Adding upstream version 115.7.0esr.upstream/115.7.0esr

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

98 files changed, 20595 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/congestion_controller/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/BUILD.gn
new file mode 100644
index 0000000000..33f5508137
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/BUILD.gn
@@ -0,0 +1,68 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+config("bwe_test_logging") {
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+}
+
+rtc_library("congestion_controller") {
+  visibility = [ "*" ]
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "include/receive_side_congestion_controller.h",
+    "receive_side_congestion_controller.cc",
+    "remb_throttler.cc",
+    "remb_throttler.h",
+  ]
+
+  deps = [
+    "../../api/transport:network_control",
+    "../../api/units:data_rate",
+    "../../api/units:time_delta",
+    "../../api/units:timestamp",
+    "../../rtc_base:logging",
+    "../../rtc_base:macromagic",
+    "../../rtc_base/synchronization:mutex",
+    "../pacing",
+    "../remote_bitrate_estimator",
+    "../rtp_rtcp:rtp_rtcp_format",
+  ]
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  rtc_library("congestion_controller_unittests") {
+    testonly = true
+
+    sources = [
+      "receive_side_congestion_controller_unittest.cc",
+      "remb_throttler_unittest.cc",
+    ]
+    deps = [
+      ":congestion_controller",
+      "../../api/test/network_emulation",
+      "../../api/test/network_emulation:create_cross_traffic",
+      "../../api/units:data_rate",
+      "../../api/units:time_delta",
+      "../../api/units:timestamp",
+      "../../system_wrappers",
+      "../../test:test_support",
+      "../../test/scenario",
+      "../pacing",
+      "goog_cc:estimators",
+      "goog_cc:goog_cc_unittests",
+      "pcc:pcc_unittests",
+      "rtp:congestion_controller_unittests",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/DEPS b/third_party/libwebrtc/modules/congestion_controller/DEPS
new file mode 100644
index 0000000000..2ed9952e22
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/DEPS
@@ -0,0 +1,5 @@
+include_rules = [
+  "+logging/rtc_event_log",
+  "+system_wrappers",
+  "+video",
+]
diff --git a/third_party/libwebrtc/modules/congestion_controller/OWNERS b/third_party/libwebrtc/modules/congestion_controller/OWNERS
new file mode 100644
index 0000000000..9a836bad06
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/OWNERS
@@ -0,0 +1,7 @@
+danilchap@webrtc.org
+linderborg@webrtc.org
+stefan@webrtc.org
+terelius@webrtc.org
+mflodman@webrtc.org
+yinwa@webrtc.org
+perkj@webrtc.org
diff --git a/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build
new file mode 100644
index 0000000000..9633f1ef48
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/congestion_controller_gn/moz.build
@@ -0,0 +1,234 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("congestion_controller_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/goog_cc/BUILD.gn
new file mode 100644
index 0000000000..150201e1bd
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/BUILD.gn
@@ -0,0 +1,369 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+config("bwe_test_logging") {
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+}
+
+rtc_library("goog_cc") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "goog_cc_network_control.cc",
+    "goog_cc_network_control.h",
+  ]
+
+  deps = [
+    ":alr_detector",
+    ":delay_based_bwe",
+    ":estimators",
+    ":loss_based_bwe_v2",
+    ":probe_controller",
+    ":pushback_controller",
+    ":send_side_bwe",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:field_trial_based_config",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../logging:rtc_event_pacing",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/experiments:alr_experiment",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../rtc_base/experiments:rate_control_settings",
+    "../../../system_wrappers",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("link_capacity_estimator") {
+  sources = [
+    "link_capacity_estimator.cc",
+    "link_capacity_estimator.h",
+  ]
+  deps = [
+    "../../../api/units:data_rate",
+    "../../../rtc_base:safe_minmax",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("pushback_controller") {
+  sources = [
+    "congestion_window_pushback_controller.cc",
+    "congestion_window_pushback_controller.h",
+  ]
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_size",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/experiments:rate_control_settings",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("alr_detector") {
+  sources = [
+    "alr_detector.cc",
+    "alr_detector.h",
+  ]
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:field_trial_based_config",
+    "../../../logging:rtc_event_pacing",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:timeutils",
+    "../../../rtc_base/experiments:alr_experiment",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../pacing:interval_budget",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+rtc_library("estimators") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "acknowledged_bitrate_estimator.cc",
+    "acknowledged_bitrate_estimator.h",
+    "acknowledged_bitrate_estimator_interface.cc",
+    "acknowledged_bitrate_estimator_interface.h",
+    "bitrate_estimator.cc",
+    "bitrate_estimator.h",
+    "delay_increase_detector_interface.h",
+    "probe_bitrate_estimator.cc",
+    "probe_bitrate_estimator.h",
+    "robust_throughput_estimator.cc",
+    "robust_throughput_estimator.h",
+    "trendline_estimator.cc",
+    "trendline_estimator.h",
+  ]
+
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:rtc_numerics",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("loss_based_bwe_v2") {
+  sources = [
+    "loss_based_bwe_v2.cc",
+    "loss_based_bwe_v2.h",
+  ]
+  deps = [
+    "../../../api:array_view",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("loss_based_bwe_v1") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "loss_based_bandwidth_estimation.cc",
+    "loss_based_bandwidth_estimation.h",
+  ]
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:checks",
+    "../../../rtc_base/experiments:field_trial_parser",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+}
+
+rtc_library("send_side_bwe") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "send_side_bandwidth_estimation.cc",
+    "send_side_bandwidth_estimation.h",
+  ]
+  deps = [
+    ":loss_based_bwe_v1",
+    ":loss_based_bwe_v2",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../system_wrappers:field_trial",
+    "../../../system_wrappers:metrics",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("delay_based_bwe") {
+  configs += [ ":bwe_test_logging" ]
+  sources = [
+    "delay_based_bwe.cc",
+    "delay_based_bwe.h",
+    "inter_arrival_delta.cc",
+    "inter_arrival_delta.h",
+  ]
+
+  deps = [
+    ":estimators",
+    "../../../api:field_trials_view",
+    "../../../api:network_state_predictor_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:race_checker",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../system_wrappers:metrics",
+    "../../pacing",
+    "../../remote_bitrate_estimator",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("probe_controller") {
+  sources = [
+    "probe_controller.cc",
+    "probe_controller.h",
+  ]
+
+  deps = [
+    "../../../api:field_trials_view",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../logging:rtc_event_bwe",
+    "../../../logging:rtc_event_pacing",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base/experiments:field_trial_parser",
+    "../../../system_wrappers:metrics",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:core_headers",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("test_goog_cc_printer") {
+    testonly = true
+    sources = [
+      "test/goog_cc_printer.cc",
+      "test/goog_cc_printer.h",
+    ]
+    deps = [
+      ":alr_detector",
+      ":delay_based_bwe",
+      ":estimators",
+      ":goog_cc",
+      "../../../api/rtc_event_log",
+      "../../../api/transport:goog_cc",
+      "../../../api/transport:network_control",
+      "../../../api/units:timestamp",
+      "../../../rtc_base:checks",
+      "../../../test/logging:log_writer",
+      "../../remote_bitrate_estimator",
+    ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/strings",
+      "//third_party/abseil-cpp/absl/types:optional",
+    ]
+  }
+  if (!build_with_chromium) {
+    rtc_library("goog_cc_unittests") {
+      testonly = true
+
+      sources = [
+        "acknowledged_bitrate_estimator_unittest.cc",
+        "alr_detector_unittest.cc",
+        "congestion_window_pushback_controller_unittest.cc",
+        "delay_based_bwe_unittest.cc",
+        "delay_based_bwe_unittest_helper.cc",
+        "delay_based_bwe_unittest_helper.h",
+        "goog_cc_network_control_unittest.cc",
+        "loss_based_bwe_v2_test.cc",
+        "probe_bitrate_estimator_unittest.cc",
+        "probe_controller_unittest.cc",
+        "robust_throughput_estimator_unittest.cc",
+        "send_side_bandwidth_estimation_unittest.cc",
+        "trendline_estimator_unittest.cc",
+      ]
+      deps = [
+        ":alr_detector",
+        ":delay_based_bwe",
+        ":estimators",
+        ":goog_cc",
+        ":loss_based_bwe_v2",
+        ":probe_controller",
+        ":pushback_controller",
+        ":send_side_bwe",
+        "../../../api:field_trials_view",
+        "../../../api:network_state_predictor_api",
+        "../../../api/rtc_event_log",
+        "../../../api/test/network_emulation",
+        "../../../api/test/network_emulation:create_cross_traffic",
+        "../../../api/transport:field_trial_based_config",
+        "../../../api/transport:goog_cc",
+        "../../../api/transport:network_control",
+        "../../../api/units:data_rate",
+        "../../../api/units:data_size",
+        "../../../api/units:time_delta",
+        "../../../api/units:timestamp",
+        "../../../logging:mocks",
+        "../../../logging:rtc_event_bwe",
+        "../../../rtc_base:checks",
+        "../../../rtc_base:logging",
+        "../../../rtc_base:random",
+        "../../../rtc_base:rtc_base_tests_utils",
+        "../../../rtc_base:stringutils",
+        "../../../rtc_base/experiments:alr_experiment",
+        "../../../system_wrappers",
+        "../../../test:explicit_key_value_config",
+        "../../../test:field_trial",
+        "../../../test:test_support",
+        "../../../test/scenario",
+        "../../pacing",
+        "//testing/gmock",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/strings:strings" ]
+    }
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
new file mode 100644
index 0000000000..08b42a8168
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
@@ -0,0 +1,70 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
+    const FieldTrialsView* key_value_config)
+    : AcknowledgedBitrateEstimator(
+          key_value_config,
+          std::make_unique<BitrateEstimator>(key_value_config)) {}
+
+AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
+    const FieldTrialsView* key_value_config,
+    std::unique_ptr<BitrateEstimator> bitrate_estimator)
+    : in_alr_(false), bitrate_estimator_(std::move(bitrate_estimator)) {}
+
+void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
+    const std::vector<PacketResult>& packet_feedback_vector) {
+  RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
+                            packet_feedback_vector.end(),
+                            PacketResult::ReceiveTimeOrder()));
+  for (const auto& packet : packet_feedback_vector) {
+    if (alr_ended_time_ && packet.sent_packet.send_time > *alr_ended_time_) {
+      bitrate_estimator_->ExpectFastRateChange();
+      alr_ended_time_.reset();
+    }
+    DataSize acknowledged_estimate = packet.sent_packet.size;
+    acknowledged_estimate += packet.sent_packet.prior_unacked_data;
+    bitrate_estimator_->Update(packet.receive_time, acknowledged_estimate,
+                               in_alr_);
+  }
+}
+
+absl::optional<DataRate> AcknowledgedBitrateEstimator::bitrate() const {
+  return bitrate_estimator_->bitrate();
+}
+
+absl::optional<DataRate> AcknowledgedBitrateEstimator::PeekRate() const {
+  return bitrate_estimator_->PeekRate();
+}
+
+void AcknowledgedBitrateEstimator::SetAlrEndedTime(Timestamp alr_ended_time) {
+  alr_ended_time_.emplace(alr_ended_time);
+}
+
+void AcknowledgedBitrateEstimator::SetAlr(bool in_alr) {
+  in_alr_ = in_alr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
new file mode 100644
index 0000000000..d10846ab3a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
+
+namespace webrtc {
+
+class AcknowledgedBitrateEstimator
+    : public AcknowledgedBitrateEstimatorInterface {
+ public:
+  AcknowledgedBitrateEstimator(
+      const FieldTrialsView* key_value_config,
+      std::unique_ptr<BitrateEstimator> bitrate_estimator);
+
+  explicit AcknowledgedBitrateEstimator(
+      const FieldTrialsView* key_value_config);
+  ~AcknowledgedBitrateEstimator() override;
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) override;
+  absl::optional<DataRate> bitrate() const override;
+  absl::optional<DataRate> PeekRate() const override;
+  void SetAlr(bool in_alr) override;
+  void SetAlrEndedTime(Timestamp alr_ended_time) override;
+
+ private:
+  absl::optional<Timestamp> alr_ended_time_;
+  bool in_alr_;
+  std::unique_ptr<BitrateEstimator> bitrate_estimator_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc
new file mode 100644
index 0000000000..c043353a7a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc
@@ -0,0 +1,92 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+
+#include <algorithm>
+
+#include "api/units/time_delta.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr char RobustThroughputEstimatorSettings::kKey[];
+
+RobustThroughputEstimatorSettings::RobustThroughputEstimatorSettings(
+    const FieldTrialsView* key_value_config) {
+  Parser()->Parse(
+      key_value_config->Lookup(RobustThroughputEstimatorSettings::kKey));
+  if (window_packets < 10 || 1000 < window_packets) {
+    RTC_LOG(LS_WARNING) << "Window size must be between 10 and 1000 packets";
+    window_packets = 20;
+  }
+  if (max_window_packets < 10 || 1000 < max_window_packets) {
+    RTC_LOG(LS_WARNING)
+        << "Max window size must be between 10 and 1000 packets";
+    max_window_packets = 500;
+  }
+  max_window_packets = std::max(max_window_packets, window_packets);
+
+  if (required_packets < 10 || 1000 < required_packets) {
+    RTC_LOG(LS_WARNING) << "Required number of initial packets must be between "
+                           "10 and 1000 packets";
+    required_packets = 10;
+  }
+  required_packets = std::min(required_packets, window_packets);
+
+  if (min_window_duration < TimeDelta::Millis(100) ||
+      TimeDelta::Millis(3000) < min_window_duration) {
+    RTC_LOG(LS_WARNING) << "Window duration must be between 100 and 3000 ms";
+    min_window_duration = TimeDelta::Millis(750);
+  }
+  if (max_window_duration < TimeDelta::Seconds(1) ||
+      TimeDelta::Seconds(15) < max_window_duration) {
+    RTC_LOG(LS_WARNING) << "Max window duration must be between 1 and 15 s";
+    max_window_duration = TimeDelta::Seconds(5);
+  }
+  min_window_duration = std::min(min_window_duration, max_window_duration);
+
+  if (unacked_weight < 0.0 || 1.0 < unacked_weight) {
+    RTC_LOG(LS_WARNING)
+        << "Weight for prior unacked size must be between 0 and 1.";
+    unacked_weight = 1.0;
+  }
+}
+
+std::unique_ptr<StructParametersParser>
+RobustThroughputEstimatorSettings::Parser() {
+  return StructParametersParser::Create(
+      "enabled", &enabled,                          //
+      "window_packets", &window_packets,            //
+      "max_window_packets", &max_window_packets,    //
+      "window_duration", &min_window_duration,      //
+      "max_window_duration", &max_window_duration,  //
+      "required_packets", &required_packets,        //
+      "unacked_weight", &unacked_weight);
+}
+
+AcknowledgedBitrateEstimatorInterface::
+    ~AcknowledgedBitrateEstimatorInterface() {}
+
+std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+AcknowledgedBitrateEstimatorInterface::Create(
+    const FieldTrialsView* key_value_config) {
+  RobustThroughputEstimatorSettings simplified_estimator_settings(
+      key_value_config);
+  if (simplified_estimator_settings.enabled) {
+    return std::make_unique<RobustThroughputEstimator>(
+        simplified_estimator_settings);
+  }
+  return std::make_unique<AcknowledgedBitrateEstimator>(key_value_config);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h
new file mode 100644
index 0000000000..515af1efc9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
+
+#include <stddef.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+struct RobustThroughputEstimatorSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-RobustThroughputEstimatorSettings";
+
+  RobustThroughputEstimatorSettings() = delete;
+  explicit RobustThroughputEstimatorSettings(
+      const FieldTrialsView* key_value_config);
+
+  bool enabled = false;  // Set to true to use RobustThroughputEstimator.
+
+  // The estimator keeps the smallest window containing at least
+  // `window_packets` and at least the packets received during the last
+  // `min_window_duration` milliseconds.
+  // (This means that it may store more than `window_packets` at high bitrates,
+  // and a longer duration than `min_window_duration` at low bitrates.)
+  // However, if will never store more than kMaxPackets (for performance
+  // reasons), and never longer than max_window_duration (to avoid very old
+  // packets influencing the estimate for example when sending is paused).
+  unsigned window_packets = 20;
+  unsigned max_window_packets = 500;
+  TimeDelta min_window_duration = TimeDelta::Seconds(1);
+  TimeDelta max_window_duration = TimeDelta::Seconds(5);
+
+  // The estimator window requires at least `required_packets` packets
+  // to produce an estimate.
+  unsigned required_packets = 10;
+
+  // If audio packets aren't included in allocation (i.e. the
+  // estimated available bandwidth is divided only among the video
+  // streams), then `unacked_weight` should be set to 0.
+  // If audio packets are included in allocation, but not in bandwidth
+  // estimation (i.e. they don't have transport-wide sequence numbers,
+  // but we nevertheless divide the estimated available bandwidth among
+  // both audio and video streams), then `unacked_weight` should be set to 1.
+  // If all packets have transport-wide sequence numbers, then the value
+  // of `unacked_weight` doesn't matter.
+  double unacked_weight = 1.0;
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class AcknowledgedBitrateEstimatorInterface {
+ public:
+  static std::unique_ptr<AcknowledgedBitrateEstimatorInterface> Create(
+      const FieldTrialsView* key_value_config);
+  virtual ~AcknowledgedBitrateEstimatorInterface();
+
+  virtual void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) = 0;
+  virtual absl::optional<DataRate> bitrate() const = 0;
+  virtual absl::optional<DataRate> PeekRate() const = 0;
+  virtual void SetAlr(bool in_alr) = 0;
+  virtual void SetAlrEndedTime(Timestamp alr_ended_time) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
new file mode 100644
index 0000000000..e5b733b119
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
@@ -0,0 +1,136 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/transport/field_trial_based_config.h"
+#include "rtc_base/fake_clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::InSequence;
+using ::testing::NiceMock;
+using ::testing::Return;
+
+namespace webrtc {
+
+namespace {
+
+constexpr int64_t kFirstArrivalTimeMs = 10;
+constexpr int64_t kFirstSendTimeMs = 10;
+constexpr uint16_t kSequenceNumber = 1;
+constexpr size_t kPayloadSize = 10;
+
+class MockBitrateEstimator : public BitrateEstimator {
+ public:
+  using BitrateEstimator::BitrateEstimator;
+  MOCK_METHOD(void,
+              Update,
+              (Timestamp at_time, DataSize data_size, bool in_alr),
+              (override));
+  MOCK_METHOD(absl::optional<DataRate>, bitrate, (), (const, override));
+  MOCK_METHOD(void, ExpectFastRateChange, (), (override));
+};
+
+struct AcknowledgedBitrateEstimatorTestStates {
+  FieldTrialBasedConfig field_trial_config;
+  std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator;
+  MockBitrateEstimator* mock_bitrate_estimator;
+};
+
+AcknowledgedBitrateEstimatorTestStates CreateTestStates() {
+  AcknowledgedBitrateEstimatorTestStates states;
+  auto mock_bitrate_estimator =
+      std::make_unique<MockBitrateEstimator>(&states.field_trial_config);
+  states.mock_bitrate_estimator = mock_bitrate_estimator.get();
+  states.acknowledged_bitrate_estimator =
+      std::make_unique<AcknowledgedBitrateEstimator>(
+          &states.field_trial_config, std::move(mock_bitrate_estimator));
+  return states;
+}
+
+std::vector<PacketResult> CreateFeedbackVector() {
+  std::vector<PacketResult> packet_feedback_vector(2);
+  packet_feedback_vector[0].receive_time =
+      Timestamp::Millis(kFirstArrivalTimeMs);
+  packet_feedback_vector[0].sent_packet.send_time =
+      Timestamp::Millis(kFirstSendTimeMs);
+  packet_feedback_vector[0].sent_packet.sequence_number = kSequenceNumber;
+  packet_feedback_vector[0].sent_packet.size = DataSize::Bytes(kPayloadSize);
+  packet_feedback_vector[1].receive_time =
+      Timestamp::Millis(kFirstArrivalTimeMs + 10);
+  packet_feedback_vector[1].sent_packet.send_time =
+      Timestamp::Millis(kFirstSendTimeMs + 10);
+  packet_feedback_vector[1].sent_packet.sequence_number = kSequenceNumber;
+  packet_feedback_vector[1].sent_packet.size =
+      DataSize::Bytes(kPayloadSize + 10);
+  return packet_feedback_vector;
+}
+
+}  // anonymous namespace
+
+TEST(TestAcknowledgedBitrateEstimator, UpdateBandwidth) {
+  auto states = CreateTestStates();
+  auto packet_feedback_vector = CreateFeedbackVector();
+  {
+    InSequence dummy;
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[0].receive_time,
+                       packet_feedback_vector[0].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[1].receive_time,
+                       packet_feedback_vector[1].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+  }
+  states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+      packet_feedback_vector);
+}
+
+TEST(TestAcknowledgedBitrateEstimator, ExpectFastRateChangeWhenLeftAlr) {
+  auto states = CreateTestStates();
+  auto packet_feedback_vector = CreateFeedbackVector();
+  {
+    InSequence dummy;
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[0].receive_time,
+                       packet_feedback_vector[0].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+    EXPECT_CALL(*states.mock_bitrate_estimator, ExpectFastRateChange())
+        .Times(1);
+    EXPECT_CALL(*states.mock_bitrate_estimator,
+                Update(packet_feedback_vector[1].receive_time,
+                       packet_feedback_vector[1].sent_packet.size,
+                       /*in_alr*/ false))
+        .Times(1);
+  }
+  states.acknowledged_bitrate_estimator->SetAlrEndedTime(
+      Timestamp::Millis(kFirstArrivalTimeMs + 1));
+  states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+      packet_feedback_vector);
+}
+
+TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
+  auto states = CreateTestStates();
+  absl::optional<DataRate> return_value = DataRate::KilobitsPerSec(42);
+  EXPECT_CALL(*states.mock_bitrate_estimator, bitrate())
+      .Times(1)
+      .WillOnce(Return(return_value));
+  EXPECT_EQ(return_value, states.acknowledged_bitrate_estimator->bitrate());
+}
+
+}  // namespace webrtc*/
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc
new file mode 100644
index 0000000000..f1e649b7cd
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc
@@ -0,0 +1,111 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+
+#include <cstdint>
+#include <cstdio>
+#include <memory>
+
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+namespace {
+AlrDetectorConfig GetConfigFromTrials(const FieldTrialsView* key_value_config) {
+  RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled(*key_value_config));
+  absl::optional<AlrExperimentSettings> experiment_settings =
+      AlrExperimentSettings::CreateFromFieldTrial(
+          *key_value_config,
+          AlrExperimentSettings::kScreenshareProbingBweExperimentName);
+  if (!experiment_settings) {
+    experiment_settings = AlrExperimentSettings::CreateFromFieldTrial(
+        *key_value_config,
+        AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
+  }
+  AlrDetectorConfig conf;
+  if (experiment_settings) {
+    conf.bandwidth_usage_ratio =
+        experiment_settings->alr_bandwidth_usage_percent / 100.0;
+    conf.start_budget_level_ratio =
+        experiment_settings->alr_start_budget_level_percent / 100.0;
+    conf.stop_budget_level_ratio =
+        experiment_settings->alr_stop_budget_level_percent / 100.0;
+  }
+  conf.Parser()->Parse(
+      key_value_config->Lookup("WebRTC-AlrDetectorParameters"));
+  return conf;
+}
+}  //  namespace
+
+std::unique_ptr<StructParametersParser> AlrDetectorConfig::Parser() {
+  return StructParametersParser::Create(   //
+      "bw_usage", &bandwidth_usage_ratio,  //
+      "start", &start_budget_level_ratio,  //
+      "stop", &stop_budget_level_ratio);
+}
+
+AlrDetector::AlrDetector(AlrDetectorConfig config, RtcEventLog* event_log)
+    : conf_(config), alr_budget_(0, true), event_log_(event_log) {}
+
+AlrDetector::AlrDetector(const FieldTrialsView* key_value_config)
+    : AlrDetector(GetConfigFromTrials(key_value_config), nullptr) {}
+
+AlrDetector::AlrDetector(const FieldTrialsView* key_value_config,
+                         RtcEventLog* event_log)
+    : AlrDetector(GetConfigFromTrials(key_value_config), event_log) {}
+AlrDetector::~AlrDetector() {}
+
+void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t send_time_ms) {
+  if (!last_send_time_ms_.has_value()) {
+    last_send_time_ms_ = send_time_ms;
+    // Since the duration for sending the bytes is unknwon, return without
+    // updating alr state.
+    return;
+  }
+  int64_t delta_time_ms = send_time_ms - *last_send_time_ms_;
+  last_send_time_ms_ = send_time_ms;
+
+  alr_budget_.UseBudget(bytes_sent);
+  alr_budget_.IncreaseBudget(delta_time_ms);
+  bool state_changed = false;
+  if (alr_budget_.budget_ratio() > conf_.start_budget_level_ratio &&
+      !alr_started_time_ms_) {
+    alr_started_time_ms_.emplace(rtc::TimeMillis());
+    state_changed = true;
+  } else if (alr_budget_.budget_ratio() < conf_.stop_budget_level_ratio &&
+             alr_started_time_ms_) {
+    state_changed = true;
+    alr_started_time_ms_.reset();
+  }
+  if (event_log_ && state_changed) {
+    event_log_->Log(
+        std::make_unique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
+  }
+}
+
+void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
+  RTC_DCHECK(bitrate_bps);
+  int target_rate_kbps =
+      static_cast<double>(bitrate_bps) * conf_.bandwidth_usage_ratio / 1000;
+  alr_budget_.set_target_rate_kbps(target_rate_kbps);
+}
+
+absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
+    const {
+  return alr_started_time_ms_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.h
new file mode 100644
index 0000000000..5e7a3e1075
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "modules/pacing/interval_budget.h"
+#include "rtc_base/experiments/alr_experiment.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+struct AlrDetectorConfig {
+  // Sent traffic ratio as a function of network capacity used to determine
+  // application-limited region. ALR region start when bandwidth usage drops
+  // below kAlrStartUsageRatio and ends when it raises above
+  // kAlrEndUsageRatio. NOTE: This is intentionally conservative at the moment
+  // until BW adjustments of application limited region is fine tuned.
+  double bandwidth_usage_ratio = 0.65;
+  double start_budget_level_ratio = 0.80;
+  double stop_budget_level_ratio = 0.50;
+  std::unique_ptr<StructParametersParser> Parser();
+};
+// Application limited region detector is a class that utilizes signals of
+// elapsed time and bytes sent to estimate whether network traffic is
+// currently limited by the application's ability to generate traffic.
+//
+// AlrDetector provides a signal that can be utilized to adjust
+// estimate bandwidth.
+// Note: This class is not thread-safe.
+class AlrDetector {
+ public:
+  AlrDetector(AlrDetectorConfig config, RtcEventLog* event_log);
+  explicit AlrDetector(const FieldTrialsView* key_value_config);
+  AlrDetector(const FieldTrialsView* key_value_config, RtcEventLog* event_log);
+  ~AlrDetector();
+
+  void OnBytesSent(size_t bytes_sent, int64_t send_time_ms);
+
+  // Set current estimated bandwidth.
+  void SetEstimatedBitrate(int bitrate_bps);
+
+  // Returns time in milliseconds when the current application-limited region
+  // started or empty result if the sender is currently not application-limited.
+  absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  const AlrDetectorConfig conf_;
+
+  absl::optional<int64_t> last_send_time_ms_;
+
+  IntervalBudget alr_budget_;
+  absl::optional<int64_t> alr_started_time_ms_;
+
+  RtcEventLog* event_log_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build
new file mode 100644
index 0000000000..f6520bf358
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("alr_detector_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
new file mode 100644
index 0000000000..eac19d0081
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+
+#include "api/transport/field_trial_based_config.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/alr_experiment.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace {
+
+constexpr int kEstimatedBitrateBps = 300000;
+
+}  // namespace
+
+namespace webrtc {
+namespace {
+class SimulateOutgoingTrafficIn {
+ public:
+  explicit SimulateOutgoingTrafficIn(AlrDetector* alr_detector,
+                                     int64_t* timestamp_ms)
+      : alr_detector_(alr_detector), timestamp_ms_(timestamp_ms) {
+    RTC_CHECK(alr_detector_);
+  }
+
+  SimulateOutgoingTrafficIn& ForTimeMs(int time_ms) {
+    interval_ms_ = time_ms;
+    ProduceTraffic();
+    return *this;
+  }
+
+  SimulateOutgoingTrafficIn& AtPercentOfEstimatedBitrate(int usage_percentage) {
+    usage_percentage_.emplace(usage_percentage);
+    ProduceTraffic();
+    return *this;
+  }
+
+ private:
+  void ProduceTraffic() {
+    if (!interval_ms_ || !usage_percentage_)
+      return;
+    const int kTimeStepMs = 10;
+    for (int t = 0; t < *interval_ms_; t += kTimeStepMs) {
+      *timestamp_ms_ += kTimeStepMs;
+      alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
+                                     kTimeStepMs / (8 * 100 * 1000),
+                                 *timestamp_ms_);
+    }
+    int remainder_ms = *interval_ms_ % kTimeStepMs;
+    if (remainder_ms > 0) {
+      *timestamp_ms_ += kTimeStepMs;
+      alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
+                                     remainder_ms / (8 * 100 * 1000),
+                                 *timestamp_ms_);
+    }
+  }
+  AlrDetector* const alr_detector_;
+  int64_t* timestamp_ms_;
+  absl::optional<int> interval_ms_;
+  absl::optional<int> usage_percentage_;
+};
+}  // namespace
+
+TEST(AlrDetectorTest, AlrDetection) {
+  FieldTrialBasedConfig field_trials;
+  int64_t timestamp_ms = 1000;
+  AlrDetector alr_detector(&field_trials);
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Stay in non-ALR state when usage is close to 100%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(90);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that we ALR starts when bitrate drops below 20%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1500)
+      .AtPercentOfEstimatedBitrate(20);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that ALR ends when usage is above 65%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(4000)
+      .AtPercentOfEstimatedBitrate(100);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+TEST(AlrDetectorTest, ShortSpike) {
+  FieldTrialBasedConfig field_trials;
+  int64_t timestamp_ms = 1000;
+  AlrDetector alr_detector(&field_trials);
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that we ALR starts when bitrate drops below 20%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(20);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // Verify that we stay in ALR region even after a short bitrate spike.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(100)
+      .AtPercentOfEstimatedBitrate(150);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR ends when usage is above 65%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(3000)
+      .AtPercentOfEstimatedBitrate(100);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+TEST(AlrDetectorTest, BandwidthEstimateChanges) {
+  FieldTrialBasedConfig field_trials;
+  int64_t timestamp_ms = 1000;
+  AlrDetector alr_detector(&field_trials);
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR starts when bitrate drops below 20%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(20);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // When bandwidth estimate drops the detector should stay in ALR mode and quit
+  // it shortly afterwards as the sender continues sending the same amount of
+  // traffic. This is necessary to ensure that ProbeController can still react
+  // to the BWE drop by initiating a new probe.
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps / 5);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(50);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+TEST(AlrDetectorTest, ParseControlFieldTrial) {
+  webrtc::test::ScopedFieldTrials scoped_field_trial(
+      "WebRTC-ProbingScreenshareBwe/Control/");
+  absl::optional<AlrExperimentSettings> parsed_params =
+      AlrExperimentSettings::CreateFromFieldTrial(
+          FieldTrialBasedConfig(), "WebRTC-ProbingScreenshareBwe");
+  EXPECT_FALSE(static_cast<bool>(parsed_params));
+}
+
+TEST(AlrDetectorTest, ParseActiveFieldTrial) {
+  webrtc::test::ScopedFieldTrials scoped_field_trial(
+      "WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
+  absl::optional<AlrExperimentSettings> parsed_params =
+      AlrExperimentSettings::CreateFromFieldTrial(
+          FieldTrialBasedConfig(), "WebRTC-ProbingScreenshareBwe");
+  ASSERT_TRUE(static_cast<bool>(parsed_params));
+  EXPECT_EQ(1.1f, parsed_params->pacing_factor);
+  EXPECT_EQ(2875, parsed_params->max_paced_queue_time);
+  EXPECT_EQ(85, parsed_params->alr_bandwidth_usage_percent);
+  EXPECT_EQ(20, parsed_params->alr_start_budget_level_percent);
+  EXPECT_EQ(-20, parsed_params->alr_stop_budget_level_percent);
+  EXPECT_EQ(1, parsed_params->group_id);
+}
+
+TEST(AlrDetectorTest, ParseAlrSpecificFieldTrial) {
+  webrtc::test::ScopedFieldTrials scoped_field_trial(
+      "WebRTC-AlrDetectorParameters/"
+      "bw_usage:90%,start:0%,stop:-10%/");
+  FieldTrialBasedConfig field_trials;
+  AlrDetector alr_detector(&field_trials);
+  int64_t timestamp_ms = 1000;
+  alr_detector.SetEstimatedBitrate(kEstimatedBitrateBps);
+
+  // Start in non-ALR state.
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR does not start at 100% utilization.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(1000)
+      .AtPercentOfEstimatedBitrate(100);
+  EXPECT_FALSE(alr_detector.GetApplicationLimitedRegionStartTime());
+
+  // ALR does start at 85% utilization.
+  // Overused 10% above so it should take about 2s to reach a budget level of
+  // 0%.
+  SimulateOutgoingTrafficIn(&alr_detector, &timestamp_ms)
+      .ForTimeMs(2100)
+      .AtPercentOfEstimatedBitrate(85);
+  EXPECT_TRUE(alr_detector.GetApplicationLimitedRegionStartTime());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
new file mode 100644
index 0000000000..9c68e48886
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
@@ -0,0 +1,166 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
+
+#include <stdio.h>
+
+#include <algorithm>
+#include <cmath>
+#include <string>
+
+#include "api/units/data_rate.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kInitialRateWindowMs = 500;
+constexpr int kRateWindowMs = 150;
+constexpr int kMinRateWindowMs = 150;
+constexpr int kMaxRateWindowMs = 1000;
+
+const char kBweThroughputWindowConfig[] = "WebRTC-BweThroughputWindowConfig";
+
+}  // namespace
+
+BitrateEstimator::BitrateEstimator(const FieldTrialsView* key_value_config)
+    : sum_(0),
+      initial_window_ms_("initial_window_ms",
+                         kInitialRateWindowMs,
+                         kMinRateWindowMs,
+                         kMaxRateWindowMs),
+      noninitial_window_ms_("window_ms",
+                            kRateWindowMs,
+                            kMinRateWindowMs,
+                            kMaxRateWindowMs),
+      uncertainty_scale_("scale", 10.0),
+      uncertainty_scale_in_alr_("scale_alr", uncertainty_scale_),
+      small_sample_uncertainty_scale_("scale_small", uncertainty_scale_),
+      small_sample_threshold_("small_thresh", DataSize::Zero()),
+      uncertainty_symmetry_cap_("symmetry_cap", DataRate::Zero()),
+      estimate_floor_("floor", DataRate::Zero()),
+      current_window_ms_(0),
+      prev_time_ms_(-1),
+      bitrate_estimate_kbps_(-1.0f),
+      bitrate_estimate_var_(50.0f) {
+  // E.g WebRTC-BweThroughputWindowConfig/initial_window_ms:350,window_ms:250/
+  ParseFieldTrial(
+      {&initial_window_ms_, &noninitial_window_ms_, &uncertainty_scale_,
+       &uncertainty_scale_in_alr_, &small_sample_uncertainty_scale_,
+       &small_sample_threshold_, &uncertainty_symmetry_cap_, &estimate_floor_},
+      key_value_config->Lookup(kBweThroughputWindowConfig));
+}
+
+BitrateEstimator::~BitrateEstimator() = default;
+
+void BitrateEstimator::Update(Timestamp at_time, DataSize amount, bool in_alr) {
+  int rate_window_ms = noninitial_window_ms_.Get();
+  // We use a larger window at the beginning to get a more stable sample that
+  // we can use to initialize the estimate.
+  if (bitrate_estimate_kbps_ < 0.f)
+    rate_window_ms = initial_window_ms_.Get();
+  bool is_small_sample = false;
+  float bitrate_sample_kbps = UpdateWindow(at_time.ms(), amount.bytes(),
+                                           rate_window_ms, &is_small_sample);
+  if (bitrate_sample_kbps < 0.0f)
+    return;
+  if (bitrate_estimate_kbps_ < 0.0f) {
+    // This is the very first sample we get. Use it to initialize the estimate.
+    bitrate_estimate_kbps_ = bitrate_sample_kbps;
+    return;
+  }
+  // Optionally use higher uncertainty for very small samples to avoid dropping
+  // estimate and for samples obtained in ALR.
+  float scale = uncertainty_scale_;
+  if (is_small_sample && bitrate_sample_kbps < bitrate_estimate_kbps_) {
+    scale = small_sample_uncertainty_scale_;
+  } else if (in_alr && bitrate_sample_kbps < bitrate_estimate_kbps_) {
+    // Optionally use higher uncertainty for samples obtained during ALR.
+    scale = uncertainty_scale_in_alr_;
+  }
+  // Define the sample uncertainty as a function of how far away it is from the
+  // current estimate. With low values of uncertainty_symmetry_cap_ we add more
+  // uncertainty to increases than to decreases. For higher values we approach
+  // symmetry.
+  float sample_uncertainty =
+      scale * std::abs(bitrate_estimate_kbps_ - bitrate_sample_kbps) /
+      (bitrate_estimate_kbps_ +
+       std::min(bitrate_sample_kbps,
+                uncertainty_symmetry_cap_.Get().kbps<float>()));
+
+  float sample_var = sample_uncertainty * sample_uncertainty;
+  // Update a bayesian estimate of the rate, weighting it lower if the sample
+  // uncertainty is large.
+  // The bitrate estimate uncertainty is increased with each update to model
+  // that the bitrate changes over time.
+  float pred_bitrate_estimate_var = bitrate_estimate_var_ + 5.f;
+  bitrate_estimate_kbps_ = (sample_var * bitrate_estimate_kbps_ +
+                            pred_bitrate_estimate_var * bitrate_sample_kbps) /
+                           (sample_var + pred_bitrate_estimate_var);
+  bitrate_estimate_kbps_ =
+      std::max(bitrate_estimate_kbps_, estimate_floor_.Get().kbps<float>());
+  bitrate_estimate_var_ = sample_var * pred_bitrate_estimate_var /
+                          (sample_var + pred_bitrate_estimate_var);
+  BWE_TEST_LOGGING_PLOT(1, "acknowledged_bitrate", at_time.ms(),
+                        bitrate_estimate_kbps_ * 1000);
+}
+
+float BitrateEstimator::UpdateWindow(int64_t now_ms,
+                                     int bytes,
+                                     int rate_window_ms,
+                                     bool* is_small_sample) {
+  RTC_DCHECK(is_small_sample != nullptr);
+  // Reset if time moves backwards.
+  if (now_ms < prev_time_ms_) {
+    prev_time_ms_ = -1;
+    sum_ = 0;
+    current_window_ms_ = 0;
+  }
+  if (prev_time_ms_ >= 0) {
+    current_window_ms_ += now_ms - prev_time_ms_;
+    // Reset if nothing has been received for more than a full window.
+    if (now_ms - prev_time_ms_ > rate_window_ms) {
+      sum_ = 0;
+      current_window_ms_ %= rate_window_ms;
+    }
+  }
+  prev_time_ms_ = now_ms;
+  float bitrate_sample = -1.0f;
+  if (current_window_ms_ >= rate_window_ms) {
+    *is_small_sample = sum_ < small_sample_threshold_->bytes();
+    bitrate_sample = 8.0f * sum_ / static_cast<float>(rate_window_ms);
+    current_window_ms_ -= rate_window_ms;
+    sum_ = 0;
+  }
+  sum_ += bytes;
+  return bitrate_sample;
+}
+
+absl::optional<DataRate> BitrateEstimator::bitrate() const {
+  if (bitrate_estimate_kbps_ < 0.f)
+    return absl::nullopt;
+  return DataRate::KilobitsPerSec(bitrate_estimate_kbps_);
+}
+
+absl::optional<DataRate> BitrateEstimator::PeekRate() const {
+  if (current_window_ms_ > 0)
+    return DataSize::Bytes(sum_) / TimeDelta::Millis(current_window_ms_);
+  return absl::nullopt;
+}
+
+void BitrateEstimator::ExpectFastRateChange() {
+  // By setting the bitrate-estimate variance to a higher value we allow the
+  // bitrate to change fast for the next few samples.
+  bitrate_estimate_var_ += 200;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
new file mode 100644
index 0000000000..a6f985800e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+// Computes a bayesian estimate of the throughput given acks containing
+// the arrival time and payload size. Samples which are far from the current
+// estimate or are based on few packets are given a smaller weight, as they
+// are considered to be more likely to have been caused by, e.g., delay spikes
+// unrelated to congestion.
+class BitrateEstimator {
+ public:
+  explicit BitrateEstimator(const FieldTrialsView* key_value_config);
+  virtual ~BitrateEstimator();
+  virtual void Update(Timestamp at_time, DataSize amount, bool in_alr);
+
+  virtual absl::optional<DataRate> bitrate() const;
+  absl::optional<DataRate> PeekRate() const;
+
+  virtual void ExpectFastRateChange();
+
+ private:
+  float UpdateWindow(int64_t now_ms,
+                     int bytes,
+                     int rate_window_ms,
+                     bool* is_small_sample);
+  int sum_;
+  FieldTrialConstrained<int> initial_window_ms_;
+  FieldTrialConstrained<int> noninitial_window_ms_;
+  FieldTrialParameter<double> uncertainty_scale_;
+  FieldTrialParameter<double> uncertainty_scale_in_alr_;
+  FieldTrialParameter<double> small_sample_uncertainty_scale_;
+  FieldTrialParameter<DataSize> small_sample_threshold_;
+  FieldTrialParameter<DataRate> uncertainty_symmetry_cap_;
+  FieldTrialParameter<DataRate> estimate_floor_;
+  int64_t current_window_ms_;
+  int64_t prev_time_ms_;
+  float bitrate_estimate_kbps_;
+  float bitrate_estimate_var_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
new file mode 100644
index 0000000000..2f188f30ca
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+
+#include <inttypes.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+
+namespace webrtc {
+
+CongestionWindowPushbackController::CongestionWindowPushbackController(
+    const FieldTrialsView* key_value_config)
+    : add_pacing_(
+          absl::StartsWith(key_value_config->Lookup(
+                               "WebRTC-AddPacingToCongestionWindowPushback"),
+                           "Enabled")),
+      min_pushback_target_bitrate_bps_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config)
+              .CongestionWindowMinPushbackTargetBitrateBps()),
+      current_data_window_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config)
+              .CongestionWindowInitialDataWindow()) {}
+
+void CongestionWindowPushbackController::UpdateOutstandingData(
+    int64_t outstanding_bytes) {
+  outstanding_bytes_ = outstanding_bytes;
+}
+void CongestionWindowPushbackController::UpdatePacingQueue(
+    int64_t pacing_bytes) {
+  pacing_bytes_ = pacing_bytes;
+}
+
+void CongestionWindowPushbackController::SetDataWindow(DataSize data_window) {
+  current_data_window_ = data_window;
+}
+
+uint32_t CongestionWindowPushbackController::UpdateTargetBitrate(
+    uint32_t bitrate_bps) {
+  if (!current_data_window_ || current_data_window_->IsZero())
+    return bitrate_bps;
+  int64_t total_bytes = outstanding_bytes_;
+  if (add_pacing_)
+    total_bytes += pacing_bytes_;
+  double fill_ratio =
+      total_bytes / static_cast<double>(current_data_window_->bytes());
+  if (fill_ratio > 1.5) {
+    encoding_rate_ratio_ *= 0.9;
+  } else if (fill_ratio > 1) {
+    encoding_rate_ratio_ *= 0.95;
+  } else if (fill_ratio < 0.1) {
+    encoding_rate_ratio_ = 1.0;
+  } else {
+    encoding_rate_ratio_ *= 1.05;
+    encoding_rate_ratio_ = std::min(encoding_rate_ratio_, 1.0);
+  }
+  uint32_t adjusted_target_bitrate_bps =
+      static_cast<uint32_t>(bitrate_bps * encoding_rate_ratio_);
+
+  // Do not adjust below the minimum pushback bitrate but do obey if the
+  // original estimate is below it.
+  bitrate_bps = adjusted_target_bitrate_bps < min_pushback_target_bitrate_bps_
+                    ? std::min(bitrate_bps, min_pushback_target_bitrate_bps_)
+                    : adjusted_target_bitrate_bps;
+  return bitrate_bps;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
new file mode 100644
index 0000000000..ea9ed97c3d
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/units/data_size.h"
+
+namespace webrtc {
+
+// This class enables pushback from congestion window directly to video encoder.
+// When the congestion window is filling up, the video encoder target bitrate
+// will be reduced accordingly to accommodate the network changes. To avoid
+// pausing video too frequently, a minimum encoder target bitrate threshold is
+// used to prevent video pause due to a full congestion window.
+class CongestionWindowPushbackController {
+ public:
+  explicit CongestionWindowPushbackController(
+      const FieldTrialsView* key_value_config);
+  void UpdateOutstandingData(int64_t outstanding_bytes);
+  void UpdatePacingQueue(int64_t pacing_bytes);
+  uint32_t UpdateTargetBitrate(uint32_t bitrate_bps);
+  void SetDataWindow(DataSize data_window);
+
+ private:
+  const bool add_pacing_;
+  const uint32_t min_pushback_target_bitrate_bps_;
+  absl::optional<DataSize> current_data_window_;
+  int64_t outstanding_bytes_ = 0;
+  int64_t pacing_bytes_ = 0;
+  double encoding_rate_ratio_ = 1.0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc
new file mode 100644
index 0000000000..62dde02323
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller_unittest.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+
+#include <memory>
+
+#include "api/transport/field_trial_based_config.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+
+namespace webrtc {
+namespace test {
+
+class CongestionWindowPushbackControllerTest : public ::testing::Test {
+ public:
+  CongestionWindowPushbackControllerTest() {
+    cwnd_controller_.reset(
+        new CongestionWindowPushbackController(&field_trial_config_));
+  }
+
+ protected:
+  FieldTrialBasedConfig field_trial_config_;
+
+  std::unique_ptr<CongestionWindowPushbackController> cwnd_controller_;
+};
+
+TEST_F(CongestionWindowPushbackControllerTest, FullCongestionWindow) {
+  cwnd_controller_->UpdateOutstandingData(100000);
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(72000u, bitrate_bps);
+
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(static_cast<uint32_t>(72000 * 0.9 * 0.9), bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, NormalCongestionWindow) {
+  cwnd_controller_->UpdateOutstandingData(199999);
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(200000));
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(80000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, LowBitrate) {
+  cwnd_controller_->UpdateOutstandingData(100000);
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+
+  uint32_t bitrate_bps = 35000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(static_cast<uint32_t>(35000 * 0.9), bitrate_bps);
+
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(20000));
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(30000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, NoPushbackOnDataWindowUnset) {
+  cwnd_controller_->UpdateOutstandingData(1e8);  // Large number
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_EQ(80000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, PushbackOnInititialDataWindow) {
+  test::ScopedFieldTrials trials("WebRTC-CongestionWindow/InitWin:100000/");
+  cwnd_controller_.reset(
+      new CongestionWindowPushbackController(&field_trial_config_));
+  cwnd_controller_->UpdateOutstandingData(1e8);  // Large number
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_GT(80000u, bitrate_bps);
+}
+
+TEST_F(CongestionWindowPushbackControllerTest, PushbackDropFrame) {
+  test::ScopedFieldTrials trials("WebRTC-CongestionWindow/DropFrame:true/");
+  cwnd_controller_.reset(
+      new CongestionWindowPushbackController(&field_trial_config_));
+  cwnd_controller_->UpdateOutstandingData(1e8);  // Large number
+  cwnd_controller_->SetDataWindow(DataSize::Bytes(50000));
+
+  uint32_t bitrate_bps = 80000;
+  bitrate_bps = cwnd_controller_->UpdateTargetBitrate(bitrate_bps);
+  EXPECT_GT(80000u, bitrate_bps);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
new file mode 100644
index 0000000000..07ac599148
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
@@ -0,0 +1,305 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstdio>
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "absl/strings/match.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/units/time_delta.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kStreamTimeOut = TimeDelta::Seconds(2);
+constexpr TimeDelta kSendTimeGroupLength = TimeDelta::Millis(5);
+
+// This ssrc is used to fulfill the current API but will be removed
+// after the API has been changed.
+constexpr uint32_t kFixedSsrc = 0;
+}  // namespace
+
+constexpr char BweSeparateAudioPacketsSettings::kKey[];
+
+BweSeparateAudioPacketsSettings::BweSeparateAudioPacketsSettings(
+    const FieldTrialsView* key_value_config) {
+  Parser()->Parse(
+      key_value_config->Lookup(BweSeparateAudioPacketsSettings::kKey));
+}
+
+std::unique_ptr<StructParametersParser>
+BweSeparateAudioPacketsSettings::Parser() {
+  return StructParametersParser::Create(      //
+      "enabled", &enabled,                    //
+      "packet_threshold", &packet_threshold,  //
+      "time_threshold", &time_threshold);
+}
+
+DelayBasedBwe::Result::Result()
+    : updated(false),
+      probe(false),
+      target_bitrate(DataRate::Zero()),
+      recovered_from_overuse(false) {}
+
+DelayBasedBwe::DelayBasedBwe(const FieldTrialsView* key_value_config,
+                             RtcEventLog* event_log,
+                             NetworkStatePredictor* network_state_predictor)
+    : event_log_(event_log),
+      key_value_config_(key_value_config),
+      separate_audio_(key_value_config),
+      audio_packets_since_last_video_(0),
+      last_video_packet_recv_time_(Timestamp::MinusInfinity()),
+      network_state_predictor_(network_state_predictor),
+      video_delay_detector_(
+          new TrendlineEstimator(key_value_config_, network_state_predictor_)),
+      audio_delay_detector_(
+          new TrendlineEstimator(key_value_config_, network_state_predictor_)),
+      active_delay_detector_(video_delay_detector_.get()),
+      last_seen_packet_(Timestamp::MinusInfinity()),
+      uma_recorded_(false),
+      rate_control_(key_value_config, /*send_side=*/true),
+      prev_bitrate_(DataRate::Zero()),
+      prev_state_(BandwidthUsage::kBwNormal) {
+  RTC_LOG(LS_INFO)
+      << "Initialized DelayBasedBwe with separate audio overuse detection"
+      << separate_audio_.Parser()->Encode();
+}
+
+DelayBasedBwe::~DelayBasedBwe() {}
+
+DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
+    const TransportPacketsFeedback& msg,
+    absl::optional<DataRate> acked_bitrate,
+    absl::optional<DataRate> probe_bitrate,
+    absl::optional<NetworkStateEstimate> network_estimate,
+    bool in_alr) {
+  RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
+
+  auto packet_feedback_vector = msg.SortedByReceiveTime();
+  // TODO(holmer): An empty feedback vector here likely means that
+  // all acks were too late and that the send time history had
+  // timed out. We should reduce the rate when this occurs.
+  if (packet_feedback_vector.empty()) {
+    RTC_LOG(LS_WARNING) << "Very late feedback received.";
+    return DelayBasedBwe::Result();
+  }
+
+  if (!uma_recorded_) {
+    RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
+                              BweNames::kSendSideTransportSeqNum,
+                              BweNames::kBweNamesMax);
+    uma_recorded_ = true;
+  }
+  bool delayed_feedback = true;
+  bool recovered_from_overuse = false;
+  BandwidthUsage prev_detector_state = active_delay_detector_->State();
+  for (const auto& packet_feedback : packet_feedback_vector) {
+    delayed_feedback = false;
+    IncomingPacketFeedback(packet_feedback, msg.feedback_time);
+    if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
+        active_delay_detector_->State() == BandwidthUsage::kBwNormal) {
+      recovered_from_overuse = true;
+    }
+    prev_detector_state = active_delay_detector_->State();
+  }
+
+  if (delayed_feedback) {
+    // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
+    // against building very large network queues.
+    return Result();
+  }
+  rate_control_.SetInApplicationLimitedRegion(in_alr);
+  rate_control_.SetNetworkStateEstimate(network_estimate);
+  return MaybeUpdateEstimate(acked_bitrate, probe_bitrate,
+                             std::move(network_estimate),
+                             recovered_from_overuse, in_alr, msg.feedback_time);
+}
+
+void DelayBasedBwe::IncomingPacketFeedback(const PacketResult& packet_feedback,
+                                           Timestamp at_time) {
+  // Reset if the stream has timed out.
+  if (last_seen_packet_.IsInfinite() ||
+      at_time - last_seen_packet_ > kStreamTimeOut) {
+    video_inter_arrival_delta_ =
+        std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
+    audio_inter_arrival_delta_ =
+        std::make_unique<InterArrivalDelta>(kSendTimeGroupLength);
+
+    video_delay_detector_.reset(
+        new TrendlineEstimator(key_value_config_, network_state_predictor_));
+    audio_delay_detector_.reset(
+        new TrendlineEstimator(key_value_config_, network_state_predictor_));
+    active_delay_detector_ = video_delay_detector_.get();
+  }
+  last_seen_packet_ = at_time;
+
+  // As an alternative to ignoring small packets, we can separate audio and
+  // video packets for overuse detection.
