Adding upstream version 110.0.1.upstream/110.0.1upstream

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

83 files changed, 16670 insertions, 0 deletions

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..b0f1f0f86f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/BUILD.gn
@@ -0,0 +1,923 @@
+# 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 (!build_with_chromium) {
+  group("e2e") {
+    testonly = true
+
+    deps = [
+      ":encoded_image_data_injector_api",
+      ":example_video_quality_analyzer",
+      ":quality_analyzing_video_decoder",
+      ":quality_analyzing_video_encoder",
+      ":single_process_encoded_image_data_injector",
+      ":video_frame_tracking_id_injector",
+    ]
+    if (rtc_include_tests) {
+      deps += [
+        ":peerconnection_quality_test",
+        ":test_peer",
+        ":video_quality_analyzer_injection_helper",
+      ]
+    }
+  }
+
+  if (rtc_include_tests) {
+    group("e2e_unittests") {
+      testonly = true
+
+      deps = [
+        ":default_video_quality_analyzer_frames_comparator_test",
+        ":default_video_quality_analyzer_stream_state_test",
+        ":default_video_quality_analyzer_test",
+        ":multi_reader_queue_test",
+        ":names_collection_test",
+        ":peer_connection_e2e_smoke_test",
+        ":single_process_encoded_image_data_injector_unittest",
+        ":stats_poller_test",
+        ":video_frame_tracking_id_injector_unittest",
+      ]
+    }
+  }
+
+  rtc_library("peer_connection_quality_test_params") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "peer_connection_quality_test_params.h" ]
+
+    deps = [
+      "../../../api:callfactory_api",
+      "../../../api:fec_controller_api",
+      "../../../api:field_trials_view",
+      "../../../api:libjingle_peerconnection_api",
+      "../../../api:packet_socket_factory",
+      "../../../api:peer_connection_quality_test_fixture_api",
+      "../../../api/audio:audio_mixer_api",
+      "../../../api/rtc_event_log",
+      "../../../api/task_queue",
+      "../../../api/transport:network_control",
+      "../../../api/video_codecs:video_codecs_api",
+      "../../../modules/audio_processing:api",
+      "../../../p2p:rtc_p2p",
+      "../../../rtc_base",
+      "../../../rtc_base:threading",
+    ]
+  }
+
+  rtc_library("encoded_image_data_injector_api") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "analyzer/video/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") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "analyzer/video/single_process_encoded_image_data_injector.cc",
+      "analyzer/video/single_process_encoded_image_data_injector.h",
+    ]
+
+    deps = [
+      ":encoded_image_data_injector_api",
+      "../../../api/video:encoded_image",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base/synchronization:mutex",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+  }
+
+  rtc_library("video_frame_tracking_id_injector") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "analyzer/video/video_frame_tracking_id_injector.cc",
+      "analyzer/video/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") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "analyzer/video/simulcast_dummy_buffer_helper.cc",
+      "analyzer/video/simulcast_dummy_buffer_helper.h",
+    ]
+    deps = [ "../../../api/video:video_frame" ]
+  }
+
+  rtc_library("quality_analyzing_video_decoder") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "analyzer/video/quality_analyzing_video_decoder.cc",
+      "analyzer/video/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:video_rtp_headers",
+      "../../../api/video_codecs:video_codecs_api",
+      "../../../modules/video_coding:video_codec_interface",
+      "../../../rtc_base:criticalsection",
+      "../../../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") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "analyzer/video/quality_analyzing_video_encoder.cc",
+      "analyzer/video/quality_analyzing_video_encoder.h",
+    ]
+    deps = [
+      ":encoded_image_data_injector_api",
+      "../../../api:video_quality_analyzer_api",
+      "../../../api/video:encoded_image",
+      "../../../api/video:video_frame",
+      "../../../api/video:video_rtp_headers",
+      "../../../api/video_codecs:video_codecs_api",
+      "../../../modules/video_coding:video_codec_interface",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:logging",
+      "../../../rtc_base/synchronization:mutex",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  if (rtc_include_tests) {
+    rtc_library("video_quality_analyzer_injection_helper") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [
+        "analyzer/video/video_quality_analyzer_injection_helper.cc",
+        "analyzer/video/video_quality_analyzer_injection_helper.h",
+      ]
+      deps = [
+        ":encoded_image_data_injector_api",
+        ":quality_analyzing_video_decoder",
+        ":quality_analyzing_video_encoder",
+        ":simulcast_dummy_buffer_helper",
+        "../..:fixed_fps_video_frame_writer_adapter",
+        "../..:test_renderer",
+        "../../../api:array_view",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api:stats_observer_interface",
+        "../../../api:video_quality_analyzer_api",
+        "../../../api/video:video_frame",
+        "../../../api/video:video_rtp_headers",
+        "../../../api/video_codecs:video_codecs_api",
+        "../../../rtc_base:criticalsection",
+        "../../../rtc_base:stringutils",
+        "../../../rtc_base/synchronization:mutex",
+        "../../../system_wrappers",
+        "../../../test:video_test_common",
+        "../../../test:video_test_support",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/memory",
+        "//third_party/abseil-cpp/absl/strings",
+      ]
+    }
+
+    rtc_library("echo_emulation") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [
+        "echo/echo_emulation.cc",
+        "echo/echo_emulation.h",
+      ]
+      deps = [
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../modules/audio_device:audio_device_impl",
+        "../../../rtc_base:swap_queue",
+      ]
+    }
+
+    rtc_library("test_peer") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [
+        "test_peer.cc",
+        "test_peer.h",
+      ]
+      deps = [
+        ":peer_configurer",
+        ":peer_connection_quality_test_params",
+        ":stats_provider",
+        "../../../api:frame_generator_api",
+        "../../../api:function_view",
+        "../../../api:libjingle_peerconnection_api",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api:scoped_refptr",
+        "../../../api:sequence_checker",
+        "../../../api/task_queue:pending_task_safety_flag",
+        "../../../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") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [
+        "test_peer_factory.cc",
+        "test_peer_factory.h",
+      ]
+      deps = [
+        ":echo_emulation",
+        ":peer_configurer",
+        ":peer_connection_quality_test_params",
+        ":quality_analyzing_video_encoder",
+        ":test_peer",
+        ":video_quality_analyzer_injection_helper",
+        "../..:copy_to_file_audio_capturer",
+        "../../../api:create_time_controller",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api:time_controller",
+        "../../../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_engine_defaults",
+        "../../../modules/audio_device:audio_device_impl",
+        "../../../modules/audio_processing/aec_dump",
+        "../../../p2p:rtc_p2p",
+        "../../../rtc_base:rtc_task_queue",
+      ]
+      absl_deps = [
+        "//third_party/abseil-cpp/absl/memory",
+        "//third_party/abseil-cpp/absl/strings",
+      ]
+    }
+
+    rtc_library("media_helper") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [
+        "media/media_helper.cc",
+        "media/media_helper.h",
+        "media/test_video_capturer_video_track_source.h",
+      ]
+      deps = [
+        ":peer_configurer",
+        ":test_peer",
+        ":video_quality_analyzer_injection_helper",
+        "../..:fileutils",
+        "../..:platform_video_capturer",
+        "../..:video_test_common",
+        "../../../api:create_frame_generator",
+        "../../../api:frame_generator_api",
+        "../../../api:media_stream_interface",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api/video:video_frame",
+        "../../../pc:session_description",
+        "../../../pc:video_track_source",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:variant" ]
+    }
+
+    rtc_library("peer_configurer") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [
+        "peer_configurer.cc",
+        "peer_configurer.h",
+      ]
+      deps = [
+        ":peer_connection_quality_test_params",
+        "../..:fileutils",
+        "../../../api:callfactory_api",
+        "../../../api:create_peer_connection_quality_test_frame_generator",
+        "../../../api:fec_controller_api",
+        "../../../api:packet_socket_factory",
+        "../../../api:peer_connection_quality_test_fixture_api",
+        "../../../api/audio:audio_mixer_api",
+        "../../../api/rtc_event_log",
+        "../../../api/task_queue",
+        "../../../api/transport:network_control",
+        "../../../api/video_codecs:video_codecs_api",
+        "../../../modules/audio_processing:api",
+        "../../../rtc_base",
+        "../../../rtc_base:macromagic",
+        "../../../rtc_base:threading",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    }
+
+    rtc_library("test_activities_executor") {
+      visibility = [ "*" ]
+      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") {
+      visibility = [ "*" ]
+      testonly = true
+
+      sources = [
+        "peer_connection_quality_test.cc",
+        "peer_connection_quality_test.h",
+      ]
+      deps = [
+        ":analyzer_helper",
+        ":cross_media_metrics_reporter",
+        ":default_audio_quality_analyzer",
+        ":default_video_quality_analyzer",
+        ":media_helper",
+        ":peer_configurer",
+        ":peer_connection_quality_test_params",
+        ":sdp_changer",
+        ":single_process_encoded_image_data_injector",
+        ":stats_poller",
+        ":test_activities_executor",
+        ":test_peer",
+        ":test_peer_factory",
+        ":video_frame_tracking_id_injector",
+        ":video_quality_analyzer_injection_helper",
+        ":video_quality_metrics_reporter",
+        "../..: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/units:time_delta",
+        "../../../api/units:timestamp",
+        "../../../pc:pc_test_utils",
+        "../../../pc:sdp_utils",
+        "../../../rtc_base",
+        "../../../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",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    }
+
+    rtc_library("single_process_encoded_image_data_injector_unittest") {
+      testonly = true
+      sources = [
+        "analyzer/video/single_process_encoded_image_data_injector_unittest.cc",
+      ]
+      deps = [
+        ":single_process_encoded_image_data_injector",
+        "../../../api/video:encoded_image",
+        "../../../rtc_base:buffer",
+        "../../../test:test_support",
+      ]
+    }
+
+    rtc_library("video_frame_tracking_id_injector_unittest") {
+      testonly = true
+      sources =
+          [ "analyzer/video/video_frame_tracking_id_injector_unittest.cc" ]
+      deps = [
+        ":video_frame_tracking_id_injector",
+        "../../../api/video:encoded_image",
+        "../../../rtc_base:buffer",
+        "../../../test:test_support",
+      ]
+    }
+
+    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",
+        ":default_video_quality_analyzer",
+        ":default_video_quality_analyzer_shared",
+        ":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/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",
+        "../../../rtc_base:gunit_helpers",
+        "../../../rtc_base:logging",
+        "../../../rtc_base:rtc_event",
+        "../../../system_wrappers:field_trial",
+        "../../../test:field_trial",
+        "../../../test:fileutils",
+        "../../../test:test_support",
+      ]
+      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("stats_provider") {
+      visibility = [ "*" ]
+      testonly = true
+      sources = [ "stats_provider.h" ]
+      deps = [ "../../../api:rtc_stats_api" ]
+    }
+
+    rtc_library("stats_poller") {
+      visibility = [ "*" ]
+      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("default_video_quality_analyzer_test") {
+      testonly = true
+      sources = [ "analyzer/video/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/video:encoded_image",
+        "../../../api/video:video_frame",
+        "../../../common_video",
+        "../../../modules/rtp_rtcp:rtp_rtcp_format",
+        "../../../rtc_base:stringutils",
+        "../../../rtc_tools:video_quality_analysis",
+        "../../../system_wrappers",
+      ]
+    }
+
+    rtc_library("default_video_quality_analyzer_frames_comparator_test") {
+      testonly = true
+      sources = [ "analyzer/video/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 = [ "analyzer/video/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 = [ "analyzer/video/multi_reader_queue_test.cc" ]
+      deps = [
+        ":multi_reader_queue",
+        "../../../test:test_support",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    }
+
+    rtc_library("default_video_quality_analyzer_stream_state_test") {
+      testonly = true
+      sources = [
+        "analyzer/video/default_video_quality_analyzer_stream_state_test.cc",
+      ]
+      deps = [
+        ":default_video_quality_analyzer_internal",
+        "../../../api/units:timestamp",
+        "../../../test:test_support",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+    }
+  }
+
+  rtc_library("analyzer_helper") {
+    visibility = [ "*" ]
+    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" ]
+  }
+
+  rtc_library("default_audio_quality_analyzer") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "analyzer/audio/default_audio_quality_analyzer.cc",
+      "analyzer/audio/default_audio_quality_analyzer.h",
+    ]
+
+    deps = [
+      "../..:perf_test",
+      "../../../api:audio_quality_analyzer_api",
+      "../../../api:rtc_stats_api",
+      "../../../api:stats_observer_interface",
+      "../../../api:track_id_stream_info_map",
+      "../../../api/numerics",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:rtc_numerics",
+      "../../../rtc_base/synchronization:mutex",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("example_video_quality_analyzer") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "analyzer/video/example_video_quality_analyzer.cc",
+      "analyzer/video/example_video_quality_analyzer.h",
+    ]
+
+    deps = [
+      "../../../api:array_view",
+      "../../../api:video_quality_analyzer_api",
+      "../../../api/video:encoded_image",
+      "../../../api/video:video_frame",
+      "../../../api/video:video_rtp_headers",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:logging",
+      "../../../rtc_base/synchronization:mutex",
+    ]
+  }
+
+  rtc_library("video_quality_metrics_reporter") {
+    visibility = [ "*" ]
+
+    testonly = true
+    sources = [
+      "analyzer/video/video_quality_metrics_reporter.cc",
+      "analyzer/video/video_quality_metrics_reporter.h",
+    ]
+    deps = [
+      "../..:perf_test",
+      "../../../api:peer_connection_quality_test_fixture_api",
+      "../../../api:rtc_stats_api",
+      "../../../api:track_id_stream_info_map",
+      "../../../api/numerics",
+      "../../../api/units:data_rate",
+      "../../../api/units:data_size",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:rtc_numerics",
+      "../../../rtc_base/synchronization:mutex",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("default_video_quality_analyzer") {
+    visibility = [ "*" ]
+
+    testonly = true
+    sources = [
+      "analyzer/video/default_video_quality_analyzer.cc",
+      "analyzer/video/default_video_quality_analyzer.h",
+    ]
+
+    deps = [
+      ":default_video_quality_analyzer_internal",
+      ":default_video_quality_analyzer_shared",
+      "../..:perf_test",
+      "../../../api:array_view",
+      "../../../api:video_quality_analyzer_api",
+      "../../../api/numerics",
+      "../../../api/units:data_size",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../api/video:encoded_image",
+      "../../../api/video:video_frame",
+      "../../../api/video:video_frame_type",
+      "../../../api/video:video_rtp_headers",
+      "../../../common_video",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:criticalsection",
+      "../../../rtc_base:logging",
+      "../../../rtc_base:macromagic",
+      "../../../rtc_base:platform_thread",
+      "../../../rtc_base:rtc_event",
+      "../../../rtc_base:rtc_numerics",
+      "../../../rtc_base:stringutils",
+      "../../../rtc_base:timeutils",
