;
+}
+
+YuvFrameReaderImpl::YuvFrameReaderImpl(std::string filepath,
+                                       Resolution resolution,
+                                       RepeatMode repeat_mode)
+    : filepath_(filepath),
+      resolution_(resolution),
+      repeat_mode_(repeat_mode),
+      num_frames_(0),
+      frame_num_(0),
+      frame_size_bytes_(0),
+      header_size_bytes_(0),
+      file_(nullptr) {}
+
+YuvFrameReaderImpl::~YuvFrameReaderImpl() {
+  if (file_ != nullptr) {
+    fclose(file_);
+    file_ = nullptr;
+  }
+}
+
+void YuvFrameReaderImpl::Init() {
+  RTC_CHECK_GT(resolution_.width, 0) << "Width must be positive";
+  RTC_CHECK_GT(resolution_.height, 0) << "Height must be positive";
+  frame_size_bytes_ =
+      CalcBufferSize(VideoType::kI420, resolution_.width, resolution_.height);
+
+  file_ = fopen(filepath_.c_str(), "rb");
+  RTC_CHECK(file_ != NULL) << "Cannot open " << filepath_;
+
+  size_t file_size_bytes = GetFileSize(filepath_);
+  RTC_CHECK_GT(file_size_bytes, 0u) << "File " << filepath_ << " is empty";
+
+  num_frames_ = static_cast<int>(file_size_bytes / frame_size_bytes_);
+  RTC_CHECK_GT(num_frames_, 0u) << "File " << filepath_ << " is too small";
+}
+
+rtc::scoped_refptr<I420Buffer> YuvFrameReaderImpl::PullFrame() {
+  return PullFrame(/*frame_num=*/nullptr);
+}
+
+rtc::scoped_refptr<I420Buffer> YuvFrameReaderImpl::PullFrame(int* frame_num) {
+  return PullFrame(frame_num, resolution_, /*framerate_scale=*/kNoScale);
+}
+
+rtc::scoped_refptr<I420Buffer> YuvFrameReaderImpl::PullFrame(
+    int* frame_num,
+    Resolution resolution,
+    Ratio framerate_scale) {
+  frame_num_ += framerate_scaler_.Skip(framerate_scale);
+  auto buffer = ReadFrame(frame_num_, resolution);
+  if (frame_num != nullptr) {
+    *frame_num = frame_num_;
+  }
+  return buffer;
+}
+
+rtc::scoped_refptr<I420Buffer> YuvFrameReaderImpl::ReadFrame(int frame_num) {
+  return ReadFrame(frame_num, resolution_);
+}
+
+rtc::scoped_refptr<I420Buffer> YuvFrameReaderImpl::ReadFrame(
+    int frame_num,
+    Resolution resolution) {
+  int wrapped_num = WrapFrameNum(frame_num, num_frames_, repeat_mode_);
+  if (wrapped_num >= num_frames_) {
+    RTC_CHECK_EQ(RepeatMode::kSingle, repeat_mode_);
+    return nullptr;
+  }
+  fseek(file_, header_size_bytes_ + wrapped_num * frame_size_bytes_, SEEK_SET);
+  auto buffer = ReadI420Buffer(resolution_.width, resolution_.height, file_);
+  RTC_CHECK(buffer != nullptr);
+
+  return Scale(buffer, resolution);
+}
+
+std::unique_ptr<FrameReader> CreateYuvFrameReader(std::string filepath,
+                                                  Resolution resolution) {
+  return CreateYuvFrameReader(filepath, resolution,
+                              YuvFrameReaderImpl::RepeatMode::kSingle);
+}
+
+std::unique_ptr<FrameReader> CreateYuvFrameReader(
+    std::string filepath,
+    Resolution resolution,
+    YuvFrameReaderImpl::RepeatMode repeat_mode) {
+  YuvFrameReaderImpl* frame_reader =
+      new YuvFrameReaderImpl(filepath, resolution, repeat_mode);
+  frame_reader->Init();
+  return std::unique_ptr<FrameReader>(frame_reader);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/yuv_frame_reader_unittest.cc b/third_party/libwebrtc/test/testsupport/yuv_frame_reader_unittest.cc
new file mode 100644
index 0000000000..b9ea2d0c46
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/yuv_frame_reader_unittest.cc
@@ -0,0 +1,146 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdint.h>
+#include <stdio.h>
+
+#include <memory>
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_reader.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+using Ratio = FrameReader::Ratio;
+using RepeatMode = YuvFrameReaderImpl::RepeatMode;
+
+constexpr Resolution kResolution({.width = 1, .height = 1});
+constexpr char kFrameContent[3][3] = {{0, 1, 2}, {1, 2, 3}, {2, 3, 4}};
+constexpr int kNumFrames = sizeof(kFrameContent) / sizeof(kFrameContent[0]);
+}  // namespace
+
+class YuvFrameReaderTest : public ::testing::Test {
+ protected:
+  YuvFrameReaderTest() = default;
+  ~YuvFrameReaderTest() override = default;
+
+  void SetUp() override {
+    filepath_ = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                           "yuv_frame_reader_unittest");
+    FILE* file = fopen(filepath_.c_str(), "wb");
+    fwrite(kFrameContent, 1, sizeof(kFrameContent), file);
+    fclose(file);
+
+    reader_ = CreateYuvFrameReader(filepath_, kResolution);
+  }
+
+  void TearDown() override { remove(filepath_.c_str()); }
+
+  std::string filepath_;
+  std::unique_ptr<FrameReader> reader_;
+};
+
+TEST_F(YuvFrameReaderTest, num_frames) {
+  EXPECT_EQ(kNumFrames, reader_->num_frames());
+}
+
+TEST_F(YuvFrameReaderTest, PullFrame_frameContent) {
+  rtc::scoped_refptr<I420BufferInterface> buffer = reader_->PullFrame();
+  EXPECT_EQ(kFrameContent[0][0], *buffer->DataY());
+  EXPECT_EQ(kFrameContent[0][1], *buffer->DataU());
+  EXPECT_EQ(kFrameContent[0][2], *buffer->DataV());
+}
+
+TEST_F(YuvFrameReaderTest, ReadFrame_randomOrder) {
+  std::vector<int> expected_frames = {2, 0, 1};
+  std::vector<int> actual_frames;
+  for (int frame_num : expected_frames) {
+    rtc::scoped_refptr<I420BufferInterface> buffer =
+        reader_->ReadFrame(frame_num);
+    actual_frames.push_back(*buffer->DataY());
+  }
+  EXPECT_EQ(expected_frames, actual_frames);
+}
+
+TEST_F(YuvFrameReaderTest, PullFrame_scale) {
+  rtc::scoped_refptr<I420BufferInterface> buffer = reader_->PullFrame(
+      /*pulled_frame_num=*/nullptr, Resolution({.width = 2, .height = 2}),
+      FrameReader::kNoScale);
+  EXPECT_EQ(2, buffer->width());
+  EXPECT_EQ(2, buffer->height());
+}
+
+class YuvFrameReaderRepeatModeTest
+    : public YuvFrameReaderTest,
+      public ::testing::WithParamInterface<
+          std::tuple<RepeatMode, std::vector<int>>> {};
+
+TEST_P(YuvFrameReaderRepeatModeTest, PullFrame) {
+  RepeatMode mode = std::get<0>(GetParam());
+  std::vector<int> expected_frames = std::get<1>(GetParam());
+
+  reader_ = CreateYuvFrameReader(filepath_, kResolution, mode);
+  std::vector<int> read_frames;
+  for (size_t i = 0; i < expected_frames.size(); ++i) {
+    rtc::scoped_refptr<I420BufferInterface> buffer = reader_->PullFrame();
+    read_frames.push_back(*buffer->DataY());
+  }
+  EXPECT_EQ(expected_frames, read_frames);
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    YuvFrameReaderTest,
+    YuvFrameReaderRepeatModeTest,
+    ::testing::ValuesIn(
+        {std::make_tuple(RepeatMode::kSingle, std::vector<int>{0, 1, 2}),
+         std::make_tuple(RepeatMode::kRepeat,
+                         std::vector<int>{0, 1, 2, 0, 1, 2}),
+         std::make_tuple(RepeatMode::kPingPong,
+                         std::vector<int>{0, 1, 2, 1, 0, 1, 2})}));
+
+class YuvFrameReaderFramerateScaleTest
+    : public YuvFrameReaderTest,
+      public ::testing::WithParamInterface<
+          std::tuple<Ratio, std::vector<int>>> {};
+
+TEST_P(YuvFrameReaderFramerateScaleTest, PullFrame) {
+  Ratio framerate_scale = std::get<0>(GetParam());
+  std::vector<int> expected_frames = std::get<1>(GetParam());
+
+  std::vector<int> actual_frames;
+  for (size_t i = 0; i < expected_frames.size(); ++i) {
+    int pulled_frame;
+    rtc::scoped_refptr<I420BufferInterface> buffer =
+        reader_->PullFrame(&pulled_frame, kResolution, framerate_scale);
+    actual_frames.push_back(pulled_frame);
+  }
+  EXPECT_EQ(expected_frames, actual_frames);
+}
+
+INSTANTIATE_TEST_SUITE_P(YuvFrameReaderTest,
+                         YuvFrameReaderFramerateScaleTest,
+                         ::testing::ValuesIn({
+                             std::make_tuple(Ratio({.num = 1, .den = 2}),
+                                             std::vector<int>{0, 2, 4}),
+                             std::make_tuple(Ratio({.num = 2, .den = 3}),
+                                             std::vector<int>{0, 1, 3, 4, 6}),
+                             std::make_tuple(Ratio({.num = 2, .den = 1}),
+                                             std::vector<int>{0, 0, 1, 1}),
+                         }));
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/yuv_frame_writer.cc b/third_party/libwebrtc/test/testsupport/yuv_frame_writer.cc
new file mode 100644
index 0000000000..e5e0a6ba7f
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/yuv_frame_writer.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <string>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+YuvFrameWriterImpl::YuvFrameWriterImpl(std::string output_filename,
+                                       int width,
+                                       int height)
+    : output_filename_(output_filename),
+      frame_length_in_bytes_(0),
+      width_(width),
+      height_(height),
+      output_file_(nullptr) {}
+
+YuvFrameWriterImpl::~YuvFrameWriterImpl() {
+  Close();
+}
+
+bool YuvFrameWriterImpl::Init() {
+  if (width_ <= 0 || height_ <= 0) {
+    RTC_LOG(LS_ERROR) << "Frame width and height must be positive.";
+    return false;
+  }
+  frame_length_in_bytes_ =
+      width_ * height_ + 2 * ((width_ + 1) / 2) * ((height_ + 1) / 2);
+
+  output_file_ = fopen(output_filename_.c_str(), "wb");
+  if (output_file_ == nullptr) {
+    RTC_LOG(LS_ERROR) << "Couldn't open output file: "
+                      << output_filename_.c_str();
+    return false;
+  }
+  return true;
+}
+
+bool YuvFrameWriterImpl::WriteFrame(const uint8_t* frame_buffer) {
+  RTC_DCHECK(frame_buffer);
+  if (output_file_ == nullptr) {
+    RTC_LOG(LS_ERROR) << "YuvFrameWriterImpl is not initialized.";
+    return false;
+  }
+  size_t bytes_written =
+      fwrite(frame_buffer, 1, frame_length_in_bytes_, output_file_);
+  if (bytes_written != frame_length_in_bytes_) {
+    RTC_LOG(LS_ERROR) << "Cound't write frame to file: "
+                      << output_filename_.c_str();
+    return false;
+  }
+  return true;
+}
+
+void YuvFrameWriterImpl::Close() {
+  if (output_file_ != nullptr) {
+    fclose(output_file_);
+    output_file_ = nullptr;
+  }
+}
+
+size_t YuvFrameWriterImpl::FrameLength() {
+  return frame_length_in_bytes_;
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/testsupport/yuv_frame_writer_unittest.cc b/third_party/libwebrtc/test/testsupport/yuv_frame_writer_unittest.cc
new file mode 100644
index 0000000000..13ed715b9e
--- /dev/null
+++ b/third_party/libwebrtc/test/testsupport/yuv_frame_writer_unittest.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdint.h>
+#include <stdio.h>
+#include <string.h>
+
+#include <memory>
+#include <string>
+
+#include "test/gtest.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/frame_writer.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const size_t kFrameWidth = 50;
+const size_t kFrameHeight = 20;
+const size_t kFrameLength = 3 * kFrameWidth * kFrameHeight / 2;  // I420.