+  DelayIncreaseDetectorInterface* delay_detector_for_packet =
+      video_delay_detector_.get();
+  if (separate_audio_.enabled) {
+    if (packet_feedback.sent_packet.audio) {
+      delay_detector_for_packet = audio_delay_detector_.get();
+      audio_packets_since_last_video_++;
+      if (audio_packets_since_last_video_ > separate_audio_.packet_threshold &&
+          packet_feedback.receive_time - last_video_packet_recv_time_ >
+              separate_audio_.time_threshold) {
+        active_delay_detector_ = audio_delay_detector_.get();
+      }
+    } else {
+      audio_packets_since_last_video_ = 0;
+      last_video_packet_recv_time_ =
+          std::max(last_video_packet_recv_time_, packet_feedback.receive_time);
+      active_delay_detector_ = video_delay_detector_.get();
+    }
+  }
+  DataSize packet_size = packet_feedback.sent_packet.size;
+
+  TimeDelta send_delta = TimeDelta::Zero();
+  TimeDelta recv_delta = TimeDelta::Zero();
+  int size_delta = 0;
+
+  InterArrivalDelta* inter_arrival_for_packet =
+      (separate_audio_.enabled && packet_feedback.sent_packet.audio)
+          ? audio_inter_arrival_delta_.get()
+          : video_inter_arrival_delta_.get();
+  bool calculated_deltas = inter_arrival_for_packet->ComputeDeltas(
+      packet_feedback.sent_packet.send_time, packet_feedback.receive_time,
+      at_time, packet_size.bytes(), &send_delta, &recv_delta, &size_delta);
+
+  delay_detector_for_packet->Update(recv_delta.ms<double>(),
+                                    send_delta.ms<double>(),
+                                    packet_feedback.sent_packet.send_time.ms(),
+                                    packet_feedback.receive_time.ms(),
+                                    packet_size.bytes(), calculated_deltas);
+}
+
+DataRate DelayBasedBwe::TriggerOveruse(Timestamp at_time,
+                                       absl::optional<DataRate> link_capacity) {
+  RateControlInput input(BandwidthUsage::kBwOverusing, link_capacity);
+  return rate_control_.Update(&input, at_time);
+}
+
+DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
+    absl::optional<DataRate> acked_bitrate,
+    absl::optional<DataRate> probe_bitrate,
+    absl::optional<NetworkStateEstimate> state_estimate,
+    bool recovered_from_overuse,
+    bool in_alr,
+    Timestamp at_time) {
+  Result result;
+
+  // Currently overusing the bandwidth.
+  if (active_delay_detector_->State() == BandwidthUsage::kBwOverusing) {
+    if (acked_bitrate &&
+        rate_control_.TimeToReduceFurther(at_time, *acked_bitrate)) {
+      result.updated =
+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
+    } else if (!acked_bitrate && rate_control_.ValidEstimate() &&
+               rate_control_.InitialTimeToReduceFurther(at_time)) {
+      // Overusing before we have a measured acknowledged bitrate. Reduce send
+      // rate by 50% every 200 ms.
+      // TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
+      // so that we (almost) always have a bitrate estimate.
+      rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, at_time);
+      result.updated = true;
+      result.probe = false;
+      result.target_bitrate = rate_control_.LatestEstimate();
+    }
+  } else {
+    if (probe_bitrate) {
+      result.probe = true;
+      result.updated = true;
+      rate_control_.SetEstimate(*probe_bitrate, at_time);
+      result.target_bitrate = rate_control_.LatestEstimate();
+    } else {
+      result.updated =
+          UpdateEstimate(at_time, acked_bitrate, &result.target_bitrate);
+      result.recovered_from_overuse = recovered_from_overuse;
+    }
+  }
+  BandwidthUsage detector_state = active_delay_detector_->State();
+  if ((result.updated && prev_bitrate_ != result.target_bitrate) ||
+      detector_state != prev_state_) {
+    DataRate bitrate = result.updated ? result.target_bitrate : prev_bitrate_;
+
+    BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", at_time.ms(), bitrate.bps());
+
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventBweUpdateDelayBased>(
+          bitrate.bps(), detector_state));
+    }
+
+    prev_bitrate_ = bitrate;
+    prev_state_ = detector_state;
+  }
+
+  result.delay_detector_state = detector_state;
+  return result;
+}
+
+bool DelayBasedBwe::UpdateEstimate(Timestamp at_time,
+                                   absl::optional<DataRate> acked_bitrate,
+                                   DataRate* target_rate) {
+  const RateControlInput input(active_delay_detector_->State(), acked_bitrate);
+  *target_rate = rate_control_.Update(&input, at_time);
+  return rate_control_.ValidEstimate();
+}
+
+void DelayBasedBwe::OnRttUpdate(TimeDelta avg_rtt) {
+  rate_control_.SetRtt(avg_rtt);
+}
+
+bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
+                                   DataRate* bitrate) const {
+  // Currently accessed from both the process thread (see
+  // ModuleRtpRtcpImpl::Process()) and the configuration thread (see
+  // Call::GetStats()). Should in the future only be accessed from a single
+  // thread.
+  RTC_DCHECK(ssrcs);
+  RTC_DCHECK(bitrate);
+  if (!rate_control_.ValidEstimate())
+    return false;
+
+  *ssrcs = {kFixedSsrc};
+  *bitrate = rate_control_.LatestEstimate();
+  return true;
+}
+
+void DelayBasedBwe::SetStartBitrate(DataRate start_bitrate) {
+  RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: "
+                   << ToString(start_bitrate);
+  rate_control_.SetStartBitrate(start_bitrate);
+}
+
+void DelayBasedBwe::SetMinBitrate(DataRate min_bitrate) {
+  // Called from both the configuration thread and the network thread. Shouldn't
+  // be called from the network thread in the future.
+  rate_control_.SetMinBitrate(min_bitrate);
+}
+
+TimeDelta DelayBasedBwe::GetExpectedBwePeriod() const {
+  return rate_control_.GetExpectedBandwidthPeriod();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
new file mode 100644
index 0000000000..e91a1dff54
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
@@ -0,0 +1,133 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "modules/remote_bitrate_estimator/inter_arrival.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/race_checker.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+struct BweSeparateAudioPacketsSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-SeparateAudioPackets";
+
+  BweSeparateAudioPacketsSettings() = default;
+  explicit BweSeparateAudioPacketsSettings(
+      const FieldTrialsView* key_value_config);
+
+  bool enabled = false;
+  int packet_threshold = 10;
+  TimeDelta time_threshold = TimeDelta::Seconds(1);
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class DelayBasedBwe {
+ public:
+  struct Result {
+    Result();
+    ~Result() = default;
+    bool updated;
+    bool probe;
+    DataRate target_bitrate = DataRate::Zero();
+    bool recovered_from_overuse;
+    BandwidthUsage delay_detector_state;
+  };
+
+  explicit DelayBasedBwe(const FieldTrialsView* key_value_config,
+                         RtcEventLog* event_log,
+                         NetworkStatePredictor* network_state_predictor);
+
+  DelayBasedBwe() = delete;
+  DelayBasedBwe(const DelayBasedBwe&) = delete;
+  DelayBasedBwe& operator=(const DelayBasedBwe&) = delete;
+
+  virtual ~DelayBasedBwe();
+
+  Result IncomingPacketFeedbackVector(
+      const TransportPacketsFeedback& msg,
+      absl::optional<DataRate> acked_bitrate,
+      absl::optional<DataRate> probe_bitrate,
+      absl::optional<NetworkStateEstimate> network_estimate,
+      bool in_alr);
+  void OnRttUpdate(TimeDelta avg_rtt);
+  bool LatestEstimate(std::vector<uint32_t>* ssrcs, DataRate* bitrate) const;
+  void SetStartBitrate(DataRate start_bitrate);
+  void SetMinBitrate(DataRate min_bitrate);
+  TimeDelta GetExpectedBwePeriod() const;
+  DataRate TriggerOveruse(Timestamp at_time,
+                          absl::optional<DataRate> link_capacity);
+  DataRate last_estimate() const { return prev_bitrate_; }
+  BandwidthUsage last_state() const { return prev_state_; }
+
+ private:
+  friend class GoogCcStatePrinter;
+  void IncomingPacketFeedback(const PacketResult& packet_feedback,
+                              Timestamp at_time);
+  Result MaybeUpdateEstimate(
+      absl::optional<DataRate> acked_bitrate,
+      absl::optional<DataRate> probe_bitrate,
+      absl::optional<NetworkStateEstimate> state_estimate,
+      bool recovered_from_overuse,
+      bool in_alr,
+      Timestamp at_time);
+  // Updates the current remote rate estimate and returns true if a valid
+  // estimate exists.
+  bool UpdateEstimate(Timestamp at_time,
+                      absl::optional<DataRate> acked_bitrate,
+                      DataRate* target_rate);
+
+  rtc::RaceChecker network_race_;
+  RtcEventLog* const event_log_;
+  const FieldTrialsView* const key_value_config_;
+
+  // Alternatively, run two separate overuse detectors for audio and video,
+  // and fall back to the audio one if we haven't seen a video packet in a
+  // while.
+  BweSeparateAudioPacketsSettings separate_audio_;
+  int64_t audio_packets_since_last_video_;
+  Timestamp last_video_packet_recv_time_;
+
+  NetworkStatePredictor* network_state_predictor_;
+  std::unique_ptr<InterArrival> video_inter_arrival_;
+  std::unique_ptr<InterArrivalDelta> video_inter_arrival_delta_;
+  std::unique_ptr<DelayIncreaseDetectorInterface> video_delay_detector_;
+  std::unique_ptr<InterArrival> audio_inter_arrival_;
+  std::unique_ptr<InterArrivalDelta> audio_inter_arrival_delta_;
+  std::unique_ptr<DelayIncreaseDetectorInterface> audio_delay_detector_;
+  DelayIncreaseDetectorInterface* active_delay_detector_;
+
+  Timestamp last_seen_packet_;
+  bool uma_recorded_;
+  AimdRateControl rate_control_;
+  DataRate prev_bitrate_;
+  BandwidthUsage prev_state_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build
new file mode 100644
index 0000000000..15f5a583ef
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_gn/moz.build
@@ -0,0 +1,234 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("delay_based_bwe_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
new file mode 100644
index 0000000000..b7dc6aae47
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
@@ -0,0 +1,309 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+
+#include <string>
+
+#include "api/transport/network_types.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kNumProbesCluster0 = 5;
+constexpr int kNumProbesCluster1 = 8;
+const PacedPacketInfo kPacingInfo0(0, kNumProbesCluster0, 2000);
+const PacedPacketInfo kPacingInfo1(1, kNumProbesCluster1, 4000);
+constexpr float kTargetUtilizationFraction = 0.95f;
+}  // namespace
+
+TEST_F(DelayBasedBweTest, ProbeDetection) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(10);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo0);
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+
+  // Second burst sent at 8 * 1000 / 5 = 1600 kbps.
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GT(bitrate_observer_.latest_bitrate(), 1500000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionNonPacedPackets) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps, but with every other packet
+  // not being paced which could mess things up.
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(5);
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, now_ms, 1000, kPacingInfo0);
+    // Non-paced packet, arriving 5 ms after.
+    clock_.AdvanceTimeMilliseconds(5);
+    IncomingFeedback(now_ms, now_ms, 100, PacedPacketInfo());
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GT(bitrate_observer_.latest_bitrate(), 800000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionFasterArrival) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 10 = 800 kbps.
+  // Arriving at 8 * 1000 / 5 = 1600 kbps.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster0; ++i) {
+    clock_.AdvanceTimeMilliseconds(1);
+    send_time_ms += 10;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo0);
+  }
+
+  EXPECT_FALSE(bitrate_observer_.updated());
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrival) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // First burst sent at 8 * 1000 / 5 = 1600 kbps.
+  // Arriving at 8 * 1000 / 7 = 1142 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(7);
+    send_time_ms += 5;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 1140000u, 10000u);
+}
+
+TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrivalHighBitrate) {
+  int64_t now_ms = clock_.TimeInMilliseconds();
+  // Burst sent at 8 * 1000 / 1 = 8000 kbps.
+  // Arriving at 8 * 1000 / 2 = 4000 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
+  int64_t send_time_ms = 0;
+  for (int i = 0; i < kNumProbesCluster1; ++i) {
+    clock_.AdvanceTimeMilliseconds(2);
+    send_time_ms += 1;
+    now_ms = clock_.TimeInMilliseconds();
+    IncomingFeedback(now_ms, send_time_ms, 1000, kPacingInfo1);
+  }
+
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 4000000u, 10000u);
+}
+
+TEST_F(DelayBasedBweTest, GetExpectedBwePeriodMs) {
+  auto default_interval = bitrate_estimator_->GetExpectedBwePeriod();
+  EXPECT_GT(default_interval.ms(), 0);
+  CapacityDropTestHelper(1, true, 333, 0);
+  auto interval = bitrate_estimator_->GetExpectedBwePeriod();
+  EXPECT_GT(interval.ms(), 0);
+  EXPECT_NE(interval.ms(), default_interval.ms());
+}
+
+TEST_F(DelayBasedBweTest, InitialBehavior) {
+  InitialBehaviorTestHelper(730000);
+}
+
+TEST_F(DelayBasedBweTest, RateIncreaseReordering) {
+  RateIncreaseReorderingTestHelper(730000);
+}
+TEST_F(DelayBasedBweTest, RateIncreaseRtpTimestamps) {
+  RateIncreaseRtpTimestampsTestHelper(622);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropOneStream) {
+  CapacityDropTestHelper(1, false, 300, 0);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropPosOffsetChange) {
+  CapacityDropTestHelper(1, false, 867, 30000);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropNegOffsetChange) {
+  CapacityDropTestHelper(1, false, 933, -30000);
+}
+
+TEST_F(DelayBasedBweTest, CapacityDropOneStreamWrap) {
+  CapacityDropTestHelper(1, true, 333, 0);
+}
+
+TEST_F(DelayBasedBweTest, TestTimestampGrouping) {
+  TestTimestampGroupingTestHelper();
+}
+
+TEST_F(DelayBasedBweTest, TestShortTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after 35 seconds. This
+  // will make abs send time wrap, so if streams aren't timed out properly
+  // the next 30 seconds of packets will be out of order.
+  TestWrappingHelper(35);
+}
+
+TEST_F(DelayBasedBweTest, TestLongTimeoutAndWrap) {
+  // Simulate a client leaving and rejoining the call after some multiple of
+  // 64 seconds later. This will cause a zero difference in abs send times due
+  // to the wrap, but a big difference in arrival time, if streams aren't
+  // properly timed out.
+  TestWrappingHelper(10 * 64);
+}
+
+TEST_F(DelayBasedBweTest, TestInitialOveruse) {
+  const DataRate kStartBitrate = DataRate::KilobitsPerSec(300);
+  const DataRate kInitialCapacity = DataRate::KilobitsPerSec(200);
+  const uint32_t kDummySsrc = 0;
+  // High FPS to ensure that we send a lot of packets in a short time.
+  const int kFps = 90;
+
+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate.bps()));
+  stream_generator_->set_capacity_bps(kInitialCapacity.bps());
+
+  // Needed to initialize the AimdRateControl.
+  bitrate_estimator_->SetStartBitrate(kStartBitrate);
+
+  // Produce 30 frames (in 1/3 second) and give them to the estimator.
+  int64_t bitrate_bps = kStartBitrate.bps();
+  bool seen_overuse = false;
+  for (int i = 0; i < 30; ++i) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    // The purpose of this test is to ensure that we back down even if we don't
+    // have any acknowledged bitrate estimate yet. Hence, if the test works
+    // as expected, we should not have a measured bitrate yet.
+    EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate().has_value());
+    if (overuse) {
+      EXPECT_TRUE(bitrate_observer_.updated());
+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
+                  15000);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      seen_overuse = true;
+      break;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+  }
+  EXPECT_TRUE(seen_overuse);
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
+              15000);
+}
+
+TEST_F(DelayBasedBweTest, TestTimestampPrecisionHandling) {
+  // This test does some basic checks to make sure that timestamps with higher
+  // than millisecond precision are handled properly and do not cause any
+  // problems in the estimator. Specifically, previously reported in
+  // webrtc:14023 and described in more details there, the rounding to the
+  // nearest milliseconds caused discrepancy in the accumulated delay. This lead
+  // to false-positive overuse detection.
+  // Technical details of the test:
+  // Send times(ms): 0.000,  9.725, 20.000, 29.725, 40.000, 49.725, ...
+  // Recv times(ms): 0.500, 10.000, 20.500, 30.000, 40.500, 50.000, ...
+  // Send deltas(ms):   9.750,  10.250,  9.750, 10.250,  9.750, ...
+  // Recv deltas(ms):   9.500,  10.500,  9.500, 10.500,  9.500, ...
+  // There is no delay building up between the send times and the receive times,
+  // therefore this case should never lead to an overuse detection. However, if
+  // the time deltas were accidentally rounded to the nearest milliseconds, then
+  // all the send deltas would be equal to 10ms while some recv deltas would
+  // round up to 11ms which would lead in a false illusion of delay build up.
+  uint32_t last_bitrate = bitrate_observer_.latest_bitrate();
+  for (int i = 0; i < 1000; ++i) {
+    clock_.AdvanceTimeMicroseconds(500);
+    IncomingFeedback(clock_.CurrentTime(),
+                     clock_.CurrentTime() - TimeDelta::Micros(500), 1000,
+                     PacedPacketInfo());
+    clock_.AdvanceTimeMicroseconds(9500);
+    IncomingFeedback(clock_.CurrentTime(),
+                     clock_.CurrentTime() - TimeDelta::Micros(250), 1000,
+                     PacedPacketInfo());
+    clock_.AdvanceTimeMicroseconds(10000);
+
+    // The bitrate should never decrease in this test.
+    EXPECT_LE(last_bitrate, bitrate_observer_.latest_bitrate());
+    last_bitrate = bitrate_observer_.latest_bitrate();
+  }
+}
+
+class DelayBasedBweTestWithBackoffTimeoutExperiment : public DelayBasedBweTest {
+ public:
+  DelayBasedBweTestWithBackoffTimeoutExperiment()
+      : DelayBasedBweTest(
+            "WebRTC-BweAimdRateControlConfig/initial_backoff_interval:200ms/") {
+  }
+};
+
+// This test subsumes and improves DelayBasedBweTest.TestInitialOveruse above.
+TEST_F(DelayBasedBweTestWithBackoffTimeoutExperiment, TestInitialOveruse) {
+  const DataRate kStartBitrate = DataRate::KilobitsPerSec(300);
+  const DataRate kInitialCapacity = DataRate::KilobitsPerSec(200);
+  const uint32_t kDummySsrc = 0;
+  // High FPS to ensure that we send a lot of packets in a short time.
+  const int kFps = 90;
+
+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate.bps()));
+  stream_generator_->set_capacity_bps(kInitialCapacity.bps());
+
+  // Needed to initialize the AimdRateControl.
+  bitrate_estimator_->SetStartBitrate(kStartBitrate);
+
+  // Produce 30 frames (in 1/3 second) and give them to the estimator.
+  int64_t bitrate_bps = kStartBitrate.bps();
+  bool seen_overuse = false;
+  for (int frames = 0; frames < 30 && !seen_overuse; ++frames) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    // The purpose of this test is to ensure that we back down even if we don't
+    // have any acknowledged bitrate estimate yet. Hence, if the test works
+    // as expected, we should not have a measured bitrate yet.
+    EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate().has_value());
+    if (overuse) {
+      EXPECT_TRUE(bitrate_observer_.updated());
+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate.bps() / 2,
+                  15000);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      seen_overuse = true;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+  }
+  EXPECT_TRUE(seen_overuse);
+  // Continue generating an additional 15 frames (equivalent to 167 ms) and
+  // verify that we don't back down further.
+  for (int frames = 0; frames < 15 && seen_overuse; ++frames) {
+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
+    EXPECT_FALSE(overuse);
+    if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      EXPECT_GE(bitrate_bps, kStartBitrate.bps() / 2 - 15000);
+      EXPECT_LE(bitrate_bps, kInitialCapacity.bps() + 15000);
+      bitrate_observer_.Reset();
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
new file mode 100644
index 0000000000..8618a7814e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
@@ -0,0 +1,529 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+constexpr size_t kMtu = 1200;
+constexpr uint32_t kAcceptedBitrateErrorBps = 50000;
+
+// Number of packets needed before we have a valid estimate.
+constexpr int kNumInitialPackets = 2;
+
+constexpr int kInitialProbingPackets = 5;
+
+namespace test {
+
+void TestBitrateObserver::OnReceiveBitrateChanged(uint32_t bitrate) {
+  latest_bitrate_ = bitrate;
+  updated_ = true;
+}
+
+RtpStream::RtpStream(int fps, int bitrate_bps)
+    : fps_(fps), bitrate_bps_(bitrate_bps), next_rtp_time_(0) {
+  RTC_CHECK_GT(fps_, 0);
+}
+
+// Generates a new frame for this stream. If called too soon after the
+// previous frame, no frame will be generated. The frame is split into
+// packets.
+int64_t RtpStream::GenerateFrame(int64_t time_now_us,
+                                 std::vector<PacketResult>* packets) {
+  if (time_now_us < next_rtp_time_) {
+    return next_rtp_time_;
+  }
+  RTC_CHECK(packets != NULL);
+  size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
+  size_t n_packets =
+      std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
+  size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
+  for (size_t i = 0; i < n_packets; ++i) {
+    PacketResult packet;
+    packet.sent_packet.send_time =
+        Timestamp::Micros(time_now_us + kSendSideOffsetUs);
+    packet.sent_packet.size = DataSize::Bytes(payload_size);
+    packets->push_back(packet);
+  }
+  next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
+  return next_rtp_time_;
+}
+
+// The send-side time when the next frame can be generated.
+int64_t RtpStream::next_rtp_time() const {
+  return next_rtp_time_;
+}
+
+void RtpStream::set_bitrate_bps(int bitrate_bps) {
+  ASSERT_GE(bitrate_bps, 0);
+  bitrate_bps_ = bitrate_bps;
+}
+
+int RtpStream::bitrate_bps() const {
+  return bitrate_bps_;
+}
+
+bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
+                        const std::unique_ptr<RtpStream>& rhs) {
+  return lhs->next_rtp_time_ < rhs->next_rtp_time_;
+}
+
+StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
+    : capacity_(capacity), prev_arrival_time_us_(time_now) {}
+
+StreamGenerator::~StreamGenerator() = default;
+
+// Add a new stream.
+void StreamGenerator::AddStream(RtpStream* stream) {
+  streams_.push_back(std::unique_ptr<RtpStream>(stream));
+}
+
+// Set the link capacity.
+void StreamGenerator::set_capacity_bps(int capacity_bps) {
+  ASSERT_GT(capacity_bps, 0);
+  capacity_ = capacity_bps;
+}
+
+// Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+// is decided by the current allocation ratios.
+void StreamGenerator::SetBitrateBps(int bitrate_bps) {
+  ASSERT_GE(streams_.size(), 0u);
+  int total_bitrate_before = 0;
+  for (const auto& stream : streams_) {
+    total_bitrate_before += stream->bitrate_bps();
+  }
+  int64_t bitrate_before = 0;
+  int total_bitrate_after = 0;
+  for (const auto& stream : streams_) {
+    bitrate_before += stream->bitrate_bps();
+    int64_t bitrate_after =
+        (bitrate_before * bitrate_bps + total_bitrate_before / 2) /
+        total_bitrate_before;
+    stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
+    total_bitrate_after += stream->bitrate_bps();
+  }
+  ASSERT_EQ(bitrate_before, total_bitrate_before);
+  EXPECT_EQ(total_bitrate_after, bitrate_bps);
+}
+
+// TODO(holmer): Break out the channel simulation part from this class to make
+// it possible to simulate different types of channels.
+int64_t StreamGenerator::GenerateFrame(std::vector<PacketResult>* packets,
+                                       int64_t time_now_us) {
+  RTC_CHECK(packets != NULL);
+  RTC_CHECK(packets->empty());
+  RTC_CHECK_GT(capacity_, 0);
+  auto it =
+      std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  (*it)->GenerateFrame(time_now_us, packets);
+  for (PacketResult& packet : *packets) {
+    int capacity_bpus = capacity_ / 1000;
+    int64_t required_network_time_us =
+        (8 * 1000 * packet.sent_packet.size.bytes() + capacity_bpus / 2) /
+        capacity_bpus;
+    prev_arrival_time_us_ =
+        std::max(time_now_us + required_network_time_us,
+                 prev_arrival_time_us_ + required_network_time_us);
+    packet.receive_time = Timestamp::Micros(prev_arrival_time_us_);
+  }
+  it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
+  return std::max((*it)->next_rtp_time(), time_now_us);
+}
+}  // namespace test
+
+DelayBasedBweTest::DelayBasedBweTest() : DelayBasedBweTest("") {}
+
+DelayBasedBweTest::DelayBasedBweTest(absl::string_view field_trial_string)
+    : field_trial(
+          std::make_unique<test::ScopedFieldTrials>(field_trial_string)),
+      clock_(100000000),
+      acknowledged_bitrate_estimator_(
+          AcknowledgedBitrateEstimatorInterface::Create(&field_trial_config_)),
+      probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)),
+      bitrate_estimator_(
+          new DelayBasedBwe(&field_trial_config_, nullptr, nullptr)),
+      stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
+                                                  clock_.TimeInMicroseconds())),
+      arrival_time_offset_ms_(0),
+      first_update_(true) {}
+
+DelayBasedBweTest::~DelayBasedBweTest() {}
+
+void DelayBasedBweTest::AddDefaultStream() {
+  stream_generator_->AddStream(new test::RtpStream(30, 3e5));
+}
+
+const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+                                         int64_t send_time_ms,
+                                         size_t payload_size) {
+  IncomingFeedback(arrival_time_ms, send_time_ms, payload_size,
+                   PacedPacketInfo());
+}
+
+void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
+                                         int64_t send_time_ms,
+                                         size_t payload_size,
+                                         const PacedPacketInfo& pacing_info) {
+  RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
+  IncomingFeedback(Timestamp::Millis(arrival_time_ms + arrival_time_offset_ms_),
+                   Timestamp::Millis(send_time_ms), payload_size, pacing_info);
+}
+
+void DelayBasedBweTest::IncomingFeedback(Timestamp receive_time,
+                                         Timestamp send_time,
+                                         size_t payload_size,
+                                         const PacedPacketInfo& pacing_info) {
+  PacketResult packet;
+  packet.receive_time = receive_time;
+  packet.sent_packet.send_time = send_time;
+  packet.sent_packet.size = DataSize::Bytes(payload_size);
+  packet.sent_packet.pacing_info = pacing_info;
+  if (packet.sent_packet.pacing_info.probe_cluster_id !=
+      PacedPacketInfo::kNotAProbe)
+    probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
+
+  TransportPacketsFeedback msg;
+  msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
+  msg.packet_feedbacks.push_back(packet);
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
+      msg.SortedByReceiveTime());
+  DelayBasedBwe::Result result =
+      bitrate_estimator_->IncomingPacketFeedbackVector(
+          msg, acknowledged_bitrate_estimator_->bitrate(),
+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
+          /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+  if (result.updated) {
+    bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
+  }
+}
+
+// Generates a frame of packets belonging to a stream at a given bitrate and
+// with a given ssrc. The stream is pushed through a very simple simulated
+// network, and is then given to the receive-side bandwidth estimator.
+// Returns true if an over-use was seen, false otherwise.
+// The StreamGenerator::updated() should be used to check for any changes in
+// target bitrate after the call to this function.
+bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
+                                                uint32_t bitrate_bps) {
+  stream_generator_->SetBitrateBps(bitrate_bps);
+  std::vector<PacketResult> packets;
+
+  int64_t next_time_us =
+      stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
+  if (packets.empty())
+    return false;
+
+  bool overuse = false;
+  bitrate_observer_.Reset();
+  clock_.AdvanceTimeMicroseconds(packets.back().receive_time.us() -
+                                 clock_.TimeInMicroseconds());
+  for (auto& packet : packets) {
+    RTC_CHECK_GE(packet.receive_time.ms() + arrival_time_offset_ms_, 0);
+    packet.receive_time += TimeDelta::Millis(arrival_time_offset_ms_);
+
+    if (packet.sent_packet.pacing_info.probe_cluster_id !=
+        PacedPacketInfo::kNotAProbe)
+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
+  }
+
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
+  TransportPacketsFeedback msg;
+  msg.packet_feedbacks = packets;
+  msg.feedback_time = Timestamp::Millis(clock_.TimeInMilliseconds());
+
+  DelayBasedBwe::Result result =
+      bitrate_estimator_->IncomingPacketFeedbackVector(
+          msg, acknowledged_bitrate_estimator_->bitrate(),
+          probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(),
+          /*network_estimate*/ absl::nullopt, /*in_alr*/ false);
+  if (result.updated) {
+    bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps());
+    if (!first_update_ && result.target_bitrate.bps() < bitrate_bps)
+      overuse = true;
+    first_update_ = false;
+  }
+
+  clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
+  return overuse;
+}
+
+// Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+// until it reaches `target_bitrate`.
+// Can for instance be used to run the estimator for some time to get it
+// into a steady state.
+uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
+                                           int max_number_of_frames,
+                                           uint32_t start_bitrate,
+                                           uint32_t min_bitrate,
+                                           uint32_t max_bitrate,
+                                           uint32_t target_bitrate) {
+  uint32_t bitrate_bps = start_bitrate;
+  bool bitrate_update_seen = false;
+  // Produce `number_of_frames` frames and give them to the estimator.
+  for (int i = 0; i < max_number_of_frames; ++i) {
+    bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate);
+      EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_update_seen = true;
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+    if (bitrate_update_seen && bitrate_bps > target_bitrate) {
+      break;
+    }
+  }
+  EXPECT_TRUE(bitrate_update_seen);
+  return bitrate_bps;
+}
+
+void DelayBasedBweTest::InitialBehaviorTestHelper(
+    uint32_t expected_converge_bitrate) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const PacedPacketInfo kPacingInfo(0, 5, 5000);
+  DataRate bitrate = DataRate::Zero();
+  int64_t send_time_ms = 0;
+  std::vector<uint32_t> ssrcs;
+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  EXPECT_EQ(0u, ssrcs.size());
+  clock_.AdvanceTimeMilliseconds(1000);
+  EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  EXPECT_FALSE(bitrate_observer_.updated());
+  bitrate_observer_.Reset();
+  clock_.AdvanceTimeMilliseconds(1000);
+  // Inserting packets for 5 seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    // NOTE!!! If the following line is moved under the if case then this test
+    //         wont work on windows realease bots.
+    PacedPacketInfo pacing_info =
+        i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
+
+    if (i == kNumInitialPackets) {
+      EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+      EXPECT_EQ(0u, ssrcs.size());
+      EXPECT_FALSE(bitrate_observer_.updated());
+      bitrate_observer_.Reset();
+    }
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
+                     pacing_info);
+    clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate));
+  ASSERT_EQ(1u, ssrcs.size());
+  EXPECT_EQ(kDefaultSsrc, ssrcs.front());
+  EXPECT_NEAR(expected_converge_bitrate, bitrate.bps(),
+              kAcceptedBitrateErrorBps);
+  EXPECT_TRUE(bitrate_observer_.updated());
+  bitrate_observer_.Reset();
+  EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate.bps());
+}
+
+void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
+    uint32_t expected_bitrate_bps) {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  const PacedPacketInfo kPacingInfo(0, 5, 5000);
+  int64_t send_time_ms = 0;
+  // Inserting packets for five seconds to get a valid estimate.
+  for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
+    // NOTE!!! If the following line is moved under the if case then this test
+    //         wont work on windows realease bots.
+    PacedPacketInfo pacing_info =
+        i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
+
+    // TODO(sprang): Remove this hack once the single stream estimator is gone,
+    // as it doesn't do anything in Process().
+    if (i == kNumInitialPackets) {
+      // Process after we have enough frames to get a valid input rate estimate.
+
+      EXPECT_FALSE(bitrate_observer_.updated());  // No valid estimate.
+    }
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu,
+                     pacing_info);
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+  for (int i = 0; i < 10; ++i) {
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    send_time_ms += 2 * kFrameIntervalMs;
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    IncomingFeedback(clock_.TimeInMilliseconds(),
+                     send_time_ms - kFrameIntervalMs, 1000);
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
+              kAcceptedBitrateErrorBps);
+}
+
+// Make sure we initially increase the bitrate as expected.
+void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
+    int expected_iterations) {
+  // This threshold corresponds approximately to increasing linearly with
+  // bitrate(i) = 1.04 * bitrate(i-1) + 1000
+  // until bitrate(i) > 500000, with bitrate(1) ~= 30000.
+  uint32_t bitrate_bps = 30000;
+  int iterations = 0;
+  AddDefaultStream();
+  // Feed the estimator with a stream of packets and verify that it reaches
+  // 500 kbps at the expected time.
+  while (bitrate_bps < 5e5) {
+    bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (overuse) {
+      EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    } else if (bitrate_observer_.updated()) {
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+      bitrate_observer_.Reset();
+    }
+    ++iterations;
+  }
+  ASSERT_EQ(expected_iterations, iterations);
+}
+
+void DelayBasedBweTest::CapacityDropTestHelper(
+    int number_of_streams,
+    bool wrap_time_stamp,
+    uint32_t expected_bitrate_drop_delta,
+    int64_t receiver_clock_offset_change_ms) {
+  const int kFramerate = 30;
+  const int kStartBitrate = 900e3;
+  const int kMinExpectedBitrate = 800e3;
+  const int kMaxExpectedBitrate = 1100e3;
+  const uint32_t kInitialCapacityBps = 1000e3;
+  const uint32_t kReducedCapacityBps = 500e3;
+
+  int steady_state_time = 0;
+  if (number_of_streams <= 1) {
+    steady_state_time = 10;
+    AddDefaultStream();
+  } else {
+    steady_state_time = 10 * number_of_streams;
+    int bitrate_sum = 0;
+    int kBitrateDenom = number_of_streams * (number_of_streams - 1);
+    for (int i = 0; i < number_of_streams; i++) {
+      // First stream gets half available bitrate, while the rest share the
+      // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
+      int bitrate = kStartBitrate / 2;
+      if (i > 0) {
+        bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
+      }
+      stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
+      bitrate_sum += bitrate;
+    }
+    ASSERT_EQ(bitrate_sum, kStartBitrate);
+  }
+
+  // Run in steady state to make the estimator converge.
+  stream_generator_->set_capacity_bps(kInitialCapacityBps);
+  uint32_t bitrate_bps = SteadyStateRun(
+      kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
+      kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
+  EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u);
+  bitrate_observer_.Reset();
+
+  // Add an offset to make sure the BWE can handle it.
+  arrival_time_offset_ms_ += receiver_clock_offset_change_ms;
+
+  // Reduce the capacity and verify the decrease time.
+  stream_generator_->set_capacity_bps(kReducedCapacityBps);
+  int64_t overuse_start_time = clock_.TimeInMilliseconds();
+  int64_t bitrate_drop_time = -1;
+  for (int i = 0; i < 100 * number_of_streams; ++i) {
+    GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
+    if (bitrate_drop_time == -1 &&
+        bitrate_observer_.latest_bitrate() <= kReducedCapacityBps) {
+      bitrate_drop_time = clock_.TimeInMilliseconds();
+    }
+    if (bitrate_observer_.updated())
+      bitrate_bps = bitrate_observer_.latest_bitrate();
+  }
+
+  EXPECT_NEAR(expected_bitrate_drop_delta,
+              bitrate_drop_time - overuse_start_time, 33);
+}
+
+void DelayBasedBweTest::TestTimestampGroupingTestHelper() {
+  const int kFramerate = 50;  // 50 fps to avoid rounding errors.
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  int64_t send_time_ms = 0;
+  // Initial set of frames to increase the bitrate. 6 seconds to have enough
+  // time for the first estimate to be generated and for Process() to be called.
+  for (int i = 0; i <= 6 * kFramerate; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  EXPECT_GE(bitrate_observer_.latest_bitrate(), 400000u);
+
+  // Insert batches of frames which were sent very close in time. Also simulate
+  // capacity over-use to see that we back off correctly.
+  const int kTimestampGroupLength = 15;
+  for (int i = 0; i < 100; ++i) {
+    for (int j = 0; j < kTimestampGroupLength; ++j) {
+      // Insert `kTimestampGroupLength` frames with just 1 timestamp ticks in
+      // between. Should be treated as part of the same group by the estimator.
+      IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 100);
+      clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
+      send_time_ms += 1;
+    }
+    // Increase time until next batch to simulate over-use.
+    clock_.AdvanceTimeMilliseconds(10);
+    send_time_ms += kFrameIntervalMs - kTimestampGroupLength;
+  }
+  EXPECT_TRUE(bitrate_observer_.updated());
+  // Should have reduced the estimate.
+  EXPECT_LT(bitrate_observer_.latest_bitrate(), 400000u);
+}
+
+void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
+  const int kFramerate = 100;
+  const int kFrameIntervalMs = 1000 / kFramerate;
+  int64_t send_time_ms = 0;
+
+  for (size_t i = 0; i < 3000; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  DataRate bitrate_before = DataRate::Zero();
+  std::vector<uint32_t> ssrcs;
+  bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);
+
+  clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
+  send_time_ms += silence_time_s * 1000;
+
+  for (size_t i = 0; i < 24; ++i) {
+    IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000);
+    clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
+    send_time_ms += kFrameIntervalMs;
+  }
+  DataRate bitrate_after = DataRate::Zero();
+  bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
+  EXPECT_LT(bitrate_after, bitrate_before);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
new file mode 100644
index 0000000000..d56fe892d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
@@ -0,0 +1,189 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_types.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace test {
+
+class TestBitrateObserver {
+ public:
+  TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
+  ~TestBitrateObserver() {}
+
+  void OnReceiveBitrateChanged(uint32_t bitrate);
+
+  void Reset() { updated_ = false; }
+
+  bool updated() const { return updated_; }
+
+  uint32_t latest_bitrate() const { return latest_bitrate_; }
+
+ private:
+  bool updated_;
+  uint32_t latest_bitrate_;
+};
+
+class RtpStream {
+ public:
+  enum { kSendSideOffsetUs = 1000000 };
+
+  RtpStream(int fps, int bitrate_bps);
+
+  RtpStream(const RtpStream&) = delete;
+  RtpStream& operator=(const RtpStream&) = delete;
+
+  // Generates a new frame for this stream. If called too soon after the
+  // previous frame, no frame will be generated. The frame is split into
+  // packets.
+  int64_t GenerateFrame(int64_t time_now_us,
+                        std::vector<PacketResult>* packets);
+
+  // The send-side time when the next frame can be generated.
+  int64_t next_rtp_time() const;
+
+  void set_bitrate_bps(int bitrate_bps);
+
+  int bitrate_bps() const;
+
+  static bool Compare(const std::unique_ptr<RtpStream>& lhs,
+                      const std::unique_ptr<RtpStream>& rhs);
+
+ private:
+  int fps_;
+  int bitrate_bps_;
+  int64_t next_rtp_time_;
+};
+
+class StreamGenerator {
+ public:
+  StreamGenerator(int capacity, int64_t time_now);
+  ~StreamGenerator();
+
+  StreamGenerator(const StreamGenerator&) = delete;
+  StreamGenerator& operator=(const StreamGenerator&) = delete;
+
+  // Add a new stream.
+  void AddStream(RtpStream* stream);
+
+  // Set the link capacity.
+  void set_capacity_bps(int capacity_bps);
+
+  // Divides `bitrate_bps` among all streams. The allocated bitrate per stream
+  // is decided by the initial allocation ratios.
+  void SetBitrateBps(int bitrate_bps);
+
+  // Set the RTP timestamp offset for the stream identified by `ssrc`.
+  void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
+
+  // TODO(holmer): Break out the channel simulation part from this class to make
+  // it possible to simulate different types of channels.
+  int64_t GenerateFrame(std::vector<PacketResult>* packets,
+                        int64_t time_now_us);
+
+ private:
+  // Capacity of the simulated channel in bits per second.
+  int capacity_;
+  // The time when the last packet arrived.
+  int64_t prev_arrival_time_us_;
+  // All streams being transmitted on this simulated channel.
+  std::vector<std::unique_ptr<RtpStream>> streams_;
+};
+}  // namespace test
+
+class DelayBasedBweTest : public ::testing::Test {
+ public:
+  DelayBasedBweTest();
+  explicit DelayBasedBweTest(absl::string_view field_trial_string);
+  ~DelayBasedBweTest() override;
+
+ protected:
+  void AddDefaultStream();
+
+  // Helpers to insert a single packet into the delay-based BWE.
+  void IncomingFeedback(int64_t arrival_time_ms,
+                        int64_t send_time_ms,
+                        size_t payload_size);
+  void IncomingFeedback(int64_t arrival_time_ms,
+                        int64_t send_time_ms,
+                        size_t payload_size,
+                        const PacedPacketInfo& pacing_info);
+  void IncomingFeedback(Timestamp receive_time,
+                        Timestamp send_time,
+                        size_t payload_size,
+                        const PacedPacketInfo& pacing_info);
+
+  // Generates a frame of packets belonging to a stream at a given bitrate and
+  // with a given ssrc. The stream is pushed through a very simple simulated
+  // network, and is then given to the receive-side bandwidth estimator.
+  // Returns true if an over-use was seen, false otherwise.
+  // The StreamGenerator::updated() should be used to check for any changes in
+  // target bitrate after the call to this function.
+  bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);
+
+  // Run the bandwidth estimator with a stream of `number_of_frames` frames, or
+  // until it reaches `target_bitrate`.
+  // Can for instance be used to run the estimator for some time to get it
+  // into a steady state.
+  uint32_t SteadyStateRun(uint32_t ssrc,
+                          int number_of_frames,
+                          uint32_t start_bitrate,
+                          uint32_t min_bitrate,
+                          uint32_t max_bitrate,
+                          uint32_t target_bitrate);
+
+  void TestTimestampGroupingTestHelper();
+
+  void TestWrappingHelper(int silence_time_s);
+
+  void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
+  void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
+  void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
+  void CapacityDropTestHelper(int number_of_streams,
+                              bool wrap_time_stamp,
+                              uint32_t expected_bitrate_drop_delta,
+                              int64_t receiver_clock_offset_change_ms);
+
+  static const uint32_t kDefaultSsrc;
+  FieldTrialBasedConfig field_trial_config_;
+
+  std::unique_ptr<test::ScopedFieldTrials>
+      field_trial;        // Must be initialized first.
+  SimulatedClock clock_;  // Time at the receiver.
+  test::TestBitrateObserver bitrate_observer_;
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator_;
+  const std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
+  std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
+  std::unique_ptr<test::StreamGenerator> stream_generator_;
+  int64_t arrival_time_offset_ms_;
+  bool first_update_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h
new file mode 100644
index 0000000000..fc12cff7d5
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
+
+#include <stdint.h>
+
+#include "api/network_state_predictor.h"
+
+namespace webrtc {
+
+class DelayIncreaseDetectorInterface {
+ public:
+  DelayIncreaseDetectorInterface() {}
+  virtual ~DelayIncreaseDetectorInterface() {}
+
+  DelayIncreaseDetectorInterface(const DelayIncreaseDetectorInterface&) =
+      delete;
+  DelayIncreaseDetectorInterface& operator=(
+      const DelayIncreaseDetectorInterface&) = delete;
+
+  // Update the detector with a new sample. The deltas should represent deltas
+  // between timestamp groups as defined by the InterArrival class.
+  virtual void Update(double recv_delta_ms,
+                      double send_delta_ms,
+                      int64_t send_time_ms,
+                      int64_t arrival_time_ms,
+                      size_t packet_size,
+                      bool calculated_deltas) = 0;
+
+  virtual BandwidthUsage State() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build
new file mode 100644
index 0000000000..b15062af41
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/estimators_gn/moz.build
@@ -0,0 +1,238 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("estimators_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build
new file mode 100644
index 0000000000..973d4bf32f
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("goog_cc_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
new file mode 100644
index 0000000000..3a9de8c4dc
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
@@ -0,0 +1,725 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
+
+#include <inttypes.h>
+#include <stdio.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <numeric>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/network_state_predictor.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_remote_estimate.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+// From RTCPSender video report interval.
+constexpr TimeDelta kLossUpdateInterval = TimeDelta::Millis(1000);
+
+// Pacing-rate relative to our target send rate.
+// Multiplicative factor that is applied to the target bitrate to calculate
+// the number of bytes that can be transmitted per interval.
+// Increasing this factor will result in lower delays in cases of bitrate
+// overshoots from the encoder.
+constexpr float kDefaultPaceMultiplier = 2.5f;
+
+// If the probe result is far below the current throughput estimate
+// it's unlikely that the probe is accurate, so we don't want to drop too far.
+// However, if we actually are overusing, we want to drop to something slightly
+// below the current throughput estimate to drain the network queues.
+constexpr double kProbeDropThroughputFraction = 0.85;
+
+bool IsEnabled(const FieldTrialsView* config, absl::string_view key) {
+  return absl::StartsWith(config->Lookup(key), "Enabled");
+}
+
+bool IsNotDisabled(const FieldTrialsView* config, absl::string_view key) {
+  return !absl::StartsWith(config->Lookup(key), "Disabled");
+}
+
+BandwidthLimitedCause GetBandwidthLimitedCause(
+    LossBasedState loss_based_state,
+    BandwidthUsage bandwidth_usage,
+    bool not_probe_if_delay_increased) {
+  if (not_probe_if_delay_increased &&
+      (bandwidth_usage == BandwidthUsage::kBwOverusing ||
+       bandwidth_usage == BandwidthUsage::kBwUnderusing)) {
+    return BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased;
+  }
+  switch (loss_based_state) {
+    case LossBasedState::kDecreasing:
+      return BandwidthLimitedCause::kLossLimitedBweDecreasing;
+    case LossBasedState::kIncreasing:
+      return BandwidthLimitedCause::kLossLimitedBweIncreasing;
+    default:
+      return BandwidthLimitedCause::kDelayBasedLimited;
+  }
+}
+
+}  // namespace
+
+GoogCcNetworkController::GoogCcNetworkController(NetworkControllerConfig config,
+                                                 GoogCcConfig goog_cc_config)
+    : key_value_config_(config.key_value_config ? config.key_value_config
+                                                : &trial_based_config_),
+      event_log_(config.event_log),
+      packet_feedback_only_(goog_cc_config.feedback_only),
+      safe_reset_on_route_change_("Enabled"),
+      safe_reset_acknowledged_rate_("ack"),
+      use_min_allocatable_as_lower_bound_(
+          IsNotDisabled(key_value_config_, "WebRTC-Bwe-MinAllocAsLowerBound")),
+      ignore_probes_lower_than_network_estimate_(IsNotDisabled(
+          key_value_config_,
+          "WebRTC-Bwe-IgnoreProbesLowerThanNetworkStateEstimate")),
+      limit_probes_lower_than_throughput_estimate_(
+          IsEnabled(key_value_config_,
+                    "WebRTC-Bwe-LimitProbesLowerThanThroughputEstimate")),
+      rate_control_settings_(
+          RateControlSettings::ParseFromKeyValueConfig(key_value_config_)),
+      pace_at_max_of_bwe_and_lower_link_capacity_(
+          IsEnabled(key_value_config_,
+                    "WebRTC-Bwe-PaceAtMaxOfBweAndLowerLinkCapacity")),
+      probe_controller_(
+          new ProbeController(key_value_config_, config.event_log)),
+      congestion_window_pushback_controller_(
+          rate_control_settings_.UseCongestionWindowPushback()
+              ? std::make_unique<CongestionWindowPushbackController>(
+                    key_value_config_)
+              : nullptr),
+      bandwidth_estimation_(
+          std::make_unique<SendSideBandwidthEstimation>(key_value_config_,
+                                                        event_log_)),
+      alr_detector_(
+          std::make_unique<AlrDetector>(key_value_config_, config.event_log)),
+      probe_bitrate_estimator_(new ProbeBitrateEstimator(config.event_log)),
+      network_estimator_(std::move(goog_cc_config.network_state_estimator)),
+      network_state_predictor_(
+          std::move(goog_cc_config.network_state_predictor)),
+      delay_based_bwe_(new DelayBasedBwe(key_value_config_,
+                                         event_log_,
+                                         network_state_predictor_.get())),
+      acknowledged_bitrate_estimator_(
+          AcknowledgedBitrateEstimatorInterface::Create(key_value_config_)),
+      initial_config_(config),
+      last_loss_based_target_rate_(*config.constraints.starting_rate),
+      last_pushback_target_rate_(last_loss_based_target_rate_),
+      last_stable_target_rate_(last_loss_based_target_rate_),
+      pacing_factor_(config.stream_based_config.pacing_factor.value_or(
+          kDefaultPaceMultiplier)),
+      min_total_allocated_bitrate_(
+          config.stream_based_config.min_total_allocated_bitrate.value_or(
+              DataRate::Zero())),
+      max_padding_rate_(config.stream_based_config.max_padding_rate.value_or(
+          DataRate::Zero())) {
+  RTC_DCHECK(config.constraints.at_time.IsFinite());
+  ParseFieldTrial(
+      {&safe_reset_on_route_change_, &safe_reset_acknowledged_rate_},
+      key_value_config_->Lookup("WebRTC-Bwe-SafeResetOnRouteChange"));
+  if (delay_based_bwe_)
+    delay_based_bwe_->SetMinBitrate(kCongestionControllerMinBitrate);
+}
+
+GoogCcNetworkController::~GoogCcNetworkController() {}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = probe_controller_->OnNetworkAvailability(msg);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkRouteChange(
+    NetworkRouteChange msg) {
+  if (safe_reset_on_route_change_) {
+    absl::optional<DataRate> estimated_bitrate;
+    if (safe_reset_acknowledged_rate_) {
+      estimated_bitrate = acknowledged_bitrate_estimator_->bitrate();
+      if (!estimated_bitrate)
+        estimated_bitrate = acknowledged_bitrate_estimator_->PeekRate();
+    } else {
+      estimated_bitrate = bandwidth_estimation_->target_rate();
+    }
+    if (estimated_bitrate) {
+      if (msg.constraints.starting_rate) {
+        msg.constraints.starting_rate =
+            std::min(*msg.constraints.starting_rate, *estimated_bitrate);
+      } else {
+        msg.constraints.starting_rate = estimated_bitrate;
+      }
+    }
+  }
+
+  acknowledged_bitrate_estimator_ =
+      AcknowledgedBitrateEstimatorInterface::Create(key_value_config_);
+  probe_bitrate_estimator_.reset(new ProbeBitrateEstimator(event_log_));
+  if (network_estimator_)
+    network_estimator_->OnRouteChange(msg);
+  delay_based_bwe_.reset(new DelayBasedBwe(key_value_config_, event_log_,
+                                           network_state_predictor_.get()));
+  bandwidth_estimation_->OnRouteChange();
+  probe_controller_->Reset(msg.at_time);
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = ResetConstraints(msg.constraints);
+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnProcessInterval(
+    ProcessInterval msg) {
+  NetworkControlUpdate update;
+  if (initial_config_) {
+    update.probe_cluster_configs =
+        ResetConstraints(initial_config_->constraints);
+    update.pacer_config = GetPacingRates(msg.at_time);
+
+    if (initial_config_->stream_based_config.requests_alr_probing) {
+      probe_controller_->EnablePeriodicAlrProbing(
+          *initial_config_->stream_based_config.requests_alr_probing);
+    }
+    absl::optional<DataRate> total_bitrate =
+        initial_config_->stream_based_config.max_total_allocated_bitrate;
+    if (total_bitrate) {
+      auto probes = probe_controller_->OnMaxTotalAllocatedBitrate(
+          *total_bitrate, msg.at_time);
+      update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                          probes.begin(), probes.end());
+    }
+    initial_config_.reset();
+  }
+  if (congestion_window_pushback_controller_ && msg.pacer_queue) {
+    congestion_window_pushback_controller_->UpdatePacingQueue(
+        msg.pacer_queue->bytes());
+  }
+  bandwidth_estimation_->UpdateEstimate(msg.at_time);
+  absl::optional<int64_t> start_time_ms =
+      alr_detector_->GetApplicationLimitedRegionStartTime();
+  probe_controller_->SetAlrStartTimeMs(start_time_ms);
+
+  auto probes = probe_controller_->Process(msg.at_time);
+  update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                      probes.begin(), probes.end());
+
+  if (rate_control_settings_.UseCongestionWindow() &&
+      last_packet_received_time_.IsFinite() && !feedback_max_rtts_.empty()) {
+    UpdateCongestionWindowSize();
+  }
+  if (congestion_window_pushback_controller_ && current_data_window_) {
+    congestion_window_pushback_controller_->SetDataWindow(
+        *current_data_window_);
+  } else {
+    update.congestion_window = current_data_window_;
+  }
+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnRemoteBitrateReport(
+    RemoteBitrateReport msg) {
+  if (packet_feedback_only_) {
+    RTC_LOG(LS_ERROR) << "Received REMB for packet feedback only GoogCC";
+    return NetworkControlUpdate();
+  }
+  bandwidth_estimation_->UpdateReceiverEstimate(msg.receive_time,
+                                                msg.bandwidth);
+  BWE_TEST_LOGGING_PLOT(1, "REMB_kbps", msg.receive_time.ms(),
+                        msg.bandwidth.bps() / 1000);
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnRoundTripTimeUpdate(
+    RoundTripTimeUpdate msg) {
+  if (packet_feedback_only_ || msg.smoothed)
+    return NetworkControlUpdate();
+  RTC_DCHECK(!msg.round_trip_time.IsZero());
+  if (delay_based_bwe_)
+    delay_based_bwe_->OnRttUpdate(msg.round_trip_time);
+  bandwidth_estimation_->UpdateRtt(msg.round_trip_time, msg.receive_time);
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnSentPacket(
+    SentPacket sent_packet) {
+  alr_detector_->OnBytesSent(sent_packet.size.bytes(),
+                             sent_packet.send_time.ms());
+  acknowledged_bitrate_estimator_->SetAlr(
+      alr_detector_->GetApplicationLimitedRegionStartTime().has_value());
+
+  if (!first_packet_sent_) {
+    first_packet_sent_ = true;
+    // Initialize feedback time to send time to allow estimation of RTT until
+    // first feedback is received.