+      "../../../rtc_base/synchronization:mutex",
+      "../../../rtc_tools:video_quality_analysis",
+      "../../../system_wrappers",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  # 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",
+    ]
+
+    testonly = true
+    sources = [
+      "analyzer/video/default_video_quality_analyzer_cpu_measurer.cc",
+      "analyzer/video/default_video_quality_analyzer_cpu_measurer.h",
+      "analyzer/video/default_video_quality_analyzer_frame_in_flight.cc",
+      "analyzer/video/default_video_quality_analyzer_frame_in_flight.h",
+      "analyzer/video/default_video_quality_analyzer_frames_comparator.cc",
+      "analyzer/video/default_video_quality_analyzer_frames_comparator.h",
+      "analyzer/video/default_video_quality_analyzer_internal_shared_objects.cc",
+      "analyzer/video/default_video_quality_analyzer_internal_shared_objects.h",
+      "analyzer/video/default_video_quality_analyzer_stream_state.cc",
+      "analyzer/video/default_video_quality_analyzer_stream_state.h",
+      "analyzer/video/names_collection.cc",
+      "analyzer/video/names_collection.h",
+    ]
+
+    deps = [
+      ":default_video_quality_analyzer_shared",
+      ":multi_reader_queue",
+      "../../../api:array_view",
+      "../../../api:scoped_refptr",
+      "../../../api/numerics: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:system_wrappers",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings:strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+
+  rtc_library("default_video_quality_analyzer_shared") {
+    visibility = [ "*" ]
+
+    testonly = true
+    sources = [
+      "analyzer/video/default_video_quality_analyzer_shared_objects.cc",
+      "analyzer/video/default_video_quality_analyzer_shared_objects.h",
+    ]
+
+    deps = [
+      "../../../api/numerics:numerics",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:stringutils",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+  }
+
+  rtc_library("network_quality_metrics_reporter") {
+    visibility = [ "*" ]
+    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/units:data_size",
+      "../../../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") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "stats_based_network_quality_metrics_reporter.cc",
+      "stats_based_network_quality_metrics_reporter.h",
+    ]
+    deps = [
+      "../..: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/numerics",
+      "../../../api/test/network_emulation",
+      "../../../api/units:data_rate",
+      "../../../api/units:data_size",
+      "../../../api/units:timestamp",
+      "../../../rtc_base",
+      "../../../rtc_base:ip_address",
+      "../../../rtc_base:rtc_event",
+      "../../../rtc_base:stringutils",
+      "../../../rtc_base/synchronization:mutex",
+      "../../../system_wrappers:field_trial",
+    ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  }
+
+  rtc_library("cross_media_metrics_reporter") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "cross_media_metrics_reporter.cc",
+      "cross_media_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/numerics",
+      "../../../api/units:timestamp",
+      "../../../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") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [
+      "sdp/sdp_changer.cc",
+      "sdp/sdp_changer.h",
+    ]
+    deps = [
+      "../../../api:array_view",
+      "../../../api:libjingle_peerconnection_api",
+      "../../../api:peer_connection_quality_test_fixture_api",
+      "../../../api:rtp_parameters",
+      "../../../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",
+    ]
+  }
+
+  rtc_library("multi_reader_queue") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "analyzer/video/multi_reader_queue.h" ]
+    deps = [ "../../../rtc_base:checks" ]
+    absl_deps = [ "//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..0577bccb62
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
@@ -0,0 +1,156 @@
+/*
+ *  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 "rtc_base/logging.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+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);
+
+    const std::string stream_label = std::string(
+        analyzer_helper_->GetStreamLabelFromTrackId(*stat->track_identifier));
+
+    MutexLock lock(&lock_);
+    StatsSample prev_sample = last_stats_sample_[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_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_label] = sample;
+  }
+}
+
+std::string DefaultAudioQualityAnalyzer::GetTestCaseName(
+    const std::string& stream_label) const {
+  return test_case_name_ + "/" + stream_label;
+}
+
+void DefaultAudioQualityAnalyzer::Stop() {
+  using ::webrtc::test::ImproveDirection;
+  MutexLock lock(&lock_);
+  for (auto& item : streams_stats_) {
+    ReportResult("expand_rate", item.first, item.second.expand_rate, "unitless",
+                 ImproveDirection::kSmallerIsBetter);
+    ReportResult("accelerate_rate", item.first, item.second.accelerate_rate,
+                 "unitless", ImproveDirection::kSmallerIsBetter);
+    ReportResult("preemptive_rate", item.first, item.second.preemptive_rate,
+                 "unitless", ImproveDirection::kSmallerIsBetter);
+    ReportResult("speech_expand_rate", item.first,
+                 item.second.speech_expand_rate, "unitless",
+                 ImproveDirection::kSmallerIsBetter);
+    ReportResult("average_jitter_buffer_delay_ms", item.first,
+                 item.second.average_jitter_buffer_delay_ms, "ms",
+                 ImproveDirection::kNone);
+    ReportResult("preferred_buffer_size_ms", item.first,
+                 item.second.preferred_buffer_size_ms, "ms",
+                 ImproveDirection::kNone);
+  }
+}
+
+std::map<std::string, AudioStreamStats>
+DefaultAudioQualityAnalyzer::GetAudioStreamsStats() const {
+  MutexLock lock(&lock_);
+  return streams_stats_;
+}
+
+void DefaultAudioQualityAnalyzer::ReportResult(
+    const std::string& metric_name,
+    const std::string& stream_label,
+    const SamplesStatsCounter& counter,
+    const std::string& unit,
+    webrtc::test::ImproveDirection improve_direction) const {
+  test::PrintResultMeanAndError(
+      metric_name, /*modifier=*/"", GetTestCaseName(stream_label),
+      counter.IsEmpty() ? 0 : counter.GetAverage(),
+      counter.IsEmpty() ? 0 : counter.GetStandardDeviation(), unit,
+      /*important=*/false, improve_direction);
+}
+
+}  // 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..4ad0dd3da2
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.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_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/track_id_stream_info_map.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/testsupport/perf_test.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:
+  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;
+  void ReportResult(const std::string& metric_name,
+                    const std::string& stream_label,
+                    const SamplesStatsCounter& counter,
+                    const std::string& unit,
+                    webrtc::test::ImproveDirection improve_direction) const;
+
+  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, 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/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..e31843c8eb
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.cc
@@ -0,0 +1,1036 @@
+/*
+ *  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 <memory>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/units/time_delta.h"
+#include "api/video/i420_buffer.h"
+#include "api/video/video_frame.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_tools/frame_analyzer/video_geometry_aligner.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"
+
+namespace webrtc {
+namespace {
+
+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;
+}
+
+absl::string_view ToString(FrameDropPhase phase) {
+  switch (phase) {
+    case FrameDropPhase::kBeforeEncoder:
+      return "kBeforeEncoder";
+    case FrameDropPhase::kByEncoder:
+      return "kByEncoder";
+    case FrameDropPhase::kTransport:
+      return "kTransport";
+    case FrameDropPhase::kAfterDecoder:
+      return "kAfterDecoder";
+    case FrameDropPhase::kLastValue:
+      RTC_CHECK(false) << "FrameDropPhase::kLastValue mustn't be used";
+  }
+}
+
+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,
+    DefaultVideoQualityAnalyzerOptions options)
+    : options_(options),
+      clock_(clock),
+      frames_comparator_(clock, cpu_measurer_, options) {}
+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";
+    }
+  }
+  return frame_id;
+}
+
+void DefaultVideoQualityAnalyzer::OnFramePreEncode(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame) {
+  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());
+}
+
+void DefaultVideoQualityAnalyzer::OnFrameEncoded(
+    absl::string_view peer_name,
+    uint16_t frame_id,
+    const webrtc::EncodedImage& encoded_image,
+    const EncoderStats& stats) {
+  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, used_encoder);
+}
+
+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) {
+  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()));
+}
+
+void DefaultVideoQualityAnalyzer::OnFrameDecoded(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame,
+    const DecoderStats& stats) {
+  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 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_.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, used_decoder);
+}
+
+void DefaultVideoQualityAnalyzer::OnFrameRendered(
+    absl::string_view peer_name,
+    const webrtc::VideoFrame& frame) {
+  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(), frame.width(),
+                                   frame.height());
+
+  // 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);
+  }
+}
+
+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) {
+  RTC_LOG(LS_ERROR) << "Decoder error for frame_id=" << frame_id
+                    << ", code=" << error_code;
+}
+
+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;
+    }
+    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;
+      for (size_t peer_index : peers_->GetPresentIndexes()) {
+        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());
+        }
+
+        // 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));
+
+          if (frame.HaveAllPeersReceived()) {
+            captured_frames_in_flight_.erase(it);
+          }
+        }
+      }
+    }
+  }
+  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 =
+        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::ReportResults() {
+  using ::webrtc::test::ImproveDirection;
+
+  MutexLock lock(&mutex_);
+  for (auto& item : frames_comparator_.stream_stats()) {
+    ReportResults(GetTestCaseName(ToMetricName(item.first)), item.second,
+                  stream_frame_counters_.at(item.first));
+  }
+  test::PrintResult("cpu_usage", "", test_label_.c_str(), GetCpuUsagePercent(),
+                    "%", false, ImproveDirection::kSmallerIsBetter);
+  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;
+}
+
+void DefaultVideoQualityAnalyzer::ReportResults(
+    const std::string& test_case_name,
+    const StreamStats& stats,
+    const FrameCounters& frame_counters) {
+  using ::webrtc::test::ImproveDirection;
+  TimeDelta test_duration = Now() - start_time_;
+
+  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;
+  }
+
+  ReportResult("psnr", test_case_name, stats.psnr, "dB",
+               ImproveDirection::kBiggerIsBetter);
+  ReportResult("ssim", test_case_name, stats.ssim, "unitless",
+               ImproveDirection::kBiggerIsBetter);
+  ReportResult("transport_time", test_case_name, stats.transport_time_ms, "ms",
+               ImproveDirection::kSmallerIsBetter);
+  ReportResult("total_delay_incl_transport", test_case_name,
+               stats.total_delay_incl_transport_ms, "ms",
+               ImproveDirection::kSmallerIsBetter);
+  ReportResult("time_between_rendered_frames", test_case_name,
+               stats.time_between_rendered_frames_ms, "ms",
+               ImproveDirection::kSmallerIsBetter);
+  test::PrintResult("harmonic_framerate", "", test_case_name,
+                    harmonic_framerate_fps, "Hz", /*important=*/false,
+                    ImproveDirection::kBiggerIsBetter);
+  test::PrintResult("encode_frame_rate", "", test_case_name,
+                    stats.encode_frame_rate.IsEmpty()
+                        ? 0
+                        : stats.encode_frame_rate.GetEventsPerSecond(),
+                    "Hz", /*important=*/false,
+                    ImproveDirection::kBiggerIsBetter);
+  ReportResult("encode_time", test_case_name, stats.encode_time_ms, "ms",
+               ImproveDirection::kSmallerIsBetter);
+  ReportResult("time_between_freezes", test_case_name,
+               stats.time_between_freezes_ms, "ms",
+               ImproveDirection::kBiggerIsBetter);
+  ReportResult("freeze_time_ms", test_case_name, stats.freeze_time_ms, "ms",
+               ImproveDirection::kSmallerIsBetter);
+  ReportResult("pixels_per_frame", test_case_name,
+               stats.resolution_of_rendered_frame, "count",
+               ImproveDirection::kBiggerIsBetter);
+  test::PrintResult("min_psnr", "", test_case_name,
+                    stats.psnr.IsEmpty() ? 0 : stats.psnr.GetMin(), "dB",
+                    /*important=*/false, ImproveDirection::kBiggerIsBetter);
+  ReportResult("decode_time", test_case_name, stats.decode_time_ms, "ms",
+               ImproveDirection::kSmallerIsBetter);
+  ReportResult("receive_to_render_time", test_case_name,
+               stats.receive_to_render_time_ms, "ms",
+               ImproveDirection::kSmallerIsBetter);
+  test::PrintResult("dropped_frames", "", test_case_name,
+                    frame_counters.dropped, "count",
+                    /*important=*/false, ImproveDirection::kSmallerIsBetter);
+  test::PrintResult("frames_in_flight", "", test_case_name,
+                    frame_counters.captured - frame_counters.rendered -
+                        frame_counters.dropped,
+                    "count",
+                    /*important=*/false, ImproveDirection::kSmallerIsBetter);
+  test::PrintResult("rendered_frames", "", test_case_name,
+                    frame_counters.rendered, "count", /*important=*/false,
+                    ImproveDirection::kBiggerIsBetter);
+  ReportResult("max_skipped", test_case_name, stats.skipped_between_rendered,
+               "count", ImproveDirection::kSmallerIsBetter);
+  ReportResult("target_encode_bitrate", test_case_name,
+               stats.target_encode_bitrate / kBitsInByte, "bytesPerSecond",
+               ImproveDirection::kNone);
+  test::PrintResult("actual_encode_bitrate", "", test_case_name,
+                    static_cast<double>(stats.total_encoded_images_payload) /
+                        test_duration.seconds<double>(),
+                    "bytesPerSecond", /*important=*/false,
+                    ImproveDirection::kNone);
+
+  if (options_.report_detailed_frame_stats) {
+    test::PrintResult("num_encoded_frames", "", test_case_name,
+                      frame_counters.encoded, "count",
+                      /*important=*/false, ImproveDirection::kBiggerIsBetter);
+    test::PrintResult("num_decoded_frames", "", test_case_name,
+                      frame_counters.decoded, "count",
+                      /*important=*/false, ImproveDirection::kBiggerIsBetter);
+    test::PrintResult("num_send_key_frames", "", test_case_name,
+                      stats.num_send_key_frames, "count",
+                      /*important=*/false, ImproveDirection::kBiggerIsBetter);
+    test::PrintResult("num_recv_key_frames", "", test_case_name,
+                      stats.num_recv_key_frames, "count",
+                      /*important=*/false, ImproveDirection::kBiggerIsBetter);
+
+    ReportResult("recv_key_frame_size_bytes", test_case_name,
+                 stats.recv_key_frame_size_bytes, "count",
+                 ImproveDirection::kBiggerIsBetter);
+    ReportResult("recv_delta_frame_size_bytes", test_case_name,
+                 stats.recv_delta_frame_size_bytes, "count",
+                 ImproveDirection::kBiggerIsBetter);
+  }
+}
+
+void DefaultVideoQualityAnalyzer::ReportResult(
+    const std::string& metric_name,
+    const std::string& test_case_name,
+    const SamplesStatsCounter& counter,
+    const std::string& unit,
+    webrtc::test::ImproveDirection improve_direction) {
+  test::PrintResult(metric_name, /*modifier=*/"", test_case_name, counter, unit,
+                    /*important=*/false, improve_direction);
+}
+
+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_->size() <= 2 && key.sender != key.receiver) {
+    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..5aec7c1e26
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h
@@ -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.