+}  // namespace
+
+class YuvFrameWriterTest : public ::testing::Test {
+ protected:
+  YuvFrameWriterTest() = default;
+  ~YuvFrameWriterTest() override = default;
+
+  void SetUp() override {
+    temp_filename_ = webrtc::test::TempFilename(webrtc::test::OutputPath(),
+                                                "yuv_frame_writer_unittest");
+    frame_writer_.reset(
+        new YuvFrameWriterImpl(temp_filename_, kFrameWidth, kFrameHeight));
+    ASSERT_TRUE(frame_writer_->Init());
+  }
+
+  void TearDown() override { remove(temp_filename_.c_str()); }
+
+  std::unique_ptr<FrameWriter> frame_writer_;
+  std::string temp_filename_;
+};
+
+TEST_F(YuvFrameWriterTest, InitSuccess) {}
+
+TEST_F(YuvFrameWriterTest, FrameLength) {
+  EXPECT_EQ(kFrameLength, frame_writer_->FrameLength());
+}
+
+TEST_F(YuvFrameWriterTest, WriteFrame) {
+  uint8_t buffer[kFrameLength];
+  memset(buffer, 9, kFrameLength);  // Write lots of 9s to the buffer.
+  bool result = frame_writer_->WriteFrame(buffer);
+  ASSERT_TRUE(result);
+
+  frame_writer_->Close();
+  EXPECT_EQ(kFrameLength, GetFileSize(temp_filename_));
+}
+
+TEST_F(YuvFrameWriterTest, WriteFrameUninitialized) {
+  uint8_t buffer[kFrameLength];
+  YuvFrameWriterImpl frame_writer(temp_filename_, kFrameWidth, kFrameHeight);
+  EXPECT_FALSE(frame_writer.WriteFrame(buffer));
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/BUILD.gn b/third_party/libwebrtc/test/time_controller/BUILD.gn
new file mode 100644
index 0000000000..b4b368a42a
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/BUILD.gn
@@ -0,0 +1,70 @@
+# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_library("time_controller") {
+  testonly = true
+  sources = [
+    "external_time_controller.cc",
+    "external_time_controller.h",
+    "real_time_controller.cc",
+    "real_time_controller.h",
+    "simulated_task_queue.cc",
+    "simulated_task_queue.h",
+    "simulated_thread.cc",
+    "simulated_thread.h",
+    "simulated_time_controller.cc",
+    "simulated_time_controller.h",
+  ]
+
+  deps = [
+    "../../api:sequence_checker",
+    "../../api:time_controller",
+    "../../api/task_queue",
+    "../../api/task_queue:default_task_queue_factory",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../rtc_base:checks",
+    "../../rtc_base:null_socket_server",
+    "../../rtc_base:platform_thread_types",
+    "../../rtc_base:rtc_base_tests_utils",
+    "../../rtc_base:rtc_event",
+    "../../rtc_base/synchronization:mutex",
+    "../../rtc_base/synchronization:yield_policy",
+    "../../system_wrappers",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/strings",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("time_controller_unittests") {
+    testonly = true
+    sources = [
+      "external_time_controller_unittest.cc",
+      "simulated_time_controller_unittest.cc",
+      "time_controller_conformance_test.cc",
+    ]
+    deps = [
+      ":time_controller",
+      "../:test_support",
+      "../../api:time_controller",
+      "../../api/units:time_delta",
+      "../../rtc_base:macromagic",
+      "../../rtc_base:rtc_event",
+      "../../rtc_base:rtc_task_queue",
+      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base:threading",
+      "../../rtc_base/synchronization:mutex",
+      "../../rtc_base/task_utils:repeating_task",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/test/time_controller/external_time_controller.cc b/third_party/libwebrtc/test/time_controller/external_time_controller.cc
new file mode 100644
index 0000000000..f652eb686c
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/external_time_controller.cc
@@ -0,0 +1,134 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/time_controller/external_time_controller.h"
+
+#include <algorithm>
+#include <map>
+#include <memory>
+#include <utility>
+
+#include "absl/functional/any_invocable.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/synchronization/yield_policy.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+// Wraps a TaskQueue so that it can reschedule the time controller whenever
+// an external call schedules a new task.
+class ExternalTimeController::TaskQueueWrapper : public TaskQueueBase {
+ public:
+  TaskQueueWrapper(ExternalTimeController* parent,
+                   std::unique_ptr<TaskQueueBase, TaskQueueDeleter> base)
+      : parent_(parent), base_(std::move(base)) {}
+
+  void PostTask(absl::AnyInvocable<void() &&> task) override {
+    parent_->UpdateTime();
+    base_->PostTask(TaskWrapper(std::move(task)));
+    parent_->ScheduleNext();
+  }
+
+  void PostDelayedTask(absl::AnyInvocable<void() &&> task,
+                       TimeDelta delay) override {
+    parent_->UpdateTime();
+    base_->PostDelayedTask(TaskWrapper(std::move(task)), delay);
+    parent_->ScheduleNext();
+  }
+
+  void PostDelayedHighPrecisionTask(absl::AnyInvocable<void() &&> task,
+                                    TimeDelta delay) override {
+    parent_->UpdateTime();
+    base_->PostDelayedHighPrecisionTask(TaskWrapper(std::move(task)), delay);
+    parent_->ScheduleNext();
+  }
+
+  void Delete() override { delete this; }
+
+ private:
+  absl::AnyInvocable<void() &&> TaskWrapper(
+      absl::AnyInvocable<void() &&> task) {
+    return [task = std::move(task), this]() mutable {
+      CurrentTaskQueueSetter current(this);
+      std::move(task)();
+    };
+  }
+
+  ExternalTimeController* const parent_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> base_;
+};
+
+ExternalTimeController::ExternalTimeController(ControlledAlarmClock* alarm)
+    : alarm_(alarm),
+      impl_(alarm_->GetClock()->CurrentTime()),
+      yield_policy_(&impl_) {
+  global_clock_.SetTime(alarm_->GetClock()->CurrentTime());
+  alarm_->SetCallback([this] { Run(); });
+}
+
+Clock* ExternalTimeController::GetClock() {
+  return alarm_->GetClock();
+}
+
+TaskQueueFactory* ExternalTimeController::GetTaskQueueFactory() {
+  return this;
+}
+
+void ExternalTimeController::AdvanceTime(TimeDelta duration) {
+  alarm_->Sleep(duration);
+}
+
+std::unique_ptr<rtc::Thread> ExternalTimeController::CreateThread(
+    const std::string& name,
+    std::unique_ptr<rtc::SocketServer> socket_server) {
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+}
+
+rtc::Thread* ExternalTimeController::GetMainThread() {
+  RTC_DCHECK_NOTREACHED();
+  return nullptr;
+}
+
+std::unique_ptr<TaskQueueBase, TaskQueueDeleter>
+ExternalTimeController::CreateTaskQueue(
+    absl::string_view name,
+    TaskQueueFactory::Priority priority) const {
+  return std::unique_ptr<TaskQueueBase, TaskQueueDeleter>(
+      new TaskQueueWrapper(const_cast<ExternalTimeController*>(this),
+                           impl_.CreateTaskQueue(name, priority)));
+}
+
+void ExternalTimeController::Run() {
+  rtc::ScopedYieldPolicy yield_policy(&impl_);
+  UpdateTime();
+  impl_.RunReadyRunners();
+  ScheduleNext();
+}
+
+void ExternalTimeController::UpdateTime() {
+  Timestamp now = alarm_->GetClock()->CurrentTime();
+  impl_.AdvanceTime(now);
+  global_clock_.SetTime(now);
+}
+
+void ExternalTimeController::ScheduleNext() {
+  RTC_DCHECK_EQ(impl_.CurrentTime(), alarm_->GetClock()->CurrentTime());
+  TimeDelta delay =
+      std::max(impl_.NextRunTime() - impl_.CurrentTime(), TimeDelta::Zero());
+  if (delay.IsFinite()) {
+    alarm_->ScheduleAlarmAt(alarm_->GetClock()->CurrentTime() + delay);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/external_time_controller.h b/third_party/libwebrtc/test/time_controller/external_time_controller.h
new file mode 100644
index 0000000000..a67f2557b4
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/external_time_controller.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TIME_CONTROLLER_EXTERNAL_TIME_CONTROLLER_H_
+#define TEST_TIME_CONTROLLER_EXTERNAL_TIME_CONTROLLER_H_
+
+#include <functional>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/time_controller.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "system_wrappers/include/clock.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+// TimeController implementation built on an external controlled alarm.
+// This implementation is used to delegate scheduling and execution to an
+// external run loop.
+class ExternalTimeController : public TimeController, public TaskQueueFactory {
+ public:
+  explicit ExternalTimeController(ControlledAlarmClock* alarm);
+
+  // Implementation of TimeController.
+  Clock* GetClock() override;
+  TaskQueueFactory* GetTaskQueueFactory() override;
+  void AdvanceTime(TimeDelta duration) override;
+  std::unique_ptr<rtc::Thread> CreateThread(
+      const std::string& name,
+      std::unique_ptr<rtc::SocketServer> socket_server) override;
+  rtc::Thread* GetMainThread() override;
+
+  // Implementation of TaskQueueFactory.
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+      absl::string_view name,
+      TaskQueueFactory::Priority priority) const override;
+
+ private:
+  class TaskQueueWrapper;
+
+  // Executes any tasks scheduled at or before the current time.  May call
+  // `ScheduleNext` to schedule the next call to `Run`.
+  void Run();
+
+  void UpdateTime();
+  void ScheduleNext();
+
+  ControlledAlarmClock* alarm_;
+  sim_time_impl::SimulatedTimeControllerImpl impl_;
+  rtc::ScopedYieldPolicy yield_policy_;
+
+  // Overrides the global rtc::Clock to ensure that it reports the same times as
+  // the time controller.
+  rtc::ScopedBaseFakeClock global_clock_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_TIME_CONTROLLER_EXTERNAL_TIME_CONTROLLER_H_
diff --git a/third_party/libwebrtc/test/time_controller/external_time_controller_unittest.cc b/third_party/libwebrtc/test/time_controller/external_time_controller_unittest.cc
new file mode 100644
index 0000000000..13d63fe8ed
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/external_time_controller_unittest.cc
@@ -0,0 +1,179 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/time_controller/external_time_controller.h"
+
+#include <atomic>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+// NOTE: Since these tests rely on real time behavior, they will be flaky
+// if run on heavily loaded systems.
+namespace webrtc {
+namespace {
+using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::MockFunction;
+using ::testing::NiceMock;
+using ::testing::Return;
+constexpr Timestamp kStartTime = Timestamp::Seconds(1000);
+
+class FakeAlarm : public ControlledAlarmClock {
+ public:
+  explicit FakeAlarm(Timestamp start_time);
+
+  Clock* GetClock() override;
+  bool ScheduleAlarmAt(Timestamp deadline) override;
+  void SetCallback(std::function<void()> callback) override;
+  void Sleep(TimeDelta duration) override;
+
+ private:
+  SimulatedClock clock_;
+  Timestamp deadline_;
+  std::function<void()> callback_;
+};
+
+FakeAlarm::FakeAlarm(Timestamp start_time)
+    : clock_(start_time),
+      deadline_(Timestamp::PlusInfinity()),
+      callback_([] {}) {}
+
+Clock* FakeAlarm::GetClock() {
+  return &clock_;
+}
+
+bool FakeAlarm::ScheduleAlarmAt(Timestamp deadline) {
+  if (deadline < deadline_) {
+    deadline_ = deadline;
+    return true;
+  }
+  return false;
+}
+
+void FakeAlarm::SetCallback(std::function<void()> callback) {
+  callback_ = callback;
+}
+
+void FakeAlarm::Sleep(TimeDelta duration) {
+  Timestamp end_time = clock_.CurrentTime() + duration;
+
+  while (deadline_ <= end_time) {
+    clock_.AdvanceTime(deadline_ - clock_.CurrentTime());
+    deadline_ = Timestamp::PlusInfinity();
+    callback_();
+  }
+
+  clock_.AdvanceTime(end_time - clock_.CurrentTime());
+}
+
+}  // namespace
+
+TEST(ExternalTimeControllerTest, TaskIsStoppedOnStop) {
+  const TimeDelta kShortInterval = TimeDelta::Millis(5);
+  const TimeDelta kLongInterval = TimeDelta::Millis(20);
+  const int kShortIntervalCount = 4;
+  const int kMargin = 1;
+  FakeAlarm alarm(kStartTime);
+  ExternalTimeController time_simulation(&alarm);
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+  std::atomic_int counter(0);
+  auto handle = RepeatingTaskHandle::Start(task_queue.Get(), [&] {
+    if (++counter >= kShortIntervalCount)
+      return kLongInterval;
+    return kShortInterval;
+  });
+  // Sleep long enough to go through the initial phase.