+    bandwidth_estimation_->UpdatePropagationRtt(sent_packet.send_time,
+                                                TimeDelta::Zero());
+  }
+  bandwidth_estimation_->OnSentPacket(sent_packet);
+
+  if (congestion_window_pushback_controller_) {
+    congestion_window_pushback_controller_->UpdateOutstandingData(
+        sent_packet.data_in_flight.bytes());
+    NetworkControlUpdate update;
+    MaybeTriggerOnNetworkChanged(&update, sent_packet.send_time);
+    return update;
+  } else {
+    return NetworkControlUpdate();
+  }
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnReceivedPacket(
+    ReceivedPacket received_packet) {
+  last_packet_received_time_ = received_packet.receive_time;
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnStreamsConfig(
+    StreamsConfig msg) {
+  NetworkControlUpdate update;
+  if (msg.requests_alr_probing) {
+    probe_controller_->EnablePeriodicAlrProbing(*msg.requests_alr_probing);
+  }
+  if (msg.max_total_allocated_bitrate) {
+    update.probe_cluster_configs =
+        probe_controller_->OnMaxTotalAllocatedBitrate(
+            *msg.max_total_allocated_bitrate, msg.at_time);
+  }
+
+  bool pacing_changed = false;
+  if (msg.pacing_factor && *msg.pacing_factor != pacing_factor_) {
+    pacing_factor_ = *msg.pacing_factor;
+    pacing_changed = true;
+  }
+  if (msg.min_total_allocated_bitrate &&
+      *msg.min_total_allocated_bitrate != min_total_allocated_bitrate_) {
+    min_total_allocated_bitrate_ = *msg.min_total_allocated_bitrate;
+    pacing_changed = true;
+
+    if (use_min_allocatable_as_lower_bound_) {
+      ClampConstraints();
+      delay_based_bwe_->SetMinBitrate(min_data_rate_);
+      bandwidth_estimation_->SetMinMaxBitrate(min_data_rate_, max_data_rate_);
+    }
+  }
+  if (msg.max_padding_rate && *msg.max_padding_rate != max_padding_rate_) {
+    max_padding_rate_ = *msg.max_padding_rate;
+    pacing_changed = true;
+  }
+
+  if (pacing_changed)
+    update.pacer_config = GetPacingRates(msg.at_time);
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTargetRateConstraints(
+    TargetRateConstraints constraints) {
+  NetworkControlUpdate update;
+  update.probe_cluster_configs = ResetConstraints(constraints);
+  MaybeTriggerOnNetworkChanged(&update, constraints.at_time);
+  return update;
+}
+
+void GoogCcNetworkController::ClampConstraints() {
+  // TODO(holmer): We should make sure the default bitrates are set to 10 kbps,
+  // and that we don't try to set the min bitrate to 0 from any applications.
+  // The congestion controller should allow a min bitrate of 0.
+  min_data_rate_ = std::max(min_target_rate_, kCongestionControllerMinBitrate);
+  if (use_min_allocatable_as_lower_bound_) {
+    min_data_rate_ = std::max(min_data_rate_, min_total_allocated_bitrate_);
+  }
+  if (max_data_rate_ < min_data_rate_) {
+    RTC_LOG(LS_WARNING) << "max bitrate smaller than min bitrate";
+    max_data_rate_ = min_data_rate_;
+  }
+  if (starting_rate_ && starting_rate_ < min_data_rate_) {
+    RTC_LOG(LS_WARNING) << "start bitrate smaller than min bitrate";
+    starting_rate_ = min_data_rate_;
+  }
+}
+
+std::vector<ProbeClusterConfig> GoogCcNetworkController::ResetConstraints(
+    TargetRateConstraints new_constraints) {
+  min_target_rate_ = new_constraints.min_data_rate.value_or(DataRate::Zero());
+  max_data_rate_ =
+      new_constraints.max_data_rate.value_or(DataRate::PlusInfinity());
+  starting_rate_ = new_constraints.starting_rate;
+  ClampConstraints();
+
+  bandwidth_estimation_->SetBitrates(starting_rate_, min_data_rate_,
+                                     max_data_rate_, new_constraints.at_time);
+
+  if (starting_rate_)
+    delay_based_bwe_->SetStartBitrate(*starting_rate_);
+  delay_based_bwe_->SetMinBitrate(min_data_rate_);
+
+  return probe_controller_->SetBitrates(
+      min_data_rate_, starting_rate_.value_or(DataRate::Zero()), max_data_rate_,
+      new_constraints.at_time);
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTransportLossReport(
+    TransportLossReport msg) {
+  if (packet_feedback_only_)
+    return NetworkControlUpdate();
+  int64_t total_packets_delta =
+      msg.packets_received_delta + msg.packets_lost_delta;
+  bandwidth_estimation_->UpdatePacketsLost(
+      msg.packets_lost_delta, total_packets_delta, msg.receive_time);
+  return NetworkControlUpdate();
+}
+
+void GoogCcNetworkController::UpdateCongestionWindowSize() {
+  TimeDelta min_feedback_max_rtt = TimeDelta::Millis(
+      *std::min_element(feedback_max_rtts_.begin(), feedback_max_rtts_.end()));
+
+  const DataSize kMinCwnd = DataSize::Bytes(2 * 1500);
+  TimeDelta time_window =
+      min_feedback_max_rtt +
+      TimeDelta::Millis(
+          rate_control_settings_.GetCongestionWindowAdditionalTimeMs());
+
+  DataSize data_window = last_loss_based_target_rate_ * time_window;
+  if (current_data_window_) {
+    data_window =
+        std::max(kMinCwnd, (data_window + current_data_window_.value()) / 2);
+  } else {
+    data_window = std::max(kMinCwnd, data_window);
+  }
+  current_data_window_ = data_window;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
+    TransportPacketsFeedback report) {
+  if (report.packet_feedbacks.empty()) {
+    // TODO(bugs.webrtc.org/10125): Design a better mechanism to safe-guard
+    // against building very large network queues.
+    return NetworkControlUpdate();
+  }
+
+  if (congestion_window_pushback_controller_) {
+    congestion_window_pushback_controller_->UpdateOutstandingData(
+        report.data_in_flight.bytes());
+  }
+  TimeDelta max_feedback_rtt = TimeDelta::MinusInfinity();
+  TimeDelta min_propagation_rtt = TimeDelta::PlusInfinity();
+  Timestamp max_recv_time = Timestamp::MinusInfinity();
+
+  std::vector<PacketResult> feedbacks = report.ReceivedWithSendInfo();
+  for (const auto& feedback : feedbacks)
+    max_recv_time = std::max(max_recv_time, feedback.receive_time);
+
+  for (const auto& feedback : feedbacks) {
+    TimeDelta feedback_rtt =
+        report.feedback_time - feedback.sent_packet.send_time;
+    TimeDelta min_pending_time = max_recv_time - feedback.receive_time;
+    TimeDelta propagation_rtt = feedback_rtt - min_pending_time;
+    max_feedback_rtt = std::max(max_feedback_rtt, feedback_rtt);
+    min_propagation_rtt = std::min(min_propagation_rtt, propagation_rtt);
+  }
+
+  if (max_feedback_rtt.IsFinite()) {
+    feedback_max_rtts_.push_back(max_feedback_rtt.ms());
+    const size_t kMaxFeedbackRttWindow = 32;
+    if (feedback_max_rtts_.size() > kMaxFeedbackRttWindow)
+      feedback_max_rtts_.pop_front();
+    // TODO(srte): Use time since last unacknowledged packet.
+    bandwidth_estimation_->UpdatePropagationRtt(report.feedback_time,
+                                                min_propagation_rtt);
+  }
+  if (packet_feedback_only_) {
+    if (!feedback_max_rtts_.empty()) {
+      int64_t sum_rtt_ms =
+          std::accumulate(feedback_max_rtts_.begin(), feedback_max_rtts_.end(),
+                          static_cast<int64_t>(0));
+      int64_t mean_rtt_ms = sum_rtt_ms / feedback_max_rtts_.size();
+      if (delay_based_bwe_)
+        delay_based_bwe_->OnRttUpdate(TimeDelta::Millis(mean_rtt_ms));
+    }
+
+    TimeDelta feedback_min_rtt = TimeDelta::PlusInfinity();
+    for (const auto& packet_feedback : feedbacks) {
+      TimeDelta pending_time = packet_feedback.receive_time - max_recv_time;
+      TimeDelta rtt = report.feedback_time -
+                      packet_feedback.sent_packet.send_time - pending_time;
+      // Value used for predicting NACK round trip time in FEC controller.
+      feedback_min_rtt = std::min(rtt, feedback_min_rtt);
+    }
+    if (feedback_min_rtt.IsFinite()) {
+      bandwidth_estimation_->UpdateRtt(feedback_min_rtt, report.feedback_time);
+    }
+
+    expected_packets_since_last_loss_update_ +=
+        report.PacketsWithFeedback().size();
+    for (const auto& packet_feedback : report.PacketsWithFeedback()) {
+      if (!packet_feedback.IsReceived())
+        lost_packets_since_last_loss_update_ += 1;
+    }
+    if (report.feedback_time > next_loss_update_) {
+      next_loss_update_ = report.feedback_time + kLossUpdateInterval;
+      bandwidth_estimation_->UpdatePacketsLost(
+          lost_packets_since_last_loss_update_,
+          expected_packets_since_last_loss_update_, report.feedback_time);
+      expected_packets_since_last_loss_update_ = 0;
+      lost_packets_since_last_loss_update_ = 0;
+    }
+  }
+  absl::optional<int64_t> alr_start_time =
+      alr_detector_->GetApplicationLimitedRegionStartTime();
+
+  if (previously_in_alr_ && !alr_start_time.has_value()) {
+    int64_t now_ms = report.feedback_time.ms();
+    acknowledged_bitrate_estimator_->SetAlrEndedTime(report.feedback_time);
+    probe_controller_->SetAlrEndedTimeMs(now_ms);
+  }
+  previously_in_alr_ = alr_start_time.has_value();
+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
+      report.SortedByReceiveTime());
+  auto acknowledged_bitrate = acknowledged_bitrate_estimator_->bitrate();
+  bandwidth_estimation_->SetAcknowledgedRate(acknowledged_bitrate,
+                                             report.feedback_time);
+  for (const auto& feedback : report.SortedByReceiveTime()) {
+    if (feedback.sent_packet.pacing_info.probe_cluster_id !=
+        PacedPacketInfo::kNotAProbe) {
+      probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(feedback);
+    }
+  }
+
+  if (network_estimator_) {
+    network_estimator_->OnTransportPacketsFeedback(report);
+    auto prev_estimate = estimate_;
+    estimate_ = network_estimator_->GetCurrentEstimate();
+    // TODO(srte): Make OnTransportPacketsFeedback signal whether the state
+    // changed to avoid the need for this check.
+    if (estimate_ && (!prev_estimate || estimate_->last_feed_time !=
+                                            prev_estimate->last_feed_time)) {
+      event_log_->Log(std::make_unique<RtcEventRemoteEstimate>(
+          estimate_->link_capacity_lower, estimate_->link_capacity_upper));
+      probe_controller_->SetNetworkStateEstimate(*estimate_);
+    }
+  }
+  absl::optional<DataRate> probe_bitrate =
+      probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate();
+  if (ignore_probes_lower_than_network_estimate_ && probe_bitrate &&
+      estimate_ && *probe_bitrate < delay_based_bwe_->last_estimate() &&
+      *probe_bitrate < estimate_->link_capacity_lower) {
+    probe_bitrate.reset();
+  }
+  if (limit_probes_lower_than_throughput_estimate_ && probe_bitrate &&
+      acknowledged_bitrate) {
+    // Limit the backoff to something slightly below the acknowledged
+    // bitrate. ("Slightly below" because we want to drain the queues
+    // if we are actually overusing.)
+    // The acknowledged bitrate shouldn't normally be higher than the delay
+    // based estimate, but it could happen e.g. due to packet bursts or
+    // encoder overshoot. We use std::min to ensure that a probe result
+    // below the current BWE never causes an increase.
+    DataRate limit =
+        std::min(delay_based_bwe_->last_estimate(),
+                 *acknowledged_bitrate * kProbeDropThroughputFraction);
+    probe_bitrate = std::max(*probe_bitrate, limit);
+  }
+
+  NetworkControlUpdate update;
+  bool recovered_from_overuse = false;
+
+  DelayBasedBwe::Result result;
+  result = delay_based_bwe_->IncomingPacketFeedbackVector(
+      report, acknowledged_bitrate, probe_bitrate, estimate_,
+      alr_start_time.has_value());
+
+  if (result.updated) {
+    if (result.probe) {
+      bandwidth_estimation_->SetSendBitrate(result.target_bitrate,
+                                            report.feedback_time);
+    }
+    // Since SetSendBitrate now resets the delay-based estimate, we have to
+    // call UpdateDelayBasedEstimate after SetSendBitrate.
+    bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time,
+                                                    result.target_bitrate);
+  }
+  bandwidth_estimation_->UpdateLossBasedEstimator(
+      report, result.delay_detector_state, probe_bitrate,
+      estimate_ ? estimate_->link_capacity_upper : DataRate::PlusInfinity());
+  if (result.updated) {
+    // Update the estimate in the ProbeController, in case we want to probe.
+    MaybeTriggerOnNetworkChanged(&update, report.feedback_time);
+  }
+
+  recovered_from_overuse = result.recovered_from_overuse;
+
+  if (recovered_from_overuse) {
+    probe_controller_->SetAlrStartTimeMs(alr_start_time);
+    auto probes = probe_controller_->RequestProbe(report.feedback_time);
+    update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
+                                        probes.begin(), probes.end());
+  }
+
+  // No valid RTT could be because send-side BWE isn't used, in which case
+  // we don't try to limit the outstanding packets.
+  if (rate_control_settings_.UseCongestionWindow() &&
+      max_feedback_rtt.IsFinite()) {
+    UpdateCongestionWindowSize();
+  }
+  if (congestion_window_pushback_controller_ && current_data_window_) {
+    congestion_window_pushback_controller_->SetDataWindow(
+        *current_data_window_);
+  } else {
+    update.congestion_window = current_data_window_;
+  }
+
+  return update;
+}
+
+NetworkControlUpdate GoogCcNetworkController::OnNetworkStateEstimate(
+    NetworkStateEstimate msg) {
+  estimate_ = msg;
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate GoogCcNetworkController::GetNetworkState(
+    Timestamp at_time) const {
+  NetworkControlUpdate update;
+  update.target_rate = TargetTransferRate();
+  update.target_rate->network_estimate.at_time = at_time;
+  update.target_rate->network_estimate.loss_rate_ratio =
+      last_estimated_fraction_loss_.value_or(0) / 255.0;
+  update.target_rate->network_estimate.round_trip_time =
+      last_estimated_round_trip_time_;
+  update.target_rate->network_estimate.bwe_period =
+      delay_based_bwe_->GetExpectedBwePeriod();
+
+  update.target_rate->at_time = at_time;
+  update.target_rate->target_rate = last_pushback_target_rate_;
+  update.target_rate->stable_target_rate =
+      bandwidth_estimation_->GetEstimatedLinkCapacity();
+  update.pacer_config = GetPacingRates(at_time);
+  update.congestion_window = current_data_window_;
+  return update;
+}
+
+void GoogCcNetworkController::MaybeTriggerOnNetworkChanged(
+    NetworkControlUpdate* update,
+    Timestamp at_time) {
+  uint8_t fraction_loss = bandwidth_estimation_->fraction_loss();
+  TimeDelta round_trip_time = bandwidth_estimation_->round_trip_time();
+  DataRate loss_based_target_rate = bandwidth_estimation_->target_rate();
+  DataRate pushback_target_rate = loss_based_target_rate;
+
+  BWE_TEST_LOGGING_PLOT(1, "fraction_loss_%", at_time.ms(),
+                        (fraction_loss * 100) / 256);
+  BWE_TEST_LOGGING_PLOT(1, "rtt_ms", at_time.ms(), round_trip_time.ms());
+  BWE_TEST_LOGGING_PLOT(1, "Target_bitrate_kbps", at_time.ms(),
+                        loss_based_target_rate.kbps());
+
+  double cwnd_reduce_ratio = 0.0;
+  if (congestion_window_pushback_controller_) {
+    int64_t pushback_rate =
+        congestion_window_pushback_controller_->UpdateTargetBitrate(
+            loss_based_target_rate.bps());
+    pushback_rate = std::max<int64_t>(bandwidth_estimation_->GetMinBitrate(),
+                                      pushback_rate);
+    pushback_target_rate = DataRate::BitsPerSec(pushback_rate);
+    if (rate_control_settings_.UseCongestionWindowDropFrameOnly()) {
+      cwnd_reduce_ratio = static_cast<double>(loss_based_target_rate.bps() -
+                                              pushback_target_rate.bps()) /
+                          loss_based_target_rate.bps();
+    }
+  }
+  DataRate stable_target_rate =
+      bandwidth_estimation_->GetEstimatedLinkCapacity();
+  stable_target_rate = std::min(stable_target_rate, pushback_target_rate);
+
+  if ((loss_based_target_rate != last_loss_based_target_rate_) ||
+      (fraction_loss != last_estimated_fraction_loss_) ||
+      (round_trip_time != last_estimated_round_trip_time_) ||
+      (pushback_target_rate != last_pushback_target_rate_) ||
+      (stable_target_rate != last_stable_target_rate_)) {
+    last_loss_based_target_rate_ = loss_based_target_rate;
+    last_pushback_target_rate_ = pushback_target_rate;
+    last_estimated_fraction_loss_ = fraction_loss;
+    last_estimated_round_trip_time_ = round_trip_time;
+    last_stable_target_rate_ = stable_target_rate;
+
+    alr_detector_->SetEstimatedBitrate(loss_based_target_rate.bps());
+
+    TimeDelta bwe_period = delay_based_bwe_->GetExpectedBwePeriod();
+
+    TargetTransferRate target_rate_msg;
+    target_rate_msg.at_time = at_time;
+    if (rate_control_settings_.UseCongestionWindowDropFrameOnly()) {
+      target_rate_msg.target_rate = loss_based_target_rate;
+      target_rate_msg.cwnd_reduce_ratio = cwnd_reduce_ratio;
+    } else {
+      target_rate_msg.target_rate = pushback_target_rate;
+    }
+    target_rate_msg.stable_target_rate = stable_target_rate;
+    target_rate_msg.network_estimate.at_time = at_time;
+    target_rate_msg.network_estimate.round_trip_time = round_trip_time;
+    target_rate_msg.network_estimate.loss_rate_ratio = fraction_loss / 255.0f;
+    target_rate_msg.network_estimate.bwe_period = bwe_period;
+
+    update->target_rate = target_rate_msg;
+
+    auto probes = probe_controller_->SetEstimatedBitrate(
+        loss_based_target_rate,
+        GetBandwidthLimitedCause(
+            bandwidth_estimation_->loss_based_state(),
+            delay_based_bwe_->last_state(),
+            probe_controller_->DontProbeIfDelayIncreased()),
+        at_time);
+    update->probe_cluster_configs.insert(update->probe_cluster_configs.end(),
+                                         probes.begin(), probes.end());
+    update->pacer_config = GetPacingRates(at_time);
+    RTC_LOG(LS_VERBOSE) << "bwe " << at_time.ms() << " pushback_target_bps="
+                        << last_pushback_target_rate_.bps()
+                        << " estimate_bps=" << loss_based_target_rate.bps();
+  }
+}
+
+PacerConfig GoogCcNetworkController::GetPacingRates(Timestamp at_time) const {
+  // Pacing rate is based on target rate before congestion window pushback,
+  // because we don't want to build queues in the pacer when pushback occurs.
+  DataRate pacing_rate = DataRate::Zero();
+  if (pace_at_max_of_bwe_and_lower_link_capacity_ && estimate_) {
+    pacing_rate =
+        std::max({min_total_allocated_bitrate_, estimate_->link_capacity_lower,
+                  last_loss_based_target_rate_}) *
+        pacing_factor_;
+  } else {
+    pacing_rate =
+        std::max(min_total_allocated_bitrate_, last_loss_based_target_rate_) *
+        pacing_factor_;
+  }
+  DataRate padding_rate =
+      std::min(max_padding_rate_, last_pushback_target_rate_);
+  PacerConfig msg;
+  msg.at_time = at_time;
+  msg.time_window = TimeDelta::Seconds(1);
+  msg.data_window = pacing_rate * msg.time_window;
+  msg.pad_window = padding_rate * msg.time_window;
+  return msg;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
new file mode 100644
index 0000000000..37a064e37c
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
@@ -0,0 +1,147 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/field_trial_based_config.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/congestion_window_pushback_controller.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/experiments/rate_control_settings.h"
+
+namespace webrtc {
+struct GoogCcConfig {
+  std::unique_ptr<NetworkStateEstimator> network_state_estimator = nullptr;
+  std::unique_ptr<NetworkStatePredictor> network_state_predictor = nullptr;
+  bool feedback_only = false;
+};
+
+class GoogCcNetworkController : public NetworkControllerInterface {
+ public:
+  GoogCcNetworkController(NetworkControllerConfig config,
+                          GoogCcConfig goog_cc_config);
+
+  GoogCcNetworkController() = delete;
+  GoogCcNetworkController(const GoogCcNetworkController&) = delete;
+  GoogCcNetworkController& operator=(const GoogCcNetworkController&) = delete;
+
+  ~GoogCcNetworkController() override;
+
+  // NetworkControllerInterface
+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
+  NetworkControlUpdate OnReceivedPacket(ReceivedPacket msg) override;
+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
+  NetworkControlUpdate OnTargetRateConstraints(
+      TargetRateConstraints msg) override;
+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
+  NetworkControlUpdate OnTransportPacketsFeedback(
+      TransportPacketsFeedback msg) override;
+  NetworkControlUpdate OnNetworkStateEstimate(
+      NetworkStateEstimate msg) override;
+
+  NetworkControlUpdate GetNetworkState(Timestamp at_time) const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  std::vector<ProbeClusterConfig> ResetConstraints(
+      TargetRateConstraints new_constraints);
+  void ClampConstraints();
+  void MaybeTriggerOnNetworkChanged(NetworkControlUpdate* update,
+                                    Timestamp at_time);
+  void UpdateCongestionWindowSize();
+  PacerConfig GetPacingRates(Timestamp at_time) const;
+  const FieldTrialBasedConfig trial_based_config_;
+
+  const FieldTrialsView* const key_value_config_;
+  RtcEventLog* const event_log_;
+  const bool packet_feedback_only_;
+  FieldTrialFlag safe_reset_on_route_change_;
+  FieldTrialFlag safe_reset_acknowledged_rate_;
+  const bool use_min_allocatable_as_lower_bound_;
+  const bool ignore_probes_lower_than_network_estimate_;
+  const bool limit_probes_lower_than_throughput_estimate_;
+  const RateControlSettings rate_control_settings_;
+  const bool pace_at_max_of_bwe_and_lower_link_capacity_;
+
+  const std::unique_ptr<ProbeController> probe_controller_;
+  const std::unique_ptr<CongestionWindowPushbackController>
+      congestion_window_pushback_controller_;
+
+  std::unique_ptr<SendSideBandwidthEstimation> bandwidth_estimation_;
+  std::unique_ptr<AlrDetector> alr_detector_;
+  std::unique_ptr<ProbeBitrateEstimator> probe_bitrate_estimator_;
+  std::unique_ptr<NetworkStateEstimator> network_estimator_;
+  std::unique_ptr<NetworkStatePredictor> network_state_predictor_;
+  std::unique_ptr<DelayBasedBwe> delay_based_bwe_;
+  std::unique_ptr<AcknowledgedBitrateEstimatorInterface>
+      acknowledged_bitrate_estimator_;
+
+  absl::optional<NetworkControllerConfig> initial_config_;
+
+  DataRate min_target_rate_ = DataRate::Zero();
+  DataRate min_data_rate_ = DataRate::Zero();
+  DataRate max_data_rate_ = DataRate::PlusInfinity();
+  absl::optional<DataRate> starting_rate_;
+
+  bool first_packet_sent_ = false;
+
+  absl::optional<NetworkStateEstimate> estimate_;
+
+  Timestamp next_loss_update_ = Timestamp::MinusInfinity();
+  int lost_packets_since_last_loss_update_ = 0;
+  int expected_packets_since_last_loss_update_ = 0;
+
+  std::deque<int64_t> feedback_max_rtts_;
+
+  DataRate last_loss_based_target_rate_;
+  DataRate last_pushback_target_rate_;
+  DataRate last_stable_target_rate_;
+
+  absl::optional<uint8_t> last_estimated_fraction_loss_ = 0;
+  TimeDelta last_estimated_round_trip_time_ = TimeDelta::PlusInfinity();
+  Timestamp last_packet_received_time_ = Timestamp::MinusInfinity();
+
+  double pacing_factor_;
+  DataRate min_total_allocated_bitrate_;
+  DataRate max_padding_rate_;
+
+  bool previously_in_alr_ = false;
+
+  absl::optional<DataSize> current_data_window_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
new file mode 100644
index 0000000000..7e051f505b
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control_unittest.cc
@@ -0,0 +1,934 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <queue>
+
+#include "absl/strings/string_view.h"
+#include "api/test/network_emulation/create_cross_traffic.h"
+#include "api/test/network_emulation/cross_traffic.h"
+#include "api/transport/goog_cc_factory.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+using ::testing::NiceMock;
+
+namespace webrtc {
+namespace test {
+namespace {
+// Count dips from a constant high bandwidth level within a short window.
+int CountBandwidthDips(std::queue<DataRate> bandwidth_history,
+                       DataRate threshold) {
+  if (bandwidth_history.empty())
+    return true;
+  DataRate first = bandwidth_history.front();
+  bandwidth_history.pop();
+
+  int dips = 0;
+  bool state_high = true;
+  while (!bandwidth_history.empty()) {
+    if (bandwidth_history.front() + threshold < first && state_high) {
+      ++dips;
+      state_high = false;
+    } else if (bandwidth_history.front() == first) {
+      state_high = true;
+    } else if (bandwidth_history.front() > first) {
+      // If this is toggling we will catch it later when front becomes first.
+      state_high = false;
+    }
+    bandwidth_history.pop();
+  }
+  return dips;
+}
+GoogCcNetworkControllerFactory CreateFeedbackOnlyFactory() {
+  GoogCcFactoryConfig config;
+  config.feedback_only = true;
+  return GoogCcNetworkControllerFactory(std::move(config));
+}
+
+const uint32_t kInitialBitrateKbps = 60;
+const DataRate kInitialBitrate = DataRate::KilobitsPerSec(kInitialBitrateKbps);
+const float kDefaultPacingRate = 2.5f;
+
+CallClient* CreateVideoSendingClient(
+    Scenario* s,
+    CallClientConfig config,
+    std::vector<EmulatedNetworkNode*> send_link,
+    std::vector<EmulatedNetworkNode*> return_link) {
+  auto* client = s->CreateClient("send", std::move(config));
+  auto* route = s->CreateRoutes(client, send_link,
+                                s->CreateClient("return", CallClientConfig()),
+                                return_link);
+  s->CreateVideoStream(route->forward(), VideoStreamConfig());
+  return client;
+}
+
+NetworkRouteChange CreateRouteChange(
+    Timestamp time,
+    absl::optional<DataRate> start_rate = absl::nullopt,
+    absl::optional<DataRate> min_rate = absl::nullopt,
+    absl::optional<DataRate> max_rate = absl::nullopt) {
+  NetworkRouteChange route_change;
+  route_change.at_time = time;
+  route_change.constraints.at_time = time;
+  route_change.constraints.min_data_rate = min_rate;
+  route_change.constraints.max_data_rate = max_rate;
+  route_change.constraints.starting_rate = start_rate;
+  return route_change;
+}
+
+PacketResult CreatePacketResult(Timestamp arrival_time,
+                                Timestamp send_time,
+                                size_t payload_size,
+                                PacedPacketInfo pacing_info) {
+  PacketResult packet_result;
+  packet_result.sent_packet = SentPacket();
+  packet_result.sent_packet.send_time = send_time;
+  packet_result.sent_packet.size = DataSize::Bytes(payload_size);
+  packet_result.sent_packet.pacing_info = pacing_info;
+  packet_result.receive_time = arrival_time;
+  return packet_result;
+}
+
+// Simulate sending packets and receiving transport feedback during
+// `runtime_ms`.
+absl::optional<DataRate> PacketTransmissionAndFeedbackBlock(
+    NetworkControllerInterface* controller,
+    int64_t runtime_ms,
+    int64_t delay,
+    Timestamp& current_time) {
+  NetworkControlUpdate update;
+  absl::optional<DataRate> target_bitrate;
+  int64_t delay_buildup = 0;
+  int64_t start_time_ms = current_time.ms();
+  while (current_time.ms() - start_time_ms < runtime_ms) {
+    constexpr size_t kPayloadSize = 1000;
+    PacketResult packet =
+        CreatePacketResult(current_time + TimeDelta::Millis(delay_buildup),
+                           current_time, kPayloadSize, PacedPacketInfo());
+    delay_buildup += delay;
+    update = controller->OnSentPacket(packet.sent_packet);
+    if (update.target_rate) {
+      target_bitrate = update.target_rate->target_rate;
+    }
+    TransportPacketsFeedback feedback;
+    feedback.feedback_time = packet.receive_time;
+    feedback.packet_feedbacks.push_back(packet);
+    update = controller->OnTransportPacketsFeedback(feedback);
+    if (update.target_rate) {
+      target_bitrate = update.target_rate->target_rate;
+    }
+    current_time += TimeDelta::Millis(50);
+    update = controller->OnProcessInterval({.at_time = current_time});
+    if (update.target_rate) {
+      target_bitrate = update.target_rate->target_rate;
+    }
+  }
+  return target_bitrate;
+}
+
+// Scenarios:
+
+void UpdatesTargetRateBasedOnLinkCapacity(absl::string_view test_name = "") {
+  auto factory = CreateFeedbackOnlyFactory();
+  Scenario s("googcc_unit/target_capacity" + std::string(test_name), false);
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(1500);
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto send_net = s.CreateMutableSimulationNode([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(500);
+    c->delay = TimeDelta::Millis(100);
+    c->loss_rate = 0.0;
+  });
+  auto ret_net = s.CreateMutableSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  StatesPrinter* truth = s.CreatePrinter(
+      "send.truth.txt", TimeDelta::PlusInfinity(), {send_net->ConfigPrinter()});
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net->node()},
+                                          {ret_net->node()});
+
+  truth->PrintRow();
+  s.RunFor(TimeDelta::Seconds(25));
+  truth->PrintRow();
+  EXPECT_NEAR(client->target_rate().kbps(), 450, 100);
+
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(800);
+    c->delay = TimeDelta::Millis(100);
+  });
+
+  truth->PrintRow();
+  s.RunFor(TimeDelta::Seconds(20));
+  truth->PrintRow();
+  EXPECT_NEAR(client->target_rate().kbps(), 750, 150);
+
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(100);
+    c->delay = TimeDelta::Millis(200);
+  });
+  ret_net->UpdateConfig(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(200); });
+
+  truth->PrintRow();
+  s.RunFor(TimeDelta::Seconds(50));
+  truth->PrintRow();
+  EXPECT_NEAR(client->target_rate().kbps(), 90, 25);
+}
+
+DataRate RunRembDipScenario(absl::string_view test_name) {
+  Scenario s(test_name);
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(2000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+  auto* route = s.CreateRoutes(
+      client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+
+  s.RunFor(TimeDelta::Seconds(10));
+  EXPECT_GT(client->send_bandwidth().kbps(), 1500);
+
+  DataRate RembLimit = DataRate::KilobitsPerSec(250);
+  client->SetRemoteBitrate(RembLimit);
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_EQ(client->send_bandwidth(), RembLimit);
+
+  DataRate RembLimitLifted = DataRate::KilobitsPerSec(10000);
+  client->SetRemoteBitrate(RembLimitLifted);
+  s.RunFor(TimeDelta::Seconds(10));
+
+  return client->send_bandwidth();
+}
+
+}  // namespace
+
+class NetworkControllerTestFixture {
+ public:
+  NetworkControllerTestFixture() : factory_() {}
+
+  std::unique_ptr<NetworkControllerInterface> CreateController() {
+    NetworkControllerConfig config = InitialConfig();
+    std::unique_ptr<NetworkControllerInterface> controller =
+        factory_.Create(config);
+    return controller;
+  }
+
+ private:
+  NetworkControllerConfig InitialConfig(
+      int starting_bandwidth_kbps = kInitialBitrateKbps,
+      int min_data_rate_kbps = 0,
+      int max_data_rate_kbps = 5 * kInitialBitrateKbps) {
+    NetworkControllerConfig config;
+    config.constraints.at_time = Timestamp::Zero();
+    config.constraints.min_data_rate =
+        DataRate::KilobitsPerSec(min_data_rate_kbps);
+    config.constraints.max_data_rate =
+        DataRate::KilobitsPerSec(max_data_rate_kbps);
+    config.constraints.starting_rate =
+        DataRate::KilobitsPerSec(starting_bandwidth_kbps);
+    config.event_log = &event_log_;
+    return config;
+  }
+
+  NiceMock<MockRtcEventLog> event_log_;
+  GoogCcNetworkControllerFactory factory_;
+};
+
+TEST(GoogCcNetworkControllerTest, InitializeTargetRateOnFirstProcessInterval) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+
+  NetworkControlUpdate update =
+      controller->OnProcessInterval({.at_time = Timestamp::Millis(123456)});
+
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate);
+  EXPECT_EQ(update.pacer_config->data_rate(),
+            kInitialBitrate * kDefaultPacingRate);
+  EXPECT_EQ(update.probe_cluster_configs[0].target_data_rate,
+            kInitialBitrate * 3);
+  EXPECT_EQ(update.probe_cluster_configs[1].target_data_rate,
+            kInitialBitrate * 5);
+}
+
+TEST(GoogCcNetworkControllerTest, ReactsToChangedNetworkConditions) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  NetworkControlUpdate update =
+      controller->OnProcessInterval({.at_time = current_time});
+  update = controller->OnRemoteBitrateReport(
+      {.receive_time = current_time, .bandwidth = kInitialBitrate * 2});
+
+  current_time += TimeDelta::Millis(25);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate * 2);
+  EXPECT_EQ(update.pacer_config->data_rate(),
+            kInitialBitrate * 2 * kDefaultPacingRate);
+
+  update = controller->OnRemoteBitrateReport(
+      {.receive_time = current_time, .bandwidth = kInitialBitrate});
+  current_time += TimeDelta::Millis(25);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate);
+  EXPECT_EQ(update.pacer_config->data_rate(),
+            kInitialBitrate * kDefaultPacingRate);
+}
+
+TEST(GoogCcNetworkControllerTest, OnNetworkRouteChanged) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  DataRate new_bitrate = DataRate::BitsPerSec(200000);
+  NetworkControlUpdate update = controller->OnNetworkRouteChange(
+      CreateRouteChange(current_time, new_bitrate));
+  EXPECT_EQ(update.target_rate->target_rate, new_bitrate);
+  EXPECT_EQ(update.pacer_config->data_rate(), new_bitrate * kDefaultPacingRate);
+  EXPECT_EQ(update.probe_cluster_configs.size(), 2u);
+
+  // If the bitrate is reset to -1, the new starting bitrate will be
+  // the minimum default bitrate.
+  const DataRate kDefaultMinBitrate = DataRate::KilobitsPerSec(5);
+  update = controller->OnNetworkRouteChange(CreateRouteChange(current_time));
+  EXPECT_EQ(update.target_rate->target_rate, kDefaultMinBitrate);
+  EXPECT_NEAR(update.pacer_config->data_rate().bps<double>(),
+              kDefaultMinBitrate.bps<double>() * kDefaultPacingRate, 10);
+  EXPECT_EQ(update.probe_cluster_configs.size(), 2u);
+}
+
+TEST(GoogCcNetworkControllerTest, ProbeOnRouteChange) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  NetworkControlUpdate update = controller->OnNetworkRouteChange(
+      CreateRouteChange(current_time, 2 * kInitialBitrate, DataRate::Zero(),
+                        20 * kInitialBitrate));
+
+  EXPECT_TRUE(update.pacer_config.has_value());
+  EXPECT_EQ(update.target_rate->target_rate, kInitialBitrate * 2);
+  EXPECT_EQ(update.probe_cluster_configs.size(), 2u);
+  EXPECT_EQ(update.probe_cluster_configs[0].target_data_rate,
+            kInitialBitrate * 6);
+  EXPECT_EQ(update.probe_cluster_configs[1].target_data_rate,
+            kInitialBitrate * 12);
+
+  update = controller->OnProcessInterval({.at_time = current_time});
+}
+
+// Bandwidth estimation is updated when feedbacks are received.
+// Feedbacks which show an increasing delay cause the estimation to be reduced.
+TEST(GoogCcNetworkControllerTest, UpdatesDelayBasedEstimate) {
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  const int64_t kRunTimeMs = 6000;
+  Timestamp current_time = Timestamp::Millis(123);
+
+  // The test must run and insert packets/feedback long enough that the
+  // BWE computes a valid estimate. This is first done in an environment which
+  // simulates no bandwidth limitation, and therefore not built-up delay.
+  absl::optional<DataRate> target_bitrate_before_delay =
+      PacketTransmissionAndFeedbackBlock(controller.get(), kRunTimeMs, 0,
+                                         current_time);
+  ASSERT_TRUE(target_bitrate_before_delay.has_value());
+
+  // Repeat, but this time with a building delay, and make sure that the
+  // estimation is adjusted downwards.
+  absl::optional<DataRate> target_bitrate_after_delay =
+      PacketTransmissionAndFeedbackBlock(controller.get(), kRunTimeMs, 50,
+                                         current_time);
+  EXPECT_LT(*target_bitrate_after_delay, *target_bitrate_before_delay);
+}
+
+TEST(GoogCcNetworkControllerTest, PaceAtMaxOfLowerLinkCapacityAndBwe) {
+  ScopedFieldTrials trial(
+      "WebRTC-Bwe-PaceAtMaxOfBweAndLowerLinkCapacity/Enabled/");
+  NetworkControllerTestFixture fixture;
+  std::unique_ptr<NetworkControllerInterface> controller =
+      fixture.CreateController();
+  Timestamp current_time = Timestamp::Millis(123);
+  NetworkControlUpdate update =
+      controller->OnProcessInterval({.at_time = current_time});
+  current_time += TimeDelta::Millis(100);
+  NetworkStateEstimate network_estimate = {.link_capacity_lower =
+                                               10 * kInitialBitrate};
+  update = controller->OnNetworkStateEstimate(network_estimate);
+  // OnNetworkStateEstimate does not trigger processing a new estimate. So add a
+  // dummy loss report to trigger a BWE update in the next process interval.
+  TransportLossReport loss_report;
+  loss_report.start_time = current_time;
+  loss_report.end_time = current_time;
+  loss_report.receive_time = current_time;
+  loss_report.packets_received_delta = 50;
+  loss_report.packets_lost_delta = 1;
+  update = controller->OnTransportLossReport(loss_report);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  ASSERT_TRUE(update.pacer_config);
+  ASSERT_TRUE(update.target_rate);
+  ASSERT_LT(update.target_rate->target_rate,
+            network_estimate.link_capacity_lower);
+  EXPECT_EQ(update.pacer_config->data_rate().kbps(),
+            network_estimate.link_capacity_lower.kbps() * kDefaultPacingRate);
+
+  current_time += TimeDelta::Millis(100);
+  // Set a low link capacity estimate and verify that pacing rate is set
+  // relative to loss based/delay based estimate.
+  network_estimate = {.link_capacity_lower = 0.5 * kInitialBitrate};
+  update = controller->OnNetworkStateEstimate(network_estimate);
+  // Again, we need to inject a dummy loss report to trigger an update of the
+  // BWE in the next process interval.
+  loss_report.start_time = current_time;
+  loss_report.end_time = current_time;
+  loss_report.receive_time = current_time;
+  loss_report.packets_received_delta = 50;
+  loss_report.packets_lost_delta = 0;
+  update = controller->OnTransportLossReport(loss_report);
+  update = controller->OnProcessInterval({.at_time = current_time});
+  ASSERT_TRUE(update.target_rate);
+  ASSERT_GT(update.target_rate->target_rate,
+            network_estimate.link_capacity_lower);
+  EXPECT_EQ(update.pacer_config->data_rate().kbps(),
+            update.target_rate->target_rate.kbps() * kDefaultPacingRate);
+}
+
+// Test congestion window pushback on network delay happens.
+TEST(GoogCcScenario, CongestionWindowPushbackOnNetworkDelay) {
+  auto factory = CreateFeedbackOnlyFactory();
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:800,MinBitrate:30000/");
+  Scenario s("googcc_unit/cwnd_on_delay", false);
+  auto send_net =
+      s.CreateMutableSimulationNode([=](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(1000);
+        c->delay = TimeDelta::Millis(100);
+      });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  // Start high so bandwidth drop has max effect.
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(2000);
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+
+  auto* client = CreateVideoSendingClient(&s, std::move(config),
+                                          {send_net->node()}, {ret_net});
+
+  s.RunFor(TimeDelta::Seconds(10));
+  send_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(10));
+  s.RunFor(TimeDelta::Seconds(3));
+
+  // After 3 seconds without feedback from any sent packets, we expect that the
+  // target rate is reduced to the minimum pushback threshold
+  // kDefaultMinPushbackTargetBitrateBps, which is defined as 30 kbps in
+  // congestion_window_pushback_controller.
+  EXPECT_LT(client->target_rate().kbps(), 40);
+}
+
+// Test congestion window pushback on network delay happens.
+TEST(GoogCcScenario, CongestionWindowPushbackDropFrameOnNetworkDelay) {
+  auto factory = CreateFeedbackOnlyFactory();
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:800,MinBitrate:30000,DropFrame:true/");
+  Scenario s("googcc_unit/cwnd_on_delay", false);
+  auto send_net =
+      s.CreateMutableSimulationNode([=](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(1000);
+        c->delay = TimeDelta::Millis(100);
+      });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  // Start high so bandwidth drop has max effect.
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(2000);
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+
+  auto* client = CreateVideoSendingClient(&s, std::move(config),
+                                          {send_net->node()}, {ret_net});
+
+  s.RunFor(TimeDelta::Seconds(10));
+  send_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(10));
+  s.RunFor(TimeDelta::Seconds(3));
+
+  // As the dropframe is set, after 3 seconds without feedback from any sent
+  // packets, we expect that the target rate is not reduced by congestion
+  // window.
+  EXPECT_GT(client->target_rate().kbps(), 300);
+}
+
+TEST(GoogCcScenario, PaddingRateLimitedByCongestionWindowInTrial) {
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:200,MinBitrate:30000/");
+
+  Scenario s("googcc_unit/padding_limited", false);
+  auto send_net =
+      s.CreateMutableSimulationNode([=](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(1000);
+        c->delay = TimeDelta::Millis(100);
+      });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  // Start high so bandwidth drop has max effect.
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(2000);
+  auto* client = s.CreateClient("send", config);
+  auto* route =
+      s.CreateRoutes(client, {send_net->node()},
+                     s.CreateClient("return", CallClientConfig()), {ret_net});
+  VideoStreamConfig video;
+  video.stream.pad_to_rate = config.transport.rates.max_rate;
+  s.CreateVideoStream(route->forward(), video);
+
+  // Run for a few seconds to allow the controller to stabilize.
+  s.RunFor(TimeDelta::Seconds(10));
+
+  // Check that padding rate matches target rate.
+  EXPECT_NEAR(client->padding_rate().kbps(), client->target_rate().kbps(), 1);
+
+  // Check this is also the case when congestion window pushback kicks in.
+  send_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(1));
+  EXPECT_NEAR(client->padding_rate().kbps(), client->target_rate().kbps(), 1);
+}
+
+TEST(GoogCcScenario, LimitsToFloorIfRttIsHighInTrial) {
+  // The field trial limits maximum RTT to 2 seconds, higher RTT means that the
+  // controller backs off until it reaches the minimum configured bitrate. This
+  // allows the RTT to recover faster than the regular control mechanism would
+  // achieve.
+  const DataRate kBandwidthFloor = DataRate::KilobitsPerSec(50);
+  ScopedFieldTrials trial("WebRTC-Bwe-MaxRttLimit/limit:2s,floor:" +
+                          std::to_string(kBandwidthFloor.kbps()) + "kbps/");
+  // In the test case, we limit the capacity and add a cross traffic packet
+  // burst that blocks media from being sent. This causes the RTT to quickly
+  // increase above the threshold in the trial.
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(100);
+  const TimeDelta kBufferBloatDuration = TimeDelta::Seconds(10);
+  Scenario s("googcc_unit/limit_trial", false);
+  auto send_net = s.CreateSimulationNode([=](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(100);
+  });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+  CallClientConfig config;
+  config.transport.rates.start_rate = kLinkCapacity;
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net}, {ret_net});
+  // Run for a few seconds to allow the controller to stabilize.
+  s.RunFor(TimeDelta::Seconds(10));
+  const DataSize kBloatPacketSize = DataSize::Bytes(1000);
+  const int kBloatPacketCount =
+      static_cast<int>(kBufferBloatDuration * kLinkCapacity / kBloatPacketSize);
+  // This will cause the RTT to be large for a while.
+  s.TriggerPacketBurst({send_net}, kBloatPacketCount, kBloatPacketSize.bytes());
+  // Wait to allow the high RTT to be detected and acted upon.
+  s.RunFor(TimeDelta::Seconds(6));
+  // By now the target rate should have dropped to the minimum configured rate.
+  EXPECT_NEAR(client->target_rate().kbps(), kBandwidthFloor.kbps(), 5);
+}
+
+TEST(GoogCcScenario, UpdatesTargetRateBasedOnLinkCapacity) {
+  UpdatesTargetRateBasedOnLinkCapacity();
+}
+
+TEST(GoogCcScenario, StableEstimateDoesNotVaryInSteadyState) {
+  auto factory = CreateFeedbackOnlyFactory();
+  Scenario s("googcc_unit/stable_target", false);
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(500);
+  net_conf.delay = TimeDelta::Millis(100);
+  auto send_net = s.CreateSimulationNode(net_conf);
+  auto ret_net = s.CreateSimulationNode(net_conf);
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net}, {ret_net});
+  // Run for a while to allow the estimate to stabilize.
+  s.RunFor(TimeDelta::Seconds(30));
+  DataRate min_stable_target = DataRate::PlusInfinity();
+  DataRate max_stable_target = DataRate::MinusInfinity();
+  DataRate min_target = DataRate::PlusInfinity();
+  DataRate max_target = DataRate::MinusInfinity();
+
+  // Measure variation in steady state.
+  for (int i = 0; i < 20; ++i) {
+    auto stable_target_rate = client->stable_target_rate();
+    auto target_rate = client->target_rate();
+    EXPECT_LE(stable_target_rate, target_rate);
+
+    min_stable_target = std::min(min_stable_target, stable_target_rate);
+    max_stable_target = std::max(max_stable_target, stable_target_rate);
+    min_target = std::min(min_target, target_rate);
+    max_target = std::max(max_target, target_rate);
+    s.RunFor(TimeDelta::Seconds(1));
+  }
+  // We should expect drops by at least 15% (default backoff.)
+  EXPECT_LT(min_target / max_target, 0.85);
+  // We should expect the stable target to be more stable than the immediate one
+  EXPECT_GE(min_stable_target / max_stable_target, min_target / max_target);
+}
+
+TEST(GoogCcScenario, LossBasedControlUpdatesTargetRateBasedOnLinkCapacity) {
+  ScopedFieldTrials trial("WebRTC-Bwe-LossBasedControl/Enabled/");
+  // TODO(srte): Should the behavior be unaffected at low loss rates?
+  UpdatesTargetRateBasedOnLinkCapacity("_loss_based");
+}
+
+TEST(GoogCcScenario, LossBasedControlDoesModestBackoffToHighLoss) {
+  ScopedFieldTrials trial("WebRTC-Bwe-LossBasedControl/Enabled/");
+  Scenario s("googcc_unit/high_loss_channel", false);
+  CallClientConfig config;
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(1500);
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto send_net = s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(2000);
+    c->delay = TimeDelta::Millis(200);
+    c->loss_rate = 0.1;
+  });
+  auto ret_net = s.CreateSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(200); });
+
+  auto* client = CreateVideoSendingClient(&s, config, {send_net}, {ret_net});
+
+  s.RunFor(TimeDelta::Seconds(120));
+  // Without LossBasedControl trial, bandwidth drops to ~10 kbps.
+  EXPECT_GT(client->target_rate().kbps(), 100);
+}
+
+DataRate AverageBitrateAfterCrossInducedLoss(absl::string_view name) {
+  Scenario s(name, false);
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(100);
+  // Short queue length means that we'll induce loss when sudden TCP traffic
+  // spikes are induced. This corresponds to ca 200 ms for a packet size of 1000
+  // bytes. Such limited buffers are common on for instance wifi routers.
+  net_conf.packet_queue_length_limit = 25;
+
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+
+  auto* client = s.CreateClient("send", CallClientConfig());
+  auto* callee = s.CreateClient("return", CallClientConfig());
+  auto* route = s.CreateRoutes(client, send_net, callee, ret_net);
+  // TODO(srte): Make this work with RTX enabled or remove it.
+  auto* video = s.CreateVideoStream(route->forward(), [](VideoStreamConfig* c) {
+    c->stream.use_rtx = false;
+  });
+  s.RunFor(TimeDelta::Seconds(10));
+  for (int i = 0; i < 4; ++i) {
+    // Sends TCP cross traffic inducing loss.
+    auto* tcp_traffic = s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+        s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+        FakeTcpConfig()));
+    s.RunFor(TimeDelta::Seconds(2));
+    // Allow the ccongestion controller to recover.
+    s.net()->StopCrossTraffic(tcp_traffic);
+    s.RunFor(TimeDelta::Seconds(20));
+  }
+
+  // Querying the video stats from within the expected runtime environment
+  // (i.e. the TQ that belongs to the CallClient, not the Scenario TQ that
+  // we're currently on).
+  VideoReceiveStreamInterface::Stats video_receive_stats;
+  auto* video_stream = video->receive();
+  callee->SendTask([&video_stream, &video_receive_stats]() {
+    video_receive_stats = video_stream->GetStats();
+  });
+  return DataSize::Bytes(
+             video_receive_stats.rtp_stats.packet_counter.TotalBytes()) /
+         s.TimeSinceStart();
+}
+
+TEST(GoogCcScenario, MaintainsLowRateInSafeResetTrial) {
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(200);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+
+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled/");
+  Scenario s("googcc_unit/safe_reset_low");
+  auto* send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(10);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to stabilize.
+  s.RunFor(TimeDelta::Millis(500));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 50);
+  s.ChangeRoute(route->forward(), {send_net});
+  // Allow new settings to propagate.
+  s.RunFor(TimeDelta::Millis(100));
+  // Under the trial, the target should be unchanged for low rates.
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 50);
+}
+
+TEST(GoogCcScenario, CutsHighRateInSafeResetTrial) {
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(1000);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+
+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled/");
+  Scenario s("googcc_unit/safe_reset_high_cut");
+  auto send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to stabilize.
+  s.RunFor(TimeDelta::Millis(500));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kLinkCapacity.kbps(), 300);
+  s.ChangeRoute(route->forward(), {send_net});
+  // Allow new settings to propagate.
+  s.RunFor(TimeDelta::Millis(50));
+  // Under the trial, the target should be reset from high values.
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kStartRate.kbps(), 30);
+}
+
+TEST(GoogCcScenario, DetectsHighRateInSafeResetTrial) {
+  ScopedFieldTrials trial("WebRTC-Bwe-SafeResetOnRouteChange/Enabled,ack/");
+  const DataRate kInitialLinkCapacity = DataRate::KilobitsPerSec(200);
+  const DataRate kNewLinkCapacity = DataRate::KilobitsPerSec(800);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+
+  Scenario s("googcc_unit/safe_reset_high_detect");
+  auto* initial_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kInitialLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* new_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kNewLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {initial_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to stabilize.
+  s.RunFor(TimeDelta::Millis(2000));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kInitialLinkCapacity.kbps(), 50);
+  s.ChangeRoute(route->forward(), {new_net});
+  // Allow new settings to propagate, but not probes to be received.
+  s.RunFor(TimeDelta::Millis(50));
+  // Under the field trial, the target rate should be unchanged since it's lower
+  // than the starting rate.