+ */
+
+#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/numerics/samples_stats_counter.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 "api/video/video_frame_type.h"
+#include "rtc_base/event.h"
+#include "rtc_base/platform_thread.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"
+#include "test/testsupport/perf_test.h"
+
+namespace webrtc {
+
+class DefaultVideoQualityAnalyzer : public VideoQualityAnalyzerInterface {
+ public:
+  explicit DefaultVideoQualityAnalyzer(
+      webrtc::Clock* clock,
+      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) 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) 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_);
+
+  // Report results for all metrics for all streams.
+  void ReportResults();
+  void ReportResults(const std::string& test_case_name,
+                     const StreamStats& stats,
+                     const FrameCounters& frame_counters)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  // Report result for single metric for specified stream.
+  static void ReportResult(const std::string& metric_name,
+                           const std::string& test_case_name,
+                           const SamplesStatsCounter& counter,
+                           const std::string& unit,
+                           webrtc::test::ImproveDirection improve_direction =
+                               webrtc::test::ImproveDirection::kNone);
+  // 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_;
+
+  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..13e77b4586
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.cc
@@ -0,0 +1,194 @@
+/*
+ *  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_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);
+  }
+}
+
+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,
+                                   StreamCodecInfo used_encoder) {
+  encoded_time_ = time;
+  frame_type_ = frame_type;
+  encoded_image_size_ = encoded_image_size;
+  target_encode_bitrate_ += target_encode_bitrate;
+  // 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,
+                                   StreamCodecInfo used_decoder) {
+  receiver_stats_[peer].decode_end_time = time;
+  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,
+                                    int width,
+                                    int height) {
+  receiver_stats_[peer].rendered_time = time;
+  receiver_stats_[peer].rendered_frame_width = width;
+  receiver_stats_[peer].rendered_frame_height = height;
+}
+
+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 {
+  FrameStats stats(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_;
+
+  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.rendered_frame_width = receiver_stats->rendered_frame_width;
+    stats.rendered_frame_height = receiver_stats->rendered_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;
+  }
+  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..a1ce8faf5b
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frame_in_flight.h
@@ -0,0 +1,159 @@
+/*
+ *  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/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> rendered_frame_width = absl::nullopt;
+  absl::optional<int> rendered_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;
+};
+
+// 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,
+                      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,
+                      StreamCodecInfo used_decoder);
+
+  bool HasDecodeEndTime(size_t peer) const;
+
+  void OnFrameRendered(size_t peer,
+                       webrtc::Timestamp time,
+                       int width,
+                       int height);
+
+  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_;
+
+  // 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;
+  // 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..e216c70c6f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator.cc
@@ -0,0 +1,533 @@
+/*
+ *  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 <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"
+
+namespace webrtc {
+namespace {
+
+constexpr TimeDelta kFreezeThreshold = TimeDelta::Millis(150);
+constexpr int kMaxActiveComparisons = 10;
+
+SamplesStatsCounter::StatsSample StatsSample(double value,
+                                             Timestamp sampling_time) {
+  return SamplesStatsCounter::StatsSample{value, sampling_time};
+}
+
+SamplesStatsCounter::StatsSample StatsSample(TimeDelta duration,
+                                             Timestamp sampling_time) {
+  return SamplesStatsCounter::StatsSample{duration.ms<double>(), sampling_time};
+}
+
+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.rendered_frame_width.has_value())
+          << "Regular comparison has to have rendered_frame_width";
+      RTC_DCHECK(comparison.frame_stats.rendered_frame_height.has_value())
+          << "Regular comparison has to have rendered_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";
+      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()) {
+        RTC_DCHECK(comparison.frame_stats.received_time.IsFinite())
+            << "Dropped frame comparison has to have received_time when "
+            << "decode_end_time is set";
+        RTC_DCHECK(comparison.frame_stats.decode_start_time.IsFinite())
+            << "Dropped frame comparison has to have decode_start_time when "
+            << "decode_end_time is set";
+        RTC_DCHECK(comparison.frame_stats.used_decoder.has_value())
+            << "Dropped frame comparison has to have used_decoder when "
+            << "decode_end_time is set";
+      } else {
+        RTC_DCHECK(!comparison.frame_stats.received_time.IsFinite())
+            << "Dropped frame comparison can't have received_time when "
+            << "decode_end_time is not set";
+        RTC_DCHECK(!comparison.frame_stats.decode_start_time.IsFinite())
+            << "Dropped frame comparison can't have decode_start_time when "
+            << "decode_end_time is not set";
+        RTC_DCHECK(!comparison.frame_stats.used_decoder.has_value())
+            << "Dropped frame comparison can't have used_decoder when "
+            << "decode_end_time is not set";
+      }
+      RTC_DCHECK(!comparison.frame_stats.rendered_time.IsFinite())
+          << "Dropped frame comparison can't have rendered_time";
+      RTC_DCHECK(!comparison.frame_stats.rendered_frame_width.has_value())
+          << "Dropped frame comparison can't have rendered_frame_width";
+      RTC_DCHECK(!comparison.frame_stats.rendered_frame_height.has_value())
+          << "Dropped frame comparison can't have rendered_frame_height";
+      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";
+      RTC_DCHECK(!comparison.frame_stats.rendered_frame_width.has_value())
+          << "Frame in flight comparison can't have rendered_frame_width";
+      RTC_DCHECK(!comparison.frame_stats.rendered_frame_height.has_value())
+          << "Frame in flight comparison can't have rendered_frame_height";
+
+      if (comparison.frame_stats.decode_end_time.IsFinite()) {
+        RTC_DCHECK(comparison.frame_stats.used_decoder.has_value())
+            << "Frame in flight comparison has to have used_decoder when "
+            << "decode_end_time is set";
+        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.";
+      }
+      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()));
+    }
+  }
+}
+
+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()));
+  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()));
+  // 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(1000);
+      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++;
+  }
+
+  if (psnr > 0) {
+    stats->psnr.AddSample(StatsSample(psnr, frame_stats.rendered_time));
+  }
+  if (ssim > 0) {
+    stats->ssim.AddSample(StatsSample(ssim, frame_stats.received_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.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));
+    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));
+
+    // 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) {
+    if (frame_stats.rendered_time.IsFinite()) {
+      stats->resolution_of_rendered_frame.AddSample(
+          StatsSample(*comparison.frame_stats.rendered_frame_width *
+                          *comparison.frame_stats.rendered_frame_height,
+                      frame_stats.rendered_time));
+      stats->total_delay_incl_transport_ms.AddSample(
+          StatsSample(frame_stats.rendered_time - frame_stats.captured_time,
+                      frame_stats.received_time));
+      stats->receive_to_render_time_ms.AddSample(
+          StatsSample(frame_stats.rendered_time - frame_stats.received_time,
+                      frame_stats.rendered_time));
+    }
+    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));
+
+      // 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));
+      } 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));
+      }
+    }
+    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));
+    }
+
+    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));
+      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));
+        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));
+        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..b9b822072f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_frames_comparator_test.cc
@@ -0,0 +1,283 @@
+/*
+ *  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 <vector>
+
+#include "api/units/timestamp.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/clock.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 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;
+}
+
+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(
+    Timestamp captured_time) {
+  FrameStats frame_stats(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.rendered_frame_width = 10;
+  frame_stats.rendered_frame_height = 10;
+  return frame_stats;
+}
+
+FrameStats ShiftStatsOn(const FrameStats& stats, TimeDelta delta) {
+  FrameStats frame_stats(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.rendered_frame_width = stats.rendered_frame_width;
+  frame_stats.rendered_frame_height = stats.rendered_frame_height;
+
+  return frame_stats;
+}
+
+double GetFirstOrDie(const SamplesStatsCounter& counter) {
+  EXPECT_TRUE(!counter.IsEmpty()) << "Counter has to be not empty";
+  return counter.GetSamples()[0];
+}
+
+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();
+}
+
+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(stream_start_time);
+
+  comparator.Start(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();
+  EXPECT_DOUBLE_EQ(GetFirstOrDie(stats.at(stats_key).transport_time_ms), 20.0);
+  EXPECT_DOUBLE_EQ(
+      GetFirstOrDie(stats.at(stats_key).total_delay_incl_transport_ms), 60.0);
+  EXPECT_DOUBLE_EQ(GetFirstOrDie(stats.at(stats_key).encode_time_ms), 10.0);
+  EXPECT_DOUBLE_EQ(GetFirstOrDie(stats.at(stats_key).decode_time_ms), 0.01);
+  EXPECT_DOUBLE_EQ(GetFirstOrDie(stats.at(stats_key).receive_to_render_time_ms),
+                   30.0);
+  EXPECT_DOUBLE_EQ(
+      GetFirstOrDie(stats.at(stats_key).resolution_of_rendered_frame), 100.0);
+}
+
+TEST(DefaultVideoQualityAnalyzerFramesComparatorTest,
+     MultiFrameStatsPresentedAfterAddingTwoComparisonWith10msDelay) {
+  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(stream_start_time);
+  FrameStats frame_stats2 = FrameStatsWith10msDeltaBetweenPhasesAnd10x10Frame(
+      stream_start_time + TimeDelta::Millis(15));
+  frame_stats2.prev_frame_rendered_time = frame_stats1.rendered_time;
+
+  comparator.Start(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();
+  EXPECT_DOUBLE_EQ(
+      GetFirstOrDie(stats.at(stats_key).time_between_rendered_frames_ms), 15.0);
+  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(stream_start_time);
+  stats.push_back(frame_stats);
+  // 2nd stat
+  frame_stats = ShiftStatsOn(frame_stats, TimeDelta::Millis(15));
+  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.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.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.decode_end_time =
+      frame_stats.captured_time + TimeDelta::Millis(50);
+  frame_stats.used_decoder =
+      Vp8CodecForOneFrame(1, frame_stats.decode_end_time);
+  stats.push_back(frame_stats);
+  // 6th stat
+  frame_stats = ShiftStatsOn(frame_stats, TimeDelta::Millis(15));
+  frame_stats.rendered_time = frame_stats.captured_time + TimeDelta::Millis(60);
+  frame_stats.rendered_frame_width = 10;
+  frame_stats.rendered_frame_height = 10;
+  stats.push_back(frame_stats);
+
+  comparator.Start(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_rendered_frame.GetAverage(), 100)
+      << ToString(result_stats.resolution_of_rendered_frame);
+  EXPECT_EQ(result_stats.resolution_of_rendered_frame.NumSamples(), 1);
+
+  EXPECT_DOUBLE_EQ(result_stats.encode_frame_rate.GetEventsPerSecond(),
+                   4.0 / 45 * 1000)
+      << "There should be 4 events with interval of 15 ms";
+}
+
+}  // 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..04f653c02b
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_internal_shared_objects.h
@@ -0,0 +1,123 @@
+/*
+ *  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 "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_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 {
+  explicit FrameStats(Timestamp captured_time) : captured_time(captured_time) {}
+
+  // 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;
+
+  absl::optional<int> rendered_frame_width = absl::nullopt;
+  absl::optional<int> rendered_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;
+};
+
+// 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_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..cc17429072
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.cc
@@ -0,0 +1,121 @@
+/*
+ *  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 "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kMicrosPerSecond = 1000000;
+
+}  // namespace
+
+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..8f0afd36f7
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_shared_objects.h
@@ -0,0 +1,249 @@
+/*
+ *  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 <map>
+#include <memory>
+#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"
+
+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;
+};
+
+// 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();
+};
+
+// Represents phases where video frame can be dropped and such drop will be
+// detected by analyzer.
+enum class FrameDropPhase : int {
+  kBeforeEncoder,
+  kByEncoder,
+  kTransport,
+  kAfterDecoder,
+  // kLastValue must be the last value in this enumeration.