+  time_simulation.AdvanceTime(kShortInterval * (kShortIntervalCount + kMargin));
+  EXPECT_EQ(counter.load(), kShortIntervalCount);
+
+  task_queue.PostTask(
+      [handle = std::move(handle)]() mutable { handle.Stop(); });
+
+  // Sleep long enough that the task would run at least once more if not
+  // stopped.
+  time_simulation.AdvanceTime(kLongInterval * 2);
+  EXPECT_EQ(counter.load(), kShortIntervalCount);
+}
+
+TEST(ExternalTimeControllerTest, TaskCanStopItself) {
+  std::atomic_int counter(0);
+  FakeAlarm alarm(kStartTime);
+  ExternalTimeController time_simulation(&alarm);
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+
+  RepeatingTaskHandle handle;
+  task_queue.PostTask([&] {
+    handle = RepeatingTaskHandle::Start(task_queue.Get(), [&] {
+      ++counter;
+      handle.Stop();
+      return TimeDelta::Millis(2);
+    });
+  });
+  time_simulation.AdvanceTime(TimeDelta::Millis(10));
+  EXPECT_EQ(counter.load(), 1);
+}
+
+TEST(ExternalTimeControllerTest, YieldForTask) {
+  FakeAlarm alarm(kStartTime);
+  ExternalTimeController time_simulation(&alarm);
+
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+
+  rtc::Event event;
+  task_queue.PostTask([&] { event.Set(); });
+  EXPECT_TRUE(event.Wait(TimeDelta::Millis(200)));
+}
+
+TEST(ExternalTimeControllerTest, TasksYieldToEachOther) {
+  FakeAlarm alarm(kStartTime);
+  ExternalTimeController time_simulation(&alarm);
+
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+  rtc::TaskQueue other_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "OtherQueue", TaskQueueFactory::Priority::NORMAL));
+
+  task_queue.PostTask([&] {
+    rtc::Event event;
+    other_queue.PostTask([&] { event.Set(); });
+    EXPECT_TRUE(event.Wait(TimeDelta::Millis(200)));
+  });
+
+  time_simulation.AdvanceTime(TimeDelta::Millis(300));
+}
+
+TEST(ExternalTimeControllerTest, CurrentTaskQueue) {
+  FakeAlarm alarm(kStartTime);
+  ExternalTimeController time_simulation(&alarm);
+
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+
+  task_queue.PostTask([&] { EXPECT_TRUE(task_queue.IsCurrent()); });
+
+  time_simulation.AdvanceTime(TimeDelta::Millis(10));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/real_time_controller.cc b/third_party/libwebrtc/test/time_controller/real_time_controller.cc
new file mode 100644
index 0000000000..7cc750d6d4
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/real_time_controller.cc
@@ -0,0 +1,66 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/time_controller/real_time_controller.h"
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "rtc_base/null_socket_server.h"
+
+namespace webrtc {
+namespace {
+class MainThread : public rtc::Thread {
+ public:
+  MainThread()
+      : Thread(std::make_unique<rtc::NullSocketServer>(), false),
+        current_setter_(this) {
+    DoInit();
+  }
+  ~MainThread() {
+    Stop();
+    DoDestroy();
+  }
+
+ private:
+  CurrentThreadSetter current_setter_;
+};
+}  // namespace
+RealTimeController::RealTimeController()
+    : task_queue_factory_(CreateDefaultTaskQueueFactory()),
+      main_thread_(std::make_unique<MainThread>()) {
+  main_thread_->SetName("Main", this);
+}
+
+Clock* RealTimeController::GetClock() {
+  return Clock::GetRealTimeClock();
+}
+
+TaskQueueFactory* RealTimeController::GetTaskQueueFactory() {
+  return task_queue_factory_.get();
+}
+
+std::unique_ptr<rtc::Thread> RealTimeController::CreateThread(
+    const std::string& name,
+    std::unique_ptr<rtc::SocketServer> socket_server) {
+  if (!socket_server)
+    socket_server = std::make_unique<rtc::NullSocketServer>();
+  auto res = std::make_unique<rtc::Thread>(std::move(socket_server));
+  res->SetName(name, nullptr);
+  res->Start();
+  return res;
+}
+
+rtc::Thread* RealTimeController::GetMainThread() {
+  return main_thread_.get();
+}
+
+void RealTimeController::AdvanceTime(TimeDelta duration) {
+  main_thread_->ProcessMessages(duration.ms());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/real_time_controller.h b/third_party/libwebrtc/test/time_controller/real_time_controller.h
new file mode 100644
index 0000000000..5f02eaf85f
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/real_time_controller.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TIME_CONTROLLER_REAL_TIME_CONTROLLER_H_
+#define TEST_TIME_CONTROLLER_REAL_TIME_CONTROLLER_H_
+
+#include <functional>
+#include <memory>
+
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/time_controller.h"
+#include "api/units/time_delta.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+class RealTimeController : public TimeController {
+ public:
+  RealTimeController();
+
+  Clock* GetClock() override;
+  TaskQueueFactory* GetTaskQueueFactory() override;
+  std::unique_ptr<rtc::Thread> CreateThread(
+      const std::string& name,
+      std::unique_ptr<rtc::SocketServer> socket_server) override;
+  rtc::Thread* GetMainThread() override;
+  void AdvanceTime(TimeDelta duration) override;
+
+ private:
+  const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  const std::unique_ptr<rtc::Thread> main_thread_;
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_TIME_CONTROLLER_REAL_TIME_CONTROLLER_H_
diff --git a/third_party/libwebrtc/test/time_controller/simulated_task_queue.cc b/third_party/libwebrtc/test/time_controller/simulated_task_queue.cc
new file mode 100644
index 0000000000..3c26721845
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/simulated_task_queue.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/time_controller/simulated_task_queue.h"
+
+#include <algorithm>
+#include <utility>
+
+namespace webrtc {
+
+SimulatedTaskQueue::SimulatedTaskQueue(
+    sim_time_impl::SimulatedTimeControllerImpl* handler,
+    absl::string_view name)
+    : handler_(handler), name_(new char[name.size()]) {
+  std::copy_n(name.begin(), name.size(), name_);
+}
+
+SimulatedTaskQueue::~SimulatedTaskQueue() {
+  handler_->Unregister(this);
+  delete[] name_;
+}
+
+void SimulatedTaskQueue::Delete() {
+  // Need to destroy the tasks outside of the lock because task destruction
+  // can lead to re-entry in SimulatedTaskQueue via custom destructors.
+  std::deque<absl::AnyInvocable<void() &&>> ready_tasks;
+  std::map<Timestamp, std::vector<absl::AnyInvocable<void() &&>>> delayed_tasks;
+  {
+    MutexLock lock(&lock_);
+    ready_tasks_.swap(ready_tasks);
+    delayed_tasks_.swap(delayed_tasks);
+  }
+  ready_tasks.clear();
+  delayed_tasks.clear();
+  delete this;
+}
+
+void SimulatedTaskQueue::RunReady(Timestamp at_time) {
+  MutexLock lock(&lock_);
+  for (auto it = delayed_tasks_.begin();
+       it != delayed_tasks_.end() && it->first <= at_time;
+       it = delayed_tasks_.erase(it)) {
+    for (auto& task : it->second) {
+      ready_tasks_.push_back(std::move(task));
+    }
+  }
+  CurrentTaskQueueSetter set_current(this);
+  while (!ready_tasks_.empty()) {
+    absl::AnyInvocable<void()&&> ready = std::move(ready_tasks_.front());
+    ready_tasks_.pop_front();
+    lock_.Unlock();
+    std::move(ready)();
+    ready = nullptr;
+    lock_.Lock();
+  }
+  if (!delayed_tasks_.empty()) {
+    next_run_time_ = delayed_tasks_.begin()->first;
+  } else {
+    next_run_time_ = Timestamp::PlusInfinity();
+  }
+}
+
+void SimulatedTaskQueue::PostTask(absl::AnyInvocable<void() &&> task) {
+  MutexLock lock(&lock_);
+  ready_tasks_.push_back(std::move(task));
+  next_run_time_ = Timestamp::MinusInfinity();
+}
+
+void SimulatedTaskQueue::PostDelayedTask(absl::AnyInvocable<void() &&> task,
+                                         TimeDelta delay) {
+  PostDelayedHighPrecisionTask(std::move(task), delay);
+}
+
+void SimulatedTaskQueue::PostDelayedHighPrecisionTask(
+    absl::AnyInvocable<void() &&> task,
+    TimeDelta delay) {
+  MutexLock lock(&lock_);
+  Timestamp target_time = handler_->CurrentTime() + delay;
+  delayed_tasks_[target_time].push_back(std::move(task));
+  next_run_time_ = std::min(next_run_time_, target_time);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/simulated_task_queue.h b/third_party/libwebrtc/test/time_controller/simulated_task_queue.h
new file mode 100644
index 0000000000..3c55f15dde
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/simulated_task_queue.h
@@ -0,0 +1,65 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TIME_CONTROLLER_SIMULATED_TASK_QUEUE_H_
+#define TEST_TIME_CONTROLLER_SIMULATED_TASK_QUEUE_H_
+
+#include <deque>
+#include <map>
+#include <memory>
+#include <vector>
+
+#include "absl/functional/any_invocable.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+class SimulatedTaskQueue : public TaskQueueBase,
+                           public sim_time_impl::SimulatedSequenceRunner {
+ public:
+  SimulatedTaskQueue(sim_time_impl::SimulatedTimeControllerImpl* handler,
+                     absl::string_view name);
+
+  ~SimulatedTaskQueue();
+
+  void RunReady(Timestamp at_time) override;
+
+  Timestamp GetNextRunTime() const override {
+    MutexLock lock(&lock_);
+    return next_run_time_;
+  }
+  TaskQueueBase* GetAsTaskQueue() override { return this; }
+
+  // TaskQueueBase interface
+  void Delete() override;
+  void PostTask(absl::AnyInvocable<void() &&> task) override;
+  void PostDelayedTask(absl::AnyInvocable<void() &&> task,
+                       TimeDelta delay) override;
+  void PostDelayedHighPrecisionTask(absl::AnyInvocable<void() &&> task,
+                                    TimeDelta delay) override;
+
+ private:
+  sim_time_impl::SimulatedTimeControllerImpl* const handler_;
+  // Using char* to be debugger friendly.
+  char* name_;
+
+  mutable Mutex lock_;
+
+  std::deque<absl::AnyInvocable<void() &&>> ready_tasks_ RTC_GUARDED_BY(lock_);
+  std::map<Timestamp, std::vector<absl::AnyInvocable<void() &&>>> delayed_tasks_
+      RTC_GUARDED_BY(lock_);
+
+  Timestamp next_run_time_ RTC_GUARDED_BY(lock_) = Timestamp::PlusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // TEST_TIME_CONTROLLER_SIMULATED_TASK_QUEUE_H_
diff --git a/third_party/libwebrtc/test/time_controller/simulated_thread.cc b/third_party/libwebrtc/test/time_controller/simulated_thread.cc
new file mode 100644
index 0000000000..bdd1096327
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/simulated_thread.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/time_controller/simulated_thread.h"
+
+#include <algorithm>
+#include <utility>
+
+namespace webrtc {
+namespace {
+
+// A socket server that does nothing. It's different from NullSocketServer in
+// that it does allow sleep/wakeup. This avoids usage of an Event instance which
+// otherwise would cause issues with the simulated Yeild behavior.