+  EXPECT_NEAR(client->send_bandwidth().kbps(), kInitialLinkCapacity.kbps(), 50);
+  // However, probing should have made us detect the higher rate.
+  // NOTE: This test causes high loss rate, and the loss-based estimator reduces
+  // the bitrate, making the test fail if we wait longer than one second here.
+  s.RunFor(TimeDelta::Millis(1000));
+  EXPECT_GT(client->send_bandwidth().kbps(), kNewLinkCapacity.kbps() - 300);
+}
+
+TEST(GoogCcScenario, TargetRateReducedOnPacingBufferBuildupInTrial) {
+  // Configure strict pacing to ensure build-up.
+  ScopedFieldTrials trial(
+      "WebRTC-CongestionWindow/QueueSize:100,MinBitrate:30000/"
+      "WebRTC-Video-Pacing/factor:1.0/"
+      "WebRTC-AddPacingToCongestionWindowPushback/Enabled/");
+
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(1000);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(1000);
+
+  Scenario s("googcc_unit/pacing_buffer_buildup");
+  auto* net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(50);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow some time for the buffer to build up.
+  s.RunFor(TimeDelta::Seconds(5));
+
+  // Without trial, pacer delay reaches ~250 ms.
+  EXPECT_LT(client->GetStats().pacer_delay_ms, 150);
+}
+
+TEST(GoogCcScenario, NoBandwidthTogglingInLossControlTrial) {
+  ScopedFieldTrials trial("WebRTC-Bwe-LossBasedControl/Enabled/");
+  Scenario s("googcc_unit/no_toggling");
+  auto* send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(2000);
+    c->loss_rate = 0.2;
+    c->delay = TimeDelta::Millis(10);
+  });
+
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to initialize.
+  s.RunFor(TimeDelta::Millis(250));
+
+  std::queue<DataRate> bandwidth_history;
+  const TimeDelta step = TimeDelta::Millis(50);
+  for (TimeDelta time = TimeDelta::Zero(); time < TimeDelta::Millis(2000);
+       time += step) {
+    s.RunFor(step);
+    const TimeDelta window = TimeDelta::Millis(500);
+    if (bandwidth_history.size() >= window / step)
+      bandwidth_history.pop();
+    bandwidth_history.push(client->send_bandwidth());
+    EXPECT_LT(
+        CountBandwidthDips(bandwidth_history, DataRate::KilobitsPerSec(100)),
+        2);
+  }
+}
+
+TEST(GoogCcScenario, NoRttBackoffCollapseWhenVideoStops) {
+  ScopedFieldTrials trial("WebRTC-Bwe-MaxRttLimit/limit:2s/");
+  Scenario s("googcc_unit/rttbackoff_video_stop");
+  auto* send_net = s.CreateSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(2000);
+    c->delay = TimeDelta::Millis(100);
+  });
+
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto* route = s.CreateRoutes(
+      client, {send_net}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  auto* video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  // Allow the controller to initialize, then stop video.
+  s.RunFor(TimeDelta::Seconds(1));
+  video->send()->Stop();
+  s.RunFor(TimeDelta::Seconds(4));
+  EXPECT_GT(client->send_bandwidth().kbps(), 1000);
+}
+
+TEST(GoogCcScenario, NoCrashOnVeryLateFeedback) {
+  Scenario s;
+  auto ret_net = s.CreateMutableSimulationNode(NetworkSimulationConfig());
+  auto* route = s.CreateRoutes(
+      s.CreateClient("send", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())},
+      s.CreateClient("return", CallClientConfig()), {ret_net->node()});
+  auto* video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  s.RunFor(TimeDelta::Seconds(5));
+  // Delay feedback by several minutes. This will cause removal of the send time
+  // history for the packets as long as kSendTimeHistoryWindow is configured for
+  // a shorter time span.
+  ret_net->PauseTransmissionUntil(s.Now() + TimeDelta::Seconds(300));
+  // Stopping video stream while waiting to save test execution time.
+  video->send()->Stop();
+  s.RunFor(TimeDelta::Seconds(299));
+  // Starting to cause addition of new packet to history, which cause old
+  // packets to be removed.
+  video->send()->Start();
+  // Runs until the lost packets are received. We expect that this will run
+  // without causing any runtime failures.
+  s.RunFor(TimeDelta::Seconds(2));
+}
+
+TEST(GoogCcScenario, IsFairToTCP) {
+  Scenario s("googcc_unit/tcp_fairness");
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+  auto* route = s.CreateRoutes(
+      client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+  s.CreateVideoStream(route->forward(), VideoStreamConfig());
+  s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+      s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+      FakeTcpConfig()));
+  s.RunFor(TimeDelta::Seconds(10));
+
+  // Currently only testing for the upper limit as we in practice back out
+  // quite a lot in this scenario. If this behavior is fixed, we should add a
+  // lower bound to ensure it stays fixed.
+  EXPECT_LT(client->send_bandwidth().kbps(), 750);
+}
+
+TEST(GoogCcScenario, FastRampupOnRembCapLifted) {
+  DataRate final_estimate =
+      RunRembDipScenario("googcc_unit/default_fast_rampup_on_remb_cap_lifted");
+  EXPECT_GT(final_estimate.kbps(), 1500);
+}
+
+TEST(GoogCcScenario, FallbackToLossBasedBweWithoutPacketFeedback) {
+  const DataRate kLinkCapacity = DataRate::KilobitsPerSec(1000);
+  const DataRate kStartRate = DataRate::KilobitsPerSec(1000);
+
+  Scenario s("googcc_unit/high_loss_channel", false);
+  auto* net = s.CreateMutableSimulationNode([&](NetworkSimulationConfig* c) {
+    c->bandwidth = kLinkCapacity;
+    c->delay = TimeDelta::Millis(100);
+  });
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = kStartRate;
+  });
+  auto* route = s.CreateRoutes(
+      client, {net->node()}, s.CreateClient("return", CallClientConfig()),
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+
+  // Create a config without packet feedback.
+  VideoStreamConfig video_config;
+  video_config.stream.packet_feedback = false;
+  s.CreateVideoStream(route->forward(), video_config);
+
+  s.RunFor(TimeDelta::Seconds(20));
+  // Bandwith does not backoff because network is normal.
+  EXPECT_GE(client->target_rate().kbps(), 500);
+
+  // Update the network to create high loss ratio
+  net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->loss_rate = 0.15;
+  });
+  s.RunFor(TimeDelta::Seconds(20));
+
+  // Bandwidth decreases thanks to loss based bwe v0.
+  EXPECT_LE(client->target_rate().kbps(), 300);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc
new file mode 100644
index 0000000000..2d50d08e6a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.cc
@@ -0,0 +1,140 @@
+/*
+ *  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 "modules/congestion_controller/goog_cc/inter_arrival_delta.h"
+
+#include <algorithm>
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+static constexpr TimeDelta kBurstDeltaThreshold = TimeDelta::Millis(5);
+static constexpr TimeDelta kMaxBurstDuration = TimeDelta::Millis(100);
+constexpr TimeDelta InterArrivalDelta::kArrivalTimeOffsetThreshold;
+
+InterArrivalDelta::InterArrivalDelta(TimeDelta send_time_group_length)
+    : send_time_group_length_(send_time_group_length),
+      current_timestamp_group_(),
+      prev_timestamp_group_(),
+      num_consecutive_reordered_packets_(0) {}
+
+bool InterArrivalDelta::ComputeDeltas(Timestamp send_time,
+                                      Timestamp arrival_time,
+                                      Timestamp system_time,
+                                      size_t packet_size,
+                                      TimeDelta* send_time_delta,
+                                      TimeDelta* arrival_time_delta,
+                                      int* packet_size_delta) {
+  bool calculated_deltas = false;
+  if (current_timestamp_group_.IsFirstPacket()) {
+    // We don't have enough data to update the filter, so we store it until we
+    // have two frames of data to process.
+    current_timestamp_group_.send_time = send_time;
+    current_timestamp_group_.first_send_time = send_time;
+    current_timestamp_group_.first_arrival = arrival_time;
+  } else if (current_timestamp_group_.first_send_time > send_time) {
+    // Reordered packet.
+    return false;
+  } else if (NewTimestampGroup(arrival_time, send_time)) {
+    // First packet of a later send burst, the previous packets sample is ready.
+    if (prev_timestamp_group_.complete_time.IsFinite()) {
+      *send_time_delta =
+          current_timestamp_group_.send_time - prev_timestamp_group_.send_time;
+      *arrival_time_delta = current_timestamp_group_.complete_time -
+                            prev_timestamp_group_.complete_time;
+
+      TimeDelta system_time_delta = current_timestamp_group_.last_system_time -
+                                    prev_timestamp_group_.last_system_time;
+
+      if (*arrival_time_delta - system_time_delta >=
+          kArrivalTimeOffsetThreshold) {
+        RTC_LOG(LS_WARNING)
+            << "The arrival time clock offset has changed (diff = "
+            << arrival_time_delta->ms() - system_time_delta.ms()
+            << " ms), resetting.";
+        Reset();
+        return false;
+      }
+      if (*arrival_time_delta < TimeDelta::Zero()) {
+        // The group of packets has been reordered since receiving its local
+        // arrival timestamp.
+        ++num_consecutive_reordered_packets_;
+        if (num_consecutive_reordered_packets_ >= kReorderedResetThreshold) {
+          RTC_LOG(LS_WARNING)
+              << "Packets between send burst arrived out of order, resetting:"
+              << " arrival_time_delta_ms=" << arrival_time_delta->ms()
+              << ", send_time_delta_ms=" << send_time_delta->ms();
+          Reset();
+        }
+        return false;
+      } else {
+        num_consecutive_reordered_packets_ = 0;
+      }
+      *packet_size_delta = static_cast<int>(current_timestamp_group_.size) -
+                           static_cast<int>(prev_timestamp_group_.size);
+      calculated_deltas = true;
+    }
+    prev_timestamp_group_ = current_timestamp_group_;
+    // The new timestamp is now the current frame.
+    current_timestamp_group_.first_send_time = send_time;
+    current_timestamp_group_.send_time = send_time;
+    current_timestamp_group_.first_arrival = arrival_time;
+    current_timestamp_group_.size = 0;
+  } else {
+    current_timestamp_group_.send_time =
+        std::max(current_timestamp_group_.send_time, send_time);
+  }
+  // Accumulate the frame size.
+  current_timestamp_group_.size += packet_size;
+  current_timestamp_group_.complete_time = arrival_time;
+  current_timestamp_group_.last_system_time = system_time;
+
+  return calculated_deltas;
+}
+
+// Assumes that `timestamp` is not reordered compared to
+// `current_timestamp_group_`.
+bool InterArrivalDelta::NewTimestampGroup(Timestamp arrival_time,
+                                          Timestamp send_time) const {
+  if (current_timestamp_group_.IsFirstPacket()) {
+    return false;
+  } else if (BelongsToBurst(arrival_time, send_time)) {
+    return false;
+  } else {
+    return send_time - current_timestamp_group_.first_send_time >
+           send_time_group_length_;
+  }
+}
+
+bool InterArrivalDelta::BelongsToBurst(Timestamp arrival_time,
+                                       Timestamp send_time) const {
+  RTC_DCHECK(current_timestamp_group_.complete_time.IsFinite());
+  TimeDelta arrival_time_delta =
+      arrival_time - current_timestamp_group_.complete_time;
+  TimeDelta send_time_delta = send_time - current_timestamp_group_.send_time;
+  if (send_time_delta.IsZero())
+    return true;
+  TimeDelta propagation_delta = arrival_time_delta - send_time_delta;
+  if (propagation_delta < TimeDelta::Zero() &&
+      arrival_time_delta <= kBurstDeltaThreshold &&
+      arrival_time - current_timestamp_group_.first_arrival < kMaxBurstDuration)
+    return true;
+  return false;
+}
+
+void InterArrivalDelta::Reset() {
+  num_consecutive_reordered_packets_ = 0;
+  current_timestamp_group_ = SendTimeGroup();
+  prev_timestamp_group_ = SendTimeGroup();
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.h
new file mode 100644
index 0000000000..4046590eeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/inter_arrival_delta.h
@@ -0,0 +1,90 @@
+/*
+ *  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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
+
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+
+// Helper class to compute the inter-arrival time delta and the size delta
+// between two send bursts. This code is branched from
+// modules/remote_bitrate_estimator/inter_arrival.
+class InterArrivalDelta {
+ public:
+  // After this many packet groups received out of order InterArrival will
+  // reset, assuming that clocks have made a jump.
+  static constexpr int kReorderedResetThreshold = 3;
+  static constexpr TimeDelta kArrivalTimeOffsetThreshold =
+      TimeDelta::Seconds(3);
+
+  // A send time group is defined as all packets with a send time which are at
+  // most send_time_group_length older than the first timestamp in that
+  // group.
+  explicit InterArrivalDelta(TimeDelta send_time_group_length);
+
+  InterArrivalDelta() = delete;
+  InterArrivalDelta(const InterArrivalDelta&) = delete;
+  InterArrivalDelta& operator=(const InterArrivalDelta&) = delete;
+
+  // This function returns true if a delta was computed, or false if the current
+  // group is still incomplete or if only one group has been completed.
+  // `send_time` is the send time.
+  // `arrival_time` is the time at which the packet arrived.
+  // `packet_size` is the size of the packet.
+  // `timestamp_delta` (output) is the computed send time delta.
+  // `arrival_time_delta` (output) is the computed arrival-time delta.
+  // `packet_size_delta` (output) is the computed size delta.
+  bool ComputeDeltas(Timestamp send_time,
+                     Timestamp arrival_time,
+                     Timestamp system_time,
+                     size_t packet_size,
+                     TimeDelta* send_time_delta,
+                     TimeDelta* arrival_time_delta,
+                     int* packet_size_delta);
+
+ private:
+  struct SendTimeGroup {
+    SendTimeGroup()
+        : size(0),
+          first_send_time(Timestamp::MinusInfinity()),
+          send_time(Timestamp::MinusInfinity()),
+          first_arrival(Timestamp::MinusInfinity()),
+          complete_time(Timestamp::MinusInfinity()),
+          last_system_time(Timestamp::MinusInfinity()) {}
+
+    bool IsFirstPacket() const { return complete_time.IsInfinite(); }
+
+    size_t size;
+    Timestamp first_send_time;
+    Timestamp send_time;
+    Timestamp first_arrival;
+    Timestamp complete_time;
+    Timestamp last_system_time;
+  };
+
+  // Returns true if the last packet was the end of the current batch and the
+  // packet with `send_time` is the first of a new batch.
+  bool NewTimestampGroup(Timestamp arrival_time, Timestamp send_time) const;
+
+  bool BelongsToBurst(Timestamp arrival_time, Timestamp send_time) const;
+
+  void Reset();
+
+  const TimeDelta send_time_group_length_;
+  SendTimeGroup current_timestamp_group_;
+  SendTimeGroup prev_timestamp_group_;
+  int num_consecutive_reordered_packets_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_INTER_ARRIVAL_DELTA_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc
new file mode 100644
index 0000000000..9fd537a422
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc
@@ -0,0 +1,77 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/link_capacity_estimator.h"
+
+#include <algorithm>
+
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+LinkCapacityEstimator::LinkCapacityEstimator() {}
+
+DataRate LinkCapacityEstimator::UpperBound() const {
+  if (estimate_kbps_.has_value())
+    return DataRate::KilobitsPerSec(estimate_kbps_.value() +
+                                    3 * deviation_estimate_kbps());
+  return DataRate::Infinity();
+}
+
+DataRate LinkCapacityEstimator::LowerBound() const {
+  if (estimate_kbps_.has_value())
+    return DataRate::KilobitsPerSec(
+        std::max(0.0, estimate_kbps_.value() - 3 * deviation_estimate_kbps()));
+  return DataRate::Zero();
+}
+
+void LinkCapacityEstimator::Reset() {
+  estimate_kbps_.reset();
+}
+
+void LinkCapacityEstimator::OnOveruseDetected(DataRate acknowledged_rate) {
+  Update(acknowledged_rate, 0.05);
+}
+
+void LinkCapacityEstimator::OnProbeRate(DataRate probe_rate) {
+  Update(probe_rate, 0.5);
+}
+
+void LinkCapacityEstimator::Update(DataRate capacity_sample, double alpha) {
+  double sample_kbps = capacity_sample.kbps();
+  if (!estimate_kbps_.has_value()) {
+    estimate_kbps_ = sample_kbps;
+  } else {
+    estimate_kbps_ = (1 - alpha) * estimate_kbps_.value() + alpha * sample_kbps;
+  }
+  // Estimate the variance of the link capacity estimate and normalize the
+  // variance with the link capacity estimate.
+  const double norm = std::max(estimate_kbps_.value(), 1.0);
+  double error_kbps = estimate_kbps_.value() - sample_kbps;
+  deviation_kbps_ =
+      (1 - alpha) * deviation_kbps_ + alpha * error_kbps * error_kbps / norm;
+  // 0.4 ~= 14 kbit/s at 500 kbit/s
+  // 2.5f ~= 35 kbit/s at 500 kbit/s
+  deviation_kbps_ = rtc::SafeClamp(deviation_kbps_, 0.4f, 2.5f);
+}
+
+bool LinkCapacityEstimator::has_estimate() const {
+  return estimate_kbps_.has_value();
+}
+
+DataRate LinkCapacityEstimator::estimate() const {
+  return DataRate::KilobitsPerSec(*estimate_kbps_);
+}
+
+double LinkCapacityEstimator::deviation_estimate_kbps() const {
+  // Calculate the max bit rate std dev given the normalized
+  // variance and the current throughput bitrate. The standard deviation will
+  // only be used if estimate_kbps_ has a value.
+  return sqrt(deviation_kbps_ * estimate_kbps_.value());
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h
new file mode 100644
index 0000000000..aa23491d9d
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+class LinkCapacityEstimator {
+ public:
+  LinkCapacityEstimator();
+  DataRate UpperBound() const;
+  DataRate LowerBound() const;
+  void Reset();
+  void OnOveruseDetected(DataRate acknowledged_rate);
+  void OnProbeRate(DataRate probe_rate);
+  bool has_estimate() const;
+  DataRate estimate() const;
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Update(DataRate capacity_sample, double alpha);
+
+  double deviation_estimate_kbps() const;
+  absl::optional<double> estimate_kbps_;
+  double deviation_kbps_ = 0.4;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LINK_CAPACITY_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build
new file mode 100644
index 0000000000..829ff47d87
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator_gn/moz.build
@@ -0,0 +1,221 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("link_capacity_estimator_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc
new file mode 100644
index 0000000000..7524c84d92
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc
@@ -0,0 +1,260 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h"
+
+#include <algorithm>
+#include <string>
+#include <vector>
+
+#include "absl/strings/match.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+namespace {
+const char kBweLossBasedControl[] = "WebRTC-Bwe-LossBasedControl";
+
+// Expecting RTCP feedback to be sent with roughly 1s intervals, a 5s gap
+// indicates a channel outage.
+constexpr TimeDelta kMaxRtcpFeedbackInterval = TimeDelta::Millis(5000);
+
+// Increase slower when RTT is high.
+double GetIncreaseFactor(const LossBasedControlConfig& config, TimeDelta rtt) {
+  // Clamp the RTT
+  if (rtt < config.increase_low_rtt) {
+    rtt = config.increase_low_rtt;
+  } else if (rtt > config.increase_high_rtt) {
+    rtt = config.increase_high_rtt;
+  }
+  auto rtt_range = config.increase_high_rtt.Get() - config.increase_low_rtt;
+  if (rtt_range <= TimeDelta::Zero()) {
+    RTC_DCHECK_NOTREACHED();  // Only on misconfiguration.
+    return config.min_increase_factor;
+  }
+  auto rtt_offset = rtt - config.increase_low_rtt;
+  auto relative_offset = std::max(0.0, std::min(rtt_offset / rtt_range, 1.0));
+  auto factor_range = config.max_increase_factor - config.min_increase_factor;
+  return config.min_increase_factor + (1 - relative_offset) * factor_range;
+}
+
+double LossFromBitrate(DataRate bitrate,
+                       DataRate loss_bandwidth_balance,
+                       double exponent) {
+  if (loss_bandwidth_balance >= bitrate)
+    return 1.0;
+  return pow(loss_bandwidth_balance / bitrate, exponent);
+}
+
+DataRate BitrateFromLoss(double loss,
+                         DataRate loss_bandwidth_balance,
+                         double exponent) {
+  if (exponent <= 0) {
+    RTC_DCHECK_NOTREACHED();
+    return DataRate::Infinity();
+  }
+  if (loss < 1e-5)
+    return DataRate::Infinity();
+  return loss_bandwidth_balance * pow(loss, -1.0 / exponent);
+}
+
+double ExponentialUpdate(TimeDelta window, TimeDelta interval) {
+  // Use the convention that exponential window length (which is really
+  // infinite) is the time it takes to dampen to 1/e.
+  if (window <= TimeDelta::Zero()) {
+    RTC_DCHECK_NOTREACHED();
+    return 1.0f;
+  }
+  return 1.0f - exp(interval / window * -1.0);
+}
+
+bool IsEnabled(const webrtc::FieldTrialsView& key_value_config,
+               absl::string_view name) {
+  return absl::StartsWith(key_value_config.Lookup(name), "Enabled");
+}
+
+}  // namespace
+
+LossBasedControlConfig::LossBasedControlConfig(
+    const FieldTrialsView* key_value_config)
+    : enabled(IsEnabled(*key_value_config, kBweLossBasedControl)),
+      min_increase_factor("min_incr", 1.02),
+      max_increase_factor("max_incr", 1.08),
+      increase_low_rtt("incr_low_rtt", TimeDelta::Millis(200)),
+      increase_high_rtt("incr_high_rtt", TimeDelta::Millis(800)),
+      decrease_factor("decr", 0.99),
+      loss_window("loss_win", TimeDelta::Millis(800)),
+      loss_max_window("loss_max_win", TimeDelta::Millis(800)),
+      acknowledged_rate_max_window("ackrate_max_win", TimeDelta::Millis(800)),
+      increase_offset("incr_offset", DataRate::BitsPerSec(1000)),
+      loss_bandwidth_balance_increase("balance_incr",
+                                      DataRate::KilobitsPerSec(0.5)),
+      loss_bandwidth_balance_decrease("balance_decr",
+                                      DataRate::KilobitsPerSec(4)),
+      loss_bandwidth_balance_reset("balance_reset",
+                                   DataRate::KilobitsPerSec(0.1)),
+      loss_bandwidth_balance_exponent("exponent", 0.5),
+      allow_resets("resets", false),
+      decrease_interval("decr_intvl", TimeDelta::Millis(300)),
+      loss_report_timeout("timeout", TimeDelta::Millis(6000)) {
+  ParseFieldTrial(
+      {&min_increase_factor, &max_increase_factor, &increase_low_rtt,
+       &increase_high_rtt, &decrease_factor, &loss_window, &loss_max_window,
+       &acknowledged_rate_max_window, &increase_offset,
+       &loss_bandwidth_balance_increase, &loss_bandwidth_balance_decrease,
+       &loss_bandwidth_balance_reset, &loss_bandwidth_balance_exponent,
+       &allow_resets, &decrease_interval, &loss_report_timeout},
+      key_value_config->Lookup(kBweLossBasedControl));
+}
+LossBasedControlConfig::LossBasedControlConfig(const LossBasedControlConfig&) =
+    default;
+LossBasedControlConfig::~LossBasedControlConfig() = default;
+
+LossBasedBandwidthEstimation::LossBasedBandwidthEstimation(
+    const FieldTrialsView* key_value_config)
+    : config_(key_value_config),
+      average_loss_(0),
+      average_loss_max_(0),
+      loss_based_bitrate_(DataRate::Zero()),
+      acknowledged_bitrate_max_(DataRate::Zero()),
+      acknowledged_bitrate_last_update_(Timestamp::MinusInfinity()),
+      time_last_decrease_(Timestamp::MinusInfinity()),
+      has_decreased_since_last_loss_report_(false),
+      last_loss_packet_report_(Timestamp::MinusInfinity()),
+      last_loss_ratio_(0) {}
+
+void LossBasedBandwidthEstimation::UpdateLossStatistics(
+    const std::vector<PacketResult>& packet_results,
+    Timestamp at_time) {
+  if (packet_results.empty()) {
+    RTC_DCHECK_NOTREACHED();
+    return;
+  }
+  int loss_count = 0;
+  for (const auto& pkt : packet_results) {
+    loss_count += !pkt.IsReceived() ? 1 : 0;
+  }
+  last_loss_ratio_ = static_cast<double>(loss_count) / packet_results.size();
+  const TimeDelta time_passed = last_loss_packet_report_.IsFinite()
+                                    ? at_time - last_loss_packet_report_
+                                    : TimeDelta::Seconds(1);
+  last_loss_packet_report_ = at_time;
+  has_decreased_since_last_loss_report_ = false;
+
+  average_loss_ += ExponentialUpdate(config_.loss_window, time_passed) *
+                   (last_loss_ratio_ - average_loss_);
+  if (average_loss_ > average_loss_max_) {
+    average_loss_max_ = average_loss_;
+  } else {
+    average_loss_max_ +=
+        ExponentialUpdate(config_.loss_max_window, time_passed) *
+        (average_loss_ - average_loss_max_);
+  }
+}
+
+void LossBasedBandwidthEstimation::UpdateAcknowledgedBitrate(
+    DataRate acknowledged_bitrate,
+    Timestamp at_time) {
+  const TimeDelta time_passed =
+      acknowledged_bitrate_last_update_.IsFinite()
+          ? at_time - acknowledged_bitrate_last_update_
+          : TimeDelta::Seconds(1);
+  acknowledged_bitrate_last_update_ = at_time;
+  if (acknowledged_bitrate > acknowledged_bitrate_max_) {
+    acknowledged_bitrate_max_ = acknowledged_bitrate;
+  } else {
+    acknowledged_bitrate_max_ -=
+        ExponentialUpdate(config_.acknowledged_rate_max_window, time_passed) *
+        (acknowledged_bitrate_max_ - acknowledged_bitrate);
+  }
+}
+
+DataRate LossBasedBandwidthEstimation::Update(Timestamp at_time,
+                                              DataRate min_bitrate,
+                                              DataRate wanted_bitrate,
+                                              TimeDelta last_round_trip_time) {
+  if (loss_based_bitrate_.IsZero()) {
+    loss_based_bitrate_ = wanted_bitrate;
+  }
+
+  // Only increase if loss has been low for some time.
+  const double loss_estimate_for_increase = average_loss_max_;
+  // Avoid multiple decreases from averaging over one loss spike.
+  const double loss_estimate_for_decrease =
+      std::min(average_loss_, last_loss_ratio_);
+  const bool allow_decrease =
+      !has_decreased_since_last_loss_report_ &&
+      (at_time - time_last_decrease_ >=
+       last_round_trip_time + config_.decrease_interval);
+  // If packet lost reports are too old, dont increase bitrate.
+  const bool loss_report_valid =
+      at_time - last_loss_packet_report_ < 1.2 * kMaxRtcpFeedbackInterval;
+
+  if (loss_report_valid && config_.allow_resets &&
+      loss_estimate_for_increase < loss_reset_threshold()) {
+    loss_based_bitrate_ = wanted_bitrate;
+  } else if (loss_report_valid &&
+             loss_estimate_for_increase < loss_increase_threshold()) {
+    // Increase bitrate by RTT-adaptive ratio.
+    DataRate new_increased_bitrate =
+        min_bitrate * GetIncreaseFactor(config_, last_round_trip_time) +
+        config_.increase_offset;
+    // The bitrate that would make the loss "just high enough".
+    const DataRate new_increased_bitrate_cap = BitrateFromLoss(
+        loss_estimate_for_increase, config_.loss_bandwidth_balance_increase,
+        config_.loss_bandwidth_balance_exponent);
+    new_increased_bitrate =
+        std::min(new_increased_bitrate, new_increased_bitrate_cap);
+    loss_based_bitrate_ = std::max(new_increased_bitrate, loss_based_bitrate_);
+  } else if (loss_estimate_for_decrease > loss_decrease_threshold() &&
+             allow_decrease) {
+    // The bitrate that would make the loss "just acceptable".
+    const DataRate new_decreased_bitrate_floor = BitrateFromLoss(
+        loss_estimate_for_decrease, config_.loss_bandwidth_balance_decrease,
+        config_.loss_bandwidth_balance_exponent);
+    DataRate new_decreased_bitrate =
+        std::max(decreased_bitrate(), new_decreased_bitrate_floor);
+    if (new_decreased_bitrate < loss_based_bitrate_) {
+      time_last_decrease_ = at_time;
+      has_decreased_since_last_loss_report_ = true;
+      loss_based_bitrate_ = new_decreased_bitrate;
+    }
+  }
+  return loss_based_bitrate_;
+}
+
+void LossBasedBandwidthEstimation::Initialize(DataRate bitrate) {
+  loss_based_bitrate_ = bitrate;
+  average_loss_ = 0;
+  average_loss_max_ = 0;
+}
+
+double LossBasedBandwidthEstimation::loss_reset_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_reset,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+double LossBasedBandwidthEstimation::loss_increase_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_increase,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+double LossBasedBandwidthEstimation::loss_decrease_threshold() const {
+  return LossFromBitrate(loss_based_bitrate_,
+                         config_.loss_bandwidth_balance_decrease,
+                         config_.loss_bandwidth_balance_exponent);
+}
+
+DataRate LossBasedBandwidthEstimation::decreased_bitrate() const {
+  return config_.decrease_factor * acknowledged_bitrate_max_;
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h
new file mode 100644
index 0000000000..9f69caba89
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h
@@ -0,0 +1,97 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
+
+#include <vector>
+
+#include "api/field_trials_view.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+struct LossBasedControlConfig {
+  explicit LossBasedControlConfig(const FieldTrialsView* key_value_config);
+  LossBasedControlConfig(const LossBasedControlConfig&);
+  LossBasedControlConfig& operator=(const LossBasedControlConfig&) = default;
+  ~LossBasedControlConfig();
+  bool enabled;
+  FieldTrialParameter<double> min_increase_factor;
+  FieldTrialParameter<double> max_increase_factor;
+  FieldTrialParameter<TimeDelta> increase_low_rtt;
+  FieldTrialParameter<TimeDelta> increase_high_rtt;
+  FieldTrialParameter<double> decrease_factor;
+  FieldTrialParameter<TimeDelta> loss_window;
+  FieldTrialParameter<TimeDelta> loss_max_window;
+  FieldTrialParameter<TimeDelta> acknowledged_rate_max_window;
+  FieldTrialParameter<DataRate> increase_offset;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_increase;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_decrease;
+  FieldTrialParameter<DataRate> loss_bandwidth_balance_reset;
+  FieldTrialParameter<double> loss_bandwidth_balance_exponent;
+  FieldTrialParameter<bool> allow_resets;
+  FieldTrialParameter<TimeDelta> decrease_interval;
+  FieldTrialParameter<TimeDelta> loss_report_timeout;
+};
+
+// Estimates an upper BWE limit based on loss.
+// It requires knowledge about lost packets and acknowledged bitrate.
+// Ie, this class require transport feedback.
+class LossBasedBandwidthEstimation {
+ public:
+  explicit LossBasedBandwidthEstimation(
+      const FieldTrialsView* key_value_config);
+  // Returns the new estimate.
+  DataRate Update(Timestamp at_time,
+                  DataRate min_bitrate,
+                  DataRate wanted_bitrate,
+                  TimeDelta last_round_trip_time);
+  void UpdateAcknowledgedBitrate(DataRate acknowledged_bitrate,
+                                 Timestamp at_time);
+  void Initialize(DataRate bitrate);
+  bool Enabled() const { return config_.enabled; }
+  // Returns true if LossBasedBandwidthEstimation is enabled and have
+  // received loss statistics. Ie, this class require transport feedback.
+  bool InUse() const {
+    return Enabled() && last_loss_packet_report_.IsFinite();
+  }
+  void UpdateLossStatistics(const std::vector<PacketResult>& packet_results,
+                            Timestamp at_time);
+  DataRate GetEstimate() const { return loss_based_bitrate_; }
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Reset(DataRate bitrate);
+  double loss_increase_threshold() const;
+  double loss_decrease_threshold() const;
+  double loss_reset_threshold() const;
+
+  DataRate decreased_bitrate() const;
+
+  const LossBasedControlConfig config_;
+  double average_loss_;
+  double average_loss_max_;
+  DataRate loss_based_bitrate_;
+  DataRate acknowledged_bitrate_max_;
+  Timestamp acknowledged_bitrate_last_update_;
+  Timestamp time_last_decrease_;
+  bool has_decreased_since_last_loss_report_;
+  Timestamp last_loss_packet_report_;
+  double last_loss_ratio_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BANDWIDTH_ESTIMATION_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build
new file mode 100644
index 0000000000..aae4b82520
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v1_gn/moz.build
@@ -0,0 +1,226 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("loss_based_bwe_v1_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
new file mode 100644
index 0000000000..b6efdeee9e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc
@@ -0,0 +1,1080 @@
+/*
+ *  Copyright 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 "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+
+#include <algorithm>
+#include <cmath>
+#include <complex>
+#include <cstddef>
+#include <cstdlib>
+#include <limits>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "rtc_base/experiments/field_trial_list.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+
+bool IsValid(DataRate datarate) {
+  return datarate.IsFinite();
+}
+
+bool IsValid(absl::optional<DataRate> datarate) {
+  return datarate.has_value() && IsValid(datarate.value());
+}
+
+bool IsValid(Timestamp timestamp) {
+  return timestamp.IsFinite();
+}
+
+struct PacketResultsSummary {
+  int num_packets = 0;
+  int num_lost_packets = 0;
+  DataSize total_size = DataSize::Zero();
+  Timestamp first_send_time = Timestamp::PlusInfinity();
+  Timestamp last_send_time = Timestamp::MinusInfinity();
+};
+
+// Returns a `PacketResultsSummary` where `first_send_time` is `PlusInfinity,
+// and `last_send_time` is `MinusInfinity`, if `packet_results` is empty.
+PacketResultsSummary GetPacketResultsSummary(
+    rtc::ArrayView<const PacketResult> packet_results) {
+  PacketResultsSummary packet_results_summary;
+
+  packet_results_summary.num_packets = packet_results.size();
+  for (const PacketResult& packet : packet_results) {
+    if (!packet.IsReceived()) {
+      packet_results_summary.num_lost_packets++;
+    }
+    packet_results_summary.total_size += packet.sent_packet.size;
+    packet_results_summary.first_send_time = std::min(
+        packet_results_summary.first_send_time, packet.sent_packet.send_time);
+    packet_results_summary.last_send_time = std::max(
+        packet_results_summary.last_send_time, packet.sent_packet.send_time);
+  }
+
+  return packet_results_summary;
+}
+
+double GetLossProbability(double inherent_loss,
+                          DataRate loss_limited_bandwidth,
+                          DataRate sending_rate) {
+  if (inherent_loss < 0.0 || inherent_loss > 1.0) {
+    RTC_LOG(LS_WARNING) << "The inherent loss must be in [0,1]: "
+                        << inherent_loss;
+    inherent_loss = std::min(std::max(inherent_loss, 0.0), 1.0);
+  }
+  if (!sending_rate.IsFinite()) {
+    RTC_LOG(LS_WARNING) << "The sending rate must be finite: "
+                        << ToString(sending_rate);
+  }
+  if (!loss_limited_bandwidth.IsFinite()) {
+    RTC_LOG(LS_WARNING) << "The loss limited bandwidth must be finite: "
+                        << ToString(loss_limited_bandwidth);
+  }
+
+  double loss_probability = inherent_loss;
+  if (IsValid(sending_rate) && IsValid(loss_limited_bandwidth) &&
+      (sending_rate > loss_limited_bandwidth)) {
+    loss_probability += (1 - inherent_loss) *
+                        (sending_rate - loss_limited_bandwidth) / sending_rate;
+  }
+  return std::min(std::max(loss_probability, 1.0e-6), 1.0 - 1.0e-6);
+}
+
+}  // namespace
+
+LossBasedBweV2::LossBasedBweV2(const FieldTrialsView* key_value_config)
+    : config_(CreateConfig(key_value_config)) {
+  if (!config_.has_value()) {
+    RTC_LOG(LS_VERBOSE) << "The configuration does not specify that the "
+                           "estimator should be enabled, disabling it.";
+    return;
+  }
+  if (!IsConfigValid()) {
+    RTC_LOG(LS_WARNING)
+        << "The configuration is not valid, disabling the estimator.";
+    config_.reset();
+    return;
+  }
+
+  current_estimate_.inherent_loss = config_->initial_inherent_loss_estimate;
+  observations_.resize(config_->observation_window_size);
+  temporal_weights_.resize(config_->observation_window_size);
+  instant_upper_bound_temporal_weights_.resize(
+      config_->observation_window_size);
+  CalculateTemporalWeights();
+}
+
+bool LossBasedBweV2::IsEnabled() const {
+  return config_.has_value();
+}
+
+bool LossBasedBweV2::IsReady() const {
+  return IsEnabled() && IsValid(current_estimate_.loss_limited_bandwidth) &&
+         num_observations_ > 0;
+}
+
+LossBasedBweV2::Result LossBasedBweV2::GetLossBasedResult() const {
+  Result result;
+  result.state = current_state_;
+  if (!IsReady()) {
+    if (!IsEnabled()) {
+      RTC_LOG(LS_WARNING)
+          << "The estimator must be enabled before it can be used.";
+    } else {
+      if (!IsValid(current_estimate_.loss_limited_bandwidth)) {
+        RTC_LOG(LS_WARNING)
+            << "The estimator must be initialized before it can be used.";
+      }
+      if (num_observations_ <= 0) {
+        RTC_LOG(LS_WARNING) << "The estimator must receive enough loss "
+                               "statistics before it can be used.";
+      }
+    }
+    result.bandwidth_estimate = IsValid(delay_based_estimate_)
+                                    ? delay_based_estimate_
+                                    : DataRate::PlusInfinity();
+    return result;
+  }
+
+  if (IsValid(delay_based_estimate_)) {
+    result.bandwidth_estimate =
+        std::min({current_estimate_.loss_limited_bandwidth,
+                  GetInstantUpperBound(), delay_based_estimate_});
+  } else {
+    result.bandwidth_estimate = std::min(
+        current_estimate_.loss_limited_bandwidth, GetInstantUpperBound());
+  }
+  return result;
+}
+
+void LossBasedBweV2::SetAcknowledgedBitrate(DataRate acknowledged_bitrate) {
+  if (IsValid(acknowledged_bitrate)) {
+    acknowledged_bitrate_ = acknowledged_bitrate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The acknowledged bitrate must be finite: "
+                        << ToString(acknowledged_bitrate);
+  }
+}
+
+void LossBasedBweV2::SetBandwidthEstimate(DataRate bandwidth_estimate) {
+  if (IsValid(bandwidth_estimate)) {
+    current_estimate_.loss_limited_bandwidth = bandwidth_estimate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The bandwidth estimate must be finite: "
+                        << ToString(bandwidth_estimate);
+  }
+}
+
+void LossBasedBweV2::SetMinMaxBitrate(DataRate min_bitrate,
+                                      DataRate max_bitrate) {
+  if (IsValid(min_bitrate)) {
+    min_bitrate_ = min_bitrate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The min bitrate must be finite: "
+                        << ToString(min_bitrate);
+  }
+
+  if (IsValid(max_bitrate)) {
+    max_bitrate_ = max_bitrate;
+  } else {
+    RTC_LOG(LS_WARNING) << "The max bitrate must be finite: "
+                        << ToString(max_bitrate);
+  }
+}
+
+void LossBasedBweV2::SetProbeBitrate(absl::optional<DataRate> probe_bitrate) {
+  if (probe_bitrate.has_value() && IsValid(probe_bitrate.value())) {
+    if (!IsValid(probe_bitrate_) || probe_bitrate_ > probe_bitrate.value()) {
+      probe_bitrate_ = probe_bitrate.value();
+    }
+  }
+}
+
+void LossBasedBweV2::UpdateBandwidthEstimate(
+    rtc::ArrayView<const PacketResult> packet_results,
+    DataRate delay_based_estimate,
+    BandwidthUsage delay_detector_state,
+    absl::optional<DataRate> probe_bitrate,
+    DataRate upper_link_capacity) {
+  delay_based_estimate_ = delay_based_estimate;
+  upper_link_capacity_ = upper_link_capacity;
+  if (!IsEnabled()) {
+    RTC_LOG(LS_WARNING)
+        << "The estimator must be enabled before it can be used.";
+    return;
+  }
+  SetProbeBitrate(probe_bitrate);
+  if (packet_results.empty()) {
+    RTC_LOG(LS_VERBOSE)
+        << "The estimate cannot be updated without any loss statistics.";
+    return;
+  }
+
+  if (!PushBackObservation(packet_results, delay_detector_state)) {
+    return;
+  }
+
+  if (!IsValid(current_estimate_.loss_limited_bandwidth)) {
+    RTC_LOG(LS_VERBOSE)
+        << "The estimator must be initialized before it can be used.";
+    return;
+  }
+
+  ChannelParameters best_candidate = current_estimate_;
+  double objective_max = std::numeric_limits<double>::lowest();
+  for (ChannelParameters candidate : GetCandidates()) {
+    NewtonsMethodUpdate(candidate);
+
+    const double candidate_objective = GetObjective(candidate);
+    if (candidate_objective > objective_max) {
+      objective_max = candidate_objective;
+      best_candidate = candidate;
+    }
+  }
+  if (best_candidate.loss_limited_bandwidth <
+      current_estimate_.loss_limited_bandwidth) {
+    last_time_estimate_reduced_ = last_send_time_most_recent_observation_;
+  }
+
+  // Do not increase the estimate if the average loss is greater than current
+  // inherent loss.
+  if (GetAverageReportedLossRatio() > best_candidate.inherent_loss &&
+      config_->not_increase_if_inherent_loss_less_than_average_loss &&
+      current_estimate_.loss_limited_bandwidth <
+          best_candidate.loss_limited_bandwidth) {
+    best_candidate.loss_limited_bandwidth =
+        current_estimate_.loss_limited_bandwidth;
+  }
+
+  if (IsBandwidthLimitedDueToLoss()) {
+    // Bound the estimate increase if:
+    // 1. The estimate has been increased for less than
+    // `delayed_increase_window` ago, and
+    // 2. The best candidate is greater than bandwidth_limit_in_current_window.
+    if (recovering_after_loss_timestamp_.IsFinite() &&
+        recovering_after_loss_timestamp_ + config_->delayed_increase_window >
+            last_send_time_most_recent_observation_ &&
+        best_candidate.loss_limited_bandwidth >
+            bandwidth_limit_in_current_window_) {
+      best_candidate.loss_limited_bandwidth =
+          bandwidth_limit_in_current_window_;
+    }
+
+    bool increasing_when_loss_limited =
+        IsEstimateIncreasingWhenLossLimited(best_candidate);
+    // Bound the best candidate by the acked bitrate unless there is a recent
+    // probe result.
+    if (increasing_when_loss_limited && !IsValid(probe_bitrate_) &&
+        IsValid(acknowledged_bitrate_)) {
+      best_candidate.loss_limited_bandwidth =
+          IsValid(best_candidate.loss_limited_bandwidth)
+              ? std::min(best_candidate.loss_limited_bandwidth,
+                         config_->bandwidth_rampup_upper_bound_factor *
+                             (*acknowledged_bitrate_))
+              : config_->bandwidth_rampup_upper_bound_factor *
+                    (*acknowledged_bitrate_);
+    }
+
+    // Use probe bitrate as the estimate as probe bitrate is trusted to be
+    // correct. After being used, the probe bitrate is reset.
+    if (config_->probe_integration_enabled && IsValid(probe_bitrate_)) {
+      best_candidate.loss_limited_bandwidth =
+          std::min(probe_bitrate_, best_candidate.loss_limited_bandwidth);
+      probe_bitrate_ = DataRate::MinusInfinity();
+    }
+  }
+
+  if (IsEstimateIncreasingWhenLossLimited(best_candidate) &&
+      best_candidate.loss_limited_bandwidth < delay_based_estimate) {
+    current_state_ = LossBasedState::kIncreasing;
+  } else if (best_candidate.loss_limited_bandwidth < delay_based_estimate_) {
+    current_state_ = LossBasedState::kDecreasing;
+  } else if (best_candidate.loss_limited_bandwidth >= delay_based_estimate_) {
+    current_state_ = LossBasedState::kDelayBasedEstimate;
+  }
+  current_estimate_ = best_candidate;
+
+  if (IsBandwidthLimitedDueToLoss() &&
+      (recovering_after_loss_timestamp_.IsInfinite() ||
+       recovering_after_loss_timestamp_ + config_->delayed_increase_window <
+           last_send_time_most_recent_observation_)) {
+    bandwidth_limit_in_current_window_ =
+        std::max(kCongestionControllerMinBitrate,
+                 current_estimate_.loss_limited_bandwidth *
+                     config_->max_increase_factor);
+    recovering_after_loss_timestamp_ = last_send_time_most_recent_observation_;
+  }
+}
+
+bool LossBasedBweV2::IsEstimateIncreasingWhenLossLimited(
+    const ChannelParameters& best_candidate) {
+  return (current_estimate_.loss_limited_bandwidth <
+              best_candidate.loss_limited_bandwidth ||
+          (current_estimate_.loss_limited_bandwidth ==
+               best_candidate.loss_limited_bandwidth &&
+           current_state_ == LossBasedState::kIncreasing)) &&
+         IsBandwidthLimitedDueToLoss();
+}
+
+// Returns a `LossBasedBweV2::Config` iff the `key_value_config` specifies a
+// configuration for the `LossBasedBweV2` which is explicitly enabled.
+absl::optional<LossBasedBweV2::Config> LossBasedBweV2::CreateConfig(
+    const FieldTrialsView* key_value_config) {
+  FieldTrialParameter<bool> enabled("Enabled", true);
+  FieldTrialParameter<double> bandwidth_rampup_upper_bound_factor(
+      "BwRampupUpperBoundFactor", 1000000.0);
+  FieldTrialParameter<double> rampup_acceleration_max_factor(
+      "BwRampupAccelMaxFactor", 0.0);
+  FieldTrialParameter<TimeDelta> rampup_acceleration_maxout_time(
+      "BwRampupAccelMaxoutTime", TimeDelta::Seconds(60));
+  FieldTrialList<double> candidate_factors("CandidateFactors",
+                                           {1.02, 1.0, 0.95});
+  FieldTrialParameter<double> higher_bandwidth_bias_factor("HigherBwBiasFactor",
+                                                           0.0002);
+  FieldTrialParameter<double> higher_log_bandwidth_bias_factor(
+      "HigherLogBwBiasFactor", 0.02);
+  FieldTrialParameter<double> inherent_loss_lower_bound(
+      "InherentLossLowerBound", 1.0e-3);
+  FieldTrialParameter<double> loss_threshold_of_high_bandwidth_preference(
+      "LossThresholdOfHighBandwidthPreference", 0.15);
+  FieldTrialParameter<double> bandwidth_preference_smoothing_factor(
+      "BandwidthPreferenceSmoothingFactor", 0.002);
+  FieldTrialParameter<DataRate> inherent_loss_upper_bound_bandwidth_balance(
+      "InherentLossUpperBoundBwBalance", DataRate::KilobitsPerSec(75.0));
+  FieldTrialParameter<double> inherent_loss_upper_bound_offset(
+      "InherentLossUpperBoundOffset", 0.05);
+  FieldTrialParameter<double> initial_inherent_loss_estimate(
+      "InitialInherentLossEstimate", 0.01);
+  FieldTrialParameter<int> newton_iterations("NewtonIterations", 1);
+  FieldTrialParameter<double> newton_step_size("NewtonStepSize", 0.75);
+  FieldTrialParameter<bool> append_acknowledged_rate_candidate(
+      "AckedRateCandidate", true);
+  FieldTrialParameter<bool> append_delay_based_estimate_candidate(
+      "DelayBasedCandidate", true);
+  FieldTrialParameter<TimeDelta> observation_duration_lower_bound(
+      "ObservationDurationLowerBound", TimeDelta::Millis(250));
+  FieldTrialParameter<int> observation_window_size("ObservationWindowSize", 20);
+  FieldTrialParameter<double> sending_rate_smoothing_factor(
+      "SendingRateSmoothingFactor", 0.0);
+  FieldTrialParameter<double> instant_upper_bound_temporal_weight_factor(
+      "InstantUpperBoundTemporalWeightFactor", 0.9);
+  FieldTrialParameter<DataRate> instant_upper_bound_bandwidth_balance(
+      "InstantUpperBoundBwBalance", DataRate::KilobitsPerSec(75.0));
+  FieldTrialParameter<double> instant_upper_bound_loss_offset(
+      "InstantUpperBoundLossOffset", 0.05);
+  FieldTrialParameter<double> temporal_weight_factor("TemporalWeightFactor",
+                                                     0.9);
+  FieldTrialParameter<double> bandwidth_backoff_lower_bound_factor(
+      "BwBackoffLowerBoundFactor", 1.0);
+  FieldTrialParameter<bool> trendline_integration_enabled(
+      "TrendlineIntegrationEnabled", false);
+  FieldTrialParameter<int> trendline_observations_window_size(
+      "TrendlineObservationsWindowSize", 20);
+  FieldTrialParameter<double> max_increase_factor("MaxIncreaseFactor", 1.3);
+  FieldTrialParameter<TimeDelta> delayed_increase_window(
+      "DelayedIncreaseWindow", TimeDelta::Millis(300));
+  FieldTrialParameter<bool> use_acked_bitrate_only_when_overusing(
+      "UseAckedBitrateOnlyWhenOverusing", false);
+  FieldTrialParameter<bool>
+      not_increase_if_inherent_loss_less_than_average_loss(
+          "NotIncreaseIfInherentLossLessThanAverageLoss", true);
+  FieldTrialParameter<double> high_loss_rate_threshold("HighLossRateThreshold",
+                                                       1.0);
+  FieldTrialParameter<DataRate> bandwidth_cap_at_high_loss_rate(
+      "BandwidthCapAtHighLossRate", DataRate::KilobitsPerSec(500.0));
+  FieldTrialParameter<double> slope_of_bwe_high_loss_func(
+      "SlopeOfBweHighLossFunc", 1000);
+  FieldTrialParameter<bool> probe_integration_enabled("ProbeIntegrationEnabled",
+                                                      false);
+  FieldTrialParameter<bool> bound_by_upper_link_capacity_when_loss_limited(
+      "BoundByUpperLinkCapacityWhenLossLimited", true);
+  if (key_value_config) {
+    ParseFieldTrial({&enabled,
+                     &bandwidth_rampup_upper_bound_factor,
+                     &rampup_acceleration_max_factor,
+                     &rampup_acceleration_maxout_time,
+                     &candidate_factors,
+                     &higher_bandwidth_bias_factor,
+                     &higher_log_bandwidth_bias_factor,
+                     &inherent_loss_lower_bound,
+                     &loss_threshold_of_high_bandwidth_preference,
+                     &bandwidth_preference_smoothing_factor,
+                     &inherent_loss_upper_bound_bandwidth_balance,
+                     &inherent_loss_upper_bound_offset,
+                     &initial_inherent_loss_estimate,
+                     &newton_iterations,
+                     &newton_step_size,
+                     &append_acknowledged_rate_candidate,
+                     &append_delay_based_estimate_candidate,
+                     &observation_duration_lower_bound,
+                     &observation_window_size,
+                     &sending_rate_smoothing_factor,
+                     &instant_upper_bound_temporal_weight_factor,
+                     &instant_upper_bound_bandwidth_balance,
+                     &instant_upper_bound_loss_offset,
+                     &temporal_weight_factor,
+                     &bandwidth_backoff_lower_bound_factor,
+                     &trendline_integration_enabled,
+                     &trendline_observations_window_size,
+                     &max_increase_factor,
+                     &delayed_increase_window,
+                     &use_acked_bitrate_only_when_overusing,
+                     &not_increase_if_inherent_loss_less_than_average_loss,
+                     &probe_integration_enabled,
+                     &high_loss_rate_threshold,
+                     &bandwidth_cap_at_high_loss_rate,
+                     &slope_of_bwe_high_loss_func,
+                     &bound_by_upper_link_capacity_when_loss_limited},
+                    key_value_config->Lookup("WebRTC-Bwe-LossBasedBweV2"));
+  }
+
+  absl::optional<Config> config;
+  if (!enabled.Get()) {
+    return config;
+  }
+  config.emplace(Config());
+  config->bandwidth_rampup_upper_bound_factor =
+      bandwidth_rampup_upper_bound_factor.Get();
+  config->rampup_acceleration_max_factor = rampup_acceleration_max_factor.Get();
+  config->rampup_acceleration_maxout_time =
+      rampup_acceleration_maxout_time.Get();
+  config->candidate_factors = candidate_factors.Get();
+  config->higher_bandwidth_bias_factor = higher_bandwidth_bias_factor.Get();
+  config->higher_log_bandwidth_bias_factor =
+      higher_log_bandwidth_bias_factor.Get();
+  config->inherent_loss_lower_bound = inherent_loss_lower_bound.Get();
+  config->loss_threshold_of_high_bandwidth_preference =
+      loss_threshold_of_high_bandwidth_preference.Get();
+  config->bandwidth_preference_smoothing_factor =
+      bandwidth_preference_smoothing_factor.Get();
+  config->inherent_loss_upper_bound_bandwidth_balance =
+      inherent_loss_upper_bound_bandwidth_balance.Get();
+  config->inherent_loss_upper_bound_offset =
+      inherent_loss_upper_bound_offset.Get();
+  config->initial_inherent_loss_estimate = initial_inherent_loss_estimate.Get();
+  config->newton_iterations = newton_iterations.Get();
+  config->newton_step_size = newton_step_size.Get();
+  config->append_acknowledged_rate_candidate =
+      append_acknowledged_rate_candidate.Get();
+  config->append_delay_based_estimate_candidate =
+      append_delay_based_estimate_candidate.Get();
+  config->observation_duration_lower_bound =
+      observation_duration_lower_bound.Get();
+  config->observation_window_size = observation_window_size.Get();
+  config->sending_rate_smoothing_factor = sending_rate_smoothing_factor.Get();
+  config->instant_upper_bound_temporal_weight_factor =
+      instant_upper_bound_temporal_weight_factor.Get();
+  config->instant_upper_bound_bandwidth_balance =
+      instant_upper_bound_bandwidth_balance.Get();
+  config->instant_upper_bound_loss_offset =
+      instant_upper_bound_loss_offset.Get();
+  config->temporal_weight_factor = temporal_weight_factor.Get();
+  config->bandwidth_backoff_lower_bound_factor =
+      bandwidth_backoff_lower_bound_factor.Get();
+  config->trendline_integration_enabled = trendline_integration_enabled.Get();
+  config->trendline_observations_window_size =
+      trendline_observations_window_size.Get();
+  config->max_increase_factor = max_increase_factor.Get();
+  config->delayed_increase_window = delayed_increase_window.Get();
+  config->use_acked_bitrate_only_when_overusing =
+      use_acked_bitrate_only_when_overusing.Get();
+  config->not_increase_if_inherent_loss_less_than_average_loss =
+      not_increase_if_inherent_loss_less_than_average_loss.Get();
+  config->high_loss_rate_threshold = high_loss_rate_threshold.Get();
+  config->bandwidth_cap_at_high_loss_rate =
+      bandwidth_cap_at_high_loss_rate.Get();
+  config->slope_of_bwe_high_loss_func = slope_of_bwe_high_loss_func.Get();
+  config->probe_integration_enabled = probe_integration_enabled.Get();
+  config->bound_by_upper_link_capacity_when_loss_limited =
+      bound_by_upper_link_capacity_when_loss_limited.Get();
+
+  return config;
+}
+
+bool LossBasedBweV2::IsConfigValid() const {
+  if (!config_.has_value()) {
+    return false;
+  }
+
+  bool valid = true;
+
+  if (config_->bandwidth_rampup_upper_bound_factor <= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The bandwidth rampup upper bound factor must be greater than 1: "
+        << config_->bandwidth_rampup_upper_bound_factor;
+    valid = false;
+  }
+  if (config_->rampup_acceleration_max_factor < 0.0) {
+    RTC_LOG(LS_WARNING)
+        << "The rampup acceleration max factor must be non-negative.: "
+        << config_->rampup_acceleration_max_factor;
+    valid = false;
+  }
+  if (config_->rampup_acceleration_maxout_time <= TimeDelta::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The rampup acceleration maxout time must be above zero: "
+        << config_->rampup_acceleration_maxout_time.seconds();
+    valid = false;
+  }
+  for (double candidate_factor : config_->candidate_factors) {
+    if (candidate_factor <= 0.0) {
+      RTC_LOG(LS_WARNING) << "All candidate factors must be greater than zero: "
+                          << candidate_factor;
+      valid = false;
+    }
+  }
+
+  // Ensure that the configuration allows generation of at least one candidate
+  // other than the current estimate.