+  kLastValue
+};
+
+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 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_rendered_frame;
+  SamplesStatsCounter target_encode_bitrate;
+
+  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;
+};
+
+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;
+  // 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..40662fd31d
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer_test.cc
@@ -0,0 +1,1980 @@
+/*
+ *  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/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 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(),
+      /*audio_level=*/absl::nullopt,
+      /*absolute_capture_time=*/absl::nullopt,
+      /*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) {
+  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());
+    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);
+    }
+  }
+}
+
+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(),
+                                       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());
+  }
+
+  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(),
+                                       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());
+  }
+
+  // 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());
+  }
+
+  // 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(),
+                                       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());
+  }
+
+  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(),
+                                       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());
+  }
+
+  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(),
+                                       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());
+
+  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(),
+                                       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());
+  }
+
+  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_rendered_frame.IsEmpty());
+    EXPECT_GE(it->second.resolution_of_rendered_frame.GetMin(),
+              kFrameWidth * kFrameHeight - 1);
+    EXPECT_LE(it->second.resolution_of_rendered_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_rendered_frame.IsEmpty());
+    EXPECT_GE(it->second.resolution_of_rendered_frame.GetMin(),
+              kFrameWidth * kFrameHeight - 1);
+    EXPECT_LE(it->second.resolution_of_rendered_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(),
+                                       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());
+
+  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(),
+                                       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());
+
+    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(),
+                                       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());
+
+    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(),
+                                       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());
+  }
+
+  // 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(),
+                                       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());
+  }
+
+  // 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(),
+                                       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());
+
+  // 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());
+
+  // 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(), 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());
+  }
+
+  // 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(), 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());
+  }
+
+  // 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(),
+                                       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);
+      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(), 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());
+  }
+
+  // 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(), 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());
+  }
+
+  // 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(), 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());
+  }
+  // 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(), 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());
+
+  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(), 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());
+
+  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(), 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());
+
+  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(), 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());
+
+  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(), 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(), 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());
+
+  // 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(), 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());
+
+  // 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(), 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());
+  analyzer.UnregisterParticipantInCall("bob");
+  // 2nd simulcast layer encoded
+  analyzer.OnFrameEncoded("alice", frame.id(), FakeEncode(frame),
+                          VideoQualityAnalyzerInterface::EncoderStats());
+
+  // 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(), 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(), 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);
+}
+
+}  // 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..0b9e2cd5a2
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.cc
@@ -0,0 +1,163 @@
+/*
+ *  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_++;
+  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) {
+  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) {
+  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..f41e29e8ab
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/example_video_quality_analyzer.h
@@ -0,0 +1,99 @@
+/*
+ *  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) 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) 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_) = 0;
+  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..e29bad1ad4
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection.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_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 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..65932ea26e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/names_collection_test.cc
@@ -0,0 +1,141 @@
+/*
+ *  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)));
+}
+
+}  // 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..41b8aec8a1
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_decoder.cc
@@ -0,0 +1,270 @@
+/*
+ *  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.
+    {
+      MutexLock lock(&mutex_);
+      timestamp_to_frame_id_.erase(input_image.Timestamp());
+      decoding_images_.erase(input_image.Timestamp());
+    }
+    analyzer_->OnDecoderError(peer_name_,
+                              out.id.value_or(VideoFrame::kNotSetId), result);
+  }
+  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..d3fbab08a0
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.cc
@@ -0,0 +1,384 @@
+/*
+ *  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 "rtc_base/logging.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+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,
+    std::map<std::string, absl::optional<int>> stream_required_spatial_index,
+    EncodedImageDataInjector* injector,
+    VideoQualityAnalyzerInterface* analyzer)
+    : peer_name_(peer_name),
+      delegate_(std::move(delegate)),
+      bitrate_multiplier_(bitrate_multiplier),
+      stream_required_spatial_index_(std::move(stream_required_spatial_index)),
+      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;
+  if (codec_settings->codecType == kVideoCodecVP9) {
+    if (codec_settings->VP9().numberOfSpatialLayers > 1) {
+      switch (codec_settings->VP9().interLayerPred) {
+        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 codec_settings->VP9().interLayerPred";
+          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;
+  }
+
+  if (!discard) {
+    // Analyzer should see only encoded images, that weren't discarded. But all
+    // not discarded layers have to be passed.
+    VideoQualityAnalyzerInterface::EncoderStats stats;
+    stats.encoder_name = codec_name;
+    stats.target_encode_bitrate = target_encode_bitrate;
+    analyzer_->OnFrameEncoded(peer_name_, frame_id, encoded_image, stats);
+  }
+
+  // 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);
+  absl::optional<int> required_spatial_index =
+      stream_required_spatial_index_[stream_label];
+  if (required_spatial_index) {
+    if (*required_spatial_index == kAnalyzeAnySpatialStream) {
+      return false;
+    }
+    absl::optional<int> cur_spatial_index = encoded_image.SpatialIndex();
+    if (!cur_spatial_index) {
+      cur_spatial_index = 0;
+    }
+    RTC_CHECK(mode_ != SimulcastMode::kNormal)
+        << "Analyzing encoder is in kNormal "
+           "mode, but spatial layer/simulcast "
+           "stream met.";
+    if (mode_ == 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 != *required_spatial_index;
+    } else if (mode_ == 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 > *required_spatial_index;
+    } else if (mode_ == 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 have to be discarded.
+      if (encoded_image._frameType == VideoFrameType::kVideoFrameKey) {
+        return *cur_spatial_index > *required_spatial_index;
+      } else {
+        return *cur_spatial_index != *required_spatial_index;
+      }
+    } else {
+      RTC_DCHECK_NOTREACHED() << "Unsupported encoder mode";
+    }
+  }
+  return false;
+}
+
+QualityAnalyzingVideoEncoderFactory::QualityAnalyzingVideoEncoderFactory(
+    absl::string_view peer_name,
+    std::unique_ptr<VideoEncoderFactory> delegate,
+    double bitrate_multiplier,
+    std::map<std::string, absl::optional<int>> stream_required_spatial_index,
+    EncodedImageDataInjector* injector,
+    VideoQualityAnalyzerInterface* analyzer)
+    : peer_name_(peer_name),
+      delegate_(std::move(delegate)),
+      bitrate_multiplier_(bitrate_multiplier),
+      stream_required_spatial_index_(std::move(stream_required_spatial_index)),
+      injector_(injector),
+      analyzer_(analyzer) {}
+QualityAnalyzingVideoEncoderFactory::~QualityAnalyzingVideoEncoderFactory() =
+    default;
+
+std::vector<SdpVideoFormat>
+QualityAnalyzingVideoEncoderFactory::GetSupportedFormats() const {
+  return delegate_->GetSupportedFormats();
+}
+
+std::unique_ptr<VideoEncoder>
+QualityAnalyzingVideoEncoderFactory::CreateVideoEncoder(
+    const SdpVideoFormat& format) {
+  return std::make_unique<QualityAnalyzingVideoEncoder>(
+      peer_name_, delegate_->CreateVideoEncoder(format), bitrate_multiplier_,
+      stream_required_spatial_index_, 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..135c85c133
--- /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/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 {
+
+// Tells QualityAnalyzingVideoEncoder that it shouldn't mark any spatial stream
+// as to be discarded. In such case the top stream will be passed to
+// VideoQualityAnalyzerInterface as a reference.
+constexpr int kAnalyzeAnySpatialStream = -1;
+
+// 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:
+  QualityAnalyzingVideoEncoder(
+      absl::string_view peer_name,
+      std::unique_ptr<VideoEncoder> delegate,
+      double bitrate_multiplier,
+      std::map<std::string, absl::optional<int>> stream_required_spatial_index,
+      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 "Foo" isn't simulcast/SVC stream
+  // 2. `kAnalyzeAnySpatialStream` means all simulcast/SVC streams are required
+  // 3. Concrete value means that particular simulcast/SVC stream have to be
+  //    analyzed.
+  std::map<std::string, absl::optional<int>> stream_required_spatial_index_;
+  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,
+      std::map<std::string, absl::optional<int>> stream_required_spatial_index,
+      EncodedImageDataInjector* injector,
+      VideoQualityAnalyzerInterface* analyzer);
+  ~QualityAnalyzingVideoEncoderFactory() override;
+
+  // Methods of VideoEncoderFactory interface.
+  std::vector<SdpVideoFormat> GetSupportedFormats() 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_;
+  std::map<std::string, absl::optional<int>> stream_required_spatial_index_;
+  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..1825bfbe20
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.cc
@@ -0,0 +1,59 @@
+/*
+ *  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 <cstring>
+
+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 IsDummyFrameBuffer(
+    rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer) {
+  if (video_frame_buffer->width() != 2 || video_frame_buffer->height() != 2) {
+    return false;
+  }
+  rtc::scoped_refptr<webrtc::I420BufferInterface> buffer =
+      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..84c5abefbf
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/simulcast_dummy_buffer_helper.h
@@ -0,0 +1,28 @@
+/*
+ *  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/i420_buffer.h"
+#include "api/video/video_frame_buffer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateDummyFrameBuffer();
+
+bool IsDummyFrameBuffer(
+    rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer);
+
+}  // 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/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_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..661208e888
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.cc
@@ -0,0 +1,366 @@
+/*
+ *  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 <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/clock.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/testsupport/fixed_fps_video_frame_writer_adapter.h"
+#include "test/video_renderer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+namespace {
+
+class VideoWriter final : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  VideoWriter(test::VideoFrameWriter* video_writer, int sampling_modulo)
+      : video_writer_(video_writer), sampling_modulo_(sampling_modulo) {}
+  ~VideoWriter() override = default;
+
+  void OnFrame(const VideoFrame& frame) override {
+    if (frames_counter_++ % sampling_modulo_ != 0) {
+      return;
+    }
+    bool result = video_writer_->WriteFrame(frame);
+    RTC_CHECK(result) << "Failed to write frame";
+  }
+
+ private:
+  test::VideoFrameWriter* const video_writer_;
+  const int sampling_modulo_;
+
+  int64_t frames_counter_ = 0;
+};
+
+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::VideoFrameIdsWriter::VideoFrameIdsWriter(
+    absl::string_view file_name)
+    : file_name_(file_name) {
+  output_file_ = fopen(file_name_.c_str(), "wb");
+  RTC_CHECK(output_file_ != nullptr)
+      << "Failed to open file to dump frame ids for writing: " << file_name_;
+}
+
+VideoQualityAnalyzerInjectionHelper::VideoFrameIdsWriter::
+    ~VideoFrameIdsWriter() {
+  fclose(output_file_);
+}
+
+void VideoQualityAnalyzerInjectionHelper::VideoFrameIdsWriter::WriteFrameId(
+    uint16_t frame_id) {
+  int chars_written = fprintf(output_file_, "%d\n", frame_id);
+  RTC_CHECK_GE(chars_written, 2)
+      << "Failed to write frame id to the output file: " << file_name_;
+}
+
+VideoQualityAnalyzerInjectionHelper::VideoWriter2::VideoWriter2(
+    test::VideoFrameWriter* video_writer,
+    VideoFrameIdsWriter* frame_ids_writer,
+    int sampling_modulo)
+    : video_writer_(video_writer),
+      frame_ids_writer_(frame_ids_writer),
+      sampling_modulo_(sampling_modulo) {}
+
+void VideoQualityAnalyzerInjectionHelper::VideoWriter2::OnFrame(
+    const VideoFrame& frame) {
+  if (frames_counter_++ % sampling_modulo_ != 0) {
+    return;
+  }
+  bool result = video_writer_->WriteFrame(frame);
+  RTC_CHECK(result) << "Failed to write frame";
+  if (frame_ids_writer_) {
+    frame_ids_writer_->WriteFrameId(frame.id());
+  }
+}
+
+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,
+    std::map<std::string, absl::optional<int>> stream_required_spatial_index)
+    const {
+  return std::make_unique<QualityAnalyzingVideoEncoderFactory>(
+      peer_name, std::move(delegate), bitrate_multiplier,
+      std::move(stream_required_spatial_index), 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;
+  test::VideoFrameWriter* writer = nullptr;
+  if (config.input_dump_options.has_value()) {
+    // Using new API for video dumping.
+    writer = MaybeCreateVideoWriter(
+        config.input_dump_options->GetInputDumpFileName(*config.stream_label),
+        config);
+    RTC_CHECK(writer);
+    VideoFrameIdsWriter* frame_ids_writer = nullptr;
+    if (config.input_dump_options->export_frame_ids()) {
+      frame_ids_writers_.push_back(std::make_unique<VideoFrameIdsWriter>(
+          *config.input_dump_options->GetInputFrameIdsDumpFileName(
+              *config.stream_label)));
+      frame_ids_writer = frame_ids_writers_.back().get();
+    }
+    sinks.push_back(std::make_unique<VideoWriter2>(
+        writer, frame_ids_writer,
+        config.input_dump_options->sampling_modulo()));
+  } else {
+    // Using old API. To be removed.
+    writer = MaybeCreateVideoWriter(config.input_dump_file_name, config);
+    if (writer) {
+      sinks.push_back(std::make_unique<VideoWriter>(
+          writer, config.input_dump_sampling_modulo));
+    }
+  }
+  if (config.show_on_screen) {
+    sinks.push_back(absl::WrapUnique(
+        test::VideoRenderer::Create((*config.stream_label + "-capture").c_str(),
+                                    config.width, config.height)));
+  }
+  {
+    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<AnalyzingVideoSink>(peer_name, this);
+}
+
+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();
+  frame_ids_writers_.clear();
+}
+
+test::VideoFrameWriter*
+VideoQualityAnalyzerInjectionHelper::MaybeCreateVideoWriter(
+    absl::optional<std::string> file_name,
+    const PeerConnectionE2EQualityTestFixture::VideoConfig& config) {
+  if (!file_name.has_value()) {
+    return nullptr;
+  }
+  // TODO(titovartem) create only one file writer for simulcast video track.
+  // For now this code will be invoked for each simulcast stream separately, but
+  // only one file will be used.
+  std::unique_ptr<test::VideoFrameWriter> video_writer =
+      std::make_unique<test::Y4mVideoFrameWriterImpl>(
+          *file_name, config.width, config.height, config.fps);
+  if (config.output_dump_use_fixed_framerate) {
+    video_writer = std::make_unique<test::FixedFpsVideoFrameWriterAdapter>(
+        config.fps, clock_, std::move(video_writer));
+  }
+  test::VideoFrameWriter* out = video_writer.get();
+  video_writers_.push_back(std::move(video_writer));
+  return out;
+}
+
+void VideoQualityAnalyzerInjectionHelper::OnFrame(absl::string_view peer_name,
+                                                  const VideoFrame& frame) {
+  rtc::scoped_refptr<I420BufferInterface> i420_buffer =
+      frame.video_frame_buffer()->ToI420();
+  if (IsDummyFrameBuffer(i420_buffer)) {
+    // 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(*i420_buffer));
+  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;
+  test::VideoFrameWriter* writer = nullptr;
+  if (config.output_dump_options.has_value()) {
+    // Using new API with output directory.