+class DummySocketServer : public rtc::SocketServer {
+ public:
+  rtc::Socket* CreateSocket(int family, int type) override {
+    RTC_DCHECK_NOTREACHED();
+    return nullptr;
+  }
+  bool Wait(TimeDelta max_wait_duration, bool process_io) override {
+    RTC_CHECK(max_wait_duration.IsZero());
+    return true;
+  }
+  void WakeUp() override {}
+};
+
+}  // namespace
+
+SimulatedThread::SimulatedThread(
+    sim_time_impl::SimulatedTimeControllerImpl* handler,
+    absl::string_view name,
+    std::unique_ptr<rtc::SocketServer> socket_server)
+    : rtc::Thread(socket_server ? std::move(socket_server)
+                                : std::make_unique<DummySocketServer>()),
+      handler_(handler),
+      name_(new char[name.size()]) {
+  std::copy_n(name.begin(), name.size(), name_);
+}
+
+SimulatedThread::~SimulatedThread() {
+  handler_->Unregister(this);
+  delete[] name_;
+}
+
+void SimulatedThread::RunReady(Timestamp at_time) {
+  CurrentThreadSetter set_current(this);
+  ProcessMessages(0);
+  int delay_ms = GetDelay();
+  MutexLock lock(&lock_);
+  if (delay_ms == kForever) {
+    next_run_time_ = Timestamp::PlusInfinity();
+  } else {
+    next_run_time_ = at_time + TimeDelta::Millis(delay_ms);
+  }
+}
+
+void SimulatedThread::BlockingCall(rtc::FunctionView<void()> functor) {
+  if (IsQuitting())
+    return;
+
+  if (IsCurrent()) {
+    functor();
+  } else {
+    TaskQueueBase* yielding_from = TaskQueueBase::Current();
+    handler_->StartYield(yielding_from);
+    RunReady(Timestamp::MinusInfinity());
+    CurrentThreadSetter set_current(this);
+    functor();
+    handler_->StopYield(yielding_from);
+  }
+}
+
+void SimulatedThread::PostTask(absl::AnyInvocable<void() &&> task) {
+  rtc::Thread::PostTask(std::move(task));
+  MutexLock lock(&lock_);
+  next_run_time_ = Timestamp::MinusInfinity();
+}
+
+void SimulatedThread::PostDelayedTask(absl::AnyInvocable<void() &&> task,
+                                      TimeDelta delay) {
+  rtc::Thread::PostDelayedTask(std::move(task), delay);
+  MutexLock lock(&lock_);
+  next_run_time_ =
+      std::min(next_run_time_, Timestamp::Millis(rtc::TimeMillis()) + delay);
+}
+
+void SimulatedThread::PostDelayedHighPrecisionTask(
+    absl::AnyInvocable<void() &&> task,
+    TimeDelta delay) {
+  rtc::Thread::PostDelayedHighPrecisionTask(std::move(task), delay);
+  MutexLock lock(&lock_);
+  next_run_time_ =
+      std::min(next_run_time_, Timestamp::Millis(rtc::TimeMillis()) + delay);
+}
+
+void SimulatedThread::Stop() {
+  Thread::Quit();
+}
+
+SimulatedMainThread::SimulatedMainThread(
+    sim_time_impl::SimulatedTimeControllerImpl* handler)
+    : SimulatedThread(handler, "main", nullptr), current_setter_(this) {}
+
+SimulatedMainThread::~SimulatedMainThread() {
+  // Removes pending tasks in case they keep shared pointer references to
+  // objects whose destructor expects to run before the Thread destructor.
+  Stop();
+  DoDestroy();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/simulated_thread.h b/third_party/libwebrtc/test/time_controller/simulated_thread.h
new file mode 100644
index 0000000000..e8e08c5000
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/simulated_thread.h
@@ -0,0 +1,66 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TIME_CONTROLLER_SIMULATED_THREAD_H_
+#define TEST_TIME_CONTROLLER_SIMULATED_THREAD_H_
+
+#include <memory>
+
+#include "rtc_base/synchronization/mutex.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+
+class SimulatedThread : public rtc::Thread,
+                        public sim_time_impl::SimulatedSequenceRunner {
+ public:
+  using CurrentThreadSetter = CurrentThreadSetter;
+  SimulatedThread(sim_time_impl::SimulatedTimeControllerImpl* handler,
+                  absl::string_view name,
+                  std::unique_ptr<rtc::SocketServer> socket_server);
+  ~SimulatedThread() override;
+
+  void RunReady(Timestamp at_time) override;
+
+  Timestamp GetNextRunTime() const override {
+    MutexLock lock(&lock_);
+    return next_run_time_;
+  }
+
+  TaskQueueBase* GetAsTaskQueue() override { return this; }
+
+  // Thread interface
+  void BlockingCall(rtc::FunctionView<void()> functor) override;
+  void PostTask(absl::AnyInvocable<void() &&> task) override;
+  void PostDelayedTask(absl::AnyInvocable<void() &&> task,
+                       TimeDelta delay) override;
+  void PostDelayedHighPrecisionTask(absl::AnyInvocable<void() &&> task,
+                                    TimeDelta delay) override;
+
+  void Stop() override;
+
+ private:
+  sim_time_impl::SimulatedTimeControllerImpl* const handler_;
+  // Using char* to be debugger friendly.
+  char* name_;
+  mutable Mutex lock_;
+  Timestamp next_run_time_ RTC_GUARDED_BY(lock_) = Timestamp::PlusInfinity();
+};
+
+class SimulatedMainThread : public SimulatedThread {
+ public:
+  explicit SimulatedMainThread(
+      sim_time_impl::SimulatedTimeControllerImpl* handler);
+  ~SimulatedMainThread();
+
+ private:
+  CurrentThreadSetter current_setter_;
+};
+}  // namespace webrtc
+#endif  // TEST_TIME_CONTROLLER_SIMULATED_THREAD_H_
diff --git a/third_party/libwebrtc/test/time_controller/simulated_time_controller.cc b/third_party/libwebrtc/test/time_controller/simulated_time_controller.cc
new file mode 100644
index 0000000000..1ed2b30dc8
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/simulated_time_controller.cc
@@ -0,0 +1,224 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/time_controller/simulated_time_controller.h"
+
+#include <algorithm>
+#include <deque>
+#include <list>
+#include <memory>
+#include <string>
+#include <thread>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "test/time_controller/simulated_task_queue.h"
+#include "test/time_controller/simulated_thread.h"
+
+namespace webrtc {
+namespace {
+// Helper function to remove from a std container by value.
+template <class C>
+bool RemoveByValue(C* vec, typename C::value_type val) {
+  auto it = std::find(vec->begin(), vec->end(), val);
+  if (it == vec->end())
+    return false;
+  vec->erase(it);
+  return true;
+}
+}  // namespace
+
+namespace sim_time_impl {
+
+SimulatedTimeControllerImpl::SimulatedTimeControllerImpl(Timestamp start_time)
+    : thread_id_(rtc::CurrentThreadId()), current_time_(start_time) {}
+
+SimulatedTimeControllerImpl::~SimulatedTimeControllerImpl() = default;
+
+std::unique_ptr<TaskQueueBase, TaskQueueDeleter>
+SimulatedTimeControllerImpl::CreateTaskQueue(
+    absl::string_view name,
+    TaskQueueFactory::Priority priority) const {
+  // TODO(srte): Remove the const cast when the interface is made mutable.
+  auto mutable_this = const_cast<SimulatedTimeControllerImpl*>(this);
+  auto task_queue = std::unique_ptr<SimulatedTaskQueue, TaskQueueDeleter>(
+      new SimulatedTaskQueue(mutable_this, name));
+  mutable_this->Register(task_queue.get());
+  return task_queue;
+}
+
+std::unique_ptr<rtc::Thread> SimulatedTimeControllerImpl::CreateThread(
+    const std::string& name,
+    std::unique_ptr<rtc::SocketServer> socket_server) {
+  auto thread =
+      std::make_unique<SimulatedThread>(this, name, std::move(socket_server));
+  Register(thread.get());
+  return thread;
+}
+
+void SimulatedTimeControllerImpl::YieldExecution() {
+  if (rtc::CurrentThreadId() == thread_id_) {
+    TaskQueueBase* yielding_from = TaskQueueBase::Current();
+    // Since we might continue execution on a process thread, we should reset
+    // the thread local task queue reference. This ensures that thread checkers
+    // won't think we are executing on the yielding task queue. It also ensure
+    // that TaskQueueBase::Current() won't return the yielding task queue.
+    TokenTaskQueue::CurrentTaskQueueSetter reset_queue(nullptr);
+    // When we yield, we don't want to risk executing further tasks on the
+    // currently executing task queue. If there's a ready task that also yields,
+    // it's added to this set as well and only tasks on the remaining task
+    // queues are executed.
+    auto inserted = yielded_.insert(yielding_from);
+    RTC_DCHECK(inserted.second);
+    RunReadyRunners();
+    yielded_.erase(inserted.first);
+  }
+}
+
+void SimulatedTimeControllerImpl::RunReadyRunners() {
+  // Using a dummy thread rather than nullptr to avoid implicit thread creation
+  // by Thread::Current().
+  SimulatedThread::CurrentThreadSetter set_current(dummy_thread_.get());
+  MutexLock lock(&lock_);
+  RTC_DCHECK_EQ(rtc::CurrentThreadId(), thread_id_);
+  Timestamp current_time = CurrentTime();
+  // Clearing `ready_runners_` in case this is a recursive call:
+  // RunReadyRunners -> Run -> Event::Wait -> Yield ->RunReadyRunners
+  ready_runners_.clear();
+
+  // We repeat until we have no ready left to handle tasks posted by ready
+  // runners.
+  while (true) {
+    for (auto* runner : runners_) {
+      if (yielded_.find(runner->GetAsTaskQueue()) == yielded_.end() &&
+          runner->GetNextRunTime() <= current_time) {
+        ready_runners_.push_back(runner);
+      }
+    }
+    if (ready_runners_.empty())
+      break;
+    while (!ready_runners_.empty()) {
+      auto* runner = ready_runners_.front();
+      ready_runners_.pop_front();
+      lock_.Unlock();
+      // Note that the RunReady function might indirectly cause a call to
+      // Unregister() which will grab `lock_` again to remove items from
+      // `ready_runners_`.