+  if (!config_->append_acknowledged_rate_candidate &&
+      !config_->append_delay_based_estimate_candidate &&
+      !absl::c_any_of(config_->candidate_factors,
+                      [](double cf) { return cf != 1.0; })) {
+    RTC_LOG(LS_WARNING)
+        << "The configuration does not allow generating candidates. Specify "
+           "a candidate factor other than 1.0, allow the acknowledged rate "
+           "to be a candidate, and/or allow the delay based estimate to be a "
+           "candidate.";
+    valid = false;
+  }
+
+  if (config_->higher_bandwidth_bias_factor < 0.0) {
+    RTC_LOG(LS_WARNING)
+        << "The higher bandwidth bias factor must be non-negative: "
+        << config_->higher_bandwidth_bias_factor;
+    valid = false;
+  }
+  if (config_->inherent_loss_lower_bound < 0.0 ||
+      config_->inherent_loss_lower_bound >= 1.0) {
+    RTC_LOG(LS_WARNING) << "The inherent loss lower bound must be in [0, 1): "
+                        << config_->inherent_loss_lower_bound;
+    valid = false;
+  }
+  if (config_->loss_threshold_of_high_bandwidth_preference < 0.0 ||
+      config_->loss_threshold_of_high_bandwidth_preference >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The loss threshold of high bandwidth preference must be in [0, 1): "
+        << config_->loss_threshold_of_high_bandwidth_preference;
+    valid = false;
+  }
+  if (config_->bandwidth_preference_smoothing_factor <= 0.0 ||
+      config_->bandwidth_preference_smoothing_factor > 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The bandwidth preference smoothing factor must be in (0, 1]: "
+        << config_->bandwidth_preference_smoothing_factor;
+    valid = false;
+  }
+  if (config_->inherent_loss_upper_bound_bandwidth_balance <=
+      DataRate::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The inherent loss upper bound bandwidth balance "
+           "must be positive: "
+        << ToString(config_->inherent_loss_upper_bound_bandwidth_balance);
+    valid = false;
+  }
+  if (config_->inherent_loss_upper_bound_offset <
+          config_->inherent_loss_lower_bound ||
+      config_->inherent_loss_upper_bound_offset >= 1.0) {
+    RTC_LOG(LS_WARNING) << "The inherent loss upper bound must be greater "
+                           "than or equal to the inherent "
+                           "loss lower bound, which is "
+                        << config_->inherent_loss_lower_bound
+                        << ", and less than 1: "
+                        << config_->inherent_loss_upper_bound_offset;
+    valid = false;
+  }
+  if (config_->initial_inherent_loss_estimate < 0.0 ||
+      config_->initial_inherent_loss_estimate >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The initial inherent loss estimate must be in [0, 1): "
+        << config_->initial_inherent_loss_estimate;
+    valid = false;
+  }
+  if (config_->newton_iterations <= 0) {
+    RTC_LOG(LS_WARNING) << "The number of Newton iterations must be positive: "
+                        << config_->newton_iterations;
+    valid = false;
+  }
+  if (config_->newton_step_size <= 0.0) {
+    RTC_LOG(LS_WARNING) << "The Newton step size must be positive: "
+                        << config_->newton_step_size;
+    valid = false;
+  }
+  if (config_->observation_duration_lower_bound <= TimeDelta::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The observation duration lower bound must be positive: "
+        << ToString(config_->observation_duration_lower_bound);
+    valid = false;
+  }
+  if (config_->observation_window_size < 2) {
+    RTC_LOG(LS_WARNING) << "The observation window size must be at least 2: "
+                        << config_->observation_window_size;
+    valid = false;
+  }
+  if (config_->sending_rate_smoothing_factor < 0.0 ||
+      config_->sending_rate_smoothing_factor >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The sending rate smoothing factor must be in [0, 1): "
+        << config_->sending_rate_smoothing_factor;
+    valid = false;
+  }
+  if (config_->instant_upper_bound_temporal_weight_factor <= 0.0 ||
+      config_->instant_upper_bound_temporal_weight_factor > 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The instant upper bound temporal weight factor must be in (0, 1]"
+        << config_->instant_upper_bound_temporal_weight_factor;
+    valid = false;
+  }
+  if (config_->instant_upper_bound_bandwidth_balance <= DataRate::Zero()) {
+    RTC_LOG(LS_WARNING)
+        << "The instant upper bound bandwidth balance must be positive: "
+        << ToString(config_->instant_upper_bound_bandwidth_balance);
+    valid = false;
+  }
+  if (config_->instant_upper_bound_loss_offset < 0.0 ||
+      config_->instant_upper_bound_loss_offset >= 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The instant upper bound loss offset must be in [0, 1): "
+        << config_->instant_upper_bound_loss_offset;
+    valid = false;
+  }
+  if (config_->temporal_weight_factor <= 0.0 ||
+      config_->temporal_weight_factor > 1.0) {
+    RTC_LOG(LS_WARNING) << "The temporal weight factor must be in (0, 1]: "
+                        << config_->temporal_weight_factor;
+    valid = false;
+  }
+  if (config_->bandwidth_backoff_lower_bound_factor > 1.0) {
+    RTC_LOG(LS_WARNING)
+        << "The bandwidth backoff lower bound factor must not be greater than "
+           "1: "
+        << config_->bandwidth_backoff_lower_bound_factor;
+    valid = false;
+  }
+  if (config_->trendline_observations_window_size < 1) {
+    RTC_LOG(LS_WARNING) << "The trendline window size must be at least 1: "
+                        << config_->trendline_observations_window_size;
+    valid = false;
+  }
+  if (config_->max_increase_factor <= 0.0) {
+    RTC_LOG(LS_WARNING) << "The maximum increase factor must be positive: "
+                        << config_->max_increase_factor;
+    valid = false;
+  }
+  if (config_->delayed_increase_window <= TimeDelta::Zero()) {
+    RTC_LOG(LS_WARNING) << "The delayed increase window must be positive: "
+                        << config_->delayed_increase_window.ms();
+    valid = false;
+  }
+  if (config_->high_loss_rate_threshold <= 0.0 ||
+      config_->high_loss_rate_threshold > 1.0) {
+    RTC_LOG(LS_WARNING) << "The high loss rate threshold must be in (0, 1]: "
+                        << config_->high_loss_rate_threshold;
+    valid = false;
+  }
+  return valid;
+}
+
+double LossBasedBweV2::GetAverageReportedLossRatio() const {
+  if (num_observations_ <= 0) {
+    return 0.0;
+  }
+
+  double num_packets = 0;
+  double num_lost_packets = 0;
+  for (const Observation& observation : observations_) {
+    if (!observation.IsInitialized()) {
+      continue;
+    }
+
+    double instant_temporal_weight =
+        instant_upper_bound_temporal_weights_[(num_observations_ - 1) -
+                                              observation.id];
+    num_packets += instant_temporal_weight * observation.num_packets;
+    num_lost_packets += instant_temporal_weight * observation.num_lost_packets;
+  }
+
+  return num_lost_packets / num_packets;
+}
+
+DataRate LossBasedBweV2::GetCandidateBandwidthUpperBound() const {
+  DataRate candidate_bandwidth_upper_bound = max_bitrate_;
+  if (IsBandwidthLimitedDueToLoss() &&
+      IsValid(bandwidth_limit_in_current_window_)) {
+    candidate_bandwidth_upper_bound = bandwidth_limit_in_current_window_;
+  }
+
+  if (config_->trendline_integration_enabled) {
+    candidate_bandwidth_upper_bound =
+        std::min(GetInstantUpperBound(), candidate_bandwidth_upper_bound);
+    if (IsValid(delay_based_estimate_)) {
+      candidate_bandwidth_upper_bound =
+          std::min(delay_based_estimate_, candidate_bandwidth_upper_bound);
+    }
+  }
+
+  if (!acknowledged_bitrate_.has_value())
+    return candidate_bandwidth_upper_bound;
+
+  if (config_->rampup_acceleration_max_factor > 0.0) {
+    const TimeDelta time_since_bandwidth_reduced = std::min(
+        config_->rampup_acceleration_maxout_time,
+        std::max(TimeDelta::Zero(), last_send_time_most_recent_observation_ -
+                                        last_time_estimate_reduced_));
+    const double rampup_acceleration = config_->rampup_acceleration_max_factor *
+                                       time_since_bandwidth_reduced /
+                                       config_->rampup_acceleration_maxout_time;
+
+    candidate_bandwidth_upper_bound +=
+        rampup_acceleration * (*acknowledged_bitrate_);
+  }
+  return candidate_bandwidth_upper_bound;
+}
+
+std::vector<LossBasedBweV2::ChannelParameters> LossBasedBweV2::GetCandidates()
+    const {
+  std::vector<DataRate> bandwidths;
+  bool can_increase_bitrate = TrendlineEsimateAllowBitrateIncrease();
+  for (double candidate_factor : config_->candidate_factors) {
+    if (!can_increase_bitrate && candidate_factor > 1.0) {
+      continue;
+    }
+    bandwidths.push_back(candidate_factor *
+                         current_estimate_.loss_limited_bandwidth);
+  }
+
+  if (acknowledged_bitrate_.has_value() &&
+      config_->append_acknowledged_rate_candidate &&
+      TrendlineEsimateAllowEmergencyBackoff()) {
+    bandwidths.push_back(*acknowledged_bitrate_ *
+                         config_->bandwidth_backoff_lower_bound_factor);
+  }
+
+  if (IsValid(delay_based_estimate_) &&
+      config_->append_delay_based_estimate_candidate) {
+    if (can_increase_bitrate &&
+        delay_based_estimate_ > current_estimate_.loss_limited_bandwidth) {
+      bandwidths.push_back(delay_based_estimate_);
+    }
+  }
+
+  const DataRate candidate_bandwidth_upper_bound =
+      GetCandidateBandwidthUpperBound();
+
+  std::vector<ChannelParameters> candidates;
+  candidates.resize(bandwidths.size());
+  for (size_t i = 0; i < bandwidths.size(); ++i) {
+    ChannelParameters candidate = current_estimate_;
+    if (config_->trendline_integration_enabled) {
+      candidate.loss_limited_bandwidth =
+          std::min(bandwidths[i], candidate_bandwidth_upper_bound);
+    } else {
+      candidate.loss_limited_bandwidth = std::min(
+          bandwidths[i], std::max(current_estimate_.loss_limited_bandwidth,
+                                  candidate_bandwidth_upper_bound));
+    }
+    candidate.inherent_loss = GetFeasibleInherentLoss(candidate);
+    candidates[i] = candidate;
+  }
+  return candidates;
+}
+
+LossBasedBweV2::Derivatives LossBasedBweV2::GetDerivatives(
+    const ChannelParameters& channel_parameters) const {
+  Derivatives derivatives;
+
+  for (const Observation& observation : observations_) {
+    if (!observation.IsInitialized()) {
+      continue;
+    }
+
+    double loss_probability = GetLossProbability(
+        channel_parameters.inherent_loss,
+        channel_parameters.loss_limited_bandwidth, observation.sending_rate);
+
+    double temporal_weight =
+        temporal_weights_[(num_observations_ - 1) - observation.id];
+
+    derivatives.first +=
+        temporal_weight *
+        ((observation.num_lost_packets / loss_probability) -
+         (observation.num_received_packets / (1.0 - loss_probability)));
+    derivatives.second -=
+        temporal_weight *
+        ((observation.num_lost_packets / std::pow(loss_probability, 2)) +
+         (observation.num_received_packets /
+          std::pow(1.0 - loss_probability, 2)));
+  }
+
+  if (derivatives.second >= 0.0) {
+    RTC_LOG(LS_ERROR) << "The second derivative is mathematically guaranteed "
+                         "to be negative but is "
+                      << derivatives.second << ".";
+    derivatives.second = -1.0e-6;
+  }
+
+  return derivatives;
+}
+
+double LossBasedBweV2::GetFeasibleInherentLoss(
+    const ChannelParameters& channel_parameters) const {
+  return std::min(
+      std::max(channel_parameters.inherent_loss,
+               config_->inherent_loss_lower_bound),
+      GetInherentLossUpperBound(channel_parameters.loss_limited_bandwidth));
+}
+
+double LossBasedBweV2::GetInherentLossUpperBound(DataRate bandwidth) const {
+  if (bandwidth.IsZero()) {
+    return 1.0;
+  }
+
+  double inherent_loss_upper_bound =
+      config_->inherent_loss_upper_bound_offset +
+      config_->inherent_loss_upper_bound_bandwidth_balance / bandwidth;
+  return std::min(inherent_loss_upper_bound, 1.0);
+}
+
+double LossBasedBweV2::AdjustBiasFactor(double loss_rate,
+                                        double bias_factor) const {
+  return bias_factor *
+         (config_->loss_threshold_of_high_bandwidth_preference - loss_rate) /
+         (config_->bandwidth_preference_smoothing_factor +
+          std::abs(config_->loss_threshold_of_high_bandwidth_preference -
+                   loss_rate));
+}
+
+double LossBasedBweV2::GetHighBandwidthBias(DataRate bandwidth) const {
+  if (IsValid(bandwidth)) {
+    const double average_reported_loss_ratio = GetAverageReportedLossRatio();
+    return AdjustBiasFactor(average_reported_loss_ratio,
+                            config_->higher_bandwidth_bias_factor) *
+               bandwidth.kbps() +
+           AdjustBiasFactor(average_reported_loss_ratio,
+                            config_->higher_log_bandwidth_bias_factor) *
+               std::log(1.0 + bandwidth.kbps());
+  }
+  return 0.0;
+}
+
+double LossBasedBweV2::GetObjective(
+    const ChannelParameters& channel_parameters) const {
+  double objective = 0.0;
+
+  const double high_bandwidth_bias =
+      GetHighBandwidthBias(channel_parameters.loss_limited_bandwidth);
+
+  for (const Observation& observation : observations_) {
+    if (!observation.IsInitialized()) {
+      continue;
+    }
+
+    double loss_probability = GetLossProbability(
+        channel_parameters.inherent_loss,
+        channel_parameters.loss_limited_bandwidth, observation.sending_rate);
+
+    double temporal_weight =
+        temporal_weights_[(num_observations_ - 1) - observation.id];
+
+    objective +=
+        temporal_weight *
+        ((observation.num_lost_packets * std::log(loss_probability)) +
+         (observation.num_received_packets * std::log(1.0 - loss_probability)));
+    objective +=
+        temporal_weight * high_bandwidth_bias * observation.num_packets;
+  }
+
+  return objective;
+}
+
+DataRate LossBasedBweV2::GetSendingRate(
+    DataRate instantaneous_sending_rate) const {
+  if (num_observations_ <= 0) {
+    return instantaneous_sending_rate;
+  }
+
+  const int most_recent_observation_idx =
+      (num_observations_ - 1) % config_->observation_window_size;
+  const Observation& most_recent_observation =
+      observations_[most_recent_observation_idx];
+  DataRate sending_rate_previous_observation =
+      most_recent_observation.sending_rate;
+
+  return config_->sending_rate_smoothing_factor *
+             sending_rate_previous_observation +
+         (1.0 - config_->sending_rate_smoothing_factor) *
+             instantaneous_sending_rate;
+}
+
+DataRate LossBasedBweV2::GetInstantUpperBound() const {
+  return cached_instant_upper_bound_.value_or(max_bitrate_);
+}
+
+void LossBasedBweV2::CalculateInstantUpperBound() {
+  DataRate instant_limit = max_bitrate_;
+  const double average_reported_loss_ratio = GetAverageReportedLossRatio();
+  if (average_reported_loss_ratio > config_->instant_upper_bound_loss_offset) {
+    instant_limit = config_->instant_upper_bound_bandwidth_balance /
+                    (average_reported_loss_ratio -
+                     config_->instant_upper_bound_loss_offset);
+    if (average_reported_loss_ratio > config_->high_loss_rate_threshold) {
+      instant_limit = std::min(
+          instant_limit, DataRate::KilobitsPerSec(std::max(
+                             static_cast<double>(min_bitrate_.kbps()),
+                             config_->bandwidth_cap_at_high_loss_rate.kbps() -
+                                 config_->slope_of_bwe_high_loss_func *
+                                     average_reported_loss_ratio)));
+    }
+  }
+
+  if (IsBandwidthLimitedDueToLoss()) {
+    if (IsValid(upper_link_capacity_) &&
+        config_->bound_by_upper_link_capacity_when_loss_limited) {
+      instant_limit = std::min(instant_limit, upper_link_capacity_);
+    }
+  }
+  cached_instant_upper_bound_ = instant_limit;
+}
+
+void LossBasedBweV2::CalculateTemporalWeights() {
+  for (int i = 0; i < config_->observation_window_size; ++i) {
+    temporal_weights_[i] = std::pow(config_->temporal_weight_factor, i);
+    instant_upper_bound_temporal_weights_[i] =
+        std::pow(config_->instant_upper_bound_temporal_weight_factor, i);
+  }
+}
+
+void LossBasedBweV2::NewtonsMethodUpdate(
+    ChannelParameters& channel_parameters) const {
+  if (num_observations_ <= 0) {
+    return;
+  }
+
+  for (int i = 0; i < config_->newton_iterations; ++i) {
+    const Derivatives derivatives = GetDerivatives(channel_parameters);
+    channel_parameters.inherent_loss -=
+        config_->newton_step_size * derivatives.first / derivatives.second;
+    channel_parameters.inherent_loss =
+        GetFeasibleInherentLoss(channel_parameters);
+  }
+}
+
+bool LossBasedBweV2::TrendlineEsimateAllowBitrateIncrease() const {
+  if (!config_->trendline_integration_enabled) {
+    return true;
+  }
+
+  for (const auto& detector_state : delay_detector_states_) {
+    if (detector_state == BandwidthUsage::kBwOverusing ||
+        detector_state == BandwidthUsage::kBwUnderusing) {
+      return false;
+    }
+  }
+  return true;
+}
+
+bool LossBasedBweV2::TrendlineEsimateAllowEmergencyBackoff() const {
+  if (!config_->trendline_integration_enabled) {
+    return true;
+  }
+
+  if (!config_->use_acked_bitrate_only_when_overusing) {
+    return true;
+  }
+
+  for (const auto& detector_state : delay_detector_states_) {
+    if (detector_state == BandwidthUsage::kBwOverusing) {
+      return true;
+    }
+  }
+
+  return false;
+}
+
+bool LossBasedBweV2::PushBackObservation(
+    rtc::ArrayView<const PacketResult> packet_results,
+    BandwidthUsage delay_detector_state) {
+  delay_detector_states_.push_front(delay_detector_state);
+  if (static_cast<int>(delay_detector_states_.size()) >
+      config_->trendline_observations_window_size) {
+    delay_detector_states_.pop_back();
+  }
+
+  if (packet_results.empty()) {
+    return false;
+  }
+
+  PacketResultsSummary packet_results_summary =
+      GetPacketResultsSummary(packet_results);
+
+  partial_observation_.num_packets += packet_results_summary.num_packets;
+  partial_observation_.num_lost_packets +=
+      packet_results_summary.num_lost_packets;
+  partial_observation_.size += packet_results_summary.total_size;
+
+  // This is the first packet report we have received.
+  if (!IsValid(last_send_time_most_recent_observation_)) {
+    last_send_time_most_recent_observation_ =
+        packet_results_summary.first_send_time;
+  }
+
+  const Timestamp last_send_time = packet_results_summary.last_send_time;
+  const TimeDelta observation_duration =
+      last_send_time - last_send_time_most_recent_observation_;
+  // Too small to be meaningful.
+  if (observation_duration <= TimeDelta::Zero() ||
+      (observation_duration < config_->observation_duration_lower_bound &&
+       (delay_detector_state != BandwidthUsage::kBwOverusing ||
+        !config_->trendline_integration_enabled))) {
+    return false;
+  }
+
+  last_send_time_most_recent_observation_ = last_send_time;
+
+  Observation observation;
+  observation.num_packets = partial_observation_.num_packets;
+  observation.num_lost_packets = partial_observation_.num_lost_packets;
+  observation.num_received_packets =
+      observation.num_packets - observation.num_lost_packets;
+  observation.sending_rate =
+      GetSendingRate(partial_observation_.size / observation_duration);
+  observation.id = num_observations_++;
+  observations_[observation.id % config_->observation_window_size] =
+      observation;
+
+  partial_observation_ = PartialObservation();
+
+  CalculateInstantUpperBound();
+  return true;
+}
+
+bool LossBasedBweV2::IsBandwidthLimitedDueToLoss() const {
+  return current_state_ != LossBasedState::kDelayBasedEstimate;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
new file mode 100644
index 0000000000..9ff9cb74c6
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.h
@@ -0,0 +1,203 @@
+/*
+ *  Copyright 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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
+
+#include <cstddef>
+#include <deque>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+
+// State of the loss based estimate, which can be either increasing/decreasing
+// when network is loss limited, or equal to the delay based estimate.
+enum class LossBasedState {
+  kIncreasing = 0,
+  kDecreasing = 1,
+  kDelayBasedEstimate = 2
+};
+
+class LossBasedBweV2 {
+ public:
+  struct Result {
+    ~Result() = default;
+    DataRate bandwidth_estimate = DataRate::Zero();
+    LossBasedState state = LossBasedState::kDelayBasedEstimate;
+  };
+  // Creates a disabled `LossBasedBweV2` if the
+  // `key_value_config` is not valid.
+  explicit LossBasedBweV2(const FieldTrialsView* key_value_config);
+
+  LossBasedBweV2(const LossBasedBweV2&) = delete;
+  LossBasedBweV2& operator=(const LossBasedBweV2&) = delete;
+
+  ~LossBasedBweV2() = default;
+
+  bool IsEnabled() const;
+  // Returns true iff a BWE can be calculated, i.e., the estimator has been
+  // initialized with a BWE and then has received enough `PacketResult`s.
+  bool IsReady() const;
+
+  // Returns `DataRate::PlusInfinity` if no BWE can be calculated.
+  Result GetLossBasedResult() const;
+
+  void SetAcknowledgedBitrate(DataRate acknowledged_bitrate);
+  void SetBandwidthEstimate(DataRate bandwidth_estimate);
+  void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
+  void UpdateBandwidthEstimate(
+      rtc::ArrayView<const PacketResult> packet_results,
+      DataRate delay_based_estimate,
+      BandwidthUsage delay_detector_state,
+      absl::optional<DataRate> probe_bitrate,
+      DataRate upper_link_capacity);
+
+ private:
+  struct ChannelParameters {
+    double inherent_loss = 0.0;
+    DataRate loss_limited_bandwidth = DataRate::MinusInfinity();
+  };
+
+  struct Config {
+    double bandwidth_rampup_upper_bound_factor = 0.0;
+    double rampup_acceleration_max_factor = 0.0;
+    TimeDelta rampup_acceleration_maxout_time = TimeDelta::Zero();
+    std::vector<double> candidate_factors;
+    double higher_bandwidth_bias_factor = 0.0;
+    double higher_log_bandwidth_bias_factor = 0.0;
+    double inherent_loss_lower_bound = 0.0;
+    double loss_threshold_of_high_bandwidth_preference = 0.0;
+    double bandwidth_preference_smoothing_factor = 0.0;
+    DataRate inherent_loss_upper_bound_bandwidth_balance =
+        DataRate::MinusInfinity();
+    double inherent_loss_upper_bound_offset = 0.0;
+    double initial_inherent_loss_estimate = 0.0;
+    int newton_iterations = 0;
+    double newton_step_size = 0.0;
+    bool append_acknowledged_rate_candidate = true;
+    bool append_delay_based_estimate_candidate = false;
+    TimeDelta observation_duration_lower_bound = TimeDelta::Zero();
+    int observation_window_size = 0;
+    double sending_rate_smoothing_factor = 0.0;
+    double instant_upper_bound_temporal_weight_factor = 0.0;
+    DataRate instant_upper_bound_bandwidth_balance = DataRate::MinusInfinity();
+    double instant_upper_bound_loss_offset = 0.0;
+    double temporal_weight_factor = 0.0;
+    double bandwidth_backoff_lower_bound_factor = 0.0;
+    bool trendline_integration_enabled = false;
+    int trendline_observations_window_size = 0;
+    double max_increase_factor = 0.0;
+    TimeDelta delayed_increase_window = TimeDelta::Zero();
+    bool use_acked_bitrate_only_when_overusing = false;
+    bool not_increase_if_inherent_loss_less_than_average_loss = false;
+    double high_loss_rate_threshold = 1.0;
+    DataRate bandwidth_cap_at_high_loss_rate = DataRate::MinusInfinity();
+    double slope_of_bwe_high_loss_func = 1000.0;
+    bool probe_integration_enabled = false;
+    bool bound_by_upper_link_capacity_when_loss_limited = false;
+  };
+
+  struct Derivatives {
+    double first = 0.0;
+    double second = 0.0;
+  };
+
+  struct Observation {
+    bool IsInitialized() const { return id != -1; }
+
+    int num_packets = 0;
+    int num_lost_packets = 0;
+    int num_received_packets = 0;
+    DataRate sending_rate = DataRate::MinusInfinity();
+    int id = -1;
+  };
+
+  struct PartialObservation {
+    int num_packets = 0;
+    int num_lost_packets = 0;
+    DataSize size = DataSize::Zero();
+  };
+
+  static absl::optional<Config> CreateConfig(
+      const FieldTrialsView* key_value_config);
+  bool IsConfigValid() const;
+
+  // Returns `0.0` if not enough loss statistics have been received.
+  double GetAverageReportedLossRatio() const;
+  std::vector<ChannelParameters> GetCandidates() const;
+  DataRate GetCandidateBandwidthUpperBound() const;
+  Derivatives GetDerivatives(const ChannelParameters& channel_parameters) const;
+  double GetFeasibleInherentLoss(
+      const ChannelParameters& channel_parameters) const;
+  double GetInherentLossUpperBound(DataRate bandwidth) const;
+  double AdjustBiasFactor(double loss_rate, double bias_factor) const;
+  double GetHighBandwidthBias(DataRate bandwidth) const;
+  double GetObjective(const ChannelParameters& channel_parameters) const;
+  DataRate GetSendingRate(DataRate instantaneous_sending_rate) const;
+  DataRate GetInstantUpperBound() const;
+  void CalculateInstantUpperBound();
+
+  void CalculateTemporalWeights();
+  void NewtonsMethodUpdate(ChannelParameters& channel_parameters) const;
+
+  // Returns false if there exists a kBwOverusing or kBwUnderusing in the
+  // window.
+  bool TrendlineEsimateAllowBitrateIncrease() const;
+
+  // Returns true if there exists an overusing state in the window.
+  bool TrendlineEsimateAllowEmergencyBackoff() const;
+
+  // Returns false if no observation was created.
+  bool PushBackObservation(rtc::ArrayView<const PacketResult> packet_results,
+                           BandwidthUsage delay_detector_state);
+  void UpdateTrendlineEstimator(
+      const std::vector<PacketResult>& packet_feedbacks,
+      Timestamp at_time);
+  void UpdateDelayDetector(BandwidthUsage delay_detector_state);
+  bool IsEstimateIncreasingWhenLossLimited(
+      const ChannelParameters& best_candidate);
+  bool IsBandwidthLimitedDueToLoss() const;
+  void SetProbeBitrate(absl::optional<DataRate> probe_bitrate);
+
+  absl::optional<DataRate> acknowledged_bitrate_;
+  absl::optional<Config> config_;
+  ChannelParameters current_estimate_;
+  int num_observations_ = 0;
+  std::vector<Observation> observations_;
+  PartialObservation partial_observation_;
+  Timestamp last_send_time_most_recent_observation_ = Timestamp::PlusInfinity();
+  Timestamp last_time_estimate_reduced_ = Timestamp::MinusInfinity();
+  absl::optional<DataRate> cached_instant_upper_bound_;
+  std::vector<double> instant_upper_bound_temporal_weights_;
+  std::vector<double> temporal_weights_;
+  std::deque<BandwidthUsage> delay_detector_states_;
+  Timestamp recovering_after_loss_timestamp_ = Timestamp::MinusInfinity();
+  DataRate bandwidth_limit_in_current_window_ = DataRate::PlusInfinity();
+  DataRate min_bitrate_ = DataRate::KilobitsPerSec(1);
+  DataRate max_bitrate_ = DataRate::PlusInfinity();
+  LossBasedState current_state_ = LossBasedState::kDelayBasedEstimate;
+  DataRate probe_bitrate_ = DataRate::PlusInfinity();
+  DataRate delay_based_estimate_ = DataRate::PlusInfinity();
+  DataRate upper_link_capacity_ = DataRate::PlusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_LOSS_BASED_BWE_V2_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build
new file mode 100644
index 0000000000..5728e0c4b2
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("loss_based_bwe_v2_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc
new file mode 100644
index 0000000000..c303c29d68
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/loss_based_bwe_v2_test.cc
@@ -0,0 +1,1526 @@
+/*
+ *  Copyright 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 "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+
+#include <string>
+#include <vector>
+
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+using ::webrtc::test::ExplicitKeyValueConfig;
+
+constexpr TimeDelta kObservationDurationLowerBound = TimeDelta::Millis(200);
+constexpr TimeDelta kDelayedIncreaseWindow = TimeDelta::Millis(300);
+constexpr double kMaxIncreaseFactor = 1.5;
+
+class LossBasedBweV2Test : public ::testing::TestWithParam<bool> {
+ protected:
+  std::string Config(bool enabled,
+                     bool valid,
+                     bool trendline_integration_enabled) {
+    char buffer[1024];
+    rtc::SimpleStringBuilder config_string(buffer);
+
+    config_string << "WebRTC-Bwe-LossBasedBweV2/";
+
+    if (enabled) {
+      config_string << "Enabled:true";
+    } else {
+      config_string << "Enabled:false";
+    }
+
+    if (valid) {
+      config_string << ",BwRampupUpperBoundFactor:1.2";
+    } else {
+      config_string << ",BwRampupUpperBoundFactor:0.0";
+    }
+
+    if (trendline_integration_enabled) {
+      config_string << ",TrendlineIntegrationEnabled:true";
+    } else {
+      config_string << ",TrendlineIntegrationEnabled:false";
+    }
+
+    config_string
+        << ",CandidateFactors:1.1|1.0|0.95,HigherBwBiasFactor:0.01,"
+           "DelayBasedCandidate:true,"
+           "InherentLossLowerBound:0.001,InherentLossUpperBoundBwBalance:"
+           "14kbps,"
+           "InherentLossUpperBoundOffset:0.9,InitialInherentLossEstimate:0.01,"
+           "NewtonIterations:2,NewtonStepSize:0.4,ObservationWindowSize:15,"
+           "SendingRateSmoothingFactor:0.01,"
+           "InstantUpperBoundTemporalWeightFactor:0.97,"
+           "InstantUpperBoundBwBalance:90kbps,"
+           "InstantUpperBoundLossOffset:0.1,TemporalWeightFactor:0.98";
+
+    config_string.AppendFormat(
+        ",ObservationDurationLowerBound:%dms",
+        static_cast<int>(kObservationDurationLowerBound.ms()));
+    config_string.AppendFormat(",MaxIncreaseFactor:%f", kMaxIncreaseFactor);
+    config_string.AppendFormat(",DelayedIncreaseWindow:%dms",
+                               static_cast<int>(kDelayedIncreaseWindow.ms()));
+
+    config_string << "/";
+
+    return config_string.str();
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWithReceivedPackets(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(2);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[0].sent_packet.send_time = first_packet_timestamp;
+    enough_feedback[1].sent_packet.send_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[0].receive_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[1].receive_time =
+        first_packet_timestamp + 2 * kObservationDurationLowerBound;
+    return enough_feedback;
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWith10pLossRate(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(10);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    for (unsigned i = 0; i < enough_feedback.size(); ++i) {
+      enough_feedback[i].sent_packet.size = DataSize::Bytes(15'000);
+      enough_feedback[i].sent_packet.send_time =
+          first_packet_timestamp +
+          static_cast<int>(i) * kObservationDurationLowerBound;
+      enough_feedback[i].receive_time =
+          first_packet_timestamp +
+          static_cast<int>(i + 1) * kObservationDurationLowerBound;
+    }
+    enough_feedback[9].receive_time = Timestamp::PlusInfinity();
+    return enough_feedback;
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWith50pLossRate(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(2);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[0].sent_packet.send_time = first_packet_timestamp;
+    enough_feedback[1].sent_packet.send_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[0].receive_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[1].receive_time = Timestamp::PlusInfinity();
+    return enough_feedback;
+  }
+
+  std::vector<PacketResult> CreatePacketResultsWith100pLossRate(
+      Timestamp first_packet_timestamp) {
+    std::vector<PacketResult> enough_feedback(2);
+    enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+    enough_feedback[0].sent_packet.send_time = first_packet_timestamp;
+    enough_feedback[1].sent_packet.send_time =
+        first_packet_timestamp + kObservationDurationLowerBound;
+    enough_feedback[0].receive_time = Timestamp::PlusInfinity();
+    enough_feedback[1].receive_time = Timestamp::PlusInfinity();
+    return enough_feedback;
+  }
+};
+
+TEST_P(LossBasedBweV2Test, EnabledWhenGivenValidConfigurationValues) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  EXPECT_TRUE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, DisabledWhenGivenDisabledConfiguration) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/false, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, DisabledWhenGivenNonValidConfigurationValues) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/false,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, DisabledWhenGivenNonPositiveCandidateFactor) {
+  ExplicitKeyValueConfig key_value_config_negative_candidate_factor(
+      "WebRTC-Bwe-LossBasedBweV2/Enabled:true,CandidateFactors:-1.3|1.1/");
+  LossBasedBweV2 loss_based_bandwidth_estimator_1(
+      &key_value_config_negative_candidate_factor);
+  EXPECT_FALSE(loss_based_bandwidth_estimator_1.IsEnabled());
+
+  ExplicitKeyValueConfig key_value_config_zero_candidate_factor(
+      "WebRTC-Bwe-LossBasedBweV2/Enabled:true,CandidateFactors:0.0|1.1/");
+  LossBasedBweV2 loss_based_bandwidth_estimator_2(
+      &key_value_config_zero_candidate_factor);
+  EXPECT_FALSE(loss_based_bandwidth_estimator_2.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test,
+       DisabledWhenGivenConfigurationThatDoesNotAllowGeneratingCandidates) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "DelayBasedCandidate:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsEnabled());
+}
+
+TEST_P(LossBasedBweV2Test, ReturnsDelayBasedEstimateWhenDisabled) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/false, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      /*packet_results=*/{},
+      /*delay_based_estimate=*/DataRate::KilobitsPerSec(100),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(100));
+}
+
+TEST_P(LossBasedBweV2Test,
+       ReturnsDelayBasedEstimateWhenWhenGivenNonValidConfigurationValues) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/false,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      /*packet_results=*/{},
+      /*delay_based_estimate=*/DataRate::KilobitsPerSec(100),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(100));
+}
+
+TEST_P(LossBasedBweV2Test,
+       BandwidthEstimateGivenInitializationAndThenFeedback) {
+  std::vector<PacketResult> enough_feedback =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_TRUE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsFinite());
+}
+
+TEST_P(LossBasedBweV2Test, NoBandwidthEstimateGivenNoInitialization) {
+  std::vector<PacketResult> enough_feedback =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsPlusInfinity());
+}
+
+TEST_P(LossBasedBweV2Test, NoBandwidthEstimateGivenNotEnoughFeedback) {
+  // Create packet results where the observation duration is less than the lower
+  // bound.
+  PacketResult not_enough_feedback[2];
+  not_enough_feedback[0].sent_packet.size = DataSize::Bytes(15'000);
+  not_enough_feedback[1].sent_packet.size = DataSize::Bytes(15'000);
+  not_enough_feedback[0].sent_packet.send_time = Timestamp::Zero();
+  not_enough_feedback[1].sent_packet.send_time =
+      Timestamp::Zero() + kObservationDurationLowerBound / 2;
+  not_enough_feedback[0].receive_time =
+      Timestamp::Zero() + kObservationDurationLowerBound / 2;
+  not_enough_feedback[1].receive_time =
+      Timestamp::Zero() + kObservationDurationLowerBound;
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsPlusInfinity());
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      not_enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_FALSE(loss_based_bandwidth_estimator.IsReady());
+  EXPECT_TRUE(loss_based_bandwidth_estimator.GetLossBasedResult()
+                  .bandwidth_estimate.IsPlusInfinity());
+}
+
+TEST_P(LossBasedBweV2Test,
+       SetValueIsTheEstimateUntilAdditionalFeedbackHasBeenReceived) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_NE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_NE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test,
+       SetAcknowledgedBitrateOnlyAffectsTheBweWhenAdditionalFeedbackIsGiven) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator_1(&key_value_config);
+  LossBasedBweV2 loss_based_bandwidth_estimator_2(&key_value_config);
+
+  loss_based_bandwidth_estimator_1.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator_2.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator_1.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  loss_based_bandwidth_estimator_2.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator_1.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(660));
+
+  loss_based_bandwidth_estimator_1.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(900));
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator_1.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(660));
+
+  loss_based_bandwidth_estimator_1.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  loss_based_bandwidth_estimator_2.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_NE(
+      loss_based_bandwidth_estimator_1.GetLossBasedResult().bandwidth_estimate,
+      loss_based_bandwidth_estimator_2.GetLossBasedResult().bandwidth_estimate);
+}
+
+TEST_P(LossBasedBweV2Test,
+       BandwidthEstimateIsCappedToBeTcpFairGivenTooHighLossRate) {
+  std::vector<PacketResult> enough_feedback_no_received_packets =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_no_received_packets,
+      /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(100));
+}
+
+TEST_P(LossBasedBweV2Test, BandwidthEstimateNotIncreaseWhenNetworkUnderusing) {
+  if (!GetParam()) {
+    GTEST_SKIP() << "This test should run only if "
+                    "trendline_integration_enabled is enabled";
+  }
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwUnderusing, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+// When network is normal, estimate can increase but never be higher than
+// the delay based estimate.
+TEST_P(LossBasedBweV2Test,
+       BandwidthEstimateCappedByDelayBasedEstimateWhenNetworkNormal) {
+  // Create two packet results, network is in normal state, 100% packets are
+  // received, and no delay increase.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // If the delay based estimate is infinity, then loss based estimate increases
+  // and not bounded by delay based estimate.
+  EXPECT_GT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::KilobitsPerSec(500),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // If the delay based estimate is not infinity, then loss based estimate is
+  // bounded by delay based estimate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(500));
+}
+
+// When loss based bwe receives a strong signal of overusing and an increase in
+// loss rate, it should acked bitrate for emegency backoff.
+TEST_P(LossBasedBweV2Test, UseAckedBitrateForEmegencyBackOff) {
+  // Create two packet results, first packet has 50% loss rate, second packet
+  // has 100% loss rate.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  DataRate acked_bitrate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_bitrate);
+  // Update estimate when network is overusing, and 50% loss rate.
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwOverusing,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // Update estimate again when network is continuously overusing, and 100%
+  // loss rate.
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwOverusing,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // The estimate bitrate now is backed off based on acked bitrate.
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      acked_bitrate);
+}
+
+// When receiving the same packet feedback, loss based bwe ignores the feedback
+// and returns the current estimate.
+TEST_P(LossBasedBweV2Test, NoBweChangeIfObservationDurationUnchanged) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  // Use the same feedback and check if the estimate is unchanged.
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_EQ(estimate_2, estimate_1);
+}
+
+// When receiving feedback of packets that were sent within an observation
+// duration, and network is in the normal state, loss based bwe returns the
+// current estimate.
+TEST_P(LossBasedBweV2Test,
+       NoBweChangeIfObservationDurationIsSmallAndNetworkNormal) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_EQ(estimate_2, estimate_1);
+}
+
+// When receiving feedback of packets that were sent within an observation
+// duration, and network is in the underusing state, loss based bwe returns the
+// current estimate.
+TEST_P(LossBasedBweV2Test,
+       NoBweIncreaseIfObservationDurationIsSmallAndNetworkUnderusing) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwUnderusing, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_LE(estimate_2, estimate_1);
+}
+
+// When receiving feedback of packets that were sent within an observation
+// duration, network is overusing, and trendline integration is enabled, loss
+// based bwe updates its estimate.
+TEST_P(LossBasedBweV2Test,
+       UpdateEstimateIfObservationDurationIsSmallAndNetworkOverusing) {
+  if (!GetParam()) {
+    GTEST_SKIP() << "This test should run only if "
+                    "trendline_integration_enabled is enabled";
+  }
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_1 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwOverusing,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_LT(estimate_2, estimate_1);
+}
+
+TEST_P(LossBasedBweV2Test,
+       IncreaseToDelayBasedEstimateIfNoLossOrDelayIncrease) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          2 * kObservationDurationLowerBound);
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      delay_based_estimate);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      delay_based_estimate);
+}
+
+TEST_P(LossBasedBweV2Test,
+       IncreaseByMaxIncreaseFactorAfterLossBasedBweBacksOff) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1.5,BwRampupUpperBoundFactor:"
+      "2.0,NotIncreaseIfInherentLossLessThanAverageLoss:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  LossBasedBweV2::Result result_at_loss =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+
+  // Network recovers after loss.
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_EQ(result_after_recovery.bandwidth_estimate,
+            result_at_loss.bandwidth_estimate * 1.5);
+}
+
+TEST_P(LossBasedBweV2Test,
+       LossBasedStateIsDelayBasedEstimateAfterNetworkRecovering) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:100|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:100,"
+      "BwRampupUpperBoundFactor:"
+      "2.0,NotIncreaseIfInherentLossLessThanAverageLoss:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(600);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDecreasing);
+
+  // Network recovers after loss.
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDelayBasedEstimate);
+
+  // Network recovers continuing.
+  std::vector<PacketResult> enough_feedback_3 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound * 2);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_3, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDelayBasedEstimate);
+}
+
+TEST_P(LossBasedBweV2Test,
+       LossBasedStateIsNotDelayBasedEstimateIfDelayBasedEsimtateInfinite) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:100|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:100,"
+      "BwRampupUpperBoundFactor:"
+      "2.0/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::PlusInfinity();
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  ASSERT_EQ(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDecreasing);
+
+  // Network recovers after loss.
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  EXPECT_NE(loss_based_bandwidth_estimator.GetLossBasedResult().state,
+            LossBasedState::kDelayBasedEstimate);
+}
+
+// After loss based bwe backs off, the next estimate is capped by
+// a factor of acked bitrate.
+TEST_P(LossBasedBweV2Test,
+       IncreaseByFactorOfAckedBitrateAfterLossBasedBweBacksOff) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,LossThresholdOfHighBandwidthPreference:0.99,"
+      "BwRampupUpperBoundFactor:1.2,"
+      "InherentLossUpperBoundOffset:0.9,ObservationDurationLowerBound:200ms/");
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Change the acked bitrate to make sure that the estimate is bounded by a
+  // factor of acked bitrate.
+  DataRate acked_bitrate = DataRate::KilobitsPerSec(50);
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_bitrate);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate is capped by acked_bitrate * BwRampupUpperBoundFactor.
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+  EXPECT_EQ(estimate_2, acked_bitrate * 1.2);
+}
+
+// After loss based bwe backs off, the estimate is bounded during the delayed
+// window.
+TEST_P(LossBasedBweV2Test,
+       EstimateBitrateIsBoundedDuringDelayedWindowAfterLossBasedBweBacksOff) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow - TimeDelta::Millis(2));
+  std::vector<PacketResult> enough_feedback_3 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow - TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // Increase the acknowledged bitrate to make sure that the estimate is not
+  // capped too low.
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(5000));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate is capped by current_estimate * kMaxIncreaseFactor because
+  // it recently backed off.
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_3, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // The latest estimate is the same as the previous estimate since the sent
+  // packets were sent within the DelayedIncreaseWindow.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      estimate_2);
+}
+
+// The estimate is not bounded after the delayed increase window.
+TEST_P(LossBasedBweV2Test, KeepIncreasingEstimateAfterDelayedIncreaseWindow) {
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow - TimeDelta::Millis(1));
+  std::vector<PacketResult> enough_feedback_3 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kDelayedIncreaseWindow + TimeDelta::Millis(1));
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(300));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // Increase the acknowledged bitrate to make sure that the estimate is not
+  // capped too low.
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(
+      DataRate::KilobitsPerSec(5000));
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate is capped by current_estimate * kMaxIncreaseFactor because it
+  // recently backed off.
+  DataRate estimate_2 =
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate;
+
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_3, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+  // The estimate can continue increasing after the DelayedIncreaseWindow.
+  EXPECT_GE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      estimate_2);
+}
+
+TEST_P(LossBasedBweV2Test, NotIncreaseIfInherentLossLessThanAverageLoss) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,"
+      "NotIncreaseIfInherentLossLessThanAverageLoss:true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Do not increase the bitrate because inherent loss is less than average loss
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test,
+       SelectHighBandwidthCandidateIfLossRateIsLessThanThreshold) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|0.8,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "20,NotIncreaseIfInherentLossLessThanAverageLoss:false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Because LossThresholdOfHighBandwidthPreference is 20%, the average loss is
+  // 10%, bandwidth estimate should increase.
+  EXPECT_GT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test,
+       SelectLowBandwidthCandidateIfLossRateIsIsHigherThanThreshold) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|0.8,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Because LossThresholdOfHighBandwidthPreference is 5%, the average loss is
+  // 10%, bandwidth estimate should decrease.
+  EXPECT_LT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(600));
+}
+
+TEST_P(LossBasedBweV2Test, UseProbeResultWhenRecoveringFromLoss) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,ProbeIntegrationEnabled:true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Network recovers after loss.
+  DataRate probe_estimate = DataRate::KilobitsPerSec(300);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      probe_estimate, /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_EQ(result_after_recovery.bandwidth_estimate, probe_estimate);
+}
+
+// If BoundByUpperLinkCapacityWhenLossLimited is enabled, the estimate is
+// bounded by the upper link capacity when bandwidth is loss limited.
+TEST_P(LossBasedBweV2Test, BoundEstimateByUpperLinkCapacityWhenLossLimited) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,BoundByUpperLinkCapacityWhenLossLimited:"
+      "true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Network recovers after loss.
+  DataRate upper_link_capacity = DataRate::KilobitsPerSec(10);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt, upper_link_capacity);
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_EQ(result_after_recovery.bandwidth_estimate, upper_link_capacity);
+}
+
+// If BoundByUpperLinkCapacityWhenLossLimited is enabled, the estimate is not
+// bounded by the upper link capacity when bandwidth is not loss limited.
+TEST_P(LossBasedBweV2Test,
+       NotBoundEstimateByUpperLinkCapacityWhenNotLossLimited) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,BoundByUpperLinkCapacityWhenLossLimited:"
+      "true/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create a normal network without loss
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  DataRate upper_link_capacity = DataRate::KilobitsPerSec(10);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt, upper_link_capacity);
+
+  LossBasedBweV2::Result loss_based_result =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_GT(loss_based_result.bandwidth_estimate, upper_link_capacity);
+}
+
+// If BoundByUpperLinkCapacityWhenLossLimited is disabled, the estimate is not
+// bounded by the upper link capacity.
+TEST_P(LossBasedBweV2Test, NotBoundEstimateByUpperLinkCapacity) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.2|1|0.5,AckedRateCandidate:true,"
+      "ObservationWindowSize:2,ObservationDurationLowerBound:200ms,"
+      "InstantUpperBoundBwBalance:10000kbps,"
+      "DelayBasedCandidate:true,MaxIncreaseFactor:1000,"
+      "BwRampupUpperBoundFactor:2.0,BoundByUpperLinkCapacityWhenLossLimited:"
+      "false/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  DataRate acked_rate = DataRate::KilobitsPerSec(300);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+  loss_based_bandwidth_estimator.SetAcknowledgedBitrate(acked_rate);
+
+  // Create some loss to create the loss limited scenario.
+  std::vector<PacketResult> enough_feedback_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_1, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Network recovers after loss.
+  DataRate upper_link_capacity = DataRate::KilobitsPerSec(10);
+  std::vector<PacketResult> enough_feedback_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_2, delay_based_estimate, BandwidthUsage::kBwNormal,
+      /*probe_estimate=*/absl::nullopt, upper_link_capacity);
+
+  LossBasedBweV2::Result result_after_recovery =
+      loss_based_bandwidth_estimator.GetLossBasedResult();
+  EXPECT_GT(result_after_recovery.bandwidth_estimate, upper_link_capacity);
+}
+
+TEST_P(LossBasedBweV2Test,
+       StricterBoundUsingHighLossRateThresholdAt10pLossRate) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.09/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_10p_loss_1 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_10p_loss_2 =
+      CreatePacketResultsWith10pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_10p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // At 10% loss rate and high loss rate threshold to be 10%, cap the estimate
+  // to be 500 * 1000-0.1 = 400kbps.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(400));
+}
+
+TEST_P(LossBasedBweV2Test,
+       StricterBoundUsingHighLossRateThresholdAt50pLossRate) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.3/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_50p_loss_1 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_50p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_50p_loss_2 =
+      CreatePacketResultsWith50pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_50p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // At 50% loss rate and high loss rate threshold to be 30%, cap the estimate
+  // to be the min bitrate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+}
+
+TEST_P(LossBasedBweV2Test,
+       StricterBoundUsingHighLossRateThresholdAt100pLossRate) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.0,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.3/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_100p_loss_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_100p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_100p_loss_2 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_100p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // At 100% loss rate and high loss rate threshold to be 30%, cap the estimate
+  // to be the min bitrate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+}
+
+TEST_P(LossBasedBweV2Test, EstimateRecoversAfterHighLoss) {
+  ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-LossBasedBweV2/"
+      "Enabled:true,CandidateFactors:1.1|1.0|0.9,AckedRateCandidate:false,"
+      "ObservationWindowSize:2,"
+      "DelayBasedCandidate:true,InstantUpperBoundBwBalance:100kbps,"
+      "ObservationDurationLowerBound:200ms,HigherBwBiasFactor:1000,"
+      "HigherLogBwBiasFactor:1000,LossThresholdOfHighBandwidthPreference:0."