+    writer = MaybeCreateVideoWriter(
+        config.output_dump_options->GetOutputDumpFileName(
+            receiver_stream.stream_label, receiver_stream.peer_name),
+        config);
+    RTC_CHECK(writer);
+    VideoFrameIdsWriter* frame_ids_writer = nullptr;
+    if (config.output_dump_options->export_frame_ids()) {
+      frame_ids_writers_.push_back(std::make_unique<VideoFrameIdsWriter>(
+          *config.output_dump_options->GetOutputFrameIdsDumpFileName(
+              receiver_stream.stream_label, receiver_stream.peer_name)));
+      frame_ids_writer = frame_ids_writers_.back().get();
+    }
+    sinks.push_back(std::make_unique<VideoWriter2>(
+        writer, frame_ids_writer,
+        config.output_dump_options->sampling_modulo()));
+  } else {
+    // Using old API. To be removed.
+    absl::optional<std::string> output_dump_file_name =
+        config.output_dump_file_name;
+    if (output_dump_file_name.has_value() && peers_count_ > 2) {
+      // TODO(titovartem): make this default behavior for any amount of peers.
+      rtc::StringBuilder builder(*output_dump_file_name);
+      builder << "." << receiver_stream.peer_name;
+      output_dump_file_name = builder.str();
+    }
+    writer = MaybeCreateVideoWriter(output_dump_file_name, config);
+    if (writer) {
+      sinks.push_back(std::make_unique<VideoWriter>(
+          writer, config.output_dump_sampling_modulo));
+    }
+  }
+  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..76b8b446bc
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_analyzer_injection_helper.h
@@ -0,0 +1,195 @@
+/*
+ *  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/peerconnection_quality_test_fixture.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/encoded_image_data_injector.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:
+  using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
+
+  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,
+      std::map<std::string, absl::optional<int>> stream_required_spatial_index)
+      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 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.
+  std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> CreateVideoSink(
+      absl::string_view peer_name);
+
+  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:
+  class AnalyzingVideoSink final : public rtc::VideoSinkInterface<VideoFrame> {
+   public:
+    explicit AnalyzingVideoSink(absl::string_view peer_name,
+                                VideoQualityAnalyzerInjectionHelper* helper)
+        : peer_name_(peer_name), helper_(helper) {}
+    ~AnalyzingVideoSink() 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;
+    }
+  };
+
+  class VideoFrameIdsWriter {
+   public:
+    explicit VideoFrameIdsWriter(absl::string_view file_name);
+    ~VideoFrameIdsWriter();
+
+    void WriteFrameId(uint16_t frame_id);
+
+   private:
+    const std::string file_name_;
+    FILE* output_file_;
+  };
+
+  class VideoWriter2 final : public rtc::VideoSinkInterface<VideoFrame> {
+   public:
+    VideoWriter2(test::VideoFrameWriter* video_writer,
+                 VideoFrameIdsWriter* frame_ids_writer,
+                 int sampling_modulo);
+    ~VideoWriter2() override = default;
+
+    void OnFrame(const VideoFrame& frame) override;
+
+   private:
+    test::VideoFrameWriter* const video_writer_;
+    VideoFrameIdsWriter* const frame_ids_writer_;
+    const int sampling_modulo_;
+
+    int64_t frames_counter_ = 0;
+  };
+
+  test::VideoFrameWriter* MaybeCreateVideoWriter(
+      absl::optional<std::string> file_name,
+      const PeerConnectionE2EQualityTestFixture::VideoConfig& config);
+  // 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_;
+  std::vector<std::unique_ptr<VideoFrameIdsWriter>> frame_ids_writers_;
+
+  Mutex mutex_;
+  int peers_count_ RTC_GUARDED_BY(mutex_);
+  // Map from stream label to the video config.
+  std::map<std::string, 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..7a27ae0680
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc
@@ -0,0 +1,133 @@
+/*
+ *  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 "api/stats/rtc_stats.h"
+#include "api/stats/rtcstats_objects.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+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_us() > sample.sample_time.us()) {
+      sample.sample_time = Timestamp::Micros(s->timestamp_us());
+    }
+    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(GetTestCaseName(item.first), item.second);
+  }
+}
+
+std::string VideoQualityMetricsReporter::GetTestCaseName(
+    const std::string& stream_label) const {
+  return test_case_name_ + "/" + stream_label;
+}
+
+void VideoQualityMetricsReporter::ReportVideoBweResults(
+    const std::string& test_case_name,
+    const VideoBweStats& video_bwe_stats) {
+  ReportResult("available_send_bandwidth", test_case_name,
+               video_bwe_stats.available_send_bandwidth, "bytesPerSecond");
+  ReportResult("transmission_bitrate", test_case_name,
+               video_bwe_stats.transmission_bitrate, "bytesPerSecond");
+  ReportResult("retransmission_bitrate", test_case_name,
+               video_bwe_stats.retransmission_bitrate, "bytesPerSecond");
+}
+
+void VideoQualityMetricsReporter::ReportResult(
+    const std::string& metric_name,
+    const std::string& test_case_name,
+    const SamplesStatsCounter& counter,
+    const std::string& unit,
+    webrtc::test::ImproveDirection improve_direction) {
+  test::PrintResult(metric_name, /*modifier=*/"", test_case_name, counter, unit,
+                    /*important=*/false, improve_direction);
+}
+
+}  // 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..ff195a450e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer/video/video_quality_metrics_reporter.h
@@ -0,0 +1,86 @@
+/*
+ *  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/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"
+#include "test/testsupport/perf_test.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) : clock_(clock) {}
+  ~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& stream_label) const;
+  static void ReportVideoBweResults(const std::string& test_case_name,
+                                    const VideoBweStats& video_bwe_stats);
+  // Report result for single metric for specified stream.
+  static void ReportResult(const std::string& metric_name,
+                           const std::string& test_case_name,
+                           const SamplesStatsCounter& counter,
+                           const std::string& unit,
+                           webrtc::test::ImproveDirection improve_direction =
+                               webrtc::test::ImproveDirection::kNone);
+  Timestamp Now() const { return clock_->CurrentTime(); }
+
+  Clock* const clock_;
+
+  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..852f0a3435
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer_helper.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/pc/e2e/analyzer_helper.h"
+
+#include <utility>
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+AnalyzerHelper::AnalyzerHelper() {
+  signaling_sequence_checker_.Detach();
+}
+
+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)}});
+}
+
+const 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;
+}
+
+absl::string_view AnalyzerHelper::GetStreamLabelFromTrackId(
+    absl::string_view track_id) const {
+  return GetStreamInfoFromTrackId(track_id).stream_label;
+}
+
+absl::string_view AnalyzerHelper::GetSyncGroupLabelFromTrackId(
+    absl::string_view track_id) const {
+  return GetStreamInfoFromTrackId(track_id).sync_group;
+}
+
+}  // 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..9cebd7015e
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/analyzer_helper.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_PC_E2E_ANALYZER_HELPER_H_
+#define TEST_PC_E2E_ANALYZER_HELPER_H_
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.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 behaviour.
+class AnalyzerHelper : public TrackIdStreamInfoMap {
+ public:
+  AnalyzerHelper();
+
+  void AddTrackToStreamMapping(std::string track_id, std::string stream_label);
+  void AddTrackToStreamMapping(std::string track_id,
+                               std::string stream_label,
+                               std::string sync_group);
+
+  absl::string_view GetStreamLabelFromTrackId(
+      absl::string_view track_id) const override;
+
+  absl::string_view GetSyncGroupLabelFromTrackId(
+      absl::string_view track_id) const override;
+
+ private:
+  struct StreamInfo {
+    std::string stream_label;
+    std::string sync_group;
+  };
+
+  const StreamInfo& GetStreamInfoFromTrackId(absl::string_view track_id) const;
+
+  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..96f661fd4f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.cc
@@ -0,0 +1,129 @@
+/*
+ *  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/units/timestamp.h"
+#include "rtc_base/event.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+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<absl::string_view, std::vector<const RTCInboundRTPStreamStats*>>
+      sync_group_stats;
+  for (const auto& stat : inbound_stats) {
+    auto media_source_stat =
+        report->GetAs<RTCMediaStreamTrackStats>(*stat->track_id);
+    if (stat->estimated_playout_timestamp.ValueOrDefault(0.) > 0 &&
+        media_source_stat->track_identifier.is_defined()) {
+      sync_group_stats[reporter_helper_->GetSyncGroupLabelFromTrackId(
+                           *media_source_stat->track_identifier)]
+          .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()) {
+      auto audio_source_stat =
+          report->GetAs<RTCMediaStreamTrackStats>(*audio_stat->track_id);
+      auto video_source_stat =
+          report->GetAs<RTCMediaStreamTrackStats>(*video_stat->track_id);
+      // *_source_stat->track_identifier is always defined here because we
+      // checked it while grouping stats.
+      stats_info_[sync_group].audio_stream_label =
+          std::string(reporter_helper_->GetStreamLabelFromTrackId(
+              *audio_source_stat->track_identifier));
+      stats_info_[sync_group].video_stream_label =
+          std::string(reporter_helper_->GetStreamLabelFromTrackId(
+              *video_source_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;
+    ReportResult("audio_ahead_ms",
+                 GetTestCaseName(pair.second.audio_stream_label, sync_group),
+                 pair.second.audio_ahead_ms, "ms",
+                 webrtc::test::ImproveDirection::kSmallerIsBetter);
+    ReportResult("video_ahead_ms",
+                 GetTestCaseName(pair.second.video_stream_label, sync_group),
+                 pair.second.video_ahead_ms, "ms",
+                 webrtc::test::ImproveDirection::kSmallerIsBetter);
+  }
+}
+
+void CrossMediaMetricsReporter::ReportResult(
+    const std::string& metric_name,
+    const std::string& test_case_name,
+    const SamplesStatsCounter& counter,
+    const std::string& unit,
+    webrtc::test::ImproveDirection improve_direction) {
+  test::PrintResult(metric_name, /*modifier=*/"", test_case_name, counter, unit,
+                    /*important=*/false, improve_direction);
+}
+
+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..6ddc994d1f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/cross_media_metrics_reporter.h
@@ -0,0 +1,70 @@
+/*
+ *  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/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"
+#include "test/testsupport/perf_test.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class CrossMediaMetricsReporter
+    : public PeerConnectionE2EQualityTestFixture::QualityMetricsReporter {
+ public:
+  CrossMediaMetricsReporter() = default;
+  ~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;
+
+    std::string audio_stream_label;
+    std::string video_stream_label;
+  };
+
+  static void ReportResult(const std::string& metric_name,
+                           const std::string& test_case_name,
+                           const SamplesStatsCounter& counter,
+                           const std::string& unit,
+                           webrtc::test::ImproveDirection improve_direction =
+                               webrtc::test::ImproveDirection::kNone);
+  std::string GetTestCaseName(const std::string& stream_label,
+                              const std::string& sync_group) const;
+
+  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..f2b4be9e0d
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/echo/echo_emulation.cc
@@ -0,0 +1,115 @@
+/*
+ *  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>
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+constexpr int kSingleBufferDurationMs = 10;
+
+}  // namespace
+
+EchoEmulatingCapturer::EchoEmulatingCapturer(
+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
+    PeerConnectionE2EQualityTestFixture::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..d1d41f63a8
--- /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/peerconnection_quality_test_fixture.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,
+      PeerConnectionE2EQualityTestFixture::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 PeerConnectionE2EQualityTestFixture::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..aacdf7e5d5
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/architecture.md
@@ -0,0 +1,208 @@
+<?% config.freshness.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..ed182d833d
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/default_video_quality_analyzer.md
@@ -0,0 +1,196 @@
+<?% config.freshness.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/g3doc.lua b/third_party/libwebrtc/test/pc/e2e/g3doc/g3doc.lua
new file mode 100644
index 0000000000..981393c826
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/g3doc.lua
@@ -0,0 +1,5 @@
+config = super()
+
+config.freshness.owner = 'titovartem'
+
+return config
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..5a3c9a7e41
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/g3doc/index.md
@@ -0,0 +1,223 @@
+<?% config.freshness.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..22fcc51ffe
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/media/media_helper.cc
@@ -0,0 +1,129 @@
+/*
+ *  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 "test/frame_generator_capturer.h"
+#include "test/platform_video_capturer.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using VideoConfig =
+    ::webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::VideoConfig;
+using AudioConfig =
+    ::webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::AudioConfig;
+using CapturingDeviceIndex = ::webrtc::webrtc_pc_e2e::
+    PeerConnectionE2EQualityTestFixture::CapturingDeviceIndex;
+
+}  // namespace
+
+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();
+  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();
+    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) {
+      RtpParameters rtp_parameters = sender.value()->GetParameters();
+      for (auto& encoding_parameters : rtp_parameters.encodings) {
+        encoding_parameters.num_temporal_layers =
+            video_config.temporal_layers_count;
+      }
+      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,
+    PeerConfigurerImpl::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..4e977e3002
--- /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/peerconnection_quality_test_fixture.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/peer_configurer.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 PeerConnectionE2EQualityTestFixture::VideoConfig& video_config,
+      PeerConfigurerImpl::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/network_quality_metrics_reporter.cc b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
new file mode 100644
index 0000000000..513bdc0a5f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.cc
@@ -0,0 +1,159 @@
+/*
+ *  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 "rtc_base/event.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/testsupport/perf_test.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+constexpr int kStatsWaitTimeoutMs = 1000;
+
+// 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";
+}
+
+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.