+      runner->RunReady(current_time);
+      lock_.Lock();
+    }
+  }
+}
+
+Timestamp SimulatedTimeControllerImpl::CurrentTime() const {
+  MutexLock lock(&time_lock_);
+  return current_time_;
+}
+
+Timestamp SimulatedTimeControllerImpl::NextRunTime() const {
+  Timestamp current_time = CurrentTime();
+  Timestamp next_time = Timestamp::PlusInfinity();
+  MutexLock lock(&lock_);
+  for (auto* runner : runners_) {
+    Timestamp next_run_time = runner->GetNextRunTime();
+    if (next_run_time <= current_time)
+      return current_time;
+    next_time = std::min(next_time, next_run_time);
+  }
+  return next_time;
+}
+
+void SimulatedTimeControllerImpl::AdvanceTime(Timestamp target_time) {
+  MutexLock time_lock(&time_lock_);
+  RTC_DCHECK_GE(target_time, current_time_);
+  current_time_ = target_time;
+}
+
+void SimulatedTimeControllerImpl::Register(SimulatedSequenceRunner* runner) {
+  MutexLock lock(&lock_);
+  runners_.push_back(runner);
+}
+
+void SimulatedTimeControllerImpl::Unregister(SimulatedSequenceRunner* runner) {
+  MutexLock lock(&lock_);
+  bool removed = RemoveByValue(&runners_, runner);
+  RTC_CHECK(removed);
+  RemoveByValue(&ready_runners_, runner);
+}
+
+void SimulatedTimeControllerImpl::StartYield(TaskQueueBase* yielding_from) {
+  auto inserted = yielded_.insert(yielding_from);
+  RTC_DCHECK(inserted.second);
+}
+
+void SimulatedTimeControllerImpl::StopYield(TaskQueueBase* yielding_from) {
+  yielded_.erase(yielding_from);
+}
+
+}  // namespace sim_time_impl
+
+GlobalSimulatedTimeController::GlobalSimulatedTimeController(
+    Timestamp start_time)
+    : sim_clock_(start_time.us()), impl_(start_time), yield_policy_(&impl_) {
+  global_clock_.SetTime(start_time);
+  auto main_thread = std::make_unique<SimulatedMainThread>(&impl_);
+  impl_.Register(main_thread.get());
+  main_thread_ = std::move(main_thread);
+}
+
+GlobalSimulatedTimeController::~GlobalSimulatedTimeController() = default;
+
+Clock* GlobalSimulatedTimeController::GetClock() {
+  return &sim_clock_;
+}
+
+TaskQueueFactory* GlobalSimulatedTimeController::GetTaskQueueFactory() {
+  return &impl_;
+}
+
+std::unique_ptr<rtc::Thread> GlobalSimulatedTimeController::CreateThread(
+    const std::string& name,
+    std::unique_ptr<rtc::SocketServer> socket_server) {
+  return impl_.CreateThread(name, std::move(socket_server));
+}
+
+rtc::Thread* GlobalSimulatedTimeController::GetMainThread() {
+  return main_thread_.get();
+}
+
+void GlobalSimulatedTimeController::AdvanceTime(TimeDelta duration) {
+  rtc::ScopedYieldPolicy yield_policy(&impl_);
+  Timestamp current_time = impl_.CurrentTime();
+  Timestamp target_time = current_time + duration;
+  RTC_DCHECK_EQ(current_time.us(), rtc::TimeMicros());
+  while (current_time < target_time) {
+    impl_.RunReadyRunners();
+    Timestamp next_time = std::min(impl_.NextRunTime(), target_time);
+    impl_.AdvanceTime(next_time);
+    auto delta = next_time - current_time;
+    current_time = next_time;
+    sim_clock_.AdvanceTimeMicroseconds(delta.us());
+    global_clock_.AdvanceTime(delta);
+  }
+  // After time has been simulated up until `target_time` we also need to run
+  // tasks meant to be executed at `target_time`.
+  impl_.RunReadyRunners();
+}
+
+void GlobalSimulatedTimeController::Register(
+    sim_time_impl::SimulatedSequenceRunner* runner) {
+  impl_.Register(runner);
+}
+
+void GlobalSimulatedTimeController::Unregister(
+    sim_time_impl::SimulatedSequenceRunner* runner) {
+  impl_.Unregister(runner);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/simulated_time_controller.h b/third_party/libwebrtc/test/time_controller/simulated_time_controller.h
new file mode 100644
index 0000000000..121b9171e8
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/simulated_time_controller.h
@@ -0,0 +1,162 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_TIME_CONTROLLER_SIMULATED_TIME_CONTROLLER_H_
+#define TEST_TIME_CONTROLLER_SIMULATED_TIME_CONTROLLER_H_
+
+#include <list>
+#include <memory>
+#include <unordered_set>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/sequence_checker.h"
+#include "api/test/time_controller.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/platform_thread_types.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/synchronization/yield_policy.h"
+
+namespace webrtc {
+namespace sim_time_impl {
+class SimulatedSequenceRunner {
+ public:
+  virtual ~SimulatedSequenceRunner() = default;
+  // Provides next run time.
+  virtual Timestamp GetNextRunTime() const = 0;
+  // Runs all ready tasks and modules and updates next run time.
+  virtual void RunReady(Timestamp at_time) = 0;
+
+  // All implementations also implements TaskQueueBase in some form, but if we'd
+  // inherit from it in this interface we'd run into issues with double
+  // inheritance. Therefore we simply allow the implementations to provide a
+  // casted pointer to themself.
+  virtual TaskQueueBase* GetAsTaskQueue() = 0;
+};
+
+class SimulatedTimeControllerImpl : public TaskQueueFactory,
+                                    public rtc::YieldInterface {
+ public:
+  explicit SimulatedTimeControllerImpl(Timestamp start_time);
+  ~SimulatedTimeControllerImpl() override;
+
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> CreateTaskQueue(
+      absl::string_view name,
+      Priority priority) const RTC_LOCKS_EXCLUDED(time_lock_) override;
+
+  // Implements the YieldInterface by running ready tasks on all task queues,
+  // except that if this method is called from a task, the task queue running
+  // that task is skipped.
+  void YieldExecution() RTC_LOCKS_EXCLUDED(time_lock_, lock_) override;
+
+  // Create thread using provided `socket_server`.
+  std::unique_ptr<rtc::Thread> CreateThread(
+      const std::string& name,
+      std::unique_ptr<rtc::SocketServer> socket_server)
+      RTC_LOCKS_EXCLUDED(time_lock_, lock_);
+
+  // Runs all runners in `runners_` that has tasks or modules ready for
+  // execution.
+  void RunReadyRunners() RTC_LOCKS_EXCLUDED(time_lock_, lock_);
+  // Return `current_time_`.
+  Timestamp CurrentTime() const RTC_LOCKS_EXCLUDED(time_lock_);
+  // Return min of runner->GetNextRunTime() for runner in `runners_`.
+  Timestamp NextRunTime() const RTC_LOCKS_EXCLUDED(lock_);
+  // Set `current_time_` to `target_time`.
+  void AdvanceTime(Timestamp target_time) RTC_LOCKS_EXCLUDED(time_lock_);
+  // Adds `runner` to `runners_`.
+  void Register(SimulatedSequenceRunner* runner) RTC_LOCKS_EXCLUDED(lock_);
+  // Removes `runner` from `runners_`.
+  void Unregister(SimulatedSequenceRunner* runner) RTC_LOCKS_EXCLUDED(lock_);
+
+  // Indicates that `yielding_from` is not ready to run.
+  void StartYield(TaskQueueBase* yielding_from);
+  // Indicates that processing can be continued on `yielding_from`.
+  void StopYield(TaskQueueBase* yielding_from);
+
+ private:
+  const rtc::PlatformThreadId thread_id_;
+  const std::unique_ptr<rtc::Thread> dummy_thread_ = rtc::Thread::Create();
+  mutable Mutex time_lock_;
+  Timestamp current_time_ RTC_GUARDED_BY(time_lock_);
+  mutable Mutex lock_;
+  std::vector<SimulatedSequenceRunner*> runners_ RTC_GUARDED_BY(lock_);
+  // Used in RunReadyRunners() to keep track of ready runners that are to be
+  // processed in a round robin fashion. the reason it's a member is so that
+  // runners can removed from here by Unregister().
+  std::list<SimulatedSequenceRunner*> ready_runners_ RTC_GUARDED_BY(lock_);
+
+  // Runners on which YieldExecution has been called.
+  std::unordered_set<TaskQueueBase*> yielded_;
+};
+}  // namespace sim_time_impl
+
+// Used to satisfy sequence checkers for non task queue sequences.
+class TokenTaskQueue : public TaskQueueBase {
+ public:
+  // Promoted to public
+  using CurrentTaskQueueSetter = TaskQueueBase::CurrentTaskQueueSetter;
+
+  void Delete() override { RTC_DCHECK_NOTREACHED(); }
+  void PostTask(absl::AnyInvocable<void() &&> /*task*/) override {
+    RTC_DCHECK_NOTREACHED();
+  }
+  void PostDelayedTask(absl::AnyInvocable<void() &&> /*task*/,
+                       TimeDelta /*delay*/) override {
+    RTC_DCHECK_NOTREACHED();
+  }
+  void PostDelayedHighPrecisionTask(absl::AnyInvocable<void() &&> /*task*/,
+                                    TimeDelta /*delay*/) override {
+    RTC_DCHECK_NOTREACHED();
+  }
+};
+
+// TimeController implementation using completely simulated time. Task queues
+// and process threads created by this controller will run delayed activities
+// when AdvanceTime() is called. Overrides the global clock backing
+// rtc::TimeMillis() and rtc::TimeMicros(). Note that this is not thread safe
+// since it modifies global state.
+class GlobalSimulatedTimeController : public TimeController {
+ public:
+  explicit GlobalSimulatedTimeController(Timestamp start_time);
+  ~GlobalSimulatedTimeController() override;
+
+  Clock* GetClock() override;
+  TaskQueueFactory* GetTaskQueueFactory() override;
+  std::unique_ptr<rtc::Thread> CreateThread(
+      const std::string& name,
+      std::unique_ptr<rtc::SocketServer> socket_server) override;
+  rtc::Thread* GetMainThread() override;
+
+  void AdvanceTime(TimeDelta duration) override;
+
+  // Makes the simulated time controller aware of a custom
+  // SimulatedSequenceRunner.
+  // TODO(bugs.webrtc.org/11581): remove method once the ModuleRtpRtcpImpl2 unit
+  // test stops using it.
+  void Register(sim_time_impl::SimulatedSequenceRunner* runner);
+  // Removes a previously installed custom SimulatedSequenceRunner from the
+  // simulated time controller.
+  // TODO(bugs.webrtc.org/11581): remove method once the ModuleRtpRtcpImpl2 unit
+  // test stops using it.
+  void Unregister(sim_time_impl::SimulatedSequenceRunner* runner);
+
+ private:
+  rtc::ScopedBaseFakeClock global_clock_;
+  // Provides simulated CurrentNtpInMilliseconds()
+  SimulatedClock sim_clock_;
+  sim_time_impl::SimulatedTimeControllerImpl impl_;
+  rtc::ScopedYieldPolicy yield_policy_;
+  std::unique_ptr<rtc::Thread> main_thread_;
+};
+}  // namespace webrtc
+
+#endif  // TEST_TIME_CONTROLLER_SIMULATED_TIME_CONTROLLER_H_
diff --git a/third_party/libwebrtc/test/time_controller/simulated_time_controller_unittest.cc b/third_party/libwebrtc/test/time_controller/simulated_time_controller_unittest.cc
new file mode 100644
index 0000000000..1ee592cc7c
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/simulated_time_controller_unittest.cc
@@ -0,0 +1,149 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/time_controller/simulated_time_controller.h"
+
+#include <atomic>
+#include <memory>
+
+#include "rtc_base/event.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+// NOTE: Since these tests rely on real time behavior, they will be flaky
+// if run on heavily loaded systems.
+namespace webrtc {
+namespace {
+using ::testing::AtLeast;
+using ::testing::Invoke;
+using ::testing::MockFunction;
+using ::testing::NiceMock;
+using ::testing::Return;
+constexpr Timestamp kStartTime = Timestamp::Seconds(1000);
+}  // namespace
+
+TEST(SimulatedTimeControllerTest, TaskIsStoppedOnStop) {
+  const TimeDelta kShortInterval = TimeDelta::Millis(5);
+  const TimeDelta kLongInterval = TimeDelta::Millis(20);
+  const int kShortIntervalCount = 4;
+  const int kMargin = 1;
+  GlobalSimulatedTimeController time_simulation(kStartTime);
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+  std::atomic_int counter(0);
+  auto handle = RepeatingTaskHandle::Start(task_queue.Get(), [&] {
+    if (++counter >= kShortIntervalCount)
+      return kLongInterval;
+    return kShortInterval;
+  });
+  // Sleep long enough to go through the initial phase.
+  time_simulation.AdvanceTime(kShortInterval * (kShortIntervalCount + kMargin));
+  EXPECT_EQ(counter.load(), kShortIntervalCount);
+
+  task_queue.PostTask(
+      [handle = std::move(handle)]() mutable { handle.Stop(); });
+
+  // Sleep long enough that the task would run at least once more if not
+  // stopped.