+      "05,HighLossRateThreshold:0.3/");
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000000));
+  DataRate delay_based_estimate = DataRate::KilobitsPerSec(5000);
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(600));
+
+  std::vector<PacketResult> enough_feedback_100p_loss_1 =
+      CreatePacketResultsWith100pLossRate(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_100p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // Make sure that the estimate is set to min bitrate because of 100% loss
+  // rate.
+  EXPECT_EQ(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+
+  // Create some feedbacks with 0 loss rate to simulate network recovering.
+  std::vector<PacketResult> enough_feedback_0p_loss_1 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_0p_loss_1, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  std::vector<PacketResult> enough_feedback_0p_loss_2 =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero() +
+          kObservationDurationLowerBound * 2);
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback_0p_loss_2, delay_based_estimate,
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  // The estimate increases as network recovers.
+  EXPECT_GT(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(10));
+}
+
+TEST_P(LossBasedBweV2Test, EstimateIsNotHigherThanMaxBitrate) {
+  ExplicitKeyValueConfig key_value_config(
+      Config(/*enabled=*/true, /*valid=*/true,
+             /*trendline_integration_enabled=*/GetParam()));
+  LossBasedBweV2 loss_based_bandwidth_estimator(&key_value_config);
+  loss_based_bandwidth_estimator.SetMinMaxBitrate(
+      /*min_bitrate=*/DataRate::KilobitsPerSec(10),
+      /*max_bitrate=*/DataRate::KilobitsPerSec(1000));
+  loss_based_bandwidth_estimator.SetBandwidthEstimate(
+      DataRate::KilobitsPerSec(1000));
+  std::vector<PacketResult> enough_feedback =
+      CreatePacketResultsWithReceivedPackets(
+          /*first_packet_timestamp=*/Timestamp::Zero());
+  loss_based_bandwidth_estimator.UpdateBandwidthEstimate(
+      enough_feedback, /*delay_based_estimate=*/DataRate::PlusInfinity(),
+      BandwidthUsage::kBwNormal, /*probe_estimate=*/absl::nullopt,
+      /*upper_link_capacity=*/DataRate::PlusInfinity());
+
+  EXPECT_LE(
+      loss_based_bandwidth_estimator.GetLossBasedResult().bandwidth_estimate,
+      DataRate::KilobitsPerSec(1000));
+}
+
+INSTANTIATE_TEST_SUITE_P(LossBasedBweV2Tests,
+                         LossBasedBweV2Test,
+                         ::testing::Bool());
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
new file mode 100644
index 0000000000..a94f653157
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
@@ -0,0 +1,201 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+namespace {
+// The minumum number of probes we need to receive feedback about in percent
+// in order to have a valid estimate.
+constexpr double kMinReceivedProbesRatio = .80;
+
+// The minumum number of bytes we need to receive feedback about in percent
+// in order to have a valid estimate.
+constexpr double kMinReceivedBytesRatio = .80;
+
+// The maximum |receive rate| / |send rate| ratio for a valid estimate.
+constexpr float kMaxValidRatio = 2.0f;
+
+// The minimum |receive rate| / |send rate| ratio assuming that the link is
+// not saturated, i.e. we assume that we will receive at least
+// kMinRatioForUnsaturatedLink * |send rate| if |send rate| is less than the
+// link capacity.
+constexpr float kMinRatioForUnsaturatedLink = 0.9f;
+
+// The target utilization of the link. If we know true link capacity
+// we'd like to send at 95% of that rate.
+constexpr float kTargetUtilizationFraction = 0.95f;
+
+// The maximum time period over which the cluster history is retained.
+// This is also the maximum time period beyond which a probing burst is not
+// expected to last.
+constexpr TimeDelta kMaxClusterHistory = TimeDelta::Seconds(1);
+
+// The maximum time interval between first and the last probe on a cluster
+// on the sender side as well as the receive side.
+constexpr TimeDelta kMaxProbeInterval = TimeDelta::Seconds(1);
+
+}  // namespace
+
+ProbeBitrateEstimator::ProbeBitrateEstimator(RtcEventLog* event_log)
+    : event_log_(event_log) {}
+
+ProbeBitrateEstimator::~ProbeBitrateEstimator() = default;
+
+absl::optional<DataRate> ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
+    const PacketResult& packet_feedback) {
+  int cluster_id = packet_feedback.sent_packet.pacing_info.probe_cluster_id;
+  RTC_DCHECK_NE(cluster_id, PacedPacketInfo::kNotAProbe);
+
+  EraseOldClusters(packet_feedback.receive_time);
+
+  AggregatedCluster* cluster = &clusters_[cluster_id];
+
+  if (packet_feedback.sent_packet.send_time < cluster->first_send) {
+    cluster->first_send = packet_feedback.sent_packet.send_time;
+  }
+  if (packet_feedback.sent_packet.send_time > cluster->last_send) {
+    cluster->last_send = packet_feedback.sent_packet.send_time;
+    cluster->size_last_send = packet_feedback.sent_packet.size;
+  }
+  if (packet_feedback.receive_time < cluster->first_receive) {
+    cluster->first_receive = packet_feedback.receive_time;
+    cluster->size_first_receive = packet_feedback.sent_packet.size;
+  }
+  if (packet_feedback.receive_time > cluster->last_receive) {
+    cluster->last_receive = packet_feedback.receive_time;
+  }
+  cluster->size_total += packet_feedback.sent_packet.size;
+  cluster->num_probes += 1;
+
+  RTC_DCHECK_GT(
+      packet_feedback.sent_packet.pacing_info.probe_cluster_min_probes, 0);
+  RTC_DCHECK_GT(packet_feedback.sent_packet.pacing_info.probe_cluster_min_bytes,
+                0);
+
+  int min_probes =
+      packet_feedback.sent_packet.pacing_info.probe_cluster_min_probes *
+      kMinReceivedProbesRatio;
+  DataSize min_size =
+      DataSize::Bytes(
+          packet_feedback.sent_packet.pacing_info.probe_cluster_min_bytes) *
+      kMinReceivedBytesRatio;
+  if (cluster->num_probes < min_probes || cluster->size_total < min_size)
+    return absl::nullopt;
+
+  TimeDelta send_interval = cluster->last_send - cluster->first_send;
+  TimeDelta receive_interval = cluster->last_receive - cluster->first_receive;
+
+  if (send_interval <= TimeDelta::Zero() || send_interval > kMaxProbeInterval ||
+      receive_interval <= TimeDelta::Zero() ||
+      receive_interval > kMaxProbeInterval) {
+    RTC_LOG(LS_INFO) << "Probing unsuccessful, invalid send/receive interval"
+                        " [cluster id: "
+                     << cluster_id
+                     << "] [send interval: " << ToString(send_interval)
+                     << "]"
+                        " [receive interval: "
+                     << ToString(receive_interval) << "]";
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
+          cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
+    }
+    return absl::nullopt;
+  }
+  // Since the `send_interval` does not include the time it takes to actually
+  // send the last packet the size of the last sent packet should not be
+  // included when calculating the send bitrate.
+  RTC_DCHECK_GT(cluster->size_total, cluster->size_last_send);
+  DataSize send_size = cluster->size_total - cluster->size_last_send;
+  DataRate send_rate = send_size / send_interval;
+
+  // Since the `receive_interval` does not include the time it takes to
+  // actually receive the first packet the size of the first received packet
+  // should not be included when calculating the receive bitrate.
+  RTC_DCHECK_GT(cluster->size_total, cluster->size_first_receive);
+  DataSize receive_size = cluster->size_total - cluster->size_first_receive;
+  DataRate receive_rate = receive_size / receive_interval;
+
+  double ratio = receive_rate / send_rate;
+  if (ratio > kMaxValidRatio) {
+    RTC_LOG(LS_INFO) << "Probing unsuccessful, receive/send ratio too high"
+                        " [cluster id: "
+                     << cluster_id << "] [send: " << ToString(send_size)
+                     << " / " << ToString(send_interval) << " = "
+                     << ToString(send_rate)
+                     << "]"
+                        " [receive: "
+                     << ToString(receive_size) << " / "
+                     << ToString(receive_interval) << " = "
+                     << ToString(receive_rate)
+                     << " ]"
+                        " [ratio: "
+                     << ToString(receive_rate) << " / " << ToString(send_rate)
+                     << " = " << ratio << " > kMaxValidRatio ("
+                     << kMaxValidRatio << ")]";
+    if (event_log_) {
+      event_log_->Log(std::make_unique<RtcEventProbeResultFailure>(
+          cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
+    }
+    return absl::nullopt;
+  }
+  RTC_LOG(LS_INFO) << "Probing successful"
+                      " [cluster id: "
+                   << cluster_id << "] [send: " << ToString(send_size) << " / "
+                   << ToString(send_interval) << " = " << ToString(send_rate)
+                   << " ]"
+                      " [receive: "
+                   << ToString(receive_size) << " / "
+                   << ToString(receive_interval) << " = "
+                   << ToString(receive_rate) << "]";
+
+  DataRate res = std::min(send_rate, receive_rate);
+  // If we're receiving at significantly lower bitrate than we were sending at,
+  // it suggests that we've found the true capacity of the link. In this case,
+  // set the target bitrate slightly lower to not immediately overuse.
+  if (receive_rate < kMinRatioForUnsaturatedLink * send_rate) {
+    RTC_DCHECK_GT(send_rate, receive_rate);
+    res = kTargetUtilizationFraction * receive_rate;
+  }
+  if (event_log_) {
+    event_log_->Log(
+        std::make_unique<RtcEventProbeResultSuccess>(cluster_id, res.bps()));
+  }
+  estimated_data_rate_ = res;
+  return estimated_data_rate_;
+}
+
+absl::optional<DataRate>
+ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrate() {
+  absl::optional<DataRate> estimated_data_rate = estimated_data_rate_;
+  estimated_data_rate_.reset();
+  return estimated_data_rate;
+}
+
+void ProbeBitrateEstimator::EraseOldClusters(Timestamp timestamp) {
+  for (auto it = clusters_.begin(); it != clusters_.end();) {
+    if (it->second.last_receive + kMaxClusterHistory < timestamp) {
+      it = clusters_.erase(it);
+    } else {
+      ++it;
+    }
+  }
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
new file mode 100644
index 0000000000..d5a523b7f3
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
+
+#include <limits>
+#include <map>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+class RtcEventLog;
+
+class ProbeBitrateEstimator {
+ public:
+  explicit ProbeBitrateEstimator(RtcEventLog* event_log);
+  ~ProbeBitrateEstimator();
+
+  // Should be called for every probe packet we receive feedback about.
+  // Returns the estimated bitrate if the probe completes a valid cluster.
+  absl::optional<DataRate> HandleProbeAndEstimateBitrate(
+      const PacketResult& packet_feedback);
+
+  absl::optional<DataRate> FetchAndResetLastEstimatedBitrate();
+
+ private:
+  struct AggregatedCluster {
+    int num_probes = 0;
+    Timestamp first_send = Timestamp::PlusInfinity();
+    Timestamp last_send = Timestamp::MinusInfinity();
+    Timestamp first_receive = Timestamp::PlusInfinity();
+    Timestamp last_receive = Timestamp::MinusInfinity();
+    DataSize size_last_send = DataSize::Zero();
+    DataSize size_first_receive = DataSize::Zero();
+    DataSize size_total = DataSize::Zero();
+  };
+
+  // Erases old cluster data that was seen before `timestamp`.
+  void EraseOldClusters(Timestamp timestamp);
+
+  std::map<int, AggregatedCluster> clusters_;
+  RtcEventLog* const event_log_;
+  absl::optional<DataRate> estimated_data_rate_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
new file mode 100644
index 0000000000..6b4146d2bf
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
@@ -0,0 +1,228 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
+
+#include <stddef.h>
+
+#include "api/transport/network_types.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kDefaultMinProbes = 5;
+constexpr int kDefaultMinBytes = 5000;
+constexpr float kTargetUtilizationFraction = 0.95f;
+}  // anonymous namespace
+
+class TestProbeBitrateEstimator : public ::testing::Test {
+ public:
+  TestProbeBitrateEstimator() : probe_bitrate_estimator_(nullptr) {}
+
+  // TODO(philipel): Use PacedPacketInfo when ProbeBitrateEstimator is rewritten
+  //                 to use that information.
+  void AddPacketFeedback(int probe_cluster_id,
+                         size_t size_bytes,
+                         int64_t send_time_ms,
+                         int64_t arrival_time_ms,
+                         int min_probes = kDefaultMinProbes,
+                         int min_bytes = kDefaultMinBytes) {
+    const Timestamp kReferenceTime = Timestamp::Seconds(1000);
+    PacketResult feedback;
+    feedback.sent_packet.send_time =
+        kReferenceTime + TimeDelta::Millis(send_time_ms);
+    feedback.sent_packet.size = DataSize::Bytes(size_bytes);
+    feedback.sent_packet.pacing_info =
+        PacedPacketInfo(probe_cluster_id, min_probes, min_bytes);
+    feedback.receive_time = kReferenceTime + TimeDelta::Millis(arrival_time_ms);
+    measured_data_rate_ =
+        probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(feedback);
+  }
+
+ protected:
+  absl::optional<DataRate> measured_data_rate_;
+  ProbeBitrateEstimator probe_bitrate_estimator_;
+};
+
+TEST_F(TestProbeBitrateEstimator, OneCluster) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+
+  EXPECT_NEAR(measured_data_rate_->bps(), 800000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, OneClusterTooFewProbes) {
+  AddPacketFeedback(0, 2000, 0, 10);
+  AddPacketFeedback(0, 2000, 10, 20);
+  AddPacketFeedback(0, 2000, 20, 30);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, OneClusterTooFewBytes) {
+  const int kMinBytes = 6000;
+  AddPacketFeedback(0, 800, 0, 10, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 10, 20, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 20, 30, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 30, 40, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 800, 40, 50, kDefaultMinProbes, kMinBytes);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, SmallCluster) {
+  const int kMinBytes = 1000;
+  AddPacketFeedback(0, 150, 0, 10, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 10, 20, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 20, 30, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 30, 40, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 40, 50, kDefaultMinProbes, kMinBytes);
+  AddPacketFeedback(0, 150, 50, 60, kDefaultMinProbes, kMinBytes);
+  EXPECT_NEAR(measured_data_rate_->bps(), 120000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, LargeCluster) {
+  const int kMinProbes = 30;
+  const int kMinBytes = 312500;
+
+  int64_t send_time = 0;
+  int64_t receive_time = 5;
+  for (int i = 0; i < 25; ++i) {
+    AddPacketFeedback(0, 12500, send_time, receive_time, kMinProbes, kMinBytes);
+    ++send_time;
+    ++receive_time;
+  }
+  EXPECT_NEAR(measured_data_rate_->bps(), 100000000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, FastReceive) {
+  AddPacketFeedback(0, 1000, 0, 15);
+  AddPacketFeedback(0, 1000, 10, 30);
+  AddPacketFeedback(0, 1000, 20, 35);
+  AddPacketFeedback(0, 1000, 30, 40);
+
+  EXPECT_NEAR(measured_data_rate_->bps(), 800000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, TooFastReceive) {
+  AddPacketFeedback(0, 1000, 0, 19);
+  AddPacketFeedback(0, 1000, 10, 22);
+  AddPacketFeedback(0, 1000, 20, 25);
+  AddPacketFeedback(0, 1000, 40, 27);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, SlowReceive) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 40);
+  AddPacketFeedback(0, 1000, 20, 70);
+  AddPacketFeedback(0, 1000, 30, 85);
+  // Expected send rate = 800 kbps, expected receive rate = 320 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 320000,
+              10);
+}
+
+TEST_F(TestProbeBitrateEstimator, BurstReceive) {
+  AddPacketFeedback(0, 1000, 0, 50);
+  AddPacketFeedback(0, 1000, 10, 50);
+  AddPacketFeedback(0, 1000, 20, 50);
+  AddPacketFeedback(0, 1000, 40, 50);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, MultipleClusters) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 40, 60);
+  // Expected send rate = 600 kbps, expected receive rate = 480 kbps.
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 480000,
+              10);
+
+  AddPacketFeedback(0, 1000, 50, 60);
+  // Expected send rate = 640 kbps, expected receive rate = 640 kbps.
+  EXPECT_NEAR(measured_data_rate_->bps(), 640000, 10);
+
+  AddPacketFeedback(1, 1000, 60, 70);
+  AddPacketFeedback(1, 1000, 65, 77);
+  AddPacketFeedback(1, 1000, 70, 84);
+  AddPacketFeedback(1, 1000, 75, 90);
+  // Expected send rate = 1600 kbps, expected receive rate = 1200 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 1200000,
+              10);
+}
+
+TEST_F(TestProbeBitrateEstimator, IgnoreOldClusters) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+
+  AddPacketFeedback(1, 1000, 60, 70);
+  AddPacketFeedback(1, 1000, 65, 77);
+  AddPacketFeedback(1, 1000, 70, 84);
+  AddPacketFeedback(1, 1000, 75, 90);
+  // Expected send rate = 1600 kbps, expected receive rate = 1200 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 1200000,
+              10);
+
+  // Coming in 6s later
+  AddPacketFeedback(0, 1000, 40 + 6000, 60 + 6000);
+
+  EXPECT_FALSE(measured_data_rate_);
+}
+
+TEST_F(TestProbeBitrateEstimator, IgnoreSizeLastSendPacket) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+  AddPacketFeedback(0, 1500, 40, 50);
+  // Expected send rate = 800 kbps, expected receive rate = 900 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), 800000, 10);
+}
+
+TEST_F(TestProbeBitrateEstimator, IgnoreSizeFirstReceivePacket) {
+  AddPacketFeedback(0, 1500, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+  // Expected send rate = 933 kbps, expected receive rate = 800 kbps.
+
+  EXPECT_NEAR(measured_data_rate_->bps(), kTargetUtilizationFraction * 800000,
+              10);
+}
+
+TEST_F(TestProbeBitrateEstimator, NoLastEstimatedBitrateBps) {
+  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate());
+}
+
+TEST_F(TestProbeBitrateEstimator, FetchLastEstimatedBitrateBps) {
+  AddPacketFeedback(0, 1000, 0, 10);
+  AddPacketFeedback(0, 1000, 10, 20);
+  AddPacketFeedback(0, 1000, 20, 30);
+  AddPacketFeedback(0, 1000, 30, 40);
+
+  auto estimated_bitrate =
+      probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate();
+  EXPECT_TRUE(estimated_bitrate);
+  EXPECT_NEAR(estimated_bitrate->bps(), 800000, 10);
+  EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrate());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc
new file mode 100644
index 0000000000..1af943c4cb
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc
@@ -0,0 +1,558 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+
+#include <algorithm>
+#include <initializer_list>
+#include <memory>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+// Maximum waiting time from the time of initiating probing to getting
+// the measured results back.
+constexpr TimeDelta kMaxWaitingTimeForProbingResult = TimeDelta::Seconds(1);
+
+// Default probing bitrate limit. Applied only when the application didn't
+// specify max bitrate.
+constexpr DataRate kDefaultMaxProbingBitrate = DataRate::KilobitsPerSec(5000);
+
+// If the bitrate drops to a factor `kBitrateDropThreshold` or lower
+// and we recover within `kBitrateDropTimeoutMs`, then we'll send
+// a probe at a fraction `kProbeFractionAfterDrop` of the original bitrate.
+constexpr double kBitrateDropThreshold = 0.66;
+constexpr TimeDelta kBitrateDropTimeout = TimeDelta::Seconds(5);
+constexpr double kProbeFractionAfterDrop = 0.85;
+
+// Timeout for probing after leaving ALR. If the bitrate drops significantly,
+// (as determined by the delay based estimator) and we leave ALR, then we will
+// send a probe if we recover within `kLeftAlrTimeoutMs` ms.
+constexpr TimeDelta kAlrEndedTimeout = TimeDelta::Seconds(3);
+
+// The expected uncertainty of probe result (as a fraction of the target probe
+// This is a limit on how often probing can be done when there is a BW
+// drop detected in ALR.
+constexpr TimeDelta kMinTimeBetweenAlrProbes = TimeDelta::Seconds(5);
+
+// bitrate). Used to avoid probing if the probe bitrate is close to our current
+// estimate.
+constexpr double kProbeUncertainty = 0.05;
+
+// Use probing to recover faster after large bitrate estimate drops.
+constexpr char kBweRapidRecoveryExperiment[] =
+    "WebRTC-BweRapidRecoveryExperiment";
+
+void MaybeLogProbeClusterCreated(RtcEventLog* event_log,
+                                 const ProbeClusterConfig& probe) {
+  RTC_DCHECK(event_log);
+  if (!event_log) {
+    return;
+  }
+
+  DataSize min_data_size = probe.target_data_rate * probe.target_duration;
+  event_log->Log(std::make_unique<RtcEventProbeClusterCreated>(
+      probe.id, probe.target_data_rate.bps(), probe.target_probe_count,
+      min_data_size.bytes()));
+}
+
+}  // namespace
+
+ProbeControllerConfig::ProbeControllerConfig(
+    const FieldTrialsView* key_value_config)
+    : first_exponential_probe_scale("p1", 3.0),
+      second_exponential_probe_scale("p2", 6.0),
+      further_exponential_probe_scale("step_size", 2),
+      further_probe_threshold("further_probe_threshold", 0.7),
+      alr_probing_interval("alr_interval", TimeDelta::Seconds(5)),
+      alr_probe_scale("alr_scale", 2),
+      network_state_estimate_probing_interval("network_state_interval",
+                                              TimeDelta::PlusInfinity()),
+      probe_if_estimate_lower_than_network_state_estimate_ratio(
+          "est_lower_than_network_ratio",
+          0),
+      estimate_lower_than_network_state_estimate_probing_interval(
+          "est_lower_than_network_interval",
+          TimeDelta::Seconds(3)),
+      network_state_probe_scale("network_state_scale", 1.0),
+      network_state_probe_duration("network_state_probe_duration",
+                                   TimeDelta::Millis(15)),
+
+      probe_on_max_allocated_bitrate_change("probe_max_allocation", true),
+      first_allocation_probe_scale("alloc_p1", 1),
+      second_allocation_probe_scale("alloc_p2", 2),
+      allocation_allow_further_probing("alloc_probe_further", false),
+      allocation_probe_max("alloc_probe_max", DataRate::PlusInfinity()),
+      min_probe_packets_sent("min_probe_packets_sent", 5),
+      min_probe_duration("min_probe_duration", TimeDelta::Millis(15)),
+      limit_probe_target_rate_to_loss_bwe("limit_probe_target_rate_to_loss_bwe",
+                                          false),
+      loss_limited_probe_scale("loss_limited_scale", 1.5),
+      skip_if_estimate_larger_than_fraction_of_max(
+          "skip_if_est_larger_than_fraction_of_max",
+          0.0),
+      not_probe_if_delay_increased("not_probe_if_delay_increased", false) {
+  ParseFieldTrial({&first_exponential_probe_scale,
+                   &second_exponential_probe_scale,
+                   &further_exponential_probe_scale,
+                   &further_probe_threshold,
+                   &alr_probing_interval,
+                   &alr_probe_scale,
+                   &probe_on_max_allocated_bitrate_change,
+                   &first_allocation_probe_scale,
+                   &second_allocation_probe_scale,
+                   &allocation_allow_further_probing,
+                   &min_probe_duration,
+                   &network_state_estimate_probing_interval,
+                   &probe_if_estimate_lower_than_network_state_estimate_ratio,
+                   &estimate_lower_than_network_state_estimate_probing_interval,
+                   &network_state_probe_scale,
+                   &network_state_probe_duration,
+                   &min_probe_packets_sent,
+                   &limit_probe_target_rate_to_loss_bwe,
+                   &loss_limited_probe_scale,
+                   &skip_if_estimate_larger_than_fraction_of_max,
+                   &not_probe_if_delay_increased},
+                  key_value_config->Lookup("WebRTC-Bwe-ProbingConfiguration"));
+
+  // Specialized keys overriding subsets of WebRTC-Bwe-ProbingConfiguration
+  ParseFieldTrial(
+      {&first_exponential_probe_scale, &second_exponential_probe_scale},
+      key_value_config->Lookup("WebRTC-Bwe-InitialProbing"));
+  ParseFieldTrial({&further_exponential_probe_scale, &further_probe_threshold},
+                  key_value_config->Lookup("WebRTC-Bwe-ExponentialProbing"));
+  ParseFieldTrial(
+      {&alr_probing_interval, &alr_probe_scale, &loss_limited_probe_scale},
+      key_value_config->Lookup("WebRTC-Bwe-AlrProbing"));
+  ParseFieldTrial(
+      {&first_allocation_probe_scale, &second_allocation_probe_scale,
+       &allocation_allow_further_probing, &allocation_probe_max},
+      key_value_config->Lookup("WebRTC-Bwe-AllocationProbing"));
+  ParseFieldTrial({&min_probe_packets_sent, &min_probe_duration},
+                  key_value_config->Lookup("WebRTC-Bwe-ProbingBehavior"));
+}
+
+ProbeControllerConfig::ProbeControllerConfig(const ProbeControllerConfig&) =
+    default;
+ProbeControllerConfig::~ProbeControllerConfig() = default;
+
+ProbeController::ProbeController(const FieldTrialsView* key_value_config,
+                                 RtcEventLog* event_log)
+    : enable_periodic_alr_probing_(false),
+      in_rapid_recovery_experiment_(absl::StartsWith(
+          key_value_config->Lookup(kBweRapidRecoveryExperiment),
+          "Enabled")),
+      event_log_(event_log),
+      config_(ProbeControllerConfig(key_value_config)) {
+  Reset(Timestamp::Zero());
+}
+
+ProbeController::~ProbeController() {}
+
+std::vector<ProbeClusterConfig> ProbeController::SetBitrates(
+    DataRate min_bitrate,
+    DataRate start_bitrate,
+    DataRate max_bitrate,
+    Timestamp at_time) {
+  if (start_bitrate > DataRate::Zero()) {
+    start_bitrate_ = start_bitrate;
+    estimated_bitrate_ = start_bitrate;
+  } else if (start_bitrate_.IsZero()) {
+    start_bitrate_ = min_bitrate;
+  }
+
+  // The reason we use the variable `old_max_bitrate_pbs` is because we
+  // need to set `max_bitrate_` before we call InitiateProbing.
+  DataRate old_max_bitrate = max_bitrate_;
+  max_bitrate_ =
+      max_bitrate.IsFinite() ? max_bitrate : kDefaultMaxProbingBitrate;
+
+  switch (state_) {
+    case State::kInit:
+      if (network_available_)
+        return InitiateExponentialProbing(at_time);
+      break;
+
+    case State::kWaitingForProbingResult:
+      break;
+
+    case State::kProbingComplete:
+      // If the new max bitrate is higher than both the old max bitrate and the
+      // estimate then initiate probing.
+      if (!estimated_bitrate_.IsZero() && old_max_bitrate < max_bitrate_ &&
+          estimated_bitrate_ < max_bitrate_) {
+        return InitiateProbing(at_time, {max_bitrate_}, false);
+      }
+      break;
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::OnMaxTotalAllocatedBitrate(
+    DataRate max_total_allocated_bitrate,
+    Timestamp at_time) {
+  const bool in_alr = alr_start_time_.has_value();
+  const bool allow_allocation_probe = in_alr;
+
+  if (config_.probe_on_max_allocated_bitrate_change &&
+      state_ == State::kProbingComplete &&
+      max_total_allocated_bitrate != max_total_allocated_bitrate_ &&
+      estimated_bitrate_ < max_bitrate_ &&
+      estimated_bitrate_ < max_total_allocated_bitrate &&
+      allow_allocation_probe) {
+    max_total_allocated_bitrate_ = max_total_allocated_bitrate;
+
+    if (!config_.first_allocation_probe_scale)
+      return std::vector<ProbeClusterConfig>();
+
+    DataRate first_probe_rate = max_total_allocated_bitrate *
+                                config_.first_allocation_probe_scale.Value();
+    DataRate probe_cap = config_.allocation_probe_max.Get();
+    first_probe_rate = std::min(first_probe_rate, probe_cap);
+    std::vector<DataRate> probes = {first_probe_rate};
+    if (config_.second_allocation_probe_scale) {
+      DataRate second_probe_rate =
+          max_total_allocated_bitrate *
+          config_.second_allocation_probe_scale.Value();
+      second_probe_rate = std::min(second_probe_rate, probe_cap);
+      if (second_probe_rate > first_probe_rate)
+        probes.push_back(second_probe_rate);
+    }
+    return InitiateProbing(at_time, probes,
+                           config_.allocation_allow_further_probing.Get());
+  }
+  max_total_allocated_bitrate_ = max_total_allocated_bitrate;
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  network_available_ = msg.network_available;
+
+  if (!network_available_ && state_ == State::kWaitingForProbingResult) {
+    state_ = State::kProbingComplete;
+    min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  }
+
+  if (network_available_ && state_ == State::kInit && !start_bitrate_.IsZero())
+    return InitiateExponentialProbing(msg.at_time);
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::InitiateExponentialProbing(
+    Timestamp at_time) {
+  RTC_DCHECK(network_available_);
+  RTC_DCHECK(state_ == State::kInit);
+  RTC_DCHECK_GT(start_bitrate_, DataRate::Zero());
+
+  // When probing at 1.8 Mbps ( 6x 300), this represents a threshold of
+  // 1.2 Mbps to continue probing.
+  std::vector<DataRate> probes = {config_.first_exponential_probe_scale *
+                                  start_bitrate_};
+  if (config_.second_exponential_probe_scale &&
+      config_.second_exponential_probe_scale.GetOptional().value() > 0) {
+    probes.push_back(config_.second_exponential_probe_scale.Value() *
+                     start_bitrate_);
+  }
+  return InitiateProbing(at_time, probes, true);
+}
+
+std::vector<ProbeClusterConfig> ProbeController::SetEstimatedBitrate(
+    DataRate bitrate,
+    BandwidthLimitedCause bandwidth_limited_cause,
+    Timestamp at_time) {
+  bandwidth_limited_cause_ = bandwidth_limited_cause;
+  if (bitrate < kBitrateDropThreshold * estimated_bitrate_) {
+    time_of_last_large_drop_ = at_time;
+    bitrate_before_last_large_drop_ = estimated_bitrate_;
+  }
+  estimated_bitrate_ = bitrate;
+
+  if (state_ == State::kWaitingForProbingResult) {
+    // Continue probing if probing results indicate channel has greater
+    // capacity.
+    DataRate network_state_estimate_probe_further_limit =
+        config_.network_state_estimate_probing_interval->IsFinite() &&
+                network_estimate_
+            ? network_estimate_->link_capacity_upper *
+                  config_.further_probe_threshold
+            : DataRate::PlusInfinity();
+    RTC_LOG(LS_INFO) << "Measured bitrate: " << bitrate
+                     << " Minimum to probe further: "
+                     << min_bitrate_to_probe_further_ << " upper limit: "
+                     << network_state_estimate_probe_further_limit;
+
+    if (bitrate > min_bitrate_to_probe_further_ &&
+        bitrate <= network_state_estimate_probe_further_limit) {
+      return InitiateProbing(
+          at_time, {config_.further_exponential_probe_scale * bitrate}, true);
+    }
+  }
+  return {};
+}
+
+void ProbeController::EnablePeriodicAlrProbing(bool enable) {
+  enable_periodic_alr_probing_ = enable;
+}
+
+void ProbeController::SetAlrStartTimeMs(
+    absl::optional<int64_t> alr_start_time_ms) {
+  if (alr_start_time_ms) {
+    alr_start_time_ = Timestamp::Millis(*alr_start_time_ms);
+  } else {
+    alr_start_time_ = absl::nullopt;
+  }
+}
+void ProbeController::SetAlrEndedTimeMs(int64_t alr_end_time_ms) {
+  alr_end_time_.emplace(Timestamp::Millis(alr_end_time_ms));
+}
+
+std::vector<ProbeClusterConfig> ProbeController::RequestProbe(
+    Timestamp at_time) {
+  // Called once we have returned to normal state after a large drop in
+  // estimated bandwidth. The current response is to initiate a single probe
+  // session (if not already probing) at the previous bitrate.
+  //
+  // If the probe session fails, the assumption is that this drop was a
+  // real one from a competing flow or a network change.
+  bool in_alr = alr_start_time_.has_value();
+  bool alr_ended_recently =
+      (alr_end_time_.has_value() &&
+       at_time - alr_end_time_.value() < kAlrEndedTimeout);
+  if (in_alr || alr_ended_recently || in_rapid_recovery_experiment_) {
+    if (state_ == State::kProbingComplete) {
+      DataRate suggested_probe =
+          kProbeFractionAfterDrop * bitrate_before_last_large_drop_;
+      DataRate min_expected_probe_result =
+          (1 - kProbeUncertainty) * suggested_probe;
+      TimeDelta time_since_drop = at_time - time_of_last_large_drop_;
+      TimeDelta time_since_probe = at_time - last_bwe_drop_probing_time_;
+      if (min_expected_probe_result > estimated_bitrate_ &&
+          time_since_drop < kBitrateDropTimeout &&
+          time_since_probe > kMinTimeBetweenAlrProbes) {
+        RTC_LOG(LS_INFO) << "Detected big bandwidth drop, start probing.";
+        // Track how often we probe in response to bandwidth drop in ALR.
+        RTC_HISTOGRAM_COUNTS_10000(
+            "WebRTC.BWE.BweDropProbingIntervalInS",
+            (at_time - last_bwe_drop_probing_time_).seconds());
+        last_bwe_drop_probing_time_ = at_time;
+        return InitiateProbing(at_time, {suggested_probe}, false);
+      }
+    }
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+void ProbeController::SetNetworkStateEstimate(
+    webrtc::NetworkStateEstimate estimate) {
+  network_estimate_ = estimate;
+}
+
+void ProbeController::Reset(Timestamp at_time) {
+  network_available_ = true;
+  bandwidth_limited_cause_ = BandwidthLimitedCause::kDelayBasedLimited;
+  state_ = State::kInit;
+  min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  time_last_probing_initiated_ = Timestamp::Zero();
+  estimated_bitrate_ = DataRate::Zero();
+  network_estimate_ = absl::nullopt;
+  start_bitrate_ = DataRate::Zero();
+  max_bitrate_ = kDefaultMaxProbingBitrate;
+  Timestamp now = at_time;
+  last_bwe_drop_probing_time_ = now;
+  alr_end_time_.reset();
+  time_of_last_large_drop_ = now;
+  bitrate_before_last_large_drop_ = DataRate::Zero();
+  max_total_allocated_bitrate_ = DataRate::Zero();
+}
+
+bool ProbeController::TimeForAlrProbe(Timestamp at_time) const {
+  if (enable_periodic_alr_probing_ && alr_start_time_) {
+    Timestamp next_probe_time =
+        std::max(*alr_start_time_, time_last_probing_initiated_) +
+        config_.alr_probing_interval;
+    return at_time >= next_probe_time;
+  }
+  return false;
+}
+
+bool ProbeController::TimeForNetworkStateProbe(Timestamp at_time) const {
+  if (!network_estimate_ ||
+      network_estimate_->link_capacity_upper.IsInfinite()) {
+    return false;
+  }
+
+  bool probe_due_to_low_estimate =
+      bandwidth_limited_cause_ == BandwidthLimitedCause::kDelayBasedLimited &&
+      estimated_bitrate_ <
+          config_.probe_if_estimate_lower_than_network_state_estimate_ratio *
+              network_estimate_->link_capacity_upper;
+  if (probe_due_to_low_estimate &&
+      config_.estimate_lower_than_network_state_estimate_probing_interval
+          ->IsFinite()) {
+    Timestamp next_probe_time =
+        time_last_probing_initiated_ +
+        config_.estimate_lower_than_network_state_estimate_probing_interval;
+    return at_time >= next_probe_time;
+  }
+
+  bool periodic_probe =
+      estimated_bitrate_ < network_estimate_->link_capacity_upper;
+  if (periodic_probe &&
+      config_.network_state_estimate_probing_interval->IsFinite()) {
+    Timestamp next_probe_time = time_last_probing_initiated_ +
+                                config_.network_state_estimate_probing_interval;
+    return at_time >= next_probe_time;
+  }
+
+  return false;
+}
+
+std::vector<ProbeClusterConfig> ProbeController::Process(Timestamp at_time) {
+  if (at_time - time_last_probing_initiated_ >
+      kMaxWaitingTimeForProbingResult) {
+    if (state_ == State::kWaitingForProbingResult) {
+      RTC_LOG(LS_INFO) << "kWaitingForProbingResult: timeout";
+      state_ = State::kProbingComplete;
+      min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+    }
+  }
+  if (estimated_bitrate_.IsZero() || state_ != State::kProbingComplete) {
+    return {};
+  }
+  if (TimeForAlrProbe(at_time) || TimeForNetworkStateProbe(at_time)) {
+    return InitiateProbing(
+        at_time, {estimated_bitrate_ * config_.alr_probe_scale}, true);
+  }
+  return std::vector<ProbeClusterConfig>();
+}
+
+std::vector<ProbeClusterConfig> ProbeController::InitiateProbing(
+    Timestamp now,
+    std::vector<DataRate> bitrates_to_probe,
+    bool probe_further) {
+  if (config_.skip_if_estimate_larger_than_fraction_of_max > 0) {
+    DataRate network_estimate = network_estimate_
+                                    ? network_estimate_->link_capacity_upper
+                                    : DataRate::PlusInfinity();
+    DataRate max_probe_rate =
+        max_total_allocated_bitrate_.IsZero()
+            ? max_bitrate_
+            : std::min(max_total_allocated_bitrate_, max_bitrate_);
+    if (std::min(network_estimate, estimated_bitrate_) >
+        config_.skip_if_estimate_larger_than_fraction_of_max * max_probe_rate) {
+      state_ = State::kProbingComplete;
+      min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+      return {};
+    }
+  }
+
+  DataRate max_probe_bitrate = max_bitrate_;
+  if (max_total_allocated_bitrate_ > DataRate::Zero()) {
+    // If a max allocated bitrate has been configured, allow probing up to 2x
+    // that rate. This allows some overhead to account for bursty streams,
+    // which otherwise would have to ramp up when the overshoot is already in
+    // progress.
+    // It also avoids minor quality reduction caused by probes often being
+    // received at slightly less than the target probe bitrate.
+    max_probe_bitrate =
+        std::min(max_probe_bitrate, max_total_allocated_bitrate_ * 2);
+  }
+
+  DataRate estimate_capped_bitrate = DataRate::PlusInfinity();
+  if (config_.limit_probe_target_rate_to_loss_bwe) {
+    switch (bandwidth_limited_cause_) {
+      case BandwidthLimitedCause::kLossLimitedBweDecreasing:
+        // If bandwidth estimate is decreasing because of packet loss, do not
+        // send probes.
+        return {};
+      case BandwidthLimitedCause::kLossLimitedBweIncreasing:
+        estimate_capped_bitrate =
+            std::min(max_probe_bitrate,
+                     estimated_bitrate_ * config_.loss_limited_probe_scale);
+        break;
+      case BandwidthLimitedCause::kDelayBasedLimited:
+        break;
+      default:
+        break;
+    }
+  }
+  if (config_.not_probe_if_delay_increased &&
+      bandwidth_limited_cause_ ==
+          BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased) {
+    return {};
+  }
+
+  if (config_.network_state_estimate_probing_interval->IsFinite() &&
+      network_estimate_ && network_estimate_->link_capacity_upper.IsFinite()) {
+    if (network_estimate_->link_capacity_upper.IsZero()) {
+      RTC_LOG(LS_INFO) << "Not sending probe, Network state estimate is zero";
+      return {};
+    }
+    estimate_capped_bitrate =
+        std::min({estimate_capped_bitrate, max_probe_bitrate,
+                  network_estimate_->link_capacity_upper *
+                      config_.network_state_probe_scale});
+  }
+
+  std::vector<ProbeClusterConfig> pending_probes;
+  for (DataRate bitrate : bitrates_to_probe) {
+    RTC_DCHECK(!bitrate.IsZero());
+
+    bitrate = std::min(bitrate, estimate_capped_bitrate);
+    if (bitrate > max_probe_bitrate) {
+      bitrate = max_probe_bitrate;
+      probe_further = false;
+    }
+
+    ProbeClusterConfig config;
+    config.at_time = now;
+    config.target_data_rate = bitrate;
+    if (network_estimate_ &&
+        config_.network_state_estimate_probing_interval->IsFinite()) {
+      config.target_duration = config_.network_state_probe_duration;
+    } else {
+      config.target_duration = config_.min_probe_duration;
+    }
+
+    config.target_probe_count = config_.min_probe_packets_sent;
+    config.id = next_probe_cluster_id_;
+    next_probe_cluster_id_++;
+    MaybeLogProbeClusterCreated(event_log_, config);
+    pending_probes.push_back(config);
+  }
+  time_last_probing_initiated_ = now;
+  if (probe_further) {
+    state_ = State::kWaitingForProbingResult;
+    // Dont expect probe results to be larger than a fraction of the actual
+    // probe rate.
+    min_bitrate_to_probe_further_ =
+        std::min(estimate_capped_bitrate, (*(bitrates_to_probe.end() - 1))) *
+        config_.further_probe_threshold;
+  } else {
+    state_ = State::kProbingComplete;
+    min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  }
+  return pending_probes;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.h
new file mode 100644
index 0000000000..aa8b526ab0
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.h
@@ -0,0 +1,196 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
+
+#include <stdint.h>
+
+#include <initializer_list>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+struct ProbeControllerConfig {
+  explicit ProbeControllerConfig(const FieldTrialsView* key_value_config);
+  ProbeControllerConfig(const ProbeControllerConfig&);
+  ProbeControllerConfig& operator=(const ProbeControllerConfig&) = default;
+  ~ProbeControllerConfig();
+
+  // These parameters configure the initial probes. First we send one or two
+  // probes of sizes p1 * start_bitrate_ and p2 * start_bitrate_.
+  // Then whenever we get a bitrate estimate of at least further_probe_threshold
+  // times the size of the last sent probe we'll send another one of size
+  // step_size times the new estimate.
+  FieldTrialParameter<double> first_exponential_probe_scale;
+  FieldTrialOptional<double> second_exponential_probe_scale;
+  FieldTrialParameter<double> further_exponential_probe_scale;
+  FieldTrialParameter<double> further_probe_threshold;
+
+  // Configures how often we send ALR probes and how big they are.
+  FieldTrialParameter<TimeDelta> alr_probing_interval;
+  FieldTrialParameter<double> alr_probe_scale;
+
+  // Configures how often we send probes if NetworkStateEstimate is available.
+  FieldTrialParameter<TimeDelta> network_state_estimate_probing_interval;
+  // Periodically probe as long as the the ratio beteeen current estimate and
+  // NetworkStateEstimate is lower then this.
+  FieldTrialParameter<double>
+      probe_if_estimate_lower_than_network_state_estimate_ratio;
+  FieldTrialParameter<TimeDelta>
+      estimate_lower_than_network_state_estimate_probing_interval;
+  FieldTrialParameter<double> network_state_probe_scale;
+  // Overrides min_probe_duration if network_state_estimate_probing_interval
+  // is set and a network state estimate is known.
+  FieldTrialParameter<TimeDelta> network_state_probe_duration;
+
+  // Configures the probes emitted by changed to the allocated bitrate.
+  FieldTrialParameter<bool> probe_on_max_allocated_bitrate_change;
+  FieldTrialOptional<double> first_allocation_probe_scale;
+  FieldTrialOptional<double> second_allocation_probe_scale;
+  FieldTrialFlag allocation_allow_further_probing;
+  FieldTrialParameter<DataRate> allocation_probe_max;
+
+  // The minimum number probing packets used.
+  FieldTrialParameter<int> min_probe_packets_sent;
+  // The minimum probing duration.
+  FieldTrialParameter<TimeDelta> min_probe_duration;
+  // Periodically probe when bandwidth estimate is loss limited.
+  FieldTrialParameter<bool> limit_probe_target_rate_to_loss_bwe;
+  FieldTrialParameter<double> loss_limited_probe_scale;
+  // Dont send a probe if min(estimate, network state estimate) is larger than
+  // this fraction of the set max bitrate.
+  FieldTrialParameter<double> skip_if_estimate_larger_than_fraction_of_max;
+  // Do not send probes if network is either overusing or underusing.
+  FieldTrialParameter<bool> not_probe_if_delay_increased;
+};
+
+// Reason that bandwidth estimate is limited. Bandwidth estimate can be limited
+// by either delay based bwe, or loss based bwe when it increases/decreases the
+// estimate.
+enum class BandwidthLimitedCause {
+  kLossLimitedBweIncreasing = 0,
+  kLossLimitedBweDecreasing = 1,
+  kDelayBasedLimited = 2,
+  kDelayBasedLimitedDelayIncreased = 3,
+};
+
+// This class controls initiation of probing to estimate initial channel
+// capacity. There is also support for probing during a session when max
+// bitrate is adjusted by an application.
+class ProbeController {
+ public:
+  explicit ProbeController(const FieldTrialsView* key_value_config,
+                           RtcEventLog* event_log);
+  ~ProbeController();
+
+  ProbeController(const ProbeController&) = delete;
+  ProbeController& operator=(const ProbeController&) = delete;
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> SetBitrates(
+      DataRate min_bitrate,
+      DataRate start_bitrate,
+      DataRate max_bitrate,
+      Timestamp at_time);
+
+  // The total bitrate, as opposed to the max bitrate, is the sum of the
+  // configured bitrates for all active streams.
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig>
+  OnMaxTotalAllocatedBitrate(DataRate max_total_allocated_bitrate,
+                             Timestamp at_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> OnNetworkAvailability(
+      NetworkAvailability msg);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> SetEstimatedBitrate(
+      DataRate bitrate,
+      BandwidthLimitedCause bandwidth_limited_cause,
+      Timestamp at_time);
+
+  void EnablePeriodicAlrProbing(bool enable);
+
+  void SetAlrStartTimeMs(absl::optional<int64_t> alr_start_time);
+  void SetAlrEndedTimeMs(int64_t alr_end_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> RequestProbe(
+      Timestamp at_time);
+
+  void SetNetworkStateEstimate(webrtc::NetworkStateEstimate estimate);
+
+  // Resets the ProbeController to a state equivalent to as if it was just
+  // created EXCEPT for `enable_periodic_alr_probing_`.
+  void Reset(Timestamp at_time);
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> Process(
+      Timestamp at_time);
+
+  // Gets the value of field trial not_probe_if_delay_increased.
+  bool DontProbeIfDelayIncreased() {
+    return config_.not_probe_if_delay_increased;
+  }
+
+ private:
+  enum class State {
+    // Initial state where no probing has been triggered yet.
+    kInit,
+    // Waiting for probing results to continue further probing.
+    kWaitingForProbingResult,
+    // Probing is complete.
+    kProbingComplete,
+  };
+
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig>
+  InitiateExponentialProbing(Timestamp at_time);
+  ABSL_MUST_USE_RESULT std::vector<ProbeClusterConfig> InitiateProbing(
+      Timestamp now,
+      std::vector<DataRate> bitrates_to_probe,
+      bool probe_further);
+  bool TimeForAlrProbe(Timestamp at_time) const;
+  bool TimeForNetworkStateProbe(Timestamp at_time) const;
+
+  bool network_available_;
+  BandwidthLimitedCause bandwidth_limited_cause_ =
+      BandwidthLimitedCause::kDelayBasedLimited;
+  State state_;
+  DataRate min_bitrate_to_probe_further_ = DataRate::PlusInfinity();
+  Timestamp time_last_probing_initiated_ = Timestamp::MinusInfinity();
+  DataRate estimated_bitrate_ = DataRate::Zero();
+  absl::optional<webrtc::NetworkStateEstimate> network_estimate_;
+  DataRate start_bitrate_ = DataRate::Zero();
+  DataRate max_bitrate_ = DataRate::PlusInfinity();
+  Timestamp last_bwe_drop_probing_time_ = Timestamp::Zero();
+  absl::optional<Timestamp> alr_start_time_;
+  absl::optional<Timestamp> alr_end_time_;
+  bool enable_periodic_alr_probing_;
+  Timestamp time_of_last_large_drop_ = Timestamp::MinusInfinity();
+  DataRate bitrate_before_last_large_drop_ = DataRate::Zero();
+  DataRate max_total_allocated_bitrate_ = DataRate::Zero();
+
+  const bool in_rapid_recovery_experiment_;
+  RtcEventLog* event_log_;
+
+  int32_t next_probe_cluster_id_ = 1;
+
+  ProbeControllerConfig config_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build
new file mode 100644
index 0000000000..4f4f573cd9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("probe_controller_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
new file mode 100644
index 0000000000..e6a5c8ceef
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
@@ -0,0 +1,1131 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/probe_controller.h"
+
+#include <memory>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::NiceMock;
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+constexpr DataRate kMinBitrate = DataRate::BitsPerSec(100);
+constexpr DataRate kStartBitrate = DataRate::BitsPerSec(300);
+constexpr DataRate kMaxBitrate = DataRate::BitsPerSec(10000);
+
+constexpr TimeDelta kExponentialProbingTimeout = TimeDelta::Seconds(5);
+
+constexpr TimeDelta kAlrProbeInterval = TimeDelta::Seconds(5);
+constexpr TimeDelta kAlrEndedTimeout = TimeDelta::Seconds(3);
+constexpr TimeDelta kBitrateDropTimeout = TimeDelta::Seconds(5);
+}  // namespace
+
+class ProbeControllerFixture {
+ public:
+  explicit ProbeControllerFixture(absl::string_view field_trials = "")
+      : field_trial_config_(field_trials), clock_(100000000L) {}
+
+  std::unique_ptr<ProbeController> CreateController() {
+    return std::make_unique<ProbeController>(&field_trial_config_,
+                                             &mock_rtc_event_log);
+  }
+
+  Timestamp CurrentTime() { return clock_.CurrentTime(); }
+  void AdvanceTime(TimeDelta delta) { clock_.AdvanceTime(delta); }
+
+  ExplicitKeyValueConfig field_trial_config_;
+  SimulatedClock clock_;
+  NiceMock<MockRtcEventLog> mock_rtc_event_log;
+};
+
+TEST(ProbeControllerTest, InitiatesProbingAtStart) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_GE(probes.size(), 2u);
+}
+
+TEST(ProbeControllerTest, SetsDefaultTargetDurationAndTargetProbeCount) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  std::vector<ProbeClusterConfig> probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_GE(probes.size(), 2u);
+
+  EXPECT_EQ(probes[0].target_duration, TimeDelta::Millis(15));
+  EXPECT_EQ(probes[0].target_probe_count, 5);
+}
+
+TEST(ProbeControllerTest,
+     FieldTrialsOverrideDefaultTargetDurationAndTargetProbeCount) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingBehavior/"
+      "min_probe_packets_sent:2,min_probe_duration:123ms/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  std::vector<ProbeClusterConfig> probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_GE(probes.size(), 2u);
+
+  EXPECT_EQ(probes[0].target_duration, TimeDelta::Millis(123));
+  EXPECT_EQ(probes[0].target_probe_count, 2);
+}
+
+TEST(ProbeControllerTest, ProbeOnlyWhenNetworkIsUp) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->OnNetworkAvailability(
+      {.at_time = fixture.CurrentTime(), .network_available = false});
+  probes = probe_controller->SetBitrates(kMinBitrate, kStartBitrate,
+                                         kMaxBitrate, fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+  probes = probe_controller->OnNetworkAvailability(
+      {.at_time = fixture.CurrentTime(), .network_available = true});
+  EXPECT_GE(probes.size(), 2u);
+}
+
+TEST(ProbeControllerTest, CanConfigureInitialProbeRateFactor) {
+  ProbeControllerFixture fixture("WebRTC-Bwe-ProbingConfiguration/p1:2,p2:3/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  EXPECT_EQ(probes[0].target_data_rate, kStartBitrate * 2);
+  EXPECT_EQ(probes[1].target_data_rate, kStartBitrate * 3);
+}
+
+TEST(ProbeControllerTest, DisableSecondInitialProbeIfRateFactorZero) {
+  ProbeControllerFixture fixture("WebRTC-Bwe-ProbingConfiguration/p1:2,p2:0/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, kStartBitrate * 2);
+}
+
+TEST(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncrease) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  // Long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate + DataRate::BitsPerSec(100),
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), kMaxBitrate.bps() + 100);
+}
+
+TEST(ProbeControllerTest, ProbesOnMaxAllocatedBitrateIncreaseOnlyWhenInAlr) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate - DataRate::BitsPerSec(1),
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // Probe when in alr.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate + DataRate::BitsPerSec(1), fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  EXPECT_EQ(probes.at(0).target_data_rate, kMaxBitrate);
+
+  // Do not probe when not in alr.
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate + DataRate::BitsPerSec(2), fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, CanDisableProbingOnMaxTotalAllocatedBitrateIncrease) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "probe_max_allocation:false/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate - DataRate::BitsPerSec(1),
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+
+  // Do no probe, since probe_max_allocation:false.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate + DataRate::BitsPerSec(1), fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncreaseAtMaxBitrate) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  // Long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate + DataRate::BitsPerSec(100),
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate,
+            kMaxBitrate + DataRate::BitsPerSec(100));
+}
+
+TEST(ProbeControllerTest, TestExponentialProbing) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+
+  // Repeated probe should only be sent when estimated bitrate climbs above
+  // 0.7 * 6 * kStartBitrate = 1260.