+  std::unique_ptr<EmulatedNetworkStats> alice_stats =
+      PopulateStats(alice_network_);
+  RTC_CHECK_EQ(alice_stats->PacketsSent(), 0);
+  RTC_CHECK_EQ(alice_stats->PacketsReceived(), 0);
+  std::unique_ptr<EmulatedNetworkStats> bob_stats = PopulateStats(bob_network_);
+  RTC_CHECK_EQ(bob_stats->PacketsSent(), 0);
+  RTC_CHECK_EQ(bob_stats->PacketsReceived(), 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() {
+  std::unique_ptr<EmulatedNetworkStats> alice_stats =
+      PopulateStats(alice_network_);
+  std::unique_ptr<EmulatedNetworkStats> bob_stats = PopulateStats(bob_network_);
+  int64_t alice_packets_loss =
+      alice_stats->PacketsSent() - bob_stats->PacketsReceived();
+  int64_t bob_packets_loss =
+      bob_stats->PacketsSent() - alice_stats->PacketsReceived();
+  ReportStats("alice", std::move(alice_stats), alice_packets_loss);
+  ReportStats("bob", std::move(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);
+  }
+}
+
+std::unique_ptr<EmulatedNetworkStats>
+NetworkQualityMetricsReporter::PopulateStats(
+    EmulatedNetworkManagerInterface* network) {
+  rtc::Event wait;
+  std::unique_ptr<EmulatedNetworkStats> stats;
+  network->GetStats([&](std::unique_ptr<EmulatedNetworkStats> s) {
+    stats = std::move(s);
+    wait.Set();
+  });
+  bool stats_received = wait.Wait(kStatsWaitTimeoutMs);
+  RTC_CHECK(stats_received);
+  return stats;
+}
+
+void NetworkQualityMetricsReporter::ReportStats(
+    const std::string& network_label,
+    std::unique_ptr<EmulatedNetworkStats> stats,
+    int64_t packet_loss) {
+  ReportResult("bytes_sent", network_label, stats->BytesSent().bytes(),
+               "sizeInBytes");
+  ReportResult("packets_sent", network_label, stats->PacketsSent(), "unitless");
+  ReportResult(
+      "average_send_rate", network_label,
+      stats->PacketsSent() >= 2 ? stats->AverageSendRate().bytes_per_sec() : 0,
+      "bytesPerSecond");
+  ReportResult("bytes_discarded_no_receiver", network_label,
+               stats->BytesDropped().bytes(), "sizeInBytes");
+  ReportResult("packets_discarded_no_receiver", network_label,
+               stats->PacketsDropped(), "unitless");
+  ReportResult("bytes_received", network_label, stats->BytesReceived().bytes(),
+               "sizeInBytes");
+  ReportResult("packets_received", network_label, stats->PacketsReceived(),
+               "unitless");
+  ReportResult("average_receive_rate", network_label,
+               stats->PacketsReceived() >= 2
+                   ? stats->AverageReceiveRate().bytes_per_sec()
+                   : 0,
+               "bytesPerSecond");
+  ReportResult("sent_packets_loss", network_label, packet_loss, "unitless");
+}
+
+void NetworkQualityMetricsReporter::ReportPCStats(const std::string& pc_label,
+                                                  const PCStats& stats) {
+  ReportResult("payload_bytes_received", pc_label,
+               stats.payload_received.bytes(), "sizeInBytes");
+  ReportResult("payload_bytes_sent", pc_label, stats.payload_sent.bytes(),
+               "sizeInBytes");
+}
+
+void NetworkQualityMetricsReporter::ReportResult(
+    const std::string& metric_name,
+    const std::string& network_label,
+    const double value,
+    const std::string& unit) const {
+  test::PrintResult(metric_name, /*modifier=*/"",
+                    GetTestCaseName(network_label), value, unit,
+                    /*important=*/false);
+}
+
+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..5cedce1e7f
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/network_quality_metrics_reporter.h
@@ -0,0 +1,74 @@
+/*
+ *  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/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)
+      : alice_network_(alice_network), bob_network_(bob_network) {}
+  ~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 std::unique_ptr<EmulatedNetworkStats> PopulateStats(
+      EmulatedNetworkManagerInterface* network);
+  void ReportStats(const std::string& network_label,
+                   std::unique_ptr<EmulatedNetworkStats> stats,
+                   int64_t packet_loss);
+  void ReportPCStats(const std::string& pc_label, const PCStats& stats);
+  void ReportResult(const std::string& metric_name,
+                    const std::string& network_label,
+                    double value,
+                    const std::string& unit) const;
+  std::string GetTestCaseName(const std::string& network_label) const;
+
+  std::string test_case_name_;
+
+  EmulatedNetworkManagerInterface* alice_network_;
+  EmulatedNetworkManagerInterface* bob_network_;
+  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_configurer.cc b/third_party/libwebrtc/test/pc/e2e/peer_configurer.cc
new file mode 100644
index 0000000000..998f789cc1
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_configurer.cc
@@ -0,0 +1,179 @@
+/*
+ *  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/peer_configurer.h"
+
+#include <set>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/arraysize.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig;
+using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
+using RunParams = PeerConnectionE2EQualityTestFixture::RunParams;
+using VideoCodecConfig = PeerConnectionE2EQualityTestFixture::VideoCodecConfig;
+
+// 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
+
+DefaultNamesProvider::DefaultNamesProvider(
+    absl::string_view prefix,
+    rtc::ArrayView<const absl::string_view> default_names)
+    : prefix_(prefix), default_names_(default_names) {}
+
+void DefaultNamesProvider::MaybeSetName(absl::optional<std::string>& name) {
+  if (name.has_value()) {
+    known_names_.insert(name.value());
+  } else {
+    name = GenerateName();
+  }
+}
+
+std::string DefaultNamesProvider::GenerateName() {
+  std::string name;
+  do {
+    name = GenerateNameInternal();
+  } while (!known_names_.insert(name).second);
+  return name;
+}
+
+std::string DefaultNamesProvider::GenerateNameInternal() {
+  if (counter_ < default_names_.size()) {
+    return std::string(default_names_[counter_++]);
+  }
+  return prefix_ + std::to_string(counter_++);
+}
+
+PeerParamsPreprocessor::PeerParamsPreprocessor()
+    : peer_names_provider_("peer_", kDefaultNames) {}
+
+void PeerParamsPreprocessor::SetDefaultValuesForMissingParams(
+    PeerConfigurerImpl& 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(
+        PeerConnectionE2EQualityTestFixture::VideoCodecConfig(
+            cricket::kVp8CodecName));
+  }
+}
+
+void PeerParamsPreprocessor::ValidateParams(const PeerConfigurerImpl& 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();
+
+    if (video_config.input_dump_file_name.has_value()) {
+      RTC_CHECK_GT(video_config.input_dump_sampling_modulo, 0)
+          << "video_config.input_dump_sampling_modulo must be greater than 0";
+    }
+    if (video_config.output_dump_file_name.has_value()) {
+      RTC_CHECK_GT(video_config.output_dump_sampling_modulo, 0)
+          << "video_config.input_dump_sampling_modulo must be greater than 0";
+    }
+
+    // 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.simulcast_config->target_spatial_index) {
+        RTC_CHECK_GE(*video_config.simulcast_config->target_spatial_index, 0);
+        RTC_CHECK_LT(*video_config.simulcast_config->target_spatial_index,
+                     video_config.simulcast_config->simulcast_streams_count);
+      }
+      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 (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_configurer.h b/third_party/libwebrtc/test/pc/e2e/peer_configurer.h
new file mode 100644
index 0000000000..c8239c08d4
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_configurer.h
@@ -0,0 +1,332 @@
+/*
+ *  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_CONFIGURER_H_
+#define TEST_PC_E2E_PEER_CONFIGURER_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/async_resolver_factory.h"
+#include "api/audio/audio_mixer.h"
+#include "api/call/call_factory_interface.h"
+#include "api/fec_controller.h"
+#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/create_peer_connection_quality_test_frame_generator.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/transport/network_control.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/network.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/thread.h"
+#include "test/pc/e2e/peer_connection_quality_test_params.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+class PeerConfigurerImpl final
+    : public PeerConnectionE2EQualityTestFixture::PeerConfigurer {
+ public:
+  using VideoSource =
+      absl::variant<std::unique_ptr<test::FrameGeneratorInterface>,
+                    PeerConnectionE2EQualityTestFixture::CapturingDeviceIndex>;
+
+  PeerConfigurerImpl(rtc::Thread* network_thread,
+                     rtc::NetworkManager* network_manager,
+                     rtc::PacketSocketFactory* packet_socket_factory)
+      : components_(
+            std::make_unique<InjectableComponents>(network_thread,
+                                                   network_manager,
+                                                   packet_socket_factory)),
+        params_(std::make_unique<Params>()),
+        configurable_params_(std::make_unique<ConfigurableParams>()) {}
+
+  PeerConfigurer* SetName(absl::string_view name) override {
+    params_->name = std::string(name);
+    return this;
+  }
+
+  // Implementation of PeerConnectionE2EQualityTestFixture::PeerConfigurer.
+  PeerConfigurer* SetTaskQueueFactory(
+      std::unique_ptr<TaskQueueFactory> task_queue_factory) override {
+    components_->pcf_dependencies->task_queue_factory =
+        std::move(task_queue_factory);
+    return this;
+  }
+  PeerConfigurer* SetCallFactory(
+      std::unique_ptr<CallFactoryInterface> call_factory) override {
+    components_->pcf_dependencies->call_factory = std::move(call_factory);
+    return this;
+  }
+  PeerConfigurer* SetEventLogFactory(
+      std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory) override {
+    components_->pcf_dependencies->event_log_factory =
+        std::move(event_log_factory);
+    return this;
+  }
+  PeerConfigurer* SetFecControllerFactory(
+      std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory)
+      override {
+    components_->pcf_dependencies->fec_controller_factory =
+        std::move(fec_controller_factory);
+    return this;
+  }
+  PeerConfigurer* SetNetworkControllerFactory(
+      std::unique_ptr<NetworkControllerFactoryInterface>
+          network_controller_factory) override {
+    components_->pcf_dependencies->network_controller_factory =
+        std::move(network_controller_factory);
+    return this;
+  }
+  PeerConfigurer* SetVideoEncoderFactory(
+      std::unique_ptr<VideoEncoderFactory> video_encoder_factory) override {
+    components_->pcf_dependencies->video_encoder_factory =
+        std::move(video_encoder_factory);
+    return this;
+  }
+  PeerConfigurer* SetVideoDecoderFactory(
+      std::unique_ptr<VideoDecoderFactory> video_decoder_factory) override {
+    components_->pcf_dependencies->video_decoder_factory =
+        std::move(video_decoder_factory);
+    return this;
+  }
+
+  PeerConfigurer* SetAsyncResolverFactory(
+      std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory)
+      override {
+    components_->pc_dependencies->async_resolver_factory =
+        std::move(async_resolver_factory);
+    return this;
+  }
+  PeerConfigurer* SetRTCCertificateGenerator(
+      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator)
+      override {
+    components_->pc_dependencies->cert_generator = std::move(cert_generator);
+    return this;
+  }
+  PeerConfigurer* SetSSLCertificateVerifier(
+      std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier) override {
+    components_->pc_dependencies->tls_cert_verifier =
+        std::move(tls_cert_verifier);
+    return this;
+  }
+
+  PeerConfigurer* AddVideoConfig(
+      PeerConnectionE2EQualityTestFixture::VideoConfig config) override {
+    video_sources_.push_back(
+        CreateSquareFrameGenerator(config, /*type=*/absl::nullopt));
+    configurable_params_->video_configs.push_back(std::move(config));
+    return this;
+  }
+  PeerConfigurer* AddVideoConfig(
+      PeerConnectionE2EQualityTestFixture::VideoConfig config,
+      std::unique_ptr<test::FrameGeneratorInterface> generator) override {
+    configurable_params_->video_configs.push_back(std::move(config));
+    video_sources_.push_back(std::move(generator));
+    return this;
+  }
+  PeerConfigurer* AddVideoConfig(
+      PeerConnectionE2EQualityTestFixture::VideoConfig config,
+      PeerConnectionE2EQualityTestFixture::CapturingDeviceIndex index)
+      override {
+    configurable_params_->video_configs.push_back(std::move(config));
+    video_sources_.push_back(index);
+    return this;
+  }
+  PeerConfigurer* SetVideoSubscription(
+      PeerConnectionE2EQualityTestFixture::VideoSubscription subscription)
+      override {
+    configurable_params_->video_subscription = std::move(subscription);
+    return this;
+  }
+  PeerConfigurer* SetAudioConfig(
+      PeerConnectionE2EQualityTestFixture::AudioConfig config) override {
+    params_->audio_config = std::move(config);
+    return this;
+  }
+  PeerConfigurer* SetUseUlpFEC(bool value) override {
+    params_->use_ulp_fec = value;
+    return this;
+  }
+  PeerConfigurer* SetUseFlexFEC(bool value) override {
+    params_->use_flex_fec = value;
+    return this;
+  }
+  PeerConfigurer* SetVideoEncoderBitrateMultiplier(double multiplier) override {
+    params_->video_encoder_bitrate_multiplier = multiplier;
+    return this;
+  }
+  PeerConfigurer* SetNetEqFactory(
+      std::unique_ptr<NetEqFactory> neteq_factory) override {
+    components_->pcf_dependencies->neteq_factory = std::move(neteq_factory);
+    return this;
+  }
+  PeerConfigurer* SetAudioProcessing(
+      rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) override {
+    components_->pcf_dependencies->audio_processing = audio_processing;
+    return this;
+  }
+  PeerConfigurer* SetAudioMixer(
+      rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) override {
+    components_->pcf_dependencies->audio_mixer = audio_mixer;
+    return this;
+  }
+  PeerConfigurer* SetRtcEventLogPath(std::string path) override {
+    params_->rtc_event_log_path = std::move(path);
+    return this;
+  }
+  PeerConfigurer* SetAecDumpPath(std::string path) override {
+    params_->aec_dump_path = std::move(path);
+    return this;
+  }
+  PeerConfigurer* SetRTCConfiguration(
+      PeerConnectionInterface::RTCConfiguration configuration) override {
+    params_->rtc_configuration = std::move(configuration);
+    return this;
+  }
+  PeerConfigurer* SetRTCOfferAnswerOptions(
+      PeerConnectionInterface::RTCOfferAnswerOptions options) override {
+    params_->rtc_offer_answer_options = std::move(options);
+    return this;
+  }
+  PeerConfigurer* SetBitrateSettings(
+      BitrateSettings bitrate_settings) override {
+    params_->bitrate_settings = bitrate_settings;
+    return this;
+  }
+  PeerConfigurer* SetVideoCodecs(
+      std::vector<PeerConnectionE2EQualityTestFixture::VideoCodecConfig>
+          video_codecs) override {
+    params_->video_codecs = std::move(video_codecs);
+    return this;
+  }
+
+  PeerConfigurer* SetIceTransportFactory(
+      std::unique_ptr<IceTransportFactory> factory) override {
+    components_->pc_dependencies->ice_transport_factory = std::move(factory);
+    return this;
+  }
+
+  PeerConfigurer* SetPortAllocatorExtraFlags(uint32_t extra_flags) override {
+    params_->port_allocator_extra_flags = extra_flags;
+    return this;
+  }
+  // Implementation of PeerConnectionE2EQualityTestFixture::PeerConfigurer end.