+  time_simulation.AdvanceTime(kLongInterval * 2);
+  EXPECT_EQ(counter.load(), kShortIntervalCount);
+}
+
+TEST(SimulatedTimeControllerTest, TaskCanStopItself) {
+  std::atomic_int counter(0);
+  GlobalSimulatedTimeController time_simulation(kStartTime);
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+
+  RepeatingTaskHandle handle;
+  task_queue.PostTask([&] {
+    handle = RepeatingTaskHandle::Start(task_queue.Get(), [&] {
+      ++counter;
+      handle.Stop();
+      return TimeDelta::Millis(2);
+    });
+  });
+  time_simulation.AdvanceTime(TimeDelta::Millis(10));
+  EXPECT_EQ(counter.load(), 1);
+}
+
+TEST(SimulatedTimeControllerTest, Example) {
+  class ObjectOnTaskQueue {
+   public:
+    void DoPeriodicTask() {}
+    TimeDelta TimeUntilNextRun() { return TimeDelta::Millis(100); }
+    void StartPeriodicTask(RepeatingTaskHandle* handle,
+                           rtc::TaskQueue* task_queue) {
+      *handle = RepeatingTaskHandle::Start(task_queue->Get(), [this] {
+        DoPeriodicTask();
+        return TimeUntilNextRun();
+      });
+    }
+  };
+  GlobalSimulatedTimeController time_simulation(kStartTime);
+  rtc::TaskQueue task_queue(
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL));
+  auto object = std::make_unique<ObjectOnTaskQueue>();
+  // Create and start the periodic task.
+  RepeatingTaskHandle handle;
+  object->StartPeriodicTask(&handle, &task_queue);
+  // Restart the task
+  task_queue.PostTask(
+      [handle = std::move(handle)]() mutable { handle.Stop(); });
+  object->StartPeriodicTask(&handle, &task_queue);
+  task_queue.PostTask(
+      [handle = std::move(handle)]() mutable { handle.Stop(); });
+
+  task_queue.PostTask([object = std::move(object)] {});
+}
+
+TEST(SimulatedTimeControllerTest, DelayTaskRunOnTime) {
+  GlobalSimulatedTimeController time_simulation(kStartTime);
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue =
+      time_simulation.GetTaskQueueFactory()->CreateTaskQueue(
+          "TestQueue", TaskQueueFactory::Priority::NORMAL);
+
+  bool delay_task_executed = false;
+  task_queue->PostDelayedTask([&] { delay_task_executed = true; },
+                              TimeDelta::Millis(10));
+
+  time_simulation.AdvanceTime(TimeDelta::Millis(10));
+  EXPECT_TRUE(delay_task_executed);
+}
+
+TEST(SimulatedTimeControllerTest, ThreadYeildsOnSynchronousCall) {
+  GlobalSimulatedTimeController sim(kStartTime);
+  auto main_thread = sim.GetMainThread();
+  auto t2 = sim.CreateThread("thread", nullptr);
+  bool task_has_run = false;
+  // Posting a task to the main thread, this should not run until AdvanceTime is
+  // called.
+  main_thread->PostTask([&] { task_has_run = true; });
+  SendTask(t2.get(), [] {
+    rtc::Event yield_event;
+    // Wait() triggers YieldExecution() which will runs message processing on
+    // all threads that are not in the yielded set.
+
+    yield_event.Wait(TimeDelta::Zero());
+  });
+  // Since we are doing an invoke from the main thread, we don't expect the main
+  // thread message loop to be processed.
+  EXPECT_FALSE(task_has_run);
+  sim.AdvanceTime(TimeDelta::Seconds(1));
+  ASSERT_TRUE(task_has_run);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/time_controller/time_controller_conformance_test.cc b/third_party/libwebrtc/test/time_controller/time_controller_conformance_test.cc
new file mode 100644
index 0000000000..300dd9175c
--- /dev/null
+++ b/third_party/libwebrtc/test/time_controller/time_controller_conformance_test.cc
@@ -0,0 +1,181 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+#include <vector>
+
+#include "api/test/time_controller.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/time_controller/real_time_controller.h"
+#include "test/time_controller/simulated_time_controller.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAreArray;
+using ::testing::TestParamInfo;
+using ::testing::TestWithParam;
+using ::testing::Values;
+
+enum class TimeMode { kRealTime, kSimulated };
+
+std::unique_ptr<TimeController> CreateTimeController(TimeMode mode) {
+  switch (mode) {
+    case TimeMode::kRealTime:
+      return std::make_unique<RealTimeController>();
+    case TimeMode::kSimulated:
+      // Using an offset of 100000 to get nice fixed width and readable
+      // timestamps in typical test scenarios.
+      constexpr Timestamp kSimulatedStartTime = Timestamp::Seconds(100000);
+      return std::make_unique<GlobalSimulatedTimeController>(
+          kSimulatedStartTime);
+  }
+}
+
+std::string ParamsToString(const TestParamInfo<webrtc::TimeMode>& param) {
+  switch (param.param) {
+    case webrtc::TimeMode::kRealTime:
+      return "RealTime";
+    case webrtc::TimeMode::kSimulated:
+      return "SimulatedTime";
+    default:
+      RTC_DCHECK_NOTREACHED() << "Time mode not supported";
+  }
+}
+
+// Keeps order of executions. May be called from different threads.
+class ExecutionOrderKeeper {
+ public:
+  void Executed(int execution_id) {
+    MutexLock lock(&mutex_);
+    order_.push_back(execution_id);
+  }
+
+  std::vector<int> order() const {
+    MutexLock lock(&mutex_);
+    return order_;
+  }
+
+ private:
+  mutable Mutex mutex_;
+  std::vector<int> order_ RTC_GUARDED_BY(mutex_);
+};
+
+// Tests conformance between real time and simulated time time controller.
+class SimulatedRealTimeControllerConformanceTest
+    : public TestWithParam<webrtc::TimeMode> {};
+
+TEST_P(SimulatedRealTimeControllerConformanceTest, ThreadPostOrderTest) {
+  std::unique_ptr<TimeController> time_controller =
+      CreateTimeController(GetParam());
+  std::unique_ptr<rtc::Thread> thread = time_controller->CreateThread("thread");
+
+  // Tasks on thread have to be executed in order in which they were
+  // posted.
+  ExecutionOrderKeeper execution_order;
+  thread->PostTask([&]() { execution_order.Executed(1); });
+  thread->PostTask([&]() { execution_order.Executed(2); });
+  time_controller->AdvanceTime(TimeDelta::Millis(100));
+  EXPECT_THAT(execution_order.order(), ElementsAreArray({1, 2}));
+  // Destroy `thread` before `execution_order` to be sure `execution_order`
+  // is not accessed on the posted task after it is destroyed.
+  thread = nullptr;
+}
+
+TEST_P(SimulatedRealTimeControllerConformanceTest, ThreadPostDelayedOrderTest) {
+  std::unique_ptr<TimeController> time_controller =
+      CreateTimeController(GetParam());
+  std::unique_ptr<rtc::Thread> thread = time_controller->CreateThread("thread");
+
+  ExecutionOrderKeeper execution_order;
+  thread->PostDelayedTask([&]() { execution_order.Executed(2); },
+                          TimeDelta::Millis(500));
+  thread->PostTask([&]() { execution_order.Executed(1); });
+  time_controller->AdvanceTime(TimeDelta::Millis(600));
+  EXPECT_THAT(execution_order.order(), ElementsAreArray({1, 2}));
+  // Destroy `thread` before `execution_order` to be sure `execution_order`
+  // is not accessed on the posted task after it is destroyed.
+  thread = nullptr;
+}
+
+TEST_P(SimulatedRealTimeControllerConformanceTest, ThreadPostInvokeOrderTest) {
+  std::unique_ptr<TimeController> time_controller =
+      CreateTimeController(GetParam());
+  std::unique_ptr<rtc::Thread> thread = time_controller->CreateThread("thread");
+
+  // Tasks on thread have to be executed in order in which they were
+  // posted/invoked.
+  ExecutionOrderKeeper execution_order;
+  thread->PostTask([&]() { execution_order.Executed(1); });
+  thread->BlockingCall([&]() { execution_order.Executed(2); });
+  time_controller->AdvanceTime(TimeDelta::Millis(100));
+  EXPECT_THAT(execution_order.order(), ElementsAreArray({1, 2}));
+  // Destroy `thread` before `execution_order` to be sure `execution_order`
+  // is not accessed on the posted task after it is destroyed.
+  thread = nullptr;
+}
+
+TEST_P(SimulatedRealTimeControllerConformanceTest,
+       ThreadPostInvokeFromThreadOrderTest) {
+  std::unique_ptr<TimeController> time_controller =
+      CreateTimeController(GetParam());
+  std::unique_ptr<rtc::Thread> thread = time_controller->CreateThread("thread");
+
+  // If task is invoked from thread X on thread X it has to be executed
+  // immediately.
+  ExecutionOrderKeeper execution_order;
+  thread->PostTask([&]() {
+    thread->PostTask([&]() { execution_order.Executed(2); });
+    thread->BlockingCall([&]() { execution_order.Executed(1); });
+  });
+  time_controller->AdvanceTime(TimeDelta::Millis(100));
+  EXPECT_THAT(execution_order.order(), ElementsAreArray({1, 2}));
+  // Destroy `thread` before `execution_order` to be sure `execution_order`
+  // is not accessed on the posted task after it is destroyed.
+  thread = nullptr;
+}
+
+TEST_P(SimulatedRealTimeControllerConformanceTest,
+       TaskQueuePostEventWaitOrderTest) {
+  std::unique_ptr<TimeController> time_controller =
+      CreateTimeController(GetParam());
+  auto task_queue = time_controller->GetTaskQueueFactory()->CreateTaskQueue(
+      "task_queue", webrtc::TaskQueueFactory::Priority::NORMAL);
+
+  // Tasks on thread have to be executed in order in which they were
+  // posted/invoked.
+  ExecutionOrderKeeper execution_order;
+  rtc::Event event;
+  task_queue->PostTask([&]() { execution_order.Executed(1); });
+  task_queue->PostTask([&]() {
+    execution_order.Executed(2);
+    event.Set();
+  });
+  EXPECT_TRUE(event.Wait(/*give_up_after=*/TimeDelta::Millis(100)));
+  time_controller->AdvanceTime(TimeDelta::Millis(100));
+  EXPECT_THAT(execution_order.order(), ElementsAreArray({1, 2}));
+  // Destroy `task_queue` before `execution_order` to be sure `execution_order`
+  // is not accessed on the posted task after it is destroyed.
+  task_queue = nullptr;
+}
+
+INSTANTIATE_TEST_SUITE_P(ConformanceTest,
+                         SimulatedRealTimeControllerConformanceTest,
+                         Values(TimeMode::kRealTime, TimeMode::kSimulated),
+                         ParamsToString);
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/vcm_capturer.cc b/third_party/libwebrtc/test/vcm_capturer.cc
new file mode 100644
index 0000000000..e02fc722b2
--- /dev/null
+++ b/third_party/libwebrtc/test/vcm_capturer.cc
@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "test/vcm_capturer.h"
+
+#include <stdint.h>
+
+#include <memory>
+
+#include "modules/video_capture/video_capture_factory.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace test {
+
+VcmCapturer::VcmCapturer() : vcm_(nullptr) {}
+
+bool VcmCapturer::Init(size_t width,
+                       size_t height,
+                       size_t target_fps,
+                       size_t capture_device_index) {
+  std::unique_ptr<VideoCaptureModule::DeviceInfo> device_info(
+      VideoCaptureFactory::CreateDeviceInfo());
+
+  char device_name[256];
+  char unique_name[256];
+  if (device_info->GetDeviceName(static_cast<uint32_t>(capture_device_index),
+                                 device_name, sizeof(device_name), unique_name,
+                                 sizeof(unique_name)) != 0) {
+    Destroy();
+    return false;
+  }
+
+  vcm_ = webrtc::VideoCaptureFactory::Create(unique_name);
+  if (!vcm_) {
+    return false;
+  }
+  vcm_->RegisterCaptureDataCallback(this);
+
+  device_info->GetCapability(vcm_->CurrentDeviceName(), 0, capability_);
+
+  capability_.width = static_cast<int32_t>(width);
+  capability_.height = static_cast<int32_t>(height);
+  capability_.maxFPS = static_cast<int32_t>(target_fps);
+  capability_.videoType = VideoType::kI420;
+
+  if (vcm_->StartCapture(capability_) != 0) {
+    Destroy();
+    return false;
+  }
+
+  RTC_CHECK(vcm_->CaptureStarted());
+
+  return true;
+}
+
+VcmCapturer* VcmCapturer::Create(size_t width,
+                                 size_t height,
+                                 size_t target_fps,
+                                 size_t capture_device_index) {
+  std::unique_ptr<VcmCapturer> vcm_capturer(new VcmCapturer());
+  if (!vcm_capturer->Init(width, height, target_fps, capture_device_index)) {
+    RTC_LOG(LS_WARNING) << "Failed to create VcmCapturer(w = " << width
+                        << ", h = " << height << ", fps = " << target_fps
+                        << ")";
+    return nullptr;
+  }
+  return vcm_capturer.release();
+}
+
+void VcmCapturer::Destroy() {
+  if (!vcm_)
+    return;
+
+  vcm_->StopCapture();
+  vcm_->DeRegisterCaptureDataCallback(this);
+  // Release reference to VCM.