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1000), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1800), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 2 * 1800);
+}
+
+TEST(ProbeControllerTest, TestExponentialProbingTimeout) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  // Advance far enough to cause a time out in waiting for probing result.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1800), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, RequestProbeInAlr) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_GE(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 0.85 * 500);
+}
+
+TEST(ProbeControllerTest, RequestProbeWhenAlrEndedRecently) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetAlrEndedTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrEndedTimeout - TimeDelta::Millis(1));
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 0.85 * 500);
+}
+
+TEST(ProbeControllerTest, RequestProbeWhenAlrNotEndedRecently) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetAlrEndedTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrEndedTimeout + TimeDelta::Millis(1));
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, RequestProbeWhenBweDropNotRecent) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  fixture.AdvanceTime(kBitrateDropTimeout + TimeDelta::Millis(1));
+  probes = probe_controller->RequestProbe(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, PeriodicProbing) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  Timestamp start_time = fixture.CurrentTime();
+
+  // Expect the controller to send a new probe after 5s has passed.
+  probe_controller->SetAlrStartTimeMs(start_time.ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 1000);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  // The following probe should be sent at 10s into ALR.
+  probe_controller->SetAlrStartTimeMs(start_time.ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(4));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  probe_controller->SetAlrStartTimeMs(start_time.ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, PeriodicProbingAfterReset) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  Timestamp alr_start_time = fixture.CurrentTime();
+
+  probe_controller->SetAlrStartTimeMs(alr_start_time.ms());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->Reset(fixture.CurrentTime());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  // Since bitrates are not yet set, no probe is sent event though we are in ALR
+  // mode.
+  EXPECT_TRUE(probes.empty());
+
+  probes = probe_controller->SetBitrates(kMinBitrate, kStartBitrate,
+                                         kMaxBitrate, fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+
+  // Make sure we use `kStartBitrateBps` as the estimated bitrate
+  // until SetEstimatedBitrate is called with an updated estimate.
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, kStartBitrate * 2);
+}
+
+TEST(ProbeControllerTest, TestExponentialProbingOverflow) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  const DataRate kMbpsMultiplier = DataRate::KilobitsPerSec(1000);
+  auto probes = probe_controller->SetBitrates(kMinBitrate, 10 * kMbpsMultiplier,
+                                              100 * kMbpsMultiplier,
+                                              fixture.CurrentTime());
+  // Verify that probe bitrate is capped at the specified max bitrate.
+  probes = probe_controller->SetEstimatedBitrate(
+      60 * kMbpsMultiplier, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 100 * kMbpsMultiplier);
+  // Verify that repeated probes aren't sent.
+  probes = probe_controller->SetEstimatedBitrate(
+      100 * kMbpsMultiplier, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, TestAllocatedBitrateCap) {
+  ProbeControllerFixture fixture;
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  const DataRate kMbpsMultiplier = DataRate::KilobitsPerSec(1000);
+  const DataRate kMaxBitrate = 100 * kMbpsMultiplier;
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, 10 * kMbpsMultiplier, kMaxBitrate, fixture.CurrentTime());
+
+  // Configure ALR for periodic probing.
+  probe_controller->EnablePeriodicAlrProbing(true);
+  Timestamp alr_start_time = fixture.CurrentTime();
+  probe_controller->SetAlrStartTimeMs(alr_start_time.ms());
+
+  DataRate estimated_bitrate = kMaxBitrate / 10;
+  probes = probe_controller->SetEstimatedBitrate(
+      estimated_bitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  // Set a max allocated bitrate below the current estimate.
+  DataRate max_allocated = estimated_bitrate - 1 * kMbpsMultiplier;
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(max_allocated,
+                                                        fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());  // No probe since lower than current max.
+
+  // Probes such as ALR capped at 2x the max allocation limit.
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 2 * max_allocated);
+
+  // Remove allocation limit.
+  EXPECT_TRUE(
+      probe_controller
+          ->OnMaxTotalAllocatedBitrate(DataRate::Zero(), fixture.CurrentTime())
+          .empty());
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, estimated_bitrate * 2);
+}
+
+TEST(ProbeControllerTest, ConfigurableProbingFieldTrial) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "p1:2,p2:5,step_size:3,further_probe_threshold:0.8,"
+      "alloc_p1:2,alloc_p2,min_probe_packets_sent:2/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(kMinBitrate, kStartBitrate,
+                                              DataRate::KilobitsPerSec(5000),
+                                              fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 2u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 600);
+  EXPECT_EQ(probes[0].target_probe_count, 2);
+  EXPECT_EQ(probes[1].target_data_rate.bps(), 1500);
+  EXPECT_EQ(probes[1].target_probe_count, 2);
+
+  // Repeated probe should only be sent when estimated bitrate climbs above
+  // 0.8 * 5 * kStartBitrateBps = 1200.
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1100), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 0u);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(1250), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 3 * 1250);
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      DataRate::KilobitsPerSec(200), fixture.CurrentTime());
+  EXPECT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate.bps(), 400'000);
+}
+
+TEST(ProbeControllerTest, LimitAlrProbeWhenLossBasedBweLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500),
+      BandwidthLimitedCause::kLossLimitedBweIncreasing, fixture.CurrentTime());
+  fixture.AdvanceTime(TimeDelta::Seconds(6));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 1.5 * DataRate::BitsPerSec(500));
+
+  probes = probe_controller->SetEstimatedBitrate(
+      1.5 * DataRate::BitsPerSec(500),
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+  fixture.AdvanceTime(TimeDelta::Seconds(6));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_GT(probes[0].target_data_rate, 1.5 * 1.5 * DataRate::BitsPerSec(500));
+}
+
+TEST(ProbeControllerTest, PeriodicProbeAtUpperNetworkStateEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/network_state_interval:5s/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(5000), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::KilobitsPerSec(6);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+}
+
+TEST(ProbeControllerTest,
+     LimitProbeAtUpperNetworkStateEstimateIfLossBasedLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::BitsPerSec(700);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::BitsPerSec(500),
+      BandwidthLimitedCause::kLossLimitedBweIncreasing, fixture.CurrentTime());
+  // Expect the controller to send a new probe after 5s has passed.
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_EQ(probes[0].target_data_rate, DataRate::BitsPerSec(700));
+}
+
+TEST(ProbeControllerTest, AlrProbesLimitedByNetworkStateEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/network_state_interval:5s/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::KilobitsPerSec(6), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, kMaxBitrate);
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::BitsPerSec(8000);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+}
+
+TEST(ProbeControllerTest, CanSetLongerProbeDurationAfterNetworkStateEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,network_state_probe_duration:100ms/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      DataRate::KilobitsPerSec(5), BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_LT(probes[0].target_duration, TimeDelta::Millis(100));
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = DataRate::KilobitsPerSec(6);
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_duration, TimeDelta::Millis(100));
+}
+
+TEST(ProbeControllerTest, ProbeInAlrIfLossBasedIncreasing) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  // Probe when in alr.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes.at(0).target_data_rate, 1.5 * kStartBitrate);
+}
+
+TEST(ProbeControllerTest, ProbeFurtherInAlrIfLossBasedIncreasing) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  // Probe when in alr.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  ASSERT_EQ(probes.at(0).target_data_rate, 1.5 * kStartBitrate);
+
+  probes = probe_controller->SetEstimatedBitrate(
+      1.5 * kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_EQ(probes[0].target_data_rate, 1.5 * 1.5 * kStartBitrate);
+}
+
+TEST(ProbeControllerTest, NotProbeWhenInAlrIfLossBasedDecreases) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweDecreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  // Not probe in alr when loss based estimate decreases.
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, NotProbeIfLossBasedIncreasingOutsideAlr) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probe_controller->EnablePeriodicAlrProbing(true);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kLossLimitedBweIncreasing,
+      fixture.CurrentTime());
+
+  // Wait long enough to time out exponential probing.
+  fixture.AdvanceTime(kExponentialProbingTimeout);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  probe_controller->SetAlrStartTimeMs(absl::nullopt);
+  fixture.AdvanceTime(kAlrProbeInterval + TimeDelta::Millis(1));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, ProbeFurtherWhenLossBasedIsSameAsDelayBasedEstimate) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 5 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  DataRate probe_target_rate = probes[0].target_data_rate;
+  EXPECT_LT(probe_target_rate, state_estimate.link_capacity_upper);
+  // Expect that more probes are sent if BWE is the same as delay based
+  // estimate.
+  probes = probe_controller->SetEstimatedBitrate(
+      probe_target_rate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_EQ(probes[0].target_data_rate, 2 * probe_target_rate);
+}
+
+TEST(ProbeControllerTest, ProbeIfEstimateLowerThanNetworkStateEstimate) {
+  // Periodic probe every 1 second if estimate is lower than 50% of the
+  // NetworkStateEstimate.
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/est_lower_than_network_interval:1s,"
+      "est_lower_than_network_ratio:0.5,limit_probe_"
+      "target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  state_estimate.link_capacity_upper = kStartBitrate * 3;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_EQ(probes.size(), 1u);
+  EXPECT_GT(probes[0].target_data_rate, kStartBitrate);
+
+  // If network state not increased, send another probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+
+  // Stop probing if estimate increase. We might probe further here though.
+  probes = probe_controller->SetEstimatedBitrate(
+      2 * kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  // No more periodic probes.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, DontProbeFurtherWhenLossLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+  EXPECT_LT(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  // Expect that no more probes are sent immediately if BWE is loss limited.
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate,
+      BandwidthLimitedCause::kLossLimitedBweDecreasing, fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, ProbeFurtherWhenDelayBasedLimited) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+  EXPECT_LT(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  // Since the probe was successfull, expect to continue probing.
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+  EXPECT_EQ(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+}
+
+TEST(ProbeControllerTest,
+     ProbeFurtherIfNetworkStateEstimateIncreaseAfterProbeSent) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 1.2 * probes[0].target_data_rate / 2;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  // No immediate further probing since probe result is low.
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate / 2, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  EXPECT_LE(probes[0].target_data_rate, state_estimate.link_capacity_upper);
+  // If the network state estimate increase above the threshold to probe
+  // further, and the probe suceeed, expect a new probe.
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->SetEstimatedBitrate(
+      probes[0].target_data_rate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+
+  // But no more probes if estimate is close to the link capacity.
+  probes = probe_controller->SetEstimatedBitrate(
+      state_estimate.link_capacity_upper * 0.9,
+      BandwidthLimitedCause::kDelayBasedLimited, fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, SkipAlrProbeIfEstimateLargerThanMaxProbe) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "skip_if_est_larger_than_fraction_of_max:0.9/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // But if the max rate increase, A new probe is sent.
+  probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, 2 * kMaxBitrate, fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+}
+
+TEST(ProbeControllerTest,
+     SkipAlrProbeIfEstimateLargerThanFractionOfMaxAllocated) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "skip_if_est_larger_than_fraction_of_max:1.0/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  probe_controller->EnablePeriodicAlrProbing(true);
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate / 2, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probe_controller->SetAlrStartTimeMs(fixture.CurrentTime().ms());
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(kMaxBitrate / 2,
+                                                        fixture.CurrentTime());
+  // No probes since total allocated is not higher than the current estimate.
+  EXPECT_TRUE(probes.empty());
+  fixture.AdvanceTime(TimeDelta::Seconds(2));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // But if the max allocated increase, A new probe is sent.
+  probes = probe_controller->OnMaxTotalAllocatedBitrate(
+      kMaxBitrate / 2 + DataRate::BitsPerSec(1), fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+}
+
+TEST(ProbeControllerTest, SkipNetworkStateProbeIfEstimateLargerThanMaxProbe) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:2s,skip_if_est_larger_than_fraction_of_max:0.9/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = 2 * kMaxBitrate});
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(10));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, SendsProbeIfNetworkStateEstimateLowerThanMaxProbe) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:2s,skip_if_est_larger_than_fraction_of_max:0.9,"
+      "/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = 2 * kMaxBitrate});
+  probes = probe_controller->SetEstimatedBitrate(
+      kMaxBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+
+  // Need to wait at least two seconds before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(2100));
+
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = 2 * kStartBitrate});
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_FALSE(probes.empty());
+}
+
+TEST(ProbeControllerTest, DontSendProbeIfNetworkStateEstimateIsZero) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,limit_probe_target_rate_to_loss_bwe:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimited,
+      fixture.CurrentTime());
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = kStartBitrate});
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  probe_controller->SetNetworkStateEstimate(
+      {.link_capacity_upper = DataRate::Zero()});
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+  fixture.AdvanceTime(TimeDelta::Seconds(6));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+
+TEST(ProbeControllerTest, DontProbeIfDelayIncreased) {
+  ProbeControllerFixture fixture(
+      "WebRTC-Bwe-ProbingConfiguration/"
+      "network_state_interval:5s,not_probe_if_delay_increased:true/");
+  std::unique_ptr<ProbeController> probe_controller =
+      fixture.CreateController();
+
+  auto probes = probe_controller->SetBitrates(
+      kMinBitrate, kStartBitrate, kMaxBitrate, fixture.CurrentTime());
+  ASSERT_FALSE(probes.empty());
+
+  // Need to wait at least one second before process can trigger a new probe.
+  fixture.AdvanceTime(TimeDelta::Millis(1100));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  NetworkStateEstimate state_estimate;
+  state_estimate.link_capacity_upper = 3 * kStartBitrate;
+  probe_controller->SetNetworkStateEstimate(state_estimate);
+  probes = probe_controller->SetEstimatedBitrate(
+      kStartBitrate, BandwidthLimitedCause::kDelayBasedLimitedDelayIncreased,
+      fixture.CurrentTime());
+  ASSERT_TRUE(probes.empty());
+
+  fixture.AdvanceTime(TimeDelta::Seconds(5));
+  probes = probe_controller->Process(fixture.CurrentTime());
+  EXPECT_TRUE(probes.empty());
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build
new file mode 100644
index 0000000000..e189363d3e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/pushback_controller_gn/moz.build
@@ -0,0 +1,225 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/congestion_window_pushback_controller.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("pushback_controller_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc
new file mode 100644
index 0000000000..792a93d41e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.cc
@@ -0,0 +1,189 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+
+#include <stddef.h>
+
+#include <algorithm>
+#include <utility>
+
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+RobustThroughputEstimator::RobustThroughputEstimator(
+    const RobustThroughputEstimatorSettings& settings)
+    : settings_(settings),
+      latest_discarded_send_time_(Timestamp::MinusInfinity()) {
+  RTC_DCHECK(settings.enabled);
+}
+
+RobustThroughputEstimator::~RobustThroughputEstimator() {}
+
+bool RobustThroughputEstimator::FirstPacketOutsideWindow() {
+  if (window_.empty())
+    return false;
+  if (window_.size() > settings_.max_window_packets)
+    return true;
+  TimeDelta current_window_duration =
+      window_.back().receive_time - window_.front().receive_time;
+  if (current_window_duration > settings_.max_window_duration)
+    return true;
+  if (window_.size() > settings_.window_packets &&
+      current_window_duration > settings_.min_window_duration) {
+    return true;
+  }
+  return false;
+}
+
+void RobustThroughputEstimator::IncomingPacketFeedbackVector(
+    const std::vector<PacketResult>& packet_feedback_vector) {
+  RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
+                            packet_feedback_vector.end(),
+                            PacketResult::ReceiveTimeOrder()));
+  for (const auto& packet : packet_feedback_vector) {
+    // Ignore packets without valid send or receive times.
+    // (This should not happen in production since lost packets are filtered
+    // out before passing the feedback vector to the throughput estimator.
+    // However, explicitly handling this case makes the estimator more robust
+    // and avoids a hard-to-detect bad state.)
+    if (packet.receive_time.IsInfinite() ||
+        packet.sent_packet.send_time.IsInfinite()) {
+      continue;
+    }
+
+    // Insert the new packet.
+    window_.push_back(packet);
+    window_.back().sent_packet.prior_unacked_data =
+        window_.back().sent_packet.prior_unacked_data *
+        settings_.unacked_weight;
+    // In most cases, receive timestamps should already be in order, but in the
+    // rare case where feedback packets have been reordered, we do some swaps to
+    // ensure that the window is sorted.
+    for (size_t i = window_.size() - 1;
+         i > 0 && window_[i].receive_time < window_[i - 1].receive_time; i--) {
+      std::swap(window_[i], window_[i - 1]);
+    }
+  }
+
+  // Remove old packets.
+  while (FirstPacketOutsideWindow()) {
+    latest_discarded_send_time_ = std::max(
+        latest_discarded_send_time_, window_.front().sent_packet.send_time);
+    window_.pop_front();
+  }
+}
+
+absl::optional<DataRate> RobustThroughputEstimator::bitrate() const {
+  if (window_.empty() || window_.size() < settings_.required_packets)
+    return absl::nullopt;
+
+  TimeDelta largest_recv_gap(TimeDelta::Zero());
+  TimeDelta second_largest_recv_gap(TimeDelta::Zero());
+  for (size_t i = 1; i < window_.size(); i++) {
+    // Find receive time gaps.
+    TimeDelta gap = window_[i].receive_time - window_[i - 1].receive_time;
+    if (gap > largest_recv_gap) {
+      second_largest_recv_gap = largest_recv_gap;
+      largest_recv_gap = gap;
+    } else if (gap > second_largest_recv_gap) {
+      second_largest_recv_gap = gap;
+    }
+  }
+
+  Timestamp first_send_time = Timestamp::PlusInfinity();
+  Timestamp last_send_time = Timestamp::MinusInfinity();
+  Timestamp first_recv_time = Timestamp::PlusInfinity();
+  Timestamp last_recv_time = Timestamp::MinusInfinity();
+  DataSize recv_size = DataSize::Bytes(0);
+  DataSize send_size = DataSize::Bytes(0);
+  DataSize first_recv_size = DataSize::Bytes(0);
+  DataSize last_send_size = DataSize::Bytes(0);
+  size_t num_sent_packets_in_window = 0;
+  for (const auto& packet : window_) {
+    if (packet.receive_time < first_recv_time) {
+      first_recv_time = packet.receive_time;
+      first_recv_size =
+          packet.sent_packet.size + packet.sent_packet.prior_unacked_data;
+    }
+    last_recv_time = std::max(last_recv_time, packet.receive_time);
+    recv_size += packet.sent_packet.size;
+    recv_size += packet.sent_packet.prior_unacked_data;
+
+    if (packet.sent_packet.send_time < latest_discarded_send_time_) {
+      // If we have dropped packets from the window that were sent after
+      // this packet, then this packet was reordered. Ignore it from
+      // the send rate computation (since the send time may be very far
+      // in the past, leading to underestimation of the send rate.)
+      // However, ignoring packets creates a risk that we end up without
+      // any packets left to compute a send rate.
+      continue;
+    }
+    if (packet.sent_packet.send_time > last_send_time) {
+      last_send_time = packet.sent_packet.send_time;
+      last_send_size =
+          packet.sent_packet.size + packet.sent_packet.prior_unacked_data;
+    }
+    first_send_time = std::min(first_send_time, packet.sent_packet.send_time);
+
+    send_size += packet.sent_packet.size;
+    send_size += packet.sent_packet.prior_unacked_data;
+    ++num_sent_packets_in_window;
+  }
+
+  // Suppose a packet of size S is sent every T milliseconds.
+  // A window of N packets would contain N*S bytes, but the time difference
+  // between the first and the last packet would only be (N-1)*T. Thus, we
+  // need to remove the size of one packet to get the correct rate of S/T.
+  // Which packet to remove (if the packets have varying sizes),
+  // depends on the network model.
+  // Suppose that 2 packets with sizes s1 and s2, are received at times t1
+  // and t2, respectively. If the packets were transmitted back to back over
+  // a bottleneck with rate capacity r, then we'd expect t2 = t1 + r * s2.
+  // Thus, r = (t2-t1) / s2, so the size of the first packet doesn't affect
+  // the difference between t1 and t2.
+  // Analoguously, if the first packet is sent at time t1 and the sender
+  // paces the packets at rate r, then the second packet can be sent at time
+  // t2 = t1 + r * s1. Thus, the send rate estimate r = (t2-t1) / s1 doesn't
+  // depend on the size of the last packet.
+  recv_size -= first_recv_size;
+  send_size -= last_send_size;
+
+  // Remove the largest gap by replacing it by the second largest gap.
+  // This is to ensure that spurious "delay spikes" (i.e. when the
+  // network stops transmitting packets for a short period, followed
+  // by a burst of delayed packets), don't cause the estimate to drop.
+  // This could cause an overestimation, which we guard against by
+  // never returning an estimate above the send rate.
+  RTC_DCHECK(first_recv_time.IsFinite());
+  RTC_DCHECK(last_recv_time.IsFinite());
+  TimeDelta recv_duration = (last_recv_time - first_recv_time) -
+                            largest_recv_gap + second_largest_recv_gap;
+  recv_duration = std::max(recv_duration, TimeDelta::Millis(1));
+
+  if (num_sent_packets_in_window < settings_.required_packets) {
+    // Too few send times to calculate a reliable send rate.
+    return recv_size / recv_duration;
+  }
+
+  RTC_DCHECK(first_send_time.IsFinite());
+  RTC_DCHECK(last_send_time.IsFinite());
+  TimeDelta send_duration = last_send_time - first_send_time;
+  send_duration = std::max(send_duration, TimeDelta::Millis(1));
+
+  return std::min(send_size / send_duration, recv_size / recv_duration);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.h
new file mode 100644
index 0000000000..9d89856496
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
+
+#include <deque>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
+
+namespace webrtc {
+
+class RobustThroughputEstimator : public AcknowledgedBitrateEstimatorInterface {
+ public:
+  explicit RobustThroughputEstimator(
+      const RobustThroughputEstimatorSettings& settings);
+  ~RobustThroughputEstimator() override;
+
+  void IncomingPacketFeedbackVector(
+      const std::vector<PacketResult>& packet_feedback_vector) override;
+
+  absl::optional<DataRate> bitrate() const override;
+
+  absl::optional<DataRate> PeekRate() const override { return bitrate(); }
+  void SetAlr(bool /*in_alr*/) override {}
+  void SetAlrEndedTime(Timestamp /*alr_ended_time*/) override {}
+
+ private:
+  bool FirstPacketOutsideWindow();
+
+  const RobustThroughputEstimatorSettings settings_;
+  std::deque<PacketResult> window_;
+  Timestamp latest_discarded_send_time_ = Timestamp::MinusInfinity();
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ROBUST_THROUGHPUT_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator_unittest.cc
new file mode 100644
index 0000000000..95ac525640
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/robust_throughput_estimator_unittest.cc
@@ -0,0 +1,427 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/robust_throughput_estimator.h"
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <algorithm>
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+RobustThroughputEstimatorSettings CreateRobustThroughputEstimatorSettings(
+    absl::string_view field_trial_string) {
+  test::ExplicitKeyValueConfig trials(field_trial_string);
+  RobustThroughputEstimatorSettings settings(&trials);
+  return settings;
+}
+
+class FeedbackGenerator {
+ public:
+  std::vector<PacketResult> CreateFeedbackVector(size_t number_of_packets,
+                                                 DataSize packet_size,
+                                                 DataRate send_rate,
+                                                 DataRate recv_rate) {
+    std::vector<PacketResult> packet_feedback_vector(number_of_packets);
+    for (size_t i = 0; i < number_of_packets; i++) {
+      packet_feedback_vector[i].sent_packet.send_time = send_clock_;
+      packet_feedback_vector[i].sent_packet.sequence_number = sequence_number_;
+      packet_feedback_vector[i].sent_packet.size = packet_size;
+      send_clock_ += packet_size / send_rate;
+      recv_clock_ += packet_size / recv_rate;
+      sequence_number_ += 1;
+      packet_feedback_vector[i].receive_time = recv_clock_;
+    }
+    return packet_feedback_vector;
+  }
+
+  Timestamp CurrentReceiveClock() { return recv_clock_; }
+
+  void AdvanceReceiveClock(TimeDelta delta) { recv_clock_ += delta; }
+
+  void AdvanceSendClock(TimeDelta delta) { send_clock_ += delta; }
+
+ private:
+  Timestamp send_clock_ = Timestamp::Millis(100000);
+  Timestamp recv_clock_ = Timestamp::Millis(10000);
+  uint16_t sequence_number_ = 100;
+};
+
+TEST(RobustThroughputEstimatorTest, InitialEstimate) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  // No estimate until the estimator has enough data.
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(9, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  EXPECT_FALSE(throughput_estimator.bitrate().has_value());
+
+  // Estimate once `required_packets` packets have been received.
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      1, DataSize::Bytes(1000), send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  // Estimate remains stable when send and receive rates are stable.
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      15, DataSize::Bytes(1000), send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+}
+
+TEST(RobustThroughputEstimatorTest, EstimateAdapts) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+
+  // 1 second, 800kbps, estimate is stable.
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+  for (int i = 0; i < 10; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+
+  // 1 second, 1600kbps, estimate increases
+  send_rate = DataRate::BytesPerSec(200000);
+  recv_rate = DataRate::BytesPerSec(200000);
+  for (int i = 0; i < 20; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_GE(throughput.value(), DataRate::BytesPerSec(100000));
+    EXPECT_LE(throughput.value(), send_rate);
+  }
+
+  // 1 second, 1600kbps, estimate is stable
+  for (int i = 0; i < 20; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+
+  // 1 second, 400kbps, estimate decreases
+  send_rate = DataRate::BytesPerSec(50000);
+  recv_rate = DataRate::BytesPerSec(50000);
+  for (int i = 0; i < 5; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_LE(throughput.value(), DataRate::BytesPerSec(200000));
+    EXPECT_GE(throughput.value(), send_rate);
+  }
+
+  // 1 second, 400kbps, estimate is stable
+  send_rate = DataRate::BytesPerSec(50000);
+  recv_rate = DataRate::BytesPerSec(50000);
+  for (int i = 0; i < 5; ++i) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, CappedByReceiveRate) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(25000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              recv_rate.bytes_per_sec<double>(),
+              0.05 * recv_rate.bytes_per_sec<double>());  // Allow 5% error
+}
+
+TEST(RobustThroughputEstimatorTest, CappedBySendRate) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(50000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              send_rate.bytes_per_sec<double>(),
+              0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+}
+
+TEST(RobustThroughputEstimatorTest, DelaySpike) {
+  FeedbackGenerator feedback_generator;
+  // This test uses a 500ms window to amplify the effect
+  // of a delay spike.
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true,window_duration:500ms/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  // Delay spike. 25 packets sent, but none received.
+  feedback_generator.AdvanceReceiveClock(TimeDelta::Millis(250));
+
+  // Deliver all of the packets during the next 50 ms. (During this time,
+  // we'll have sent an additional 5 packets, so we need to receive 30
+  // packets at 1000 bytes each in 50 ms, i.e. 600000 bytes per second).
+  recv_rate = DataRate::BytesPerSec(600000);
+  // Estimate should not drop.
+  for (int i = 0; i < 30; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        1, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+                send_rate.bytes_per_sec<double>(),
+                0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+  }
+
+  // Delivery at normal rate. When the packets received before the gap
+  // has left the estimator's window, the receive rate will be high, but the
+  // estimate should be capped by the send rate.
+  recv_rate = DataRate::BytesPerSec(100000);
+  for (int i = 0; i < 20; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        5, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+                send_rate.bytes_per_sec<double>(),
+                0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, HighLoss) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+
+  // 50% loss
+  for (size_t i = 0; i < packet_feedback.size(); i++) {
+    if (i % 2 == 1) {
+      packet_feedback[i].receive_time = Timestamp::PlusInfinity();
+    }
+  }
+
+  std::sort(packet_feedback.begin(), packet_feedback.end(),
+            PacketResult::ReceiveTimeOrder());
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              send_rate.bytes_per_sec<double>() / 2,
+              0.05 * send_rate.bytes_per_sec<double>() / 2);  // Allow 5% error
+}
+
+TEST(RobustThroughputEstimatorTest, ReorderedFeedback) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  std::vector<PacketResult> delayed_feedback =
+      feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      10, DataSize::Bytes(1000), send_rate, recv_rate);
+
+  // Since we're missing some feedback, it's expected that the
+  // estimate will drop.
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_LT(throughput.value(), send_rate);
+
+  // But it should completely recover as soon as we get the feedback.
+  throughput_estimator.IncomingPacketFeedbackVector(delayed_feedback);
+  throughput = throughput_estimator.bitrate();
+  EXPECT_EQ(throughput, send_rate);
+
+  // It should then remain stable (as if the feedbacks weren't reordered.)
+  for (int i = 0; i < 10; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        15, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, DeepReordering) {
+  FeedbackGenerator feedback_generator;
+  // This test uses a 500ms window to amplify the
+  // effect of reordering.
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true,window_duration:500ms/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> delayed_packets =
+      feedback_generator.CreateFeedbackVector(1, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+
+  for (int i = 0; i < 10; i++) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+
+  // Delayed packet arrives ~1 second after it should have.
+  // Since the window is 500 ms, the delayed packet was sent ~500
+  // ms before the second oldest packet. However, the send rate
+  // should not drop.
+  delayed_packets.front().receive_time =
+      feedback_generator.CurrentReceiveClock();
+  throughput_estimator.IncomingPacketFeedbackVector(delayed_packets);
+  auto throughput = throughput_estimator.bitrate();
+  ASSERT_TRUE(throughput.has_value());
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              send_rate.bytes_per_sec<double>(),
+              0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+
+  // Thoughput should stay stable.
+  for (int i = 0; i < 10; i++) {
+    std::vector<PacketResult> packet_feedback =
+        feedback_generator.CreateFeedbackVector(10, DataSize::Bytes(1000),
+                                                send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    auto throughput = throughput_estimator.bitrate();
+    ASSERT_TRUE(throughput.has_value());
+    EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+                send_rate.bytes_per_sec<double>(),
+                0.05 * send_rate.bytes_per_sec<double>());  // Allow 5% error
+  }
+}
+
+TEST(RobustThroughputEstimatorTest, StreamPausedAndResumed) {
+  FeedbackGenerator feedback_generator;
+  RobustThroughputEstimator throughput_estimator(
+      CreateRobustThroughputEstimatorSettings(
+          "WebRTC-Bwe-RobustThroughputEstimatorSettings/"
+          "enabled:true/"));
+  DataRate send_rate(DataRate::BytesPerSec(100000));
+  DataRate recv_rate(DataRate::BytesPerSec(100000));
+
+  std::vector<PacketResult> packet_feedback =
+      feedback_generator.CreateFeedbackVector(20, DataSize::Bytes(1000),
+                                              send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  auto throughput = throughput_estimator.bitrate();
+  EXPECT_TRUE(throughput.has_value());
+  double expected_bytes_per_sec = 100 * 1000.0;
+  EXPECT_NEAR(throughput.value().bytes_per_sec<double>(),
+              expected_bytes_per_sec,
+              0.05 * expected_bytes_per_sec);  // Allow 5% error
+
+  // No packets sent or feedback received for 60s.
+  feedback_generator.AdvanceSendClock(TimeDelta::Seconds(60));
+  feedback_generator.AdvanceReceiveClock(TimeDelta::Seconds(60));
+
+  // Resume sending packets at the same rate as before. The estimate
+  // will initially be invalid, due to lack of recent data.
+  packet_feedback = feedback_generator.CreateFeedbackVector(
+      5, DataSize::Bytes(1000), send_rate, recv_rate);
+  throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+  throughput = throughput_estimator.bitrate();
+  EXPECT_FALSE(throughput.has_value());
+
+  // But be back to the normal level once we have enough data.
+  for (int i = 0; i < 4; ++i) {
+    packet_feedback = feedback_generator.CreateFeedbackVector(
+        5, DataSize::Bytes(1000), send_rate, recv_rate);
+    throughput_estimator.IncomingPacketFeedbackVector(packet_feedback);
+    throughput = throughput_estimator.bitrate();
+    EXPECT_EQ(throughput, send_rate);
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
new file mode 100644
index 0000000000..1e4db1ffaf
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
@@ -0,0 +1,695 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+
+#include <algorithm>
+#include <cstdio>
+#include <limits>
+#include <memory>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/rtc_event_log/rtc_event.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kBweIncreaseInterval = TimeDelta::Millis(1000);
+constexpr TimeDelta kBweDecreaseInterval = TimeDelta::Millis(300);
+constexpr TimeDelta kStartPhase = TimeDelta::Millis(2000);
+constexpr TimeDelta kBweConverganceTime = TimeDelta::Millis(20000);
+constexpr int kLimitNumPackets = 20;
+constexpr DataRate kDefaultMaxBitrate = DataRate::BitsPerSec(1000000000);
+constexpr TimeDelta kLowBitrateLogPeriod = TimeDelta::Millis(10000);
+constexpr TimeDelta kRtcEventLogPeriod = TimeDelta::Millis(5000);
+// Expecting that RTCP feedback is sent uniformly within [0.5, 1.5]s intervals.
+constexpr TimeDelta kMaxRtcpFeedbackInterval = TimeDelta::Millis(5000);
+
+constexpr float kDefaultLowLossThreshold = 0.02f;
+constexpr float kDefaultHighLossThreshold = 0.1f;
+constexpr DataRate kDefaultBitrateThreshold = DataRate::Zero();
+
+struct UmaRampUpMetric {
+  const char* metric_name;
+  int bitrate_kbps;
+};
+
+const UmaRampUpMetric kUmaRampupMetrics[] = {
+    {"WebRTC.BWE.RampUpTimeTo500kbpsInMs", 500},
+    {"WebRTC.BWE.RampUpTimeTo1000kbpsInMs", 1000},
+    {"WebRTC.BWE.RampUpTimeTo2000kbpsInMs", 2000}};
+const size_t kNumUmaRampupMetrics =
+    sizeof(kUmaRampupMetrics) / sizeof(kUmaRampupMetrics[0]);
+
+const char kBweLosExperiment[] = "WebRTC-BweLossExperiment";
+
+bool BweLossExperimentIsEnabled() {
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kBweLosExperiment);
+  // The experiment is enabled iff the field trial string begins with "Enabled".
+  return absl::StartsWith(experiment_string, "Enabled");
+}
+
+bool ReadBweLossExperimentParameters(float* low_loss_threshold,
+                                     float* high_loss_threshold,
+                                     uint32_t* bitrate_threshold_kbps) {
+  RTC_DCHECK(low_loss_threshold);
+  RTC_DCHECK(high_loss_threshold);
+  RTC_DCHECK(bitrate_threshold_kbps);
+  std::string experiment_string =
+      webrtc::field_trial::FindFullName(kBweLosExperiment);
+  int parsed_values =
+      sscanf(experiment_string.c_str(), "Enabled-%f,%f,%u", low_loss_threshold,
+             high_loss_threshold, bitrate_threshold_kbps);
+  if (parsed_values == 3) {
+    RTC_CHECK_GT(*low_loss_threshold, 0.0f)
+        << "Loss threshold must be greater than 0.";
+    RTC_CHECK_LE(*low_loss_threshold, 1.0f)
+        << "Loss threshold must be less than or equal to 1.";
+    RTC_CHECK_GT(*high_loss_threshold, 0.0f)
+        << "Loss threshold must be greater than 0.";
+    RTC_CHECK_LE(*high_loss_threshold, 1.0f)
+        << "Loss threshold must be less than or equal to 1.";
+    RTC_CHECK_LE(*low_loss_threshold, *high_loss_threshold)
+        << "The low loss threshold must be less than or equal to the high loss "
+           "threshold.";
+    RTC_CHECK_GE(*bitrate_threshold_kbps, 0)
+        << "Bitrate threshold can't be negative.";
+    RTC_CHECK_LT(*bitrate_threshold_kbps,
+                 std::numeric_limits<int>::max() / 1000)
+        << "Bitrate must be smaller enough to avoid overflows.";
+    return true;
+  }
+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweLossExperiment "
+                         "experiment from field trial string. Using default.";
+  *low_loss_threshold = kDefaultLowLossThreshold;
+  *high_loss_threshold = kDefaultHighLossThreshold;
+  *bitrate_threshold_kbps = kDefaultBitrateThreshold.kbps();
+  return false;
+}
+}  // namespace
+
+LinkCapacityTracker::LinkCapacityTracker()
+    : tracking_rate("rate", TimeDelta::Seconds(10)) {
+  ParseFieldTrial({&tracking_rate},
+                  field_trial::FindFullName("WebRTC-Bwe-LinkCapacity"));
+}
+
+LinkCapacityTracker::~LinkCapacityTracker() {}
+
+void LinkCapacityTracker::UpdateDelayBasedEstimate(
+    Timestamp at_time,
+    DataRate delay_based_bitrate) {
+  if (delay_based_bitrate < last_delay_based_estimate_) {
+    capacity_estimate_bps_ =
+        std::min(capacity_estimate_bps_, delay_based_bitrate.bps<double>());
+    last_link_capacity_update_ = at_time;
+  }
+  last_delay_based_estimate_ = delay_based_bitrate;
+}
+
+void LinkCapacityTracker::OnStartingRate(DataRate start_rate) {
+  if (last_link_capacity_update_.IsInfinite())
+    capacity_estimate_bps_ = start_rate.bps<double>();
+}
+
+void LinkCapacityTracker::OnRateUpdate(absl::optional<DataRate> acknowledged,
+                                       DataRate target,
+                                       Timestamp at_time) {
+  if (!acknowledged)
+    return;
+  DataRate acknowledged_target = std::min(*acknowledged, target);
+  if (acknowledged_target.bps() > capacity_estimate_bps_) {
+    TimeDelta delta = at_time - last_link_capacity_update_;
+    double alpha = delta.IsFinite() ? exp(-(delta / tracking_rate.Get())) : 0;
+    capacity_estimate_bps_ = alpha * capacity_estimate_bps_ +
+                             (1 - alpha) * acknowledged_target.bps<double>();
+  }
+  last_link_capacity_update_ = at_time;
+}
+
+void LinkCapacityTracker::OnRttBackoff(DataRate backoff_rate,
+                                       Timestamp at_time) {
+  capacity_estimate_bps_ =
+      std::min(capacity_estimate_bps_, backoff_rate.bps<double>());
+  last_link_capacity_update_ = at_time;
+}
+
+DataRate LinkCapacityTracker::estimate() const {
+  return DataRate::BitsPerSec(capacity_estimate_bps_);
+}
+
+RttBasedBackoff::RttBasedBackoff(const FieldTrialsView* key_value_config)
+    : disabled_("Disabled"),
+      configured_limit_("limit", TimeDelta::Seconds(3)),
+      drop_fraction_("fraction", 0.8),
+      drop_interval_("interval", TimeDelta::Seconds(1)),
+      bandwidth_floor_("floor", DataRate::KilobitsPerSec(5)),
+      rtt_limit_(TimeDelta::PlusInfinity()),
+      // By initializing this to plus infinity, we make sure that we never
+      // trigger rtt backoff unless packet feedback is enabled.
+      last_propagation_rtt_update_(Timestamp::PlusInfinity()),
+      last_propagation_rtt_(TimeDelta::Zero()),
+      last_packet_sent_(Timestamp::MinusInfinity()) {
+  ParseFieldTrial({&disabled_, &configured_limit_, &drop_fraction_,
+                   &drop_interval_, &bandwidth_floor_},
+                  key_value_config->Lookup("WebRTC-Bwe-MaxRttLimit"));
+  if (!disabled_) {
+    rtt_limit_ = configured_limit_.Get();
+  }
+}
+
+void RttBasedBackoff::UpdatePropagationRtt(Timestamp at_time,
+                                           TimeDelta propagation_rtt) {
+  last_propagation_rtt_update_ = at_time;
+  last_propagation_rtt_ = propagation_rtt;
+}
+
+TimeDelta RttBasedBackoff::CorrectedRtt(Timestamp at_time) const {
+  TimeDelta time_since_rtt = at_time - last_propagation_rtt_update_;
+  TimeDelta timeout_correction = time_since_rtt;
+  // Avoid timeout when no packets are being sent.
+  TimeDelta time_since_packet_sent = at_time - last_packet_sent_;
+  timeout_correction =
+      std::max(time_since_rtt - time_since_packet_sent, TimeDelta::Zero());
+  return timeout_correction + last_propagation_rtt_;
+}
+
+RttBasedBackoff::~RttBasedBackoff() = default;
+
+SendSideBandwidthEstimation::SendSideBandwidthEstimation(
+    const FieldTrialsView* key_value_config,
+    RtcEventLog* event_log)
+    : rtt_backoff_(key_value_config),
+      lost_packets_since_last_loss_update_(0),
+      expected_packets_since_last_loss_update_(0),
+      current_target_(DataRate::Zero()),
+      last_logged_target_(DataRate::Zero()),
+      min_bitrate_configured_(kCongestionControllerMinBitrate),
+      max_bitrate_configured_(kDefaultMaxBitrate),
+      last_low_bitrate_log_(Timestamp::MinusInfinity()),
+      has_decreased_since_last_fraction_loss_(false),
+      last_loss_feedback_(Timestamp::MinusInfinity()),
+      last_loss_packet_report_(Timestamp::MinusInfinity()),
+      last_fraction_loss_(0),
+      last_logged_fraction_loss_(0),
+      last_round_trip_time_(TimeDelta::Zero()),
+      receiver_limit_(DataRate::PlusInfinity()),
+      delay_based_limit_(DataRate::PlusInfinity()),
+      time_last_decrease_(Timestamp::MinusInfinity()),
+      first_report_time_(Timestamp::MinusInfinity()),
+      initially_lost_packets_(0),
+      bitrate_at_2_seconds_(DataRate::Zero()),
+      uma_update_state_(kNoUpdate),
+      uma_rtt_state_(kNoUpdate),
+      rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
+      event_log_(event_log),
+      last_rtc_event_log_(Timestamp::MinusInfinity()),
+      low_loss_threshold_(kDefaultLowLossThreshold),
+      high_loss_threshold_(kDefaultHighLossThreshold),
+      bitrate_threshold_(kDefaultBitrateThreshold),
+      loss_based_bandwidth_estimator_v1_(key_value_config),
+      loss_based_bandwidth_estimator_v2_(key_value_config),
+      loss_based_state_(LossBasedState::kDelayBasedEstimate),
+      disable_receiver_limit_caps_only_("Disabled") {
+  RTC_DCHECK(event_log);
+  if (BweLossExperimentIsEnabled()) {
+    uint32_t bitrate_threshold_kbps;
+    if (ReadBweLossExperimentParameters(&low_loss_threshold_,
+                                        &high_loss_threshold_,
+                                        &bitrate_threshold_kbps)) {
+      RTC_LOG(LS_INFO) << "Enabled BweLossExperiment with parameters "
+                       << low_loss_threshold_ << ", " << high_loss_threshold_
+                       << ", " << bitrate_threshold_kbps;
+      bitrate_threshold_ = DataRate::KilobitsPerSec(bitrate_threshold_kbps);
+    }
+  }
+  ParseFieldTrial({&disable_receiver_limit_caps_only_},
+                  key_value_config->Lookup("WebRTC-Bwe-ReceiverLimitCapsOnly"));
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate(
+        min_bitrate_configured_, max_bitrate_configured_);
+  }
+}
+
+SendSideBandwidthEstimation::~SendSideBandwidthEstimation() {}
+
+void SendSideBandwidthEstimation::OnRouteChange() {
+  lost_packets_since_last_loss_update_ = 0;
+  expected_packets_since_last_loss_update_ = 0;
+  current_target_ = DataRate::Zero();
+  min_bitrate_configured_ = kCongestionControllerMinBitrate;
+  max_bitrate_configured_ = kDefaultMaxBitrate;
+  last_low_bitrate_log_ = Timestamp::MinusInfinity();
+  has_decreased_since_last_fraction_loss_ = false;
+  last_loss_feedback_ = Timestamp::MinusInfinity();
+  last_loss_packet_report_ = Timestamp::MinusInfinity();
+  last_fraction_loss_ = 0;
+  last_logged_fraction_loss_ = 0;
+  last_round_trip_time_ = TimeDelta::Zero();
+  receiver_limit_ = DataRate::PlusInfinity();
+  delay_based_limit_ = DataRate::PlusInfinity();
+  time_last_decrease_ = Timestamp::MinusInfinity();
+  first_report_time_ = Timestamp::MinusInfinity();
+  initially_lost_packets_ = 0;
+  bitrate_at_2_seconds_ = DataRate::Zero();
+  uma_update_state_ = kNoUpdate;
+  uma_rtt_state_ = kNoUpdate;
+  last_rtc_event_log_ = Timestamp::MinusInfinity();
+}
+
+void SendSideBandwidthEstimation::SetBitrates(
+    absl::optional<DataRate> send_bitrate,
+    DataRate min_bitrate,
+    DataRate max_bitrate,
+    Timestamp at_time) {
+  SetMinMaxBitrate(min_bitrate, max_bitrate);
+  if (send_bitrate) {
+    link_capacity_.OnStartingRate(*send_bitrate);
+    SetSendBitrate(*send_bitrate, at_time);
+  }
+}
+
+void SendSideBandwidthEstimation::SetSendBitrate(DataRate bitrate,
+                                                 Timestamp at_time) {
+  RTC_DCHECK_GT(bitrate, DataRate::Zero());
+  // Reset to avoid being capped by the estimate.
+  delay_based_limit_ = DataRate::PlusInfinity();
+  UpdateTargetBitrate(bitrate, at_time);
+  // Clear last sent bitrate history so the new value can be used directly
+  // and not capped.
+  min_bitrate_history_.clear();
+}
+
+void SendSideBandwidthEstimation::SetMinMaxBitrate(DataRate min_bitrate,
+                                                   DataRate max_bitrate) {
+  min_bitrate_configured_ =
+      std::max(min_bitrate, kCongestionControllerMinBitrate);
+  if (max_bitrate > DataRate::Zero() && max_bitrate.IsFinite()) {
+    max_bitrate_configured_ = std::max(min_bitrate_configured_, max_bitrate);
+  } else {
+    max_bitrate_configured_ = kDefaultMaxBitrate;
+  }
+  loss_based_bandwidth_estimator_v2_.SetMinMaxBitrate(min_bitrate_configured_,
+                                                      max_bitrate_configured_);
+}
+
+int SendSideBandwidthEstimation::GetMinBitrate() const {
+  return min_bitrate_configured_.bps<int>();
+}
+
+DataRate SendSideBandwidthEstimation::target_rate() const {
+  DataRate target = current_target_;
+  if (!disable_receiver_limit_caps_only_)
+    target = std::min(target, receiver_limit_);
+  return std::max(min_bitrate_configured_, target);
+}
+
+LossBasedState SendSideBandwidthEstimation::loss_based_state() const {
+  return loss_based_state_;
+}
+
+DataRate SendSideBandwidthEstimation::GetEstimatedLinkCapacity() const {
+  return link_capacity_.estimate();
+}
+
+void SendSideBandwidthEstimation::UpdateReceiverEstimate(Timestamp at_time,
+                                                         DataRate bandwidth) {
+  // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no
+  // limitation.
+  receiver_limit_ = bandwidth.IsZero() ? DataRate::PlusInfinity() : bandwidth;
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(Timestamp at_time,
+                                                           DataRate bitrate) {
+  link_capacity_.UpdateDelayBasedEstimate(at_time, bitrate);
+  // TODO(srte): Ensure caller passes PlusInfinity, not zero, to represent no
+  // limitation.
+  delay_based_limit_ = bitrate.IsZero() ? DataRate::PlusInfinity() : bitrate;
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::SetAcknowledgedRate(
+    absl::optional<DataRate> acknowledged_rate,
+    Timestamp at_time) {
+  acknowledged_rate_ = acknowledged_rate;
+  if (!acknowledged_rate.has_value()) {
+    return;
+  }
+  if (LossBasedBandwidthEstimatorV1Enabled()) {
+    loss_based_bandwidth_estimator_v1_.UpdateAcknowledgedBitrate(
+        *acknowledged_rate, at_time);
+  }
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_.SetAcknowledgedBitrate(
+        *acknowledged_rate);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateLossBasedEstimator(
+    const TransportPacketsFeedback& report,
+    BandwidthUsage delay_detector_state,
+    absl::optional<DataRate> probe_bitrate,
+    DataRate upper_link_capacity) {
+  if (LossBasedBandwidthEstimatorV1Enabled()) {
+    loss_based_bandwidth_estimator_v1_.UpdateLossStatistics(
+        report.packet_feedbacks, report.feedback_time);
+  }
+  if (LossBasedBandwidthEstimatorV2Enabled()) {
+    loss_based_bandwidth_estimator_v2_.UpdateBandwidthEstimate(
+        report.packet_feedbacks, delay_based_limit_, delay_detector_state,
+        probe_bitrate, upper_link_capacity);
+    UpdateEstimate(report.feedback_time);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdatePacketsLost(int64_t packets_lost,
+                                                    int64_t number_of_packets,
+                                                    Timestamp at_time) {
+  last_loss_feedback_ = at_time;
+  if (first_report_time_.IsInfinite())
+    first_report_time_ = at_time;
+
+  // Check sequence number diff and weight loss report
+  if (number_of_packets > 0) {
+    int64_t expected =
+        expected_packets_since_last_loss_update_ + number_of_packets;
+
+    // Don't generate a loss rate until it can be based on enough packets.
+    if (expected < kLimitNumPackets) {
+      // Accumulate reports.
+      expected_packets_since_last_loss_update_ = expected;
+      lost_packets_since_last_loss_update_ += packets_lost;
+      return;
+    }
+
+    has_decreased_since_last_fraction_loss_ = false;
+    int64_t lost_q8 =
+        std::max<int64_t>(lost_packets_since_last_loss_update_ + packets_lost,
+                          0)
+        << 8;
+    last_fraction_loss_ = std::min<int>(lost_q8 / expected, 255);
+
+    // Reset accumulators.
+    lost_packets_since_last_loss_update_ = 0;
+    expected_packets_since_last_loss_update_ = 0;
+    last_loss_packet_report_ = at_time;
+    UpdateEstimate(at_time);
+  }
+
+  UpdateUmaStatsPacketsLost(at_time, packets_lost);
+}
+
+void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(Timestamp at_time,
+                                                            int packets_lost) {
+  DataRate bitrate_kbps =
+      DataRate::KilobitsPerSec((current_target_.bps() + 500) / 1000);
+  for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
+    if (!rampup_uma_stats_updated_[i] &&
+        bitrate_kbps.kbps() >= kUmaRampupMetrics[i].bitrate_kbps) {
+      RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
+                                   (at_time - first_report_time_).ms());
+      rampup_uma_stats_updated_[i] = true;
+    }
+  }
+  if (IsInStartPhase(at_time)) {
+    initially_lost_packets_ += packets_lost;
+  } else if (uma_update_state_ == kNoUpdate) {
+    uma_update_state_ = kFirstDone;
+    bitrate_at_2_seconds_ = bitrate_kbps;
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
+                         initially_lost_packets_, 0, 100, 50);
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
+                         bitrate_at_2_seconds_.kbps(), 0, 2000, 50);
+  } else if (uma_update_state_ == kFirstDone &&
+             at_time - first_report_time_ >= kBweConverganceTime) {
+    uma_update_state_ = kDone;
+    int bitrate_diff_kbps = std::max(
+        bitrate_at_2_seconds_.kbps<int>() - bitrate_kbps.kbps<int>(), 0);
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps,
+                         0, 2000, 50);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateRtt(TimeDelta rtt, Timestamp at_time) {
+  // Update RTT if we were able to compute an RTT based on this RTCP.
+  // FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
+  if (rtt > TimeDelta::Zero())
+    last_round_trip_time_ = rtt;
+
+  if (!IsInStartPhase(at_time) && uma_rtt_state_ == kNoUpdate) {
+    uma_rtt_state_ = kDone;
+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", rtt.ms<int>(), 0, 2000, 50);
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateEstimate(Timestamp at_time) {
+  if (rtt_backoff_.CorrectedRtt(at_time) > rtt_backoff_.rtt_limit_) {
+    if (at_time - time_last_decrease_ >= rtt_backoff_.drop_interval_ &&
+        current_target_ > rtt_backoff_.bandwidth_floor_) {
+      time_last_decrease_ = at_time;
+      DataRate new_bitrate =
+          std::max(current_target_ * rtt_backoff_.drop_fraction_,
+                   rtt_backoff_.bandwidth_floor_.Get());
+      link_capacity_.OnRttBackoff(new_bitrate, at_time);
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    }
+    // TODO(srte): This is likely redundant in most cases.
+    ApplyTargetLimits(at_time);
+    return;
+  }
+
+  // We trust the REMB and/or delay-based estimate during the first 2 seconds if
+  // we haven't had any packet loss reported, to allow startup bitrate probing.
+  if (last_fraction_loss_ == 0 && IsInStartPhase(at_time)) {
+    DataRate new_bitrate = current_target_;
+    // TODO(srte): We should not allow the new_bitrate to be larger than the
+    // receiver limit here.
+    if (receiver_limit_.IsFinite())
+      new_bitrate = std::max(receiver_limit_, new_bitrate);
+    if (delay_based_limit_.IsFinite())
+      new_bitrate = std::max(delay_based_limit_, new_bitrate);
+    if (LossBasedBandwidthEstimatorV1Enabled()) {
+      loss_based_bandwidth_estimator_v1_.Initialize(new_bitrate);
+    }
+    if (LossBasedBandwidthEstimatorV2Enabled()) {
+      loss_based_bandwidth_estimator_v2_.SetBandwidthEstimate(new_bitrate);
+    }
+
+    if (new_bitrate != current_target_) {
+      min_bitrate_history_.clear();
+      if (LossBasedBandwidthEstimatorV1Enabled()) {
+        min_bitrate_history_.push_back(std::make_pair(at_time, new_bitrate));
+      } else {
+        min_bitrate_history_.push_back(
+            std::make_pair(at_time, current_target_));
+      }
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    }
+  }
+  UpdateMinHistory(at_time);
+  if (last_loss_packet_report_.IsInfinite()) {
+    // No feedback received.