+
+  InjectableComponents* components() { return components_.get(); }
+  Params* params() { return params_.get(); }
+  ConfigurableParams* configurable_params() {
+    return configurable_params_.get();
+  }
+  const Params& params() const { return *params_; }
+  const ConfigurableParams& configurable_params() const {
+    return *configurable_params_;
+  }
+  std::vector<VideoSource>* video_sources() { return &video_sources_; }
+
+  // Returns InjectableComponents and transfer ownership to the caller.
+  // Can be called once.
+  std::unique_ptr<InjectableComponents> ReleaseComponents() {
+    RTC_CHECK(components_);
+    auto components = std::move(components_);
+    components_ = nullptr;
+    return components;
+  }
+
+  // Returns Params and transfer ownership to the caller.
+  // Can be called once.
+  std::unique_ptr<Params> ReleaseParams() {
+    RTC_CHECK(params_);
+    auto params = std::move(params_);
+    params_ = nullptr;
+    return params;
+  }
+
+  // Returns ConfigurableParams and transfer ownership to the caller.
+  // Can be called once.
+  std::unique_ptr<ConfigurableParams> ReleaseConfigurableParams() {
+    RTC_CHECK(configurable_params_);
+    auto configurable_params = std::move(configurable_params_);
+    configurable_params_ = nullptr;
+    return configurable_params;
+  }
+
+  // Returns video sources and transfer frame generators ownership to the
+  // caller. Can be called once.
+  std::vector<VideoSource> ReleaseVideoSources() {
+    auto video_sources = std::move(video_sources_);
+    video_sources_.clear();
+    return video_sources;
+  }
+
+ private:
+  std::unique_ptr<InjectableComponents> components_;
+  std::unique_ptr<Params> params_;
+  std::unique_ptr<ConfigurableParams> configurable_params_;
+  std::vector<VideoSource> video_sources_;
+};
+
+class 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 = {});
+
+  void MaybeSetName(absl::optional<std::string>& name);
+
+ private:
+  std::string GenerateName();
+
+  std::string GenerateNameInternal();
+
+  const std::string prefix_;
+  const rtc::ArrayView<const absl::string_view> default_names_;
+
+  std::set<std::string> known_names_;
+  size_t counter_ = 0;
+};
+
+class PeerParamsPreprocessor {
+ public:
+  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(PeerConfigurerImpl& peer);
+
+  // Validate peer's parameters, also ensure uniqueness of all video stream
+  // labels.
+  void ValidateParams(const PeerConfigurerImpl& peer);
+
+ private:
+  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_CONFIGURER_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..4ee6590419
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_e2e_smoke_test.cc
@@ -0,0 +1,472 @@
+/*
+ *  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 "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/network_emulation_manager.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:
+  using PeerConfigurer = PeerConnectionE2EQualityTestFixture::PeerConfigurer;
+  using RunParams = PeerConnectionE2EQualityTestFixture::RunParams;
+  using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
+  using VideoCodecConfig =
+      PeerConnectionE2EQualityTestFixture::VideoCodecConfig;
+  using AudioConfig = PeerConnectionE2EQualityTestFixture::AudioConfig;
+  using ScreenShareConfig =
+      PeerConnectionE2EQualityTestFixture::ScreenShareConfig;
+  using ScrollingParams = PeerConnectionE2EQualityTestFixture::ScrollingParams;
+  using VideoSimulcastConfig =
+      PeerConnectionE2EQualityTestFixture::VideoSimulcastConfig;
+  using EchoEmulationConfig =
+      PeerConnectionE2EQualityTestFixture::EchoEmulationConfig;
+
+  void SetUp() override {
+    network_emulation_ = CreateNetworkEmulationManager();
+    auto video_quality_analyzer = std::make_unique<DefaultVideoQualityAnalyzer>(
+        network_emulation_->time_controller()->GetClock());
+    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*)> configurer) {
+    fixture_->AddPeer(network->network_dependencies(), 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()));
+  RunParams run_params(TimeDelta::Seconds(2));
+  run_params.enable_flex_fec_support = true;
+  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()));
+  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()));
+
+  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(
+        TimeDelta::Millis(1800), kDefaultSlidesWidth, kDefaultSlidesHeight);
+    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, 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, 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..89d5da0d42
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.cc
@@ -0,0 +1,737 @@
+/*
+ *  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/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/stats_poller.h"
+#include "test/pc/e2e/test_peer_factory.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/perf_test.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
+using VideoCodecConfig = PeerConnectionE2EQualityTestFixture::VideoCodecConfig;
+
+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<PeerConfigurerImpl>>& 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)
+    : 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())) {
+  // 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());
+  }
+  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>();
+  }
+  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(
+    const PeerNetworkDependencies& network_dependencies,
+    rtc::FunctionView<void(PeerConfigurer*)> configurer) {
+  peer_configurations_.push_back(std::make_unique<PeerConfigurerImpl>(
+      network_dependencies.network_thread, network_dependencies.network_manager,
+      network_dependencies.packet_socket_factory));
+  configurer(peer_configurations_.back().get());
+  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<PeerConfigurerImpl> alice_configurer =
+      std::move(peer_configurations_[0]);
+  std::unique_ptr<PeerConfigurerImpl> 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 and names to correctly pass them into
+  // lambdas.
+  std::vector<VideoConfig> alice_video_configs =
+      alice_configurer->configurable_params()->video_configs;
+  std::string alice_name = alice_configurer->params()->name.value();
+  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, bob_video_configs, alice_name](
+              rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+            OnTrackCallback(alice_name, 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, alice_video_configs,
+           bob_name](rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
+            OnTrackCallback(bob_name, 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()));
+  quality_metrics_reporters_.push_back(
+      std::make_unique<CrossMediaMetricsReporter>());
+
+  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,
+    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(), 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);
+  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() {
+  test::PrintResult(*alice_->params().name + "_connected", "", test_case_name_,
+                    alice_connected_, "unitless",
+                    /*important=*/false,
+                    test::ImproveDirection::kBiggerIsBetter);
+  test::PrintResult(*bob_->params().name + "_connected", "", test_case_name_,
+                    bob_connected_, "unitless",
+                    /*important=*/false,
+                    test::ImproveDirection::kBiggerIsBetter);
+}
+
+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..7738a83329
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test.h
@@ -0,0 +1,152 @@
+/*
+ *  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/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/peer_configurer.h"
+#include "test/pc/e2e/peer_connection_quality_test_params.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 RunParams = PeerConnectionE2EQualityTestFixture::RunParams;
+  using VideoConfig = PeerConnectionE2EQualityTestFixture::VideoConfig;
+  using VideoSimulcastConfig =
+      PeerConnectionE2EQualityTestFixture::VideoSimulcastConfig;
+  using PeerConfigurer = PeerConnectionE2EQualityTestFixture::PeerConfigurer;
+  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() 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(
+      const PeerNetworkDependencies& network_dependencies,
+      rtc::FunctionView<void(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,
+                       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_;
+
+  std::vector<std::unique_ptr<PeerConfigurerImpl>> 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_params.h b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_params.h
new file mode 100644
index 0000000000..645ed7a290
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/peer_connection_quality_test_params.h
@@ -0,0 +1,159 @@
+/*
+ *  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_PARAMS_H_
+#define TEST_PC_E2E_PEER_CONNECTION_QUALITY_TEST_PARAMS_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/async_resolver_factory.h"
+#include "api/audio/audio_mixer.h"
+#include "api/call/call_factory_interface.h"
+#include "api/fec_controller.h"
+#include "api/field_trials_view.h"
+#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/transport/network_control.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "p2p/base/port_allocator.h"
+#include "rtc_base/network.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Contains most part from PeerConnectionFactoryDependencies. Also all fields
+// are optional and defaults will be provided by fixture implementation if
+// any will be omitted.
+//
+// Separate class was introduced to clarify which components can be
+// overridden. For example worker and signaling threads will be provided by
+// fixture implementation. The same is applicable to the media engine. So user
+// can override only some parts of media engine like video encoder/decoder
+// factories.
+struct PeerConnectionFactoryComponents {
+  std::unique_ptr<TaskQueueFactory> task_queue_factory;
+  std::unique_ptr<CallFactoryInterface> call_factory;
+  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
+  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
+  std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
+  std::unique_ptr<NetEqFactory> neteq_factory;
+
+  // Will be passed to MediaEngineInterface, that will be used in
+  // PeerConnectionFactory.
+  std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
+  std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
+
+  std::unique_ptr<FieldTrialsView> trials;
+
+  rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
+  rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer;
+};
+
+// Contains most parts from PeerConnectionDependencies. Also all fields are
+// optional and defaults will be provided by fixture implementation if any
+// will be omitted.
+//
+// Separate class was introduced to clarify which components can be
+// overridden. For example observer, which is required to
+// PeerConnectionDependencies, will be provided by fixture implementation,
+// so client can't inject its own. Also only network manager can be overridden
+// inside port allocator.
+struct PeerConnectionComponents {
+  PeerConnectionComponents(rtc::NetworkManager* network_manager,
+                           rtc::PacketSocketFactory* packet_socket_factory)
+      : network_manager(network_manager),
+        packet_socket_factory(packet_socket_factory) {
+    RTC_CHECK(network_manager);
+  }
+
+  rtc::NetworkManager* const network_manager;
+  rtc::PacketSocketFactory* const packet_socket_factory;
+  std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory;
+  std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
+  std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
+  std::unique_ptr<IceTransportFactory> ice_transport_factory;
+};
+
+// Contains all components, that can be overridden in peer connection. Also
+// has a network thread, that will be used to communicate with another peers.
+struct InjectableComponents {
+  InjectableComponents(rtc::Thread* network_thread,
+                       rtc::NetworkManager* network_manager,
+                       rtc::PacketSocketFactory* packet_socket_factory)
+      : network_thread(network_thread),
+        pcf_dependencies(std::make_unique<PeerConnectionFactoryComponents>()),
+        pc_dependencies(
+            std::make_unique<PeerConnectionComponents>(network_manager,
+                                                       packet_socket_factory)) {
+    RTC_CHECK(network_thread);
+  }
+
+  rtc::Thread* const network_thread;
+
+  std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies;
+  std::unique_ptr<PeerConnectionComponents> pc_dependencies;
+};
+
+// Contains information about call media streams (up to 1 audio stream and
+// unlimited amount of video streams) and rtc configuration, that will be used
+// to set up peer connection.
+struct Params {
+  // Peer name. If empty - default one will be set by the fixture.
+  absl::optional<std::string> name;
+  // If `audio_config` is set audio stream will be configured
+  absl::optional<PeerConnectionE2EQualityTestFixture::AudioConfig> audio_config;
+  // Flags to set on `cricket::PortAllocator`. These flags will be added
+  // to the default ones that are presented on the port allocator.
+  uint32_t port_allocator_extra_flags = cricket::kDefaultPortAllocatorFlags;
+  // If `rtc_event_log_path` is set, an RTCEventLog will be saved in that
+  // location and it will be available for further analysis.
+  absl::optional<std::string> rtc_event_log_path;
+  // If `aec_dump_path` is set, an AEC dump will be saved in that location and
+  // it will be available for further analysis.
+  absl::optional<std::string> aec_dump_path;
+
+  bool use_ulp_fec = false;
+  bool use_flex_fec = false;
+  // Specifies how much video encoder target bitrate should be different than
+  // target bitrate, provided by WebRTC stack. Must be greater then 0. Can be
+  // used to emulate overshooting of video encoders. This multiplier will
+  // be applied for all video encoder on both sides for all layers. Bitrate
+  // estimated by WebRTC stack will be multiplied by this multiplier and then
+  // provided into VideoEncoder::SetRates(...).
+  double video_encoder_bitrate_multiplier = 1.0;
+
+  PeerConnectionInterface::RTCConfiguration rtc_configuration;
+  PeerConnectionInterface::RTCOfferAnswerOptions rtc_offer_answer_options;
+  BitrateSettings bitrate_settings;
+  std::vector<PeerConnectionE2EQualityTestFixture::VideoCodecConfig>
+      video_codecs;
+};
+
+// Contains parameters that maybe changed by test writer during the test call.
+struct ConfigurableParams {
+  // If `video_configs` is empty - no video should be added to the test call.
+  std::vector<PeerConnectionE2EQualityTestFixture::VideoConfig> video_configs;
+
+  PeerConnectionE2EQualityTestFixture::VideoSubscription video_subscription =
+      PeerConnectionE2EQualityTestFixture::VideoSubscription()
+          .SubscribeToAllPeers();
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // TEST_PC_E2E_PEER_CONNECTION_QUALITY_TEST_PARAMS_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..b3ee3b78bb
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.cc
@@ -0,0 +1,602 @@
+/*
+ *  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 "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 {
+
+using VideoCodecConfig = PeerConnectionE2EQualityTestFixture::VideoCodecConfig;
+
+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 PeerConnectionE2EQualityTestFixture::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 PeerConnectionE2EQualityTestFixture::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..115ed5ba2c
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/sdp/sdp_changer.h
@@ -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.