+  vcm_ = nullptr;
+}
+
+VcmCapturer::~VcmCapturer() {
+  Destroy();
+}
+
+void VcmCapturer::OnFrame(const VideoFrame& frame) {
+  TestVideoCapturer::OnFrame(frame);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/vcm_capturer.h b/third_party/libwebrtc/test/vcm_capturer.h
new file mode 100644
index 0000000000..5418dc9596
--- /dev/null
+++ b/third_party/libwebrtc/test/vcm_capturer.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_VCM_CAPTURER_H_
+#define TEST_VCM_CAPTURER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/scoped_refptr.h"
+#include "modules/video_capture/video_capture.h"
+#include "test/test_video_capturer.h"
+
+namespace webrtc {
+namespace test {
+
+class VcmCapturer : public TestVideoCapturer,
+                    public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  static VcmCapturer* Create(size_t width,
+                             size_t height,
+                             size_t target_fps,
+                             size_t capture_device_index);
+  virtual ~VcmCapturer();
+
+  void OnFrame(const VideoFrame& frame) override;
+
+ private:
+  VcmCapturer();
+  bool Init(size_t width,
+            size_t height,
+            size_t target_fps,
+            size_t capture_device_index);
+  void Destroy();
+
+  rtc::scoped_refptr<VideoCaptureModule> vcm_;
+  VideoCaptureCapability capability_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_VCM_CAPTURER_H_
diff --git a/third_party/libwebrtc/test/video_codec_settings.h b/third_party/libwebrtc/test/video_codec_settings.h
new file mode 100644
index 0000000000..5ef4ed3e4a
--- /dev/null
+++ b/third_party/libwebrtc/test/video_codec_settings.h
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_VIDEO_CODEC_SETTINGS_H_
+#define TEST_VIDEO_CODEC_SETTINGS_H_
+
+#include "api/video_codecs/video_encoder.h"
+
+namespace webrtc {
+namespace test {
+
+const uint16_t kTestWidth = 352;
+const uint16_t kTestHeight = 288;
+const uint32_t kTestFrameRate = 30;
+const unsigned int kTestMinBitrateKbps = 30;
+const unsigned int kTestStartBitrateKbps = 300;
+const uint8_t kTestPayloadType = 100;
+const int64_t kTestTimingFramesDelayMs = 200;
+const uint16_t kTestOutlierFrameSizePercent = 250;
+
+static void CodecSettings(VideoCodecType codec_type, VideoCodec* settings) {
+  *settings = {};
+
+  settings->width = kTestWidth;
+  settings->height = kTestHeight;
+
+  settings->startBitrate = kTestStartBitrateKbps;
+  settings->maxBitrate = 0;
+  settings->minBitrate = kTestMinBitrateKbps;
+
+  settings->maxFramerate = kTestFrameRate;
+
+  settings->active = true;
+
+  settings->qpMax = 56;  // See webrtcvideoengine.h.
+  settings->numberOfSimulcastStreams = 0;
+
+  settings->timing_frame_thresholds = {
+      kTestTimingFramesDelayMs,
+      kTestOutlierFrameSizePercent,
+  };
+
+  settings->codecType = codec_type;
+  switch (codec_type) {
+    case kVideoCodecVP8:
+      *(settings->VP8()) = VideoEncoder::GetDefaultVp8Settings();
+      return;
+    case kVideoCodecVP9:
+      *(settings->VP9()) = VideoEncoder::GetDefaultVp9Settings();
+      return;
+    case kVideoCodecH264:
+      // TODO(brandtr): Set `qpMax` here, when the OpenH264 wrapper supports it.
+      *(settings->H264()) = VideoEncoder::GetDefaultH264Settings();
+      return;
+    default:
+      return;
+  }
+}
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_VIDEO_CODEC_SETTINGS_H_
diff --git a/third_party/libwebrtc/test/video_decoder_proxy_factory.h b/third_party/libwebrtc/test/video_decoder_proxy_factory.h
new file mode 100644
index 0000000000..6fd3805cd6
--- /dev/null
+++ b/third_party/libwebrtc/test/video_decoder_proxy_factory.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_VIDEO_DECODER_PROXY_FACTORY_H_
+#define TEST_VIDEO_DECODER_PROXY_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/video_decoder.h"
+#include "api/video_codecs/video_decoder_factory.h"
+
+namespace webrtc {
+namespace test {
+
+// A decoder factory with a single underlying VideoDecoder object, intended for
+// test purposes. Each call to CreateVideoDecoder returns a proxy for the same
+// decoder, typically an instance of FakeDecoder or MockEncoder.
+class VideoDecoderProxyFactory final : public VideoDecoderFactory {
+ public:
+  explicit VideoDecoderProxyFactory(VideoDecoder* decoder)
+      : decoder_(decoder) {}
+
+  // Unused by tests.
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
+    RTC_DCHECK_NOTREACHED();
+    return {};
+  }
+
+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
+      const SdpVideoFormat& format) override {
+    return std::make_unique<DecoderProxy>(decoder_);
+  }
+
+ private:
+  // Wrapper class, since CreateVideoDecoder needs to surrender
+  // ownership to the object it returns.
+  class DecoderProxy final : public VideoDecoder {
+   public:
+    explicit DecoderProxy(VideoDecoder* decoder) : decoder_(decoder) {}
+
+   private:
+    int32_t Decode(const EncodedImage& input_image,
+                   bool missing_frames,
+                   int64_t render_time_ms) override {
+      return decoder_->Decode(input_image, missing_frames, render_time_ms);
+    }
+    bool Configure(const Settings& settings) override {
+      return decoder_->Configure(settings);
+    }
+    int32_t RegisterDecodeCompleteCallback(
+        DecodedImageCallback* callback) override {
+      return decoder_->RegisterDecodeCompleteCallback(callback);
+    }
+    int32_t Release() override { return decoder_->Release(); }
+    DecoderInfo GetDecoderInfo() const override {
+      return decoder_->GetDecoderInfo();
+    }
+    const char* ImplementationName() const override {
+      return decoder_->ImplementationName();
+    }
+
+    VideoDecoder* const decoder_;
+  };
+
+  VideoDecoder* const decoder_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_VIDEO_DECODER_PROXY_FACTORY_H_
diff --git a/third_party/libwebrtc/test/video_encoder_nullable_proxy_factory.h b/third_party/libwebrtc/test/video_encoder_nullable_proxy_factory.h
new file mode 100644
index 0000000000..da81fff343
--- /dev/null
+++ b/third_party/libwebrtc/test/video_encoder_nullable_proxy_factory.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_VIDEO_ENCODER_NULLABLE_PROXY_FACTORY_H_
+#define TEST_VIDEO_ENCODER_NULLABLE_PROXY_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "test/video_encoder_proxy_factory.h"
+
+namespace webrtc {
+namespace test {
+
+class VideoEncoderNullableProxyFactory final : public VideoEncoderProxyFactory {
+ public:
+  explicit VideoEncoderNullableProxyFactory(
+      VideoEncoder* encoder,
+      EncoderSelectorInterface* encoder_selector)
+      : VideoEncoderProxyFactory(encoder, encoder_selector) {}
+
+  ~VideoEncoderNullableProxyFactory() override = default;
+
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override {
+    if (!encoder_) {
+      return nullptr;
+    }
+    return VideoEncoderProxyFactory::CreateVideoEncoder(format);
+  }
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_VIDEO_ENCODER_NULLABLE_PROXY_FACTORY_H_
diff --git a/third_party/libwebrtc/test/video_encoder_proxy_factory.h b/third_party/libwebrtc/test/video_encoder_proxy_factory.h
new file mode 100644
index 0000000000..cc485e993a
--- /dev/null
+++ b/third_party/libwebrtc/test/video_encoder_proxy_factory.h
@@ -0,0 +1,158 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef TEST_VIDEO_ENCODER_PROXY_FACTORY_H_
+#define TEST_VIDEO_ENCODER_PROXY_FACTORY_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+
+namespace webrtc {
+namespace test {
+
+namespace {
+const VideoEncoder::Capabilities kCapabilities(false);
+}
+
+// An encoder factory with a single underlying VideoEncoder object,
+// intended for test purposes. Each call to CreateVideoEncoder returns
+// a proxy for the same encoder, typically an instance of FakeEncoder.
+class VideoEncoderProxyFactory : public VideoEncoderFactory {
+ public:
+  explicit VideoEncoderProxyFactory(VideoEncoder* encoder)
+      : VideoEncoderProxyFactory(encoder, nullptr) {}
+
+  explicit VideoEncoderProxyFactory(VideoEncoder* encoder,
+                                    EncoderSelectorInterface* encoder_selector)
+      : encoder_(encoder),
+        encoder_selector_(encoder_selector),
+        num_simultaneous_encoder_instances_(0),
+        max_num_simultaneous_encoder_instances_(0) {
+  }
+
+  // Unused by tests.
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
+    RTC_DCHECK_NOTREACHED();
+    return {};
+  }
+
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override {
+    ++num_simultaneous_encoder_instances_;
+    max_num_simultaneous_encoder_instances_ =
+        std::max(max_num_simultaneous_encoder_instances_,
+                 num_simultaneous_encoder_instances_);
+    return std::make_unique<EncoderProxy>(encoder_, this);
+  }
+
+  std::unique_ptr<EncoderSelectorInterface> GetEncoderSelector()
+      const override {
+    if (encoder_selector_ != nullptr) {
+      return std::make_unique<EncoderSelectorProxy>(encoder_selector_);
+    }
+
+    return nullptr;
+  }
+
+  int GetMaxNumberOfSimultaneousEncoderInstances() {
+    return max_num_simultaneous_encoder_instances_;
+  }
+
+ protected:
+  void OnDestroyVideoEncoder() {
+    RTC_CHECK_GT(num_simultaneous_encoder_instances_, 0);
+    --num_simultaneous_encoder_instances_;
+  }
+
+  // Wrapper class, since CreateVideoEncoder needs to surrender
+  // ownership to the object it returns.