+ */
+
+#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/peerconnection_quality_test_fixture.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 PeerConnectionE2EQualityTestFixture::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 PeerConnectionE2EQualityTestFixture::VideoCodecConfig& first_codec);
+  LocalAndRemoteSdp PatchAnswer(
+      std::unique_ptr<SessionDescriptionInterface> answer,
+      const PeerConnectionE2EQualityTestFixture::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..94a46fbae5
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
@@ -0,0 +1,385 @@
+/*
+ *  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/stats/rtc_stats.h"
+#include "api/stats/rtcstats_objects.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/event.h"
+#include "rtc_base/ip_address.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/testsupport/perf_test.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+constexpr int kStatsWaitTimeoutMs = 1000;
+
+// 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";
+
+std::unique_ptr<EmulatedNetworkStats> PopulateStats(
+    std::vector<EmulatedEndpoint*> endpoints,
+    NetworkEmulationManager* network_emulation) {
+  rtc::Event stats_loaded;
+  std::unique_ptr<EmulatedNetworkStats> stats;
+  network_emulation->GetStats(endpoints,
+                              [&](std::unique_ptr<EmulatedNetworkStats> s) {
+                                stats = std::move(s);
+                                stats_loaded.Set();
+                              });
+  bool stats_received = stats_loaded.Wait(kStatsWaitTimeoutMs);
+  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;
+}
+
+}  // namespace
+
+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_) {
+    std::unique_ptr<EmulatedNetworkStats> stats =
+        PopulateStats(entry.second, network_emulation_);
+    RTC_CHECK_EQ(stats->PacketsSent(), 0);
+    RTC_CHECK_EQ(stats->PacketsReceived(), 0);
+  }
+}
+
+void StatsBasedNetworkQualityMetricsReporter::NetworkLayerStatsCollector::
+    AddPeer(absl::string_view peer_name,
+            std::vector<EmulatedEndpoint*> endpoints) {
+  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));
+  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,
+         StatsBasedNetworkQualityMetricsReporter::NetworkLayerStats>
+StatsBasedNetworkQualityMetricsReporter::NetworkLayerStatsCollector::
+    GetStats() {
+  MutexLock lock(&mutex_);
+  std::map<std::string, NetworkLayerStats> peer_to_stats;
+  std::map<std::string, std::vector<std::string>> sender_to_receivers;
+  for (const auto& entry : peer_endpoints_) {
+    NetworkLayerStats stats;
+    stats.stats = PopulateStats(entry.second, network_emulation_);
+    const std::string& peer_name = entry.first;
+    for (const auto& income_stats_entry :
+         stats.stats->IncomingStatsPerSource()) {
+      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);
+    }
+    peer_to_stats.emplace(peer_name, std::move(stats));
+  }
+  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));
+}
+
+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) {
+  ReportResult("bytes_discarded_no_receiver", pc_label,
+               network_layer_stats.stats->BytesDropped().bytes(),
+               "sizeInBytes");
+  ReportResult("packets_discarded_no_receiver", pc_label,
+               network_layer_stats.stats->PacketsDropped(), "unitless");
+
+  ReportResult("payload_bytes_received", pc_label,
+               pc_stats.payload_received.bytes(), "sizeInBytes");
+  ReportResult("payload_bytes_sent", pc_label, pc_stats.payload_sent.bytes(),
+               "sizeInBytes");
+
+  ReportResult("bytes_sent", pc_label, pc_stats.total_sent.bytes(),
+               "sizeInBytes");
+  ReportResult("packets_sent", pc_label, pc_stats.packets_sent, "unitless");
+  ReportResult("average_send_rate", pc_label,
+               (pc_stats.total_sent / (end_time - start_time_)).bytes_per_sec(),
+               "bytesPerSecond");
+  ReportResult("bytes_received", pc_label, pc_stats.total_received.bytes(),
+               "sizeInBytes");
+  ReportResult("packets_received", pc_label, pc_stats.packets_received,
+               "unitless");
+  ReportResult(
+      "average_receive_rate", pc_label,
+      (pc_stats.total_received / (end_time - start_time_)).bytes_per_sec(),
+      "bytesPerSecond");
+  ReportResult("sent_packets_loss", pc_label, packet_loss, "unitless");
+}
+
+void StatsBasedNetworkQualityMetricsReporter::ReportResult(
+    const std::string& metric_name,
+    const std::string& network_label,
+    const double value,
+    const std::string& unit) const {
+  test::PrintResult(metric_name, /*modifier=*/"",
+                    GetTestCaseName(network_label), value, unit,
+                    /*important=*/false);
+}
+
+void StatsBasedNetworkQualityMetricsReporter::ReportResult(
+    const std::string& metric_name,
+    const std::string& network_label,
+    const SamplesStatsCounter& value,
+    const std::string& unit) const {
+  test::PrintResult(metric_name, /*modifier=*/"",
+                    GetTestCaseName(network_label), value, unit,
+                    /*important=*/false);
+}
+
+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.stats->PacketsSent() >= 2
+                                   ? stats.stats->AverageSendRate()
+                                   : DataRate::Zero();
+  DataRate average_receive_rate = stats.stats->PacketsReceived() >= 2
+                                      ? stats.stats->AverageReceiveRate()
+                                      : DataRate::Zero();
+  rtc::StringBuilder log;
+  log << "Raw network layer statistic for [" << peer_name << "]:\n"
+      << "Local IPs:\n";
+  std::vector<rtc::IPAddress> local_ips = stats.stats->LocalAddresses();
+  for (size_t i = 0; i < local_ips.size(); ++i) {
+    log << "  " << local_ips[i].ToString() << "\n";
+  }
+  if (!stats.stats->SentPacketsSizeCounter().IsEmpty()) {
+    ReportResult("sent_packets_size", peer_name,
+                 stats.stats->SentPacketsSizeCounter(), "sizeInBytes");
+  }
+  if (!stats.stats->ReceivedPacketsSizeCounter().IsEmpty()) {
+    ReportResult("received_packets_size", peer_name,
+                 stats.stats->ReceivedPacketsSizeCounter(), "sizeInBytes");
+  }
+  if (!stats.stats->DroppedPacketsSizeCounter().IsEmpty()) {
+    ReportResult("dropped_packets_size", peer_name,
+                 stats.stats->DroppedPacketsSizeCounter(), "sizeInBytes");
+  }
+  if (!stats.stats->SentPacketsQueueWaitTimeUs().IsEmpty()) {
+    ReportResult("sent_packets_queue_wait_time_us", peer_name,
+                 stats.stats->SentPacketsQueueWaitTimeUs(), "unitless");
+  }
+
+  log << "Send statistic:\n"
+      << "  packets: " << stats.stats->PacketsSent()
+      << " bytes: " << stats.stats->BytesSent().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.stats->OutgoingStatsPerDestination()) {
+    DataRate source_average_send_rate = entry.second->PacketsSent() >= 2
+                                            ? entry.second->AverageSendRate()
+                                            : DataRate::Zero();
+    log << "(" << entry.first.ToString() << "):\n"
+        << "  packets: " << entry.second->PacketsSent()
+        << " bytes: " << entry.second->BytesSent().bytes()
+        << " avg_rate (bytes/sec): " << source_average_send_rate.bytes_per_sec()
+        << " avg_rate (bps): " << source_average_send_rate.bps() << "\n";
+    if (!entry.second->SentPacketsSizeCounter().IsEmpty()) {
+      ReportResult("sent_packets_size",
+                   peer_name + "/" + entry.first.ToString(),
+                   stats.stats->SentPacketsSizeCounter(), "sizeInBytes");
+    }
+  }
+
+  log << "Receive statistic:\n"
+      << "  packets: " << stats.stats->PacketsReceived()
+      << " bytes: " << stats.stats->BytesReceived().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.stats->IncomingStatsPerSource()) {
+    DataRate source_average_receive_rate =
+        entry.second->PacketsReceived() >= 2
+            ? entry.second->AverageReceiveRate()
+            : DataRate::Zero();
+    log << "(" << entry.first.ToString() << "):\n"
+        << "  packets: " << entry.second->PacketsReceived()
+        << " bytes: " << entry.second->BytesReceived().bytes()
+        << " avg_rate (bytes/sec): "
+        << source_average_receive_rate.bytes_per_sec()
+        << " avg_rate (bps): " << source_average_receive_rate.bps() << "\n";
+    if (!entry.second->ReceivedPacketsSizeCounter().IsEmpty()) {
+      ReportResult("received_packets_size",
+                   peer_name + "/" + entry.first.ToString(),
+                   stats.stats->ReceivedPacketsSizeCounter(), "sizeInBytes");
+    }
+    if (!entry.second->DroppedPacketsSizeCounter().IsEmpty()) {
+      ReportResult("dropped_packets_size",
+                   peer_name + "/" + entry.first.ToString(),
+                   stats.stats->DroppedPacketsSizeCounter(), "sizeInBytes");
+    }
+  }
+
+  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..a940d81df8
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/stats_based_network_quality_metrics_reporter.h
@@ -0,0 +1,130 @@
+/*
+ *  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/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:
+  // `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)
+      : collector_(std::move(peer_endpoints), network_emulation),
+        clock_(network_emulation->time_controller()->GetClock()) {}
+  ~StatsBasedNetworkQualityMetricsReporter() override = default;
+
+  void AddPeer(absl::string_view peer_name,
+               std::vector<EmulatedEndpoint*> endpoints);
+
+  // 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;
+  };
+
+  struct NetworkLayerStats {
+    std::unique_ptr<EmulatedNetworkStats> stats;
+    std::set<std::string> receivers;
+  };
+
+  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::map<std::string, NetworkLayerStats> GetStats();
+
+   private:
+    Mutex mutex_;
+    std::map<std::string, std::vector<EmulatedEndpoint*>> peer_endpoints_
+        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);
+  void ReportResult(const std::string& metric_name,
+                    const std::string& network_label,
+                    double value,
+                    const std::string& unit) const;
+  void ReportResult(const std::string& metric_name,
+                    const std::string& network_label,
+                    const SamplesStatsCounter& value,
+                    const std::string& unit) const;
+  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_;
+
+  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_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..d978f10665
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer.cc
@@ -0,0 +1,154 @@
+/*
+ *  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 "modules/audio_processing/include/audio_processing.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using VideoSubscription = ::webrtc::webrtc_pc_e2e::
+    PeerConnectionE2EQualityTestFixture::VideoSubscription;
+using VideoConfig =
+    ::webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::VideoConfig;
+
+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<PeerConfigurerImpl::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..741e2e73f1
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer.h
@@ -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.
+ */
+
+#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/peerconnection_quality_test_fixture.h"
+#include "pc/peer_connection_wrapper.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "test/pc/e2e/peer_configurer.h"
+#include "test/pc/e2e/peer_connection_quality_test_params.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(PeerConnectionE2EQualityTestFixture::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(
+      PeerConnectionE2EQualityTestFixture::VideoSubscription subscription);
+
+  void GetStats(RTCStatsCollectorCallback* callback) override;
+
+  PeerConfigurerImpl::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<PeerConfigurerImpl::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_`.
+  std::unique_ptr<rtc::Thread> worker_thread_;
+  std::unique_ptr<PeerConnectionWrapper> wrapper_;
+  std::vector<PeerConfigurerImpl::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..01388d90ea
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.cc
@@ -0,0 +1,376 @@
+/*
+ *  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/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 "test/pc/e2e/analyzer/video/quality_analyzing_video_encoder.h"
+#include "test/pc/e2e/echo/echo_emulation.h"
+#include "test/pc/e2e/peer_configurer.h"
+#include "test/testsupport/copy_to_file_audio_capturer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+using AudioConfig =
+    ::webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::AudioConfig;
+using VideoConfig =
+    ::webrtc::webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::VideoConfig;
+using EchoEmulationConfig = ::webrtc::webrtc_pc_e2e::
+    PeerConnectionE2EQualityTestFixture::EchoEmulationConfig;
+
+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 "Foo" isn't simulcast/SVC stream
+// 2. `kAnalyzeAnySpatialStream` means all simulcast/SVC streams are required
+// 3. Concrete value means that particular simulcast/SVC stream have to be
+//    analyzed.
+std::map<std::string, absl::optional<int>>
+CalculateRequiredSpatialIndexPerStream(
+    const std::vector<VideoConfig>& video_configs) {
+  std::map<std::string, absl::optional<int>> out;
+  for (auto& video_config : video_configs) {
+    // Stream label should be set by fixture implementation here.
+    RTC_DCHECK(video_config.stream_label);
+    absl::optional<int> spatial_index;
+    if (video_config.simulcast_config) {
+      spatial_index = video_config.simulcast_config->target_spatial_index;
+      if (!spatial_index) {
+        spatial_index = kAnalyzeAnySpatialStream;
+      }
+    }
+    bool res = out.insert({*video_config.stream_label, spatial_index}).second;
+    RTC_DCHECK(res) << "Duplicate video_config.stream_label="
+                    << *video_config.stream_label;
+  }
+  return out;
+}
+
+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,
+    std::map<std::string, absl::optional<int>> stream_required_spatial_index,
+    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_required_spatial_index));
+}
+
+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<PeerConfigurerImpl> configurer,
+    std::unique_ptr<MockPeerConnectionObserver> observer,
+    absl::optional<RemotePeerAudioConfig> remote_audio_config,
+    absl::optional<PeerConnectionE2EQualityTestFixture::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<PeerConfigurerImpl::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> worker_thread =
+      time_controller_.CreateThread("worker_thread");
+  // 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_, worker_thread.get(), 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(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..8d78e2f8d9
--- /dev/null
+++ b/third_party/libwebrtc/test/pc/e2e/test_peer_factory.h
@@ -0,0 +1,86 @@
+/*
+ *  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/peerconnection_quality_test_fixture.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/peer_configurer.h"
+#include "test/pc/e2e/peer_connection_quality_test_params.h"
+#include "test/pc/e2e/test_peer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+struct RemotePeerAudioConfig {
+  explicit RemotePeerAudioConfig(
+      PeerConnectionE2EQualityTestFixture::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<PeerConnectionE2EQualityTestFixture::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<PeerConfigurerImpl> configurer,
+      std::unique_ptr<MockPeerConnectionObserver> observer,
+      absl::optional<RemotePeerAudioConfig> remote_audio_config,
+      absl::optional<PeerConnectionE2EQualityTestFixture::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_