+  class EncoderProxy final : public VideoEncoder {
+   public:
+    explicit EncoderProxy(VideoEncoder* encoder,
+                          VideoEncoderProxyFactory* encoder_factory)
+        : encoder_(encoder), encoder_factory_(encoder_factory) {}
+    ~EncoderProxy() { encoder_factory_->OnDestroyVideoEncoder(); }
+
+   private:
+    void SetFecControllerOverride(
+        FecControllerOverride* fec_controller_override) override {
+      encoder_->SetFecControllerOverride(fec_controller_override);
+    }
+
+    int32_t Encode(const VideoFrame& input_image,
+                   const std::vector<VideoFrameType>* frame_types) override {
+      return encoder_->Encode(input_image, frame_types);
+    }
+
+    int32_t InitEncode(const VideoCodec* config,
+                       const Settings& settings) override {
+      return encoder_->InitEncode(config, settings);
+    }
+
+    int32_t RegisterEncodeCompleteCallback(
+        EncodedImageCallback* callback) override {
+      return encoder_->RegisterEncodeCompleteCallback(callback);
+    }
+
+    int32_t Release() override { return encoder_->Release(); }
+
+    void SetRates(const RateControlParameters& parameters) override {
+      encoder_->SetRates(parameters);
+    }
+
+    VideoEncoder::EncoderInfo GetEncoderInfo() const override {
+      return encoder_->GetEncoderInfo();
+    }
+
+    VideoEncoder* const encoder_;
+    VideoEncoderProxyFactory* const encoder_factory_;
+  };
+
+  class EncoderSelectorProxy final : public EncoderSelectorInterface {
+   public:
+    explicit EncoderSelectorProxy(EncoderSelectorInterface* encoder_selector)
+        : encoder_selector_(encoder_selector) {}
+
+    void OnCurrentEncoder(const SdpVideoFormat& format) override {
+      encoder_selector_->OnCurrentEncoder(format);
+    }
+
+    absl::optional<SdpVideoFormat> OnAvailableBitrate(
+        const DataRate& rate) override {
+      return encoder_selector_->OnAvailableBitrate(rate);
+    }
+
+    absl::optional<SdpVideoFormat> OnResolutionChange(
+        const RenderResolution& resolution) override {
+      return encoder_selector_->OnResolutionChange(resolution);
+    }
+
+    absl::optional<SdpVideoFormat> OnEncoderBroken() override {
+      return encoder_selector_->OnEncoderBroken();
+    }
+
+   private:
+    EncoderSelectorInterface* const encoder_selector_;
+  };
+
+  VideoEncoder* const encoder_;
+  EncoderSelectorInterface* const encoder_selector_;
+
+  int num_simultaneous_encoder_instances_;
+  int max_num_simultaneous_encoder_instances_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_VIDEO_ENCODER_PROXY_FACTORY_H_
diff --git a/third_party/libwebrtc/test/video_renderer.cc b/third_party/libwebrtc/test/video_renderer.cc
new file mode 100644
index 0000000000..75ab72a883
--- /dev/null
+++ b/third_party/libwebrtc/test/video_renderer.cc
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/video_renderer.h"
+
+// TODO(pbos): Android renderer
+
+namespace webrtc {
+namespace test {
+
+class NullRenderer : public VideoRenderer {
+  void OnFrame(const VideoFrame& video_frame) override {}
+};
+
+VideoRenderer* VideoRenderer::Create(const char* window_title,
+                                     size_t width,
+                                     size_t height) {
+  VideoRenderer* renderer = CreatePlatformRenderer(window_title, width, height);
+  if (renderer != nullptr)
+    return renderer;
+
+  return new NullRenderer();
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/video_renderer.h b/third_party/libwebrtc/test/video_renderer.h
new file mode 100644
index 0000000000..9e580f6f59
--- /dev/null
+++ b/third_party/libwebrtc/test/video_renderer.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_VIDEO_RENDERER_H_
+#define TEST_VIDEO_RENDERER_H_
+
+#include <stddef.h>
+
+#include "api/video/video_sink_interface.h"
+
+namespace webrtc {
+class VideoFrame;
+
+namespace test {
+class VideoRenderer : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  // Creates a platform-specific renderer if possible, or a null implementation
+  // if failing.
+  static VideoRenderer* Create(const char* window_title,
+                               size_t width,
+                               size_t height);
+  // Returns a renderer rendering to a platform specific window if possible,
+  // NULL if none can be created.
+  // Creates a platform-specific renderer if possible, returns NULL if a
+  // platform renderer could not be created. This occurs, for instance, when
+  // running without an X environment on Linux.
+  static VideoRenderer* CreatePlatformRenderer(const char* window_title,
+                                               size_t width,
+                                               size_t height);
+  virtual ~VideoRenderer() {}
+
+ protected:
+  VideoRenderer() {}
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_VIDEO_RENDERER_H_
diff --git a/third_party/libwebrtc/test/win/d3d_renderer.cc b/third_party/libwebrtc/test/win/d3d_renderer.cc
new file mode 100644
index 0000000000..5ba90fad78
--- /dev/null
+++ b/third_party/libwebrtc/test/win/d3d_renderer.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "test/win/d3d_renderer.h"
+
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+#define D3DFVF_CUSTOMVERTEX (D3DFVF_XYZ | D3DFVF_TEX1)
+
+struct D3dCustomVertex {
+  float x, y, z;
+  float u, v;
+};
+
+const char kD3DClassName[] = "d3d_renderer";
+
+VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title,
+                                                     size_t width,
+                                                     size_t height) {
+  return D3dRenderer::Create(window_title, width, height);
+}
+
+D3dRenderer::D3dRenderer(size_t width, size_t height)
+    : width_(width),
+      height_(height),
+      hwnd_(NULL),
+      d3d_(nullptr),
+      d3d_device_(nullptr),
+      texture_(nullptr),
+      vertex_buffer_(nullptr) {
+  RTC_DCHECK_GT(width, 0);
+  RTC_DCHECK_GT(height, 0);
+}
+
+D3dRenderer::~D3dRenderer() {
+  Destroy();
+}
+
+LRESULT WINAPI D3dRenderer::WindowProc(HWND hwnd,
+                                       UINT msg,
+                                       WPARAM wparam,
+                                       LPARAM lparam) {
+  if (msg == WM_DESTROY || (msg == WM_CHAR && wparam == VK_RETURN)) {
+    PostQuitMessage(0);
+    return 0;
+  }
+
+  return DefWindowProcA(hwnd, msg, wparam, lparam);
+}
+
+void D3dRenderer::Destroy() {
+  texture_ = nullptr;
+  vertex_buffer_ = nullptr;
+  d3d_device_ = nullptr;
+  d3d_ = nullptr;
+
+  if (hwnd_ != NULL) {
+    DestroyWindow(hwnd_);
+    RTC_DCHECK(!IsWindow(hwnd_));
+    hwnd_ = NULL;
+  }
+}
+
+bool D3dRenderer::Init(const char* window_title) {
+  hwnd_ = CreateWindowA(kD3DClassName, window_title, WS_OVERLAPPEDWINDOW, 0, 0,
+                        static_cast<int>(width_), static_cast<int>(height_),
+                        NULL, NULL, NULL, NULL);
+
+  if (hwnd_ == NULL) {
+    Destroy();
+    return false;
+  }
+
+  d3d_ = Direct3DCreate9(D3D_SDK_VERSION);
+  if (d3d_ == nullptr) {
+    Destroy();
+    return false;
+  }
+
+  D3DPRESENT_PARAMETERS d3d_params = {};
+
+  d3d_params.Windowed = TRUE;
+  d3d_params.SwapEffect = D3DSWAPEFFECT_COPY;
+
+  IDirect3DDevice9* d3d_device;
+  if (d3d_->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL, hwnd_,
+                         D3DCREATE_SOFTWARE_VERTEXPROCESSING, &d3d_params,
+                         &d3d_device) != D3D_OK) {
+    Destroy();
+    return false;
+  }
+  d3d_device_ = d3d_device;
+  d3d_device->Release();
+
+  IDirect3DVertexBuffer9* vertex_buffer;
+  const int kRectVertices = 4;
+  if (d3d_device_->CreateVertexBuffer(kRectVertices * sizeof(D3dCustomVertex),
+                                      0, D3DFVF_CUSTOMVERTEX, D3DPOOL_MANAGED,
+                                      &vertex_buffer, NULL) != D3D_OK) {
+    Destroy();
+    return false;
+  }
+  vertex_buffer_ = vertex_buffer;
+  vertex_buffer->Release();
+
+  d3d_device_->SetRenderState(D3DRS_CULLMODE, D3DCULL_NONE);
+  d3d_device_->SetRenderState(D3DRS_LIGHTING, FALSE);
+  Resize(width_, height_);
+
+  ShowWindow(hwnd_, SW_SHOWNOACTIVATE);
+  d3d_device_->Present(NULL, NULL, NULL, NULL);
+
+  return true;
+}
+
+D3dRenderer* D3dRenderer::Create(const char* window_title,
+                                 size_t width,
+                                 size_t height) {
+  static ATOM wc_atom = 0;
+  if (wc_atom == 0) {
+    WNDCLASSA wc = {};
+
+    wc.style = CS_HREDRAW | CS_VREDRAW;
+    wc.lpfnWndProc = WindowProc;
+    wc.hCursor = LoadCursor(NULL, IDC_ARROW);
+    wc.hbrBackground = reinterpret_cast<HBRUSH>(COLOR_WINDOW);
+    wc.lpszClassName = kD3DClassName;
+
+    wc_atom = RegisterClassA(&wc);
+    if (wc_atom == 0)
+      return nullptr;
+  }
+
+  D3dRenderer* d3d_renderer = new D3dRenderer(width, height);
+  if (!d3d_renderer->Init(window_title)) {
+    delete d3d_renderer;
+    return nullptr;
+  }
+
+  return d3d_renderer;
+}
+
+void D3dRenderer::Resize(size_t width, size_t height) {
+  width_ = width;
+  height_ = height;
+  IDirect3DTexture9* texture;
+
+  d3d_device_->CreateTexture(static_cast<UINT>(width_),
+                             static_cast<UINT>(height_), 1, 0, D3DFMT_A8R8G8B8,
+                             D3DPOOL_MANAGED, &texture, NULL);
+  texture_ = texture;
+  texture->Release();
+
+  // Vertices for the video frame to be rendered to.
+  static const D3dCustomVertex rect[] = {
+      {-1.0f, -1.0f, 0.0f, 0.0f, 1.0f},
+      {-1.0f, 1.0f, 0.0f, 0.0f, 0.0f},
+      {1.0f, -1.0f, 0.0f, 1.0f, 1.0f},
+      {1.0f, 1.0f, 0.0f, 1.0f, 0.0f},
+  };
+
+  void* buf_data;
+  if (vertex_buffer_->Lock(0, 0, &buf_data, 0) != D3D_OK)
+    return;
+
+  memcpy(buf_data, &rect, sizeof(rect));
+  vertex_buffer_->Unlock();
+}
+
+void D3dRenderer::OnFrame(const webrtc::VideoFrame& frame) {
+  if (static_cast<size_t>(frame.width()) != width_ ||
+      static_cast<size_t>(frame.height()) != height_) {
+    Resize(static_cast<size_t>(frame.width()),
+           static_cast<size_t>(frame.height()));
+  }
+
+  D3DLOCKED_RECT lock_rect;
+  if (texture_->LockRect(0, &lock_rect, NULL, 0) != D3D_OK)
+    return;
+
+  ConvertFromI420(frame, VideoType::kARGB, 0,
+                  static_cast<uint8_t*>(lock_rect.pBits));
+  texture_->UnlockRect(0);
+
+  d3d_device_->BeginScene();
+  d3d_device_->SetFVF(D3DFVF_CUSTOMVERTEX);
+  d3d_device_->SetStreamSource(0, vertex_buffer_.get(), 0,
+                               sizeof(D3dCustomVertex));
+  d3d_device_->SetTexture(0, texture_.get());
+  d3d_device_->DrawPrimitive(D3DPT_TRIANGLESTRIP, 0, 2);
+  d3d_device_->EndScene();
+
+  d3d_device_->Present(NULL, NULL, NULL, NULL);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/test/win/d3d_renderer.h b/third_party/libwebrtc/test/win/d3d_renderer.h
new file mode 100644
index 0000000000..9e5e23c328
--- /dev/null
+++ b/third_party/libwebrtc/test/win/d3d_renderer.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef TEST_WIN_D3D_RENDERER_H_
+#define TEST_WIN_D3D_RENDERER_H_
+
+#include <Windows.h>
+#include <d3d9.h>
+#pragma comment(lib, "d3d9.lib")  // located in DirectX SDK
+
+#include "api/scoped_refptr.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace test {
+
+class D3dRenderer : public VideoRenderer {
+ public:
+  static D3dRenderer* Create(const char* window_title,
+                             size_t width,
+                             size_t height);
+  virtual ~D3dRenderer();
+
+  void OnFrame(const webrtc::VideoFrame& frame) override;
+
+ private:
+  D3dRenderer(size_t width, size_t height);
+
+  static LRESULT WINAPI WindowProc(HWND hwnd,
+                                   UINT msg,
+                                   WPARAM wparam,
+                                   LPARAM lparam);
+  bool Init(const char* window_title);
+  void Resize(size_t width, size_t height);
+  void Destroy();
+
+  size_t width_, height_;
+
+  HWND hwnd_;
+  rtc::scoped_refptr<IDirect3D9> d3d_;
+  rtc::scoped_refptr<IDirect3DDevice9> d3d_device_;
+
+  rtc::scoped_refptr<IDirect3DTexture9> texture_;
+  rtc::scoped_refptr<IDirect3DVertexBuffer9> vertex_buffer_;
+};
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // TEST_WIN_D3D_RENDERER_H_