+    // TODO(srte): This is likely redundant in most cases.
+    ApplyTargetLimits(at_time);
+    return;
+  }
+
+  if (LossBasedBandwidthEstimatorV1ReadyForUse()) {
+    DataRate new_bitrate = loss_based_bandwidth_estimator_v1_.Update(
+        at_time, min_bitrate_history_.front().second, delay_based_limit_,
+        last_round_trip_time_);
+    UpdateTargetBitrate(new_bitrate, at_time);
+    return;
+  }
+
+  if (LossBasedBandwidthEstimatorV2ReadyForUse()) {
+    LossBasedBweV2::Result result =
+        loss_based_bandwidth_estimator_v2_.GetLossBasedResult();
+    loss_based_state_ = result.state;
+    UpdateTargetBitrate(result.bandwidth_estimate, at_time);
+    return;
+  }
+
+  TimeDelta time_since_loss_packet_report = at_time - last_loss_packet_report_;
+  if (time_since_loss_packet_report < 1.2 * kMaxRtcpFeedbackInterval) {
+    // We only care about loss above a given bitrate threshold.
+    float loss = last_fraction_loss_ / 256.0f;
+    // We only make decisions based on loss when the bitrate is above a
+    // threshold. This is a crude way of handling loss which is uncorrelated
+    // to congestion.
+    if (current_target_ < bitrate_threshold_ || loss <= low_loss_threshold_) {
+      // Loss < 2%: Increase rate by 8% of the min bitrate in the last
+      // kBweIncreaseInterval.
+      // Note that by remembering the bitrate over the last second one can
+      // rampup up one second faster than if only allowed to start ramping
+      // at 8% per second rate now. E.g.:
+      //   If sending a constant 100kbps it can rampup immediately to 108kbps
+      //   whenever a receiver report is received with lower packet loss.
+      //   If instead one would do: current_bitrate_ *= 1.08^(delta time),
+      //   it would take over one second since the lower packet loss to achieve
+      //   108kbps.
+      DataRate new_bitrate = DataRate::BitsPerSec(
+          min_bitrate_history_.front().second.bps() * 1.08 + 0.5);
+
+      // Add 1 kbps extra, just to make sure that we do not get stuck
+      // (gives a little extra increase at low rates, negligible at higher
+      // rates).
+      new_bitrate += DataRate::BitsPerSec(1000);
+      UpdateTargetBitrate(new_bitrate, at_time);
+      return;
+    } else if (current_target_ > bitrate_threshold_) {
+      if (loss <= high_loss_threshold_) {
+        // Loss between 2% - 10%: Do nothing.
+      } else {
+        // Loss > 10%: Limit the rate decreases to once a kBweDecreaseInterval
+        // + rtt.
+        if (!has_decreased_since_last_fraction_loss_ &&
+            (at_time - time_last_decrease_) >=
+                (kBweDecreaseInterval + last_round_trip_time_)) {
+          time_last_decrease_ = at_time;
+
+          // Reduce rate:
+          //   newRate = rate * (1 - 0.5*lossRate);
+          //   where packetLoss = 256*lossRate;
+          DataRate new_bitrate = DataRate::BitsPerSec(
+              (current_target_.bps() *
+               static_cast<double>(512 - last_fraction_loss_)) /
+              512.0);
+          has_decreased_since_last_fraction_loss_ = true;
+          UpdateTargetBitrate(new_bitrate, at_time);
+          return;
+        }
+      }
+    }
+  }
+  // TODO(srte): This is likely redundant in most cases.
+  ApplyTargetLimits(at_time);
+}
+
+void SendSideBandwidthEstimation::UpdatePropagationRtt(
+    Timestamp at_time,
+    TimeDelta propagation_rtt) {
+  rtt_backoff_.UpdatePropagationRtt(at_time, propagation_rtt);
+}
+
+void SendSideBandwidthEstimation::OnSentPacket(const SentPacket& sent_packet) {
+  // Only feedback-triggering packets will be reported here.
+  rtt_backoff_.last_packet_sent_ = sent_packet.send_time;
+}
+
+bool SendSideBandwidthEstimation::IsInStartPhase(Timestamp at_time) const {
+  return first_report_time_.IsInfinite() ||
+         at_time - first_report_time_ < kStartPhase;
+}
+
+void SendSideBandwidthEstimation::UpdateMinHistory(Timestamp at_time) {
+  // Remove old data points from history.
+  // Since history precision is in ms, add one so it is able to increase
+  // bitrate if it is off by as little as 0.5ms.
+  while (!min_bitrate_history_.empty() &&
+         at_time - min_bitrate_history_.front().first + TimeDelta::Millis(1) >
+             kBweIncreaseInterval) {
+    min_bitrate_history_.pop_front();
+  }
+
+  // Typical minimum sliding-window algorithm: Pop values higher than current
+  // bitrate before pushing it.
+  while (!min_bitrate_history_.empty() &&
+         current_target_ <= min_bitrate_history_.back().second) {
+    min_bitrate_history_.pop_back();
+  }
+
+  min_bitrate_history_.push_back(std::make_pair(at_time, current_target_));
+}
+
+DataRate SendSideBandwidthEstimation::GetUpperLimit() const {
+  DataRate upper_limit = delay_based_limit_;
+  if (disable_receiver_limit_caps_only_)
+    upper_limit = std::min(upper_limit, receiver_limit_);
+  return std::min(upper_limit, max_bitrate_configured_);
+}
+
+void SendSideBandwidthEstimation::MaybeLogLowBitrateWarning(DataRate bitrate,
+                                                            Timestamp at_time) {
+  if (at_time - last_low_bitrate_log_ > kLowBitrateLogPeriod) {
+    RTC_LOG(LS_WARNING) << "Estimated available bandwidth " << ToString(bitrate)
+                        << " is below configured min bitrate "
+                        << ToString(min_bitrate_configured_) << ".";
+    last_low_bitrate_log_ = at_time;
+  }
+}
+
+void SendSideBandwidthEstimation::MaybeLogLossBasedEvent(Timestamp at_time) {
+  if (current_target_ != last_logged_target_ ||
+      last_fraction_loss_ != last_logged_fraction_loss_ ||
+      at_time - last_rtc_event_log_ > kRtcEventLogPeriod) {
+    event_log_->Log(std::make_unique<RtcEventBweUpdateLossBased>(
+        current_target_.bps(), last_fraction_loss_,
+        expected_packets_since_last_loss_update_));
+    last_logged_fraction_loss_ = last_fraction_loss_;
+    last_logged_target_ = current_target_;
+    last_rtc_event_log_ = at_time;
+  }
+}
+
+void SendSideBandwidthEstimation::UpdateTargetBitrate(DataRate new_bitrate,
+                                                      Timestamp at_time) {
+  new_bitrate = std::min(new_bitrate, GetUpperLimit());
+  if (new_bitrate < min_bitrate_configured_) {
+    MaybeLogLowBitrateWarning(new_bitrate, at_time);
+    new_bitrate = min_bitrate_configured_;
+  }
+  current_target_ = new_bitrate;
+  MaybeLogLossBasedEvent(at_time);
+  link_capacity_.OnRateUpdate(acknowledged_rate_, current_target_, at_time);
+}
+
+void SendSideBandwidthEstimation::ApplyTargetLimits(Timestamp at_time) {
+  UpdateTargetBitrate(current_target_, at_time);
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1Enabled() const {
+  return loss_based_bandwidth_estimator_v1_.Enabled() &&
+         !LossBasedBandwidthEstimatorV2Enabled();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV1ReadyForUse()
+    const {
+  return LossBasedBandwidthEstimatorV1Enabled() &&
+         loss_based_bandwidth_estimator_v1_.InUse();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2Enabled() const {
+  return loss_based_bandwidth_estimator_v2_.IsEnabled();
+}
+
+bool SendSideBandwidthEstimation::LossBasedBandwidthEstimatorV2ReadyForUse()
+    const {
+  return LossBasedBandwidthEstimatorV2Enabled() &&
+         loss_based_bandwidth_estimator_v2_.IsReady();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
new file mode 100644
index 0000000000..77510236d3
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h
@@ -0,0 +1,210 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ *  FEC and NACK added bitrate is handled outside class
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
+
+#include <stdint.h>
+
+#include <deque>
+#include <utility>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bandwidth_estimation.h"
+#include "modules/congestion_controller/goog_cc/loss_based_bwe_v2.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+
+namespace webrtc {
+
+class RtcEventLog;
+
+class LinkCapacityTracker {
+ public:
+  LinkCapacityTracker();
+  ~LinkCapacityTracker();
+  // Call when a new delay-based estimate is available.
+  void UpdateDelayBasedEstimate(Timestamp at_time,
+                                DataRate delay_based_bitrate);
+  void OnStartingRate(DataRate start_rate);
+  void OnRateUpdate(absl::optional<DataRate> acknowledged,
+                    DataRate target,
+                    Timestamp at_time);
+  void OnRttBackoff(DataRate backoff_rate, Timestamp at_time);
+  DataRate estimate() const;
+
+ private:
+  FieldTrialParameter<TimeDelta> tracking_rate;
+  double capacity_estimate_bps_ = 0;
+  Timestamp last_link_capacity_update_ = Timestamp::MinusInfinity();
+  DataRate last_delay_based_estimate_ = DataRate::PlusInfinity();
+};
+
+class RttBasedBackoff {
+ public:
+  explicit RttBasedBackoff(const FieldTrialsView* key_value_config);
+  ~RttBasedBackoff();
+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
+  TimeDelta CorrectedRtt(Timestamp at_time) const;
+
+  FieldTrialFlag disabled_;
+  FieldTrialParameter<TimeDelta> configured_limit_;
+  FieldTrialParameter<double> drop_fraction_;
+  FieldTrialParameter<TimeDelta> drop_interval_;
+  FieldTrialParameter<DataRate> bandwidth_floor_;
+
+ public:
+  TimeDelta rtt_limit_;
+  Timestamp last_propagation_rtt_update_;
+  TimeDelta last_propagation_rtt_;
+  Timestamp last_packet_sent_;
+};
+
+class SendSideBandwidthEstimation {
+ public:
+  SendSideBandwidthEstimation() = delete;
+  SendSideBandwidthEstimation(const FieldTrialsView* key_value_config,
+                              RtcEventLog* event_log);
+  ~SendSideBandwidthEstimation();
+
+  void OnRouteChange();
+
+  DataRate target_rate() const;
+  LossBasedState loss_based_state() const;
+  uint8_t fraction_loss() const { return last_fraction_loss_; }
+  TimeDelta round_trip_time() const { return last_round_trip_time_; }
+
+  DataRate GetEstimatedLinkCapacity() const;
+  // Call periodically to update estimate.
+  void UpdateEstimate(Timestamp at_time);
+  void OnSentPacket(const SentPacket& sent_packet);
+  void UpdatePropagationRtt(Timestamp at_time, TimeDelta propagation_rtt);
+
+  // Call when we receive a RTCP message with TMMBR or REMB.
+  void UpdateReceiverEstimate(Timestamp at_time, DataRate bandwidth);
+
+  // Call when a new delay-based estimate is available.
+  void UpdateDelayBasedEstimate(Timestamp at_time, DataRate bitrate);
+
+  // Call when we receive a RTCP message with a ReceiveBlock.
+  void UpdatePacketsLost(int64_t packets_lost,
+                         int64_t number_of_packets,
+                         Timestamp at_time);
+
+  // Call when we receive a RTCP message with a ReceiveBlock.
+  void UpdateRtt(TimeDelta rtt, Timestamp at_time);
+
+  void SetBitrates(absl::optional<DataRate> send_bitrate,
+                   DataRate min_bitrate,
+                   DataRate max_bitrate,
+                   Timestamp at_time);
+  void SetSendBitrate(DataRate bitrate, Timestamp at_time);
+  void SetMinMaxBitrate(DataRate min_bitrate, DataRate max_bitrate);
+  int GetMinBitrate() const;
+  void SetAcknowledgedRate(absl::optional<DataRate> acknowledged_rate,
+                           Timestamp at_time);
+  void UpdateLossBasedEstimator(const TransportPacketsFeedback& report,
+                                BandwidthUsage delay_detector_state,
+                                absl::optional<DataRate> probe_bitrate,
+                                DataRate upper_link_capacity);
+
+ private:
+  friend class GoogCcStatePrinter;
+
+  enum UmaState { kNoUpdate, kFirstDone, kDone };
+
+  bool IsInStartPhase(Timestamp at_time) const;
+
+  void UpdateUmaStatsPacketsLost(Timestamp at_time, int packets_lost);
+
+  // Updates history of min bitrates.
+  // After this method returns min_bitrate_history_.front().second contains the
+  // min bitrate used during last kBweIncreaseIntervalMs.
+  void UpdateMinHistory(Timestamp at_time);
+
+  // Gets the upper limit for the target bitrate. This is the minimum of the
+  // delay based limit, the receiver limit and the loss based controller limit.
+  DataRate GetUpperLimit() const;
+  // Prints a warning if `bitrate` if sufficiently long time has past since last
+  // warning.
+  void MaybeLogLowBitrateWarning(DataRate bitrate, Timestamp at_time);
+  // Stores an update to the event log if the loss rate has changed, the target
+  // has changed, or sufficient time has passed since last stored event.
+  void MaybeLogLossBasedEvent(Timestamp at_time);
+
+  // Cap `bitrate` to [min_bitrate_configured_, max_bitrate_configured_] and
+  // set `current_bitrate_` to the capped value and updates the event log.
+  void UpdateTargetBitrate(DataRate bitrate, Timestamp at_time);
+  // Applies lower and upper bounds to the current target rate.
+  // TODO(srte): This seems to be called even when limits haven't changed, that
+  // should be cleaned up.
+  void ApplyTargetLimits(Timestamp at_time);
+
+  bool LossBasedBandwidthEstimatorV1Enabled() const;
+  bool LossBasedBandwidthEstimatorV2Enabled() const;
+
+  bool LossBasedBandwidthEstimatorV1ReadyForUse() const;
+  bool LossBasedBandwidthEstimatorV2ReadyForUse() const;
+
+  RttBasedBackoff rtt_backoff_;
+  LinkCapacityTracker link_capacity_;
+
+  std::deque<std::pair<Timestamp, DataRate> > min_bitrate_history_;
+
+  // incoming filters
+  int lost_packets_since_last_loss_update_;
+  int expected_packets_since_last_loss_update_;
+
+  absl::optional<DataRate> acknowledged_rate_;
+  DataRate current_target_;
+  DataRate last_logged_target_;
+  DataRate min_bitrate_configured_;
+  DataRate max_bitrate_configured_;
+  Timestamp last_low_bitrate_log_;
+
+  bool has_decreased_since_last_fraction_loss_;
+  Timestamp last_loss_feedback_;
+  Timestamp last_loss_packet_report_;
+  uint8_t last_fraction_loss_;
+  uint8_t last_logged_fraction_loss_;
+  TimeDelta last_round_trip_time_;
+
+  // The max bitrate as set by the receiver in the call. This is typically
+  // signalled using the REMB RTCP message and is used when we don't have any
+  // send side delay based estimate.
+  DataRate receiver_limit_;
+  DataRate delay_based_limit_;
+  Timestamp time_last_decrease_;
+  Timestamp first_report_time_;
+  int initially_lost_packets_;
+  DataRate bitrate_at_2_seconds_;
+  UmaState uma_update_state_;
+  UmaState uma_rtt_state_;
+  std::vector<bool> rampup_uma_stats_updated_;
+  RtcEventLog* const event_log_;
+  Timestamp last_rtc_event_log_;
+  float low_loss_threshold_;
+  float high_loss_threshold_;
+  DataRate bitrate_threshold_;
+  LossBasedBandwidthEstimation loss_based_bandwidth_estimator_v1_;
+  LossBasedBweV2 loss_based_bandwidth_estimator_v2_;
+  LossBasedState loss_based_state_;
+  FieldTrialFlag disable_receiver_limit_caps_only_;
+};
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_SEND_SIDE_BANDWIDTH_ESTIMATION_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation_unittest.cc
new file mode 100644
index 0000000000..17d1aa1ada
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation_unittest.cc
@@ -0,0 +1,206 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.h"
+
+#include "api/rtc_event_log/rtc_event.h"
+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "test/explicit_key_value_config.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+MATCHER(LossBasedBweUpdateWithBitrateOnly, "") {
+  if (arg->GetType() != RtcEvent::Type::BweUpdateLossBased) {
+    return false;
+  }
+  auto bwe_event = static_cast<RtcEventBweUpdateLossBased*>(arg);
+  return bwe_event->bitrate_bps() > 0 && bwe_event->fraction_loss() == 0;
+}
+
+MATCHER(LossBasedBweUpdateWithBitrateAndLossFraction, "") {
+  if (arg->GetType() != RtcEvent::Type::BweUpdateLossBased) {
+    return false;
+  }
+  auto bwe_event = static_cast<RtcEventBweUpdateLossBased*>(arg);
+  return bwe_event->bitrate_bps() > 0 && bwe_event->fraction_loss() > 0;
+}
+
+void TestProbing(bool use_delay_based) {
+  ::testing::NiceMock<MockRtcEventLog> event_log;
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  int64_t now_ms = 0;
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(100000),
+                       DataRate::BitsPerSec(1500000));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(200000), Timestamp::Millis(now_ms));
+
+  const int kRembBps = 1000000;
+  const int kSecondRembBps = kRembBps + 500000;
+
+  bwe.UpdatePacketsLost(/*packets_lost=*/0, /*number_of_packets=*/1,
+                        Timestamp::Millis(now_ms));
+  bwe.UpdateRtt(TimeDelta::Millis(50), Timestamp::Millis(now_ms));
+
+  // Initial REMB applies immediately.
+  if (use_delay_based) {
+    bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
+                                 DataRate::BitsPerSec(kRembBps));
+  } else {
+    bwe.UpdateReceiverEstimate(Timestamp::Millis(now_ms),
+                               DataRate::BitsPerSec(kRembBps));
+  }
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_EQ(kRembBps, bwe.target_rate().bps());
+
+  // Second REMB doesn't apply immediately.
+  now_ms += 2001;
+  if (use_delay_based) {
+    bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
+                                 DataRate::BitsPerSec(kSecondRembBps));
+  } else {
+    bwe.UpdateReceiverEstimate(Timestamp::Millis(now_ms),
+                               DataRate::BitsPerSec(kSecondRembBps));
+  }
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_EQ(kRembBps, bwe.target_rate().bps());
+}
+
+TEST(SendSideBweTest, InitialRembWithProbing) {
+  TestProbing(false);
+}
+
+TEST(SendSideBweTest, InitialDelayBasedBweWithProbing) {
+  TestProbing(true);
+}
+
+TEST(SendSideBweTest, DoesntReapplyBitrateDecreaseWithoutFollowingRemb) {
+  MockRtcEventLog event_log;
+  EXPECT_CALL(event_log, LogProxy(LossBasedBweUpdateWithBitrateOnly()))
+      .Times(1);
+  EXPECT_CALL(event_log,
+              LogProxy(LossBasedBweUpdateWithBitrateAndLossFraction()))
+      .Times(1);
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  static const int kMinBitrateBps = 100000;
+  static const int kInitialBitrateBps = 1000000;
+  int64_t now_ms = 1000;
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(kMinBitrateBps),
+                       DataRate::BitsPerSec(1500000));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kInitialBitrateBps),
+                     Timestamp::Millis(now_ms));
+
+  static const uint8_t kFractionLoss = 128;
+  static const int64_t kRttMs = 50;
+  now_ms += 10000;
+
+  EXPECT_EQ(kInitialBitrateBps, bwe.target_rate().bps());
+  EXPECT_EQ(0, bwe.fraction_loss());
+  EXPECT_EQ(0, bwe.round_trip_time().ms());
+
+  // Signal heavy loss to go down in bitrate.
+  bwe.UpdatePacketsLost(/*packets_lost=*/50, /*number_of_packets=*/100,
+                        Timestamp::Millis(now_ms));
+  bwe.UpdateRtt(TimeDelta::Millis(kRttMs), Timestamp::Millis(now_ms));
+
+  // Trigger an update 2 seconds later to not be rate limited.
+  now_ms += 1000;
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_LT(bwe.target_rate().bps(), kInitialBitrateBps);
+  // Verify that the obtained bitrate isn't hitting the min bitrate, or this
+  // test doesn't make sense. If this ever happens, update the thresholds or
+  // loss rates so that it doesn't hit min bitrate after one bitrate update.
+  EXPECT_GT(bwe.target_rate().bps(), kMinBitrateBps);
+  EXPECT_EQ(kFractionLoss, bwe.fraction_loss());
+  EXPECT_EQ(kRttMs, bwe.round_trip_time().ms());
+
+  // Triggering an update shouldn't apply further downgrade nor upgrade since
+  // there's no intermediate receiver block received indicating whether this is
+  // currently good or not.
+  int last_bitrate_bps = bwe.target_rate().bps();
+  // Trigger an update 2 seconds later to not be rate limited (but it still
+  // shouldn't update).
+  now_ms += 1000;
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+
+  EXPECT_EQ(last_bitrate_bps, bwe.target_rate().bps());
+  // The old loss rate should still be applied though.
+  EXPECT_EQ(kFractionLoss, bwe.fraction_loss());
+  EXPECT_EQ(kRttMs, bwe.round_trip_time().ms());
+}
+
+TEST(SendSideBweTest, SettingSendBitrateOverridesDelayBasedEstimate) {
+  ::testing::NiceMock<MockRtcEventLog> event_log;
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  static const int kMinBitrateBps = 10000;
+  static const int kMaxBitrateBps = 10000000;
+  static const int kInitialBitrateBps = 300000;
+  static const int kDelayBasedBitrateBps = 350000;
+  static const int kForcedHighBitrate = 2500000;
+
+  int64_t now_ms = 0;
+
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(kMinBitrateBps),
+                       DataRate::BitsPerSec(kMaxBitrateBps));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kInitialBitrateBps),
+                     Timestamp::Millis(now_ms));
+
+  bwe.UpdateDelayBasedEstimate(Timestamp::Millis(now_ms),
+                               DataRate::BitsPerSec(kDelayBasedBitrateBps));
+  bwe.UpdateEstimate(Timestamp::Millis(now_ms));
+  EXPECT_GE(bwe.target_rate().bps(), kInitialBitrateBps);
+  EXPECT_LE(bwe.target_rate().bps(), kDelayBasedBitrateBps);
+
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kForcedHighBitrate),
+                     Timestamp::Millis(now_ms));
+  EXPECT_EQ(bwe.target_rate().bps(), kForcedHighBitrate);
+}
+
+TEST(RttBasedBackoff, DefaultEnabled) {
+  test::ExplicitKeyValueConfig key_value_config("");
+  RttBasedBackoff rtt_backoff(&key_value_config);
+  EXPECT_TRUE(rtt_backoff.rtt_limit_.IsFinite());
+}
+
+TEST(RttBasedBackoff, CanBeDisabled) {
+  test::ExplicitKeyValueConfig key_value_config(
+      "WebRTC-Bwe-MaxRttLimit/Disabled/");
+  RttBasedBackoff rtt_backoff(&key_value_config);
+  EXPECT_TRUE(rtt_backoff.rtt_limit_.IsPlusInfinity());
+}
+
+TEST(SendSideBweTest, FractionLossIsNotOverflowed) {
+  MockRtcEventLog event_log;
+  test::ExplicitKeyValueConfig key_value_config("");
+  SendSideBandwidthEstimation bwe(&key_value_config, &event_log);
+  static const int kMinBitrateBps = 100000;
+  static const int kInitialBitrateBps = 1000000;
+  int64_t now_ms = 1000;
+  bwe.SetMinMaxBitrate(DataRate::BitsPerSec(kMinBitrateBps),
+                       DataRate::BitsPerSec(1500000));
+  bwe.SetSendBitrate(DataRate::BitsPerSec(kInitialBitrateBps),
+                     Timestamp::Millis(now_ms));
+
+  now_ms += 10000;
+
+  EXPECT_EQ(kInitialBitrateBps, bwe.target_rate().bps());
+  EXPECT_EQ(0, bwe.fraction_loss());
+
+  // Signal negative loss.
+  bwe.UpdatePacketsLost(/*packets_lost=*/-1, /*number_of_packets=*/100,
+                        Timestamp::Millis(now_ms));
+  EXPECT_EQ(0, bwe.fraction_loss());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build
new file mode 100644
index 0000000000..c93e412e83
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bwe_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["BWE_TEST_LOGGING_COMPILE_TIME_ENABLE"] = "0"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("send_side_bwe_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
new file mode 100644
index 0000000000..6a8849ed6d
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
@@ -0,0 +1,200 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h"
+
+#include <math.h>
+
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "modules/congestion_controller/goog_cc/alr_detector.h"
+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+void WriteTypedValue(RtcEventLogOutput* out, int value) {
+  LogWriteFormat(out, "%i", value);
+}
+void WriteTypedValue(RtcEventLogOutput* out, double value) {
+  LogWriteFormat(out, "%.6f", value);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<DataRate> value) {
+  LogWriteFormat(out, "%.0f", value ? value->bytes_per_sec<double>() : NAN);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<DataSize> value) {
+  LogWriteFormat(out, "%.0f", value ? value->bytes<double>() : NAN);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<TimeDelta> value) {
+  LogWriteFormat(out, "%.3f", value ? value->seconds<double>() : NAN);
+}
+void WriteTypedValue(RtcEventLogOutput* out, absl::optional<Timestamp> value) {
+  LogWriteFormat(out, "%.3f", value ? value->seconds<double>() : NAN);
+}
+
+template <typename F>
+class TypedFieldLogger : public FieldLogger {
+ public:
+  TypedFieldLogger(absl::string_view name, F&& getter)
+      : name_(name), getter_(std::forward<F>(getter)) {}
+  const std::string& name() const override { return name_; }
+  void WriteValue(RtcEventLogOutput* out) override {
+    WriteTypedValue(out, getter_());
+  }
+
+ private:
+  std::string name_;
+  F getter_;
+};
+
+template <typename F>
+FieldLogger* Log(absl::string_view name, F&& getter) {
+  return new TypedFieldLogger<F>(name, std::forward<F>(getter));
+}
+
+}  // namespace
+GoogCcStatePrinter::GoogCcStatePrinter() {
+  for (auto* logger : CreateLoggers()) {
+    loggers_.emplace_back(logger);
+  }
+}
+
+std::deque<FieldLogger*> GoogCcStatePrinter::CreateLoggers() {
+  auto stable_estimate = [this] {
+    return DataRate::KilobitsPerSec(
+        controller_->delay_based_bwe_->rate_control_.link_capacity_
+            .estimate_kbps_.value_or(-INFINITY));
+  };
+  auto rate_control_state = [this] {
+    return static_cast<int>(
+        controller_->delay_based_bwe_->rate_control_.rate_control_state_);
+  };
+  auto trend = [this] {
+    return reinterpret_cast<TrendlineEstimator*>(
+        controller_->delay_based_bwe_->active_delay_detector_);
+  };
+  auto acknowledged_rate = [this] {
+    return controller_->acknowledged_bitrate_estimator_->bitrate();
+  };
+  auto loss_cont = [&] {
+    return &controller_->bandwidth_estimation_
+                ->loss_based_bandwidth_estimator_v1_;
+  };
+  std::deque<FieldLogger*> loggers({
+      Log("time", [=] { return target_.at_time; }),
+      Log("rtt", [=] { return target_.network_estimate.round_trip_time; }),
+      Log("target", [=] { return target_.target_rate; }),
+      Log("stable_target", [=] { return target_.stable_target_rate; }),
+      Log("pacing", [=] { return pacing_.data_rate(); }),
+      Log("padding", [=] { return pacing_.pad_rate(); }),
+      Log("window", [=] { return congestion_window_; }),
+      Log("rate_control_state", [=] { return rate_control_state(); }),
+      Log("stable_estimate", [=] { return stable_estimate(); }),
+      Log("trendline", [=] { return trend()->prev_trend_; }),
+      Log("trendline_modified_offset",
+          [=] { return trend()->prev_modified_trend_; }),
+      Log("trendline_offset_threshold", [=] { return trend()->threshold_; }),
+      Log("acknowledged_rate", [=] { return acknowledged_rate(); }),
+      Log("est_capacity", [=] { return est_.link_capacity; }),
+      Log("est_capacity_dev", [=] { return est_.link_capacity_std_dev; }),
+      Log("est_capacity_min", [=] { return est_.link_capacity_min; }),
+      Log("est_cross_traffic", [=] { return est_.cross_traffic_ratio; }),
+      Log("est_cross_delay", [=] { return est_.cross_delay_rate; }),
+      Log("est_spike_delay", [=] { return est_.spike_delay_rate; }),
+      Log("est_pre_buffer", [=] { return est_.pre_link_buffer_delay; }),
+      Log("est_post_buffer", [=] { return est_.post_link_buffer_delay; }),
+      Log("est_propagation", [=] { return est_.propagation_delay; }),
+      Log("loss_ratio", [=] { return loss_cont()->last_loss_ratio_; }),
+      Log("loss_average", [=] { return loss_cont()->average_loss_; }),
+      Log("loss_average_max", [=] { return loss_cont()->average_loss_max_; }),
+      Log("loss_thres_inc",
+          [=] { return loss_cont()->loss_increase_threshold(); }),
+      Log("loss_thres_dec",
+          [=] { return loss_cont()->loss_decrease_threshold(); }),
+      Log("loss_dec_rate", [=] { return loss_cont()->decreased_bitrate(); }),
+      Log("loss_based_rate", [=] { return loss_cont()->loss_based_bitrate_; }),
+      Log("loss_ack_rate",
+          [=] { return loss_cont()->acknowledged_bitrate_max_; }),
+      Log("data_window", [=] { return controller_->current_data_window_; }),
+      Log("pushback_target",
+          [=] { return controller_->last_pushback_target_rate_; }),
+  });
+  return loggers;
+}
+GoogCcStatePrinter::~GoogCcStatePrinter() = default;
+
+void GoogCcStatePrinter::PrintHeaders(RtcEventLogOutput* log) {
+  int ix = 0;
+  for (const auto& logger : loggers_) {
+    if (ix++)
+      log->Write(" ");
+    log->Write(logger->name());
+  }
+  log->Write("\n");
+  log->Flush();
+}
+
+void GoogCcStatePrinter::PrintState(RtcEventLogOutput* log,
+                                    GoogCcNetworkController* controller,
+                                    Timestamp at_time) {
+  controller_ = controller;
+  auto state_update = controller_->GetNetworkState(at_time);
+  target_ = state_update.target_rate.value();
+  pacing_ = state_update.pacer_config.value();
+  if (state_update.congestion_window)
+    congestion_window_ = *state_update.congestion_window;
+  if (controller_->network_estimator_) {
+    est_ = controller_->network_estimator_->GetCurrentEstimate().value_or(
+        NetworkStateEstimate());
+  }
+
+  int ix = 0;
+  for (const auto& logger : loggers_) {
+    if (ix++)
+      log->Write(" ");
+    logger->WriteValue(log);
+  }
+
+  log->Write("\n");
+  log->Flush();
+}
+
+GoogCcDebugFactory::GoogCcDebugFactory()
+    : GoogCcDebugFactory(GoogCcFactoryConfig()) {}
+
+GoogCcDebugFactory::GoogCcDebugFactory(GoogCcFactoryConfig config)
+    : GoogCcNetworkControllerFactory(std::move(config)) {}
+
+std::unique_ptr<NetworkControllerInterface> GoogCcDebugFactory::Create(
+    NetworkControllerConfig config) {
+  RTC_CHECK(controller_ == nullptr);
+  auto controller = GoogCcNetworkControllerFactory::Create(config);
+  controller_ = static_cast<GoogCcNetworkController*>(controller.get());
+  return controller;
+}
+
+void GoogCcDebugFactory::PrintState(const Timestamp at_time) {
+  if (controller_ && log_writer_) {
+    printer_.PrintState(log_writer_.get(), controller_, at_time);
+  }
+}
+
+void GoogCcDebugFactory::AttachWriter(
+    std::unique_ptr<RtcEventLogOutput> log_writer) {
+  if (log_writer) {
+    log_writer_ = std::move(log_writer);
+    printer_.PrintHeaders(log_writer_.get());
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
new file mode 100644
index 0000000000..16fa657e71
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
+
+#include <deque>
+#include <memory>
+#include <string>
+
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/transport/goog_cc_factory.h"
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
+#include "test/logging/log_writer.h"
+
+namespace webrtc {
+
+class FieldLogger {
+ public:
+  virtual ~FieldLogger() = default;
+  virtual const std::string& name() const = 0;
+  virtual void WriteValue(RtcEventLogOutput* out) = 0;
+};
+
+class GoogCcStatePrinter {
+ public:
+  GoogCcStatePrinter();
+  GoogCcStatePrinter(const GoogCcStatePrinter&) = delete;
+  GoogCcStatePrinter& operator=(const GoogCcStatePrinter&) = delete;
+  ~GoogCcStatePrinter();
+
+  void PrintHeaders(RtcEventLogOutput* log);
+  void PrintState(RtcEventLogOutput* log,
+                  GoogCcNetworkController* controller,
+                  Timestamp at_time);
+
+ private:
+  std::deque<FieldLogger*> CreateLoggers();
+  std::deque<std::unique_ptr<FieldLogger>> loggers_;
+
+  GoogCcNetworkController* controller_ = nullptr;
+  TargetTransferRate target_;
+  PacerConfig pacing_;
+  DataSize congestion_window_ = DataSize::PlusInfinity();
+  NetworkStateEstimate est_;
+};
+
+class GoogCcDebugFactory : public GoogCcNetworkControllerFactory {
+ public:
+  GoogCcDebugFactory();
+  explicit GoogCcDebugFactory(GoogCcFactoryConfig config);
+  std::unique_ptr<NetworkControllerInterface> Create(
+      NetworkControllerConfig config) override;
+
+  void PrintState(Timestamp at_time);
+
+  void AttachWriter(std::unique_ptr<RtcEventLogOutput> log_writer);
+
+ private:
+  GoogCcStatePrinter printer_;
+  GoogCcNetworkController* controller_ = nullptr;
+  std::unique_ptr<RtcEventLogOutput> log_writer_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
new file mode 100644
index 0000000000..88182d4f80
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
@@ -0,0 +1,332 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+
+#include <math.h>
+
+#include <algorithm>
+#include <string>
+
+#include "absl/strings/match.h"
+#include "absl/types/optional.h"
+#include "api/network_state_predictor.h"
+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+namespace {
+
+// Parameters for linear least squares fit of regression line to noisy data.
+constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
+constexpr double kDefaultTrendlineThresholdGain = 4.0;
+const char kBweWindowSizeInPacketsExperiment[] =
+    "WebRTC-BweWindowSizeInPackets";
+
+size_t ReadTrendlineFilterWindowSize(const FieldTrialsView* key_value_config) {
+  std::string experiment_string =
+      key_value_config->Lookup(kBweWindowSizeInPacketsExperiment);
+  size_t window_size;
+  int parsed_values =
+      sscanf(experiment_string.c_str(), "Enabled-%zu", &window_size);
+  if (parsed_values == 1) {
+    if (window_size > 1)
+      return window_size;
+    RTC_LOG(LS_WARNING) << "Window size must be greater than 1.";
+  }
+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweWindowSizeInPackets"
+                         " experiment from field trial string. Using default.";
+  return TrendlineEstimatorSettings::kDefaultTrendlineWindowSize;
+}
+
+absl::optional<double> LinearFitSlope(
+    const std::deque<TrendlineEstimator::PacketTiming>& packets) {
+  RTC_DCHECK(packets.size() >= 2);
+  // Compute the "center of mass".
+  double sum_x = 0;
+  double sum_y = 0;
+  for (const auto& packet : packets) {
+    sum_x += packet.arrival_time_ms;
+    sum_y += packet.smoothed_delay_ms;
+  }
+  double x_avg = sum_x / packets.size();
+  double y_avg = sum_y / packets.size();
+  // Compute the slope k = \sum (x_i-x_avg)(y_i-y_avg) / \sum (x_i-x_avg)^2
+  double numerator = 0;
+  double denominator = 0;
+  for (const auto& packet : packets) {
+    double x = packet.arrival_time_ms;
+    double y = packet.smoothed_delay_ms;
+    numerator += (x - x_avg) * (y - y_avg);
+    denominator += (x - x_avg) * (x - x_avg);
+  }
+  if (denominator == 0)
+    return absl::nullopt;
+  return numerator / denominator;
+}
+
+absl::optional<double> ComputeSlopeCap(
+    const std::deque<TrendlineEstimator::PacketTiming>& packets,
+    const TrendlineEstimatorSettings& settings) {
+  RTC_DCHECK(1 <= settings.beginning_packets &&
+             settings.beginning_packets < packets.size());
+  RTC_DCHECK(1 <= settings.end_packets &&
+             settings.end_packets < packets.size());
+  RTC_DCHECK(settings.beginning_packets + settings.end_packets <=
+             packets.size());
+  TrendlineEstimator::PacketTiming early = packets[0];
+  for (size_t i = 1; i < settings.beginning_packets; ++i) {
+    if (packets[i].raw_delay_ms < early.raw_delay_ms)
+      early = packets[i];
+  }
+  size_t late_start = packets.size() - settings.end_packets;
+  TrendlineEstimator::PacketTiming late = packets[late_start];
+  for (size_t i = late_start + 1; i < packets.size(); ++i) {
+    if (packets[i].raw_delay_ms < late.raw_delay_ms)
+      late = packets[i];
+  }
+  if (late.arrival_time_ms - early.arrival_time_ms < 1) {
+    return absl::nullopt;
+  }
+  return (late.raw_delay_ms - early.raw_delay_ms) /
+             (late.arrival_time_ms - early.arrival_time_ms) +
+         settings.cap_uncertainty;
+}
+
+constexpr double kMaxAdaptOffsetMs = 15.0;
+constexpr double kOverUsingTimeThreshold = 10;
+constexpr int kMinNumDeltas = 60;
+constexpr int kDeltaCounterMax = 1000;
+
+}  // namespace
+
+constexpr char TrendlineEstimatorSettings::kKey[];
+
+TrendlineEstimatorSettings::TrendlineEstimatorSettings(
+    const FieldTrialsView* key_value_config) {
+  if (absl::StartsWith(
+          key_value_config->Lookup(kBweWindowSizeInPacketsExperiment),
+          "Enabled")) {
+    window_size = ReadTrendlineFilterWindowSize(key_value_config);
+  }
+  Parser()->Parse(key_value_config->Lookup(TrendlineEstimatorSettings::kKey));
+  if (window_size < 10 || 200 < window_size) {
+    RTC_LOG(LS_WARNING) << "Window size must be between 10 and 200 packets";
+    window_size = kDefaultTrendlineWindowSize;
+  }
+  if (enable_cap) {
+    if (beginning_packets < 1 || end_packets < 1 ||
+        beginning_packets > window_size || end_packets > window_size) {
+      RTC_LOG(LS_WARNING) << "Size of beginning and end must be between 1 and "
+                          << window_size;
+      enable_cap = false;
+      beginning_packets = end_packets = 0;
+      cap_uncertainty = 0.0;
+    }
+    if (beginning_packets + end_packets > window_size) {
+      RTC_LOG(LS_WARNING)
+          << "Size of beginning plus end can't exceed the window size";
+      enable_cap = false;
+      beginning_packets = end_packets = 0;
+      cap_uncertainty = 0.0;
+    }
+    if (cap_uncertainty < 0.0 || 0.025 < cap_uncertainty) {
+      RTC_LOG(LS_WARNING) << "Cap uncertainty must be between 0 and 0.025";
+      cap_uncertainty = 0.0;
+    }
+  }
+}
+
+std::unique_ptr<StructParametersParser> TrendlineEstimatorSettings::Parser() {
+  return StructParametersParser::Create("sort", &enable_sort,  //
+                                        "cap", &enable_cap,    //
+                                        "beginning_packets",
+                                        &beginning_packets,                   //
+                                        "end_packets", &end_packets,          //
+                                        "cap_uncertainty", &cap_uncertainty,  //
+                                        "window_size", &window_size);
+}
+
+TrendlineEstimator::TrendlineEstimator(
+    const FieldTrialsView* key_value_config,
+    NetworkStatePredictor* network_state_predictor)
+    : settings_(key_value_config),
+      smoothing_coef_(kDefaultTrendlineSmoothingCoeff),
+      threshold_gain_(kDefaultTrendlineThresholdGain),
+      num_of_deltas_(0),
+      first_arrival_time_ms_(-1),
+      accumulated_delay_(0),
+      smoothed_delay_(0),
+      delay_hist_(),
+      k_up_(0.0087),
+      k_down_(0.039),
+      overusing_time_threshold_(kOverUsingTimeThreshold),
+      threshold_(12.5),
+      prev_modified_trend_(NAN),
+      last_update_ms_(-1),
+      prev_trend_(0.0),
+      time_over_using_(-1),
+      overuse_counter_(0),
+      hypothesis_(BandwidthUsage::kBwNormal),
+      hypothesis_predicted_(BandwidthUsage::kBwNormal),
+      network_state_predictor_(network_state_predictor) {
+  RTC_LOG(LS_INFO)
+      << "Using Trendline filter for delay change estimation with settings "
+      << settings_.Parser()->Encode() << " and "
+      << (network_state_predictor_ ? "injected" : "no")
+      << " network state predictor";
+}
+
+TrendlineEstimator::~TrendlineEstimator() {}
+
+void TrendlineEstimator::UpdateTrendline(double recv_delta_ms,
+                                         double send_delta_ms,
+                                         int64_t send_time_ms,
+                                         int64_t arrival_time_ms,
+                                         size_t packet_size) {
+  const double delta_ms = recv_delta_ms - send_delta_ms;
+  ++num_of_deltas_;
+  num_of_deltas_ = std::min(num_of_deltas_, kDeltaCounterMax);
+  if (first_arrival_time_ms_ == -1)
+    first_arrival_time_ms_ = arrival_time_ms;
+
+  // Exponential backoff filter.
+  accumulated_delay_ += delta_ms;
+  BWE_TEST_LOGGING_PLOT(1, "accumulated_delay_ms", arrival_time_ms,
+                        accumulated_delay_);
+  smoothed_delay_ = smoothing_coef_ * smoothed_delay_ +
+                    (1 - smoothing_coef_) * accumulated_delay_;
+  BWE_TEST_LOGGING_PLOT(1, "smoothed_delay_ms", arrival_time_ms,
+                        smoothed_delay_);
+
+  // Maintain packet window
+  delay_hist_.emplace_back(
+      static_cast<double>(arrival_time_ms - first_arrival_time_ms_),
+      smoothed_delay_, accumulated_delay_);
+  if (settings_.enable_sort) {
+    for (size_t i = delay_hist_.size() - 1;
+         i > 0 &&
+         delay_hist_[i].arrival_time_ms < delay_hist_[i - 1].arrival_time_ms;
+         --i) {
+      std::swap(delay_hist_[i], delay_hist_[i - 1]);
+    }
+  }
+  if (delay_hist_.size() > settings_.window_size)
+    delay_hist_.pop_front();
+
+  // Simple linear regression.
+  double trend = prev_trend_;
+  if (delay_hist_.size() == settings_.window_size) {
+    // Update trend_ if it is possible to fit a line to the data. The delay
+    // trend can be seen as an estimate of (send_rate - capacity)/capacity.
+    // 0 < trend < 1   ->  the delay increases, queues are filling up
+    //   trend == 0    ->  the delay does not change
+    //   trend < 0     ->  the delay decreases, queues are being emptied
+    trend = LinearFitSlope(delay_hist_).value_or(trend);
+    if (settings_.enable_cap) {
+      absl::optional<double> cap = ComputeSlopeCap(delay_hist_, settings_);
+      // We only use the cap to filter out overuse detections, not
+      // to detect additional underuses.
+      if (trend >= 0 && cap.has_value() && trend > cap.value()) {
+        trend = cap.value();
+      }
+    }
+  }
+  BWE_TEST_LOGGING_PLOT(1, "trendline_slope", arrival_time_ms, trend);
+
+  Detect(trend, send_delta_ms, arrival_time_ms);
+}
+
+void TrendlineEstimator::Update(double recv_delta_ms,
+                                double send_delta_ms,
+                                int64_t send_time_ms,
+                                int64_t arrival_time_ms,
+                                size_t packet_size,
+                                bool calculated_deltas) {
+  if (calculated_deltas) {
+    UpdateTrendline(recv_delta_ms, send_delta_ms, send_time_ms, arrival_time_ms,
+                    packet_size);
+  }
+  if (network_state_predictor_) {
+    hypothesis_predicted_ = network_state_predictor_->Update(
+        send_time_ms, arrival_time_ms, hypothesis_);
+  }
+}
+
+BandwidthUsage TrendlineEstimator::State() const {
+  return network_state_predictor_ ? hypothesis_predicted_ : hypothesis_;
+}
+
+void TrendlineEstimator::Detect(double trend, double ts_delta, int64_t now_ms) {
+  if (num_of_deltas_ < 2) {
+    hypothesis_ = BandwidthUsage::kBwNormal;
+    return;
+  }
+  const double modified_trend =
+      std::min(num_of_deltas_, kMinNumDeltas) * trend * threshold_gain_;
+  prev_modified_trend_ = modified_trend;
+  BWE_TEST_LOGGING_PLOT(1, "T", now_ms, modified_trend);
+  BWE_TEST_LOGGING_PLOT(1, "threshold", now_ms, threshold_);
+  if (modified_trend > threshold_) {
+    if (time_over_using_ == -1) {
+      // Initialize the timer. Assume that we've been
+      // over-using half of the time since the previous
+      // sample.
+      time_over_using_ = ts_delta / 2;
+    } else {
+      // Increment timer
+      time_over_using_ += ts_delta;
+    }
+    overuse_counter_++;
+    if (time_over_using_ > overusing_time_threshold_ && overuse_counter_ > 1) {
+      if (trend >= prev_trend_) {
+        time_over_using_ = 0;
+        overuse_counter_ = 0;
+        hypothesis_ = BandwidthUsage::kBwOverusing;
+      }
+    }
+  } else if (modified_trend < -threshold_) {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwUnderusing;
+  } else {
+    time_over_using_ = -1;
+    overuse_counter_ = 0;
+    hypothesis_ = BandwidthUsage::kBwNormal;
+  }
+  prev_trend_ = trend;
+  UpdateThreshold(modified_trend, now_ms);
+}
+
+void TrendlineEstimator::UpdateThreshold(double modified_trend,
+                                         int64_t now_ms) {
+  if (last_update_ms_ == -1)
+    last_update_ms_ = now_ms;
+
+  if (fabs(modified_trend) > threshold_ + kMaxAdaptOffsetMs) {
+    // Avoid adapting the threshold to big latency spikes, caused e.g.,
+    // by a sudden capacity drop.
+    last_update_ms_ = now_ms;
+    return;
+  }
+
+  const double k = fabs(modified_trend) < threshold_ ? k_down_ : k_up_;
+  const int64_t kMaxTimeDeltaMs = 100;
+  int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
+  threshold_ += k * (fabs(modified_trend) - threshold_) * time_delta_ms;
+  threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
+  last_update_ms_ = now_ms;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
new file mode 100644
index 0000000000..ffda25df74
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <utility>
+
+#include "api/field_trials_view.h"
+#include "api/network_state_predictor.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "rtc_base/experiments/struct_parameters_parser.h"
+
+namespace webrtc {
+
+struct TrendlineEstimatorSettings {
+  static constexpr char kKey[] = "WebRTC-Bwe-TrendlineEstimatorSettings";
+  static constexpr unsigned kDefaultTrendlineWindowSize = 20;
+
+  TrendlineEstimatorSettings() = delete;
+  explicit TrendlineEstimatorSettings(const FieldTrialsView* key_value_config);
+
+  // Sort the packets in the window. Should be redundant,
+  // but then almost no cost.
+  bool enable_sort = false;
+
+  // Cap the trendline slope based on the minimum delay seen
+  // in the beginning_packets and end_packets respectively.
+  bool enable_cap = false;
+  unsigned beginning_packets = 7;
+  unsigned end_packets = 7;
+  double cap_uncertainty = 0.0;
+
+  // Size (in packets) of the window.
+  unsigned window_size = kDefaultTrendlineWindowSize;
+
+  std::unique_ptr<StructParametersParser> Parser();
+};
+
+class TrendlineEstimator : public DelayIncreaseDetectorInterface {
+ public:
+  TrendlineEstimator(const FieldTrialsView* key_value_config,
+                     NetworkStatePredictor* network_state_predictor);
+
+  ~TrendlineEstimator() override;
+
+  TrendlineEstimator(const TrendlineEstimator&) = delete;
+  TrendlineEstimator& operator=(const TrendlineEstimator&) = delete;
+
+  // Update the estimator with a new sample. The deltas should represent deltas
+  // between timestamp groups as defined by the InterArrival class.
+  void Update(double recv_delta_ms,
+              double send_delta_ms,
+              int64_t send_time_ms,
+              int64_t arrival_time_ms,
+              size_t packet_size,
+              bool calculated_deltas) override;
+
+  void UpdateTrendline(double recv_delta_ms,
+                       double send_delta_ms,
+                       int64_t send_time_ms,
+                       int64_t arrival_time_ms,
+                       size_t packet_size);
+
+  BandwidthUsage State() const override;
+
+  struct PacketTiming {
+    PacketTiming(double arrival_time_ms,
+                 double smoothed_delay_ms,
+                 double raw_delay_ms)
+        : arrival_time_ms(arrival_time_ms),
+          smoothed_delay_ms(smoothed_delay_ms),
+          raw_delay_ms(raw_delay_ms) {}
+    double arrival_time_ms;
+    double smoothed_delay_ms;
+    double raw_delay_ms;
+  };
+
+ private:
+  friend class GoogCcStatePrinter;
+  void Detect(double trend, double ts_delta, int64_t now_ms);
+
+  void UpdateThreshold(double modified_offset, int64_t now_ms);
+
+  // Parameters.
+  TrendlineEstimatorSettings settings_;
+  const double smoothing_coef_;
+  const double threshold_gain_;
+  // Used by the existing threshold.
+  int num_of_deltas_;
+  // Keep the arrival times small by using the change from the first packet.
+  int64_t first_arrival_time_ms_;
+  // Exponential backoff filtering.
+  double accumulated_delay_;
+  double smoothed_delay_;
+  // Linear least squares regression.
+  std::deque<PacketTiming> delay_hist_;
+
+  const double k_up_;
+  const double k_down_;
+  double overusing_time_threshold_;
+  double threshold_;
+  double prev_modified_trend_;
+  int64_t last_update_ms_;
+  double prev_trend_;
+  double time_over_using_;
+  int overuse_counter_;
+  BandwidthUsage hypothesis_;
+  BandwidthUsage hypothesis_predicted_;
+  NetworkStatePredictor* network_state_predictor_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc
new file mode 100644
index 0000000000..b0195abdf5
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc
@@ -0,0 +1,151 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
+
+#include <algorithm>
+#include <numeric>
+#include <vector>
+
+#include "api/transport/field_trial_based_config.h"
+#include "rtc_base/random.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+class PacketTimeGenerator {
+ public:
+  PacketTimeGenerator(int64_t initial_clock, double time_between_packets)
+      : initial_clock_(initial_clock),
+        time_between_packets_(time_between_packets),
+        packets_(0) {}
+  int64_t operator()() {
+    return initial_clock_ + time_between_packets_ * packets_++;
+  }
+
+ private:
+  const int64_t initial_clock_;
+  const double time_between_packets_;
+  size_t packets_;
+};
+
+class TrendlineEstimatorTest : public testing::Test {
+ public:
+  TrendlineEstimatorTest()
+      : send_times(kPacketCount),
+        recv_times(kPacketCount),
+        packet_sizes(kPacketCount),
+        config(),
+        estimator(&config, nullptr),
+        count(1) {
+    std::fill(packet_sizes.begin(), packet_sizes.end(), kPacketSizeBytes);
+  }
+
+  void RunTestUntilStateChange() {
+    RTC_DCHECK_EQ(send_times.size(), kPacketCount);
+    RTC_DCHECK_EQ(recv_times.size(), kPacketCount);
+    RTC_DCHECK_EQ(packet_sizes.size(), kPacketCount);
+    RTC_DCHECK_GE(count, 1);
+    RTC_DCHECK_LT(count, kPacketCount);
+
+    auto initial_state = estimator.State();
+    for (; count < kPacketCount; count++) {
+      double recv_delta = recv_times[count] - recv_times[count - 1];
+      double send_delta = send_times[count] - send_times[count - 1];
+      estimator.Update(recv_delta, send_delta, send_times[count],
+                       recv_times[count], packet_sizes[count], true);
+      if (estimator.State() != initial_state) {
+        return;
+      }
+    }
+  }
+
+ protected:
+  const size_t kPacketCount = 25;
+  const size_t kPacketSizeBytes = 1200;
+  std::vector<int64_t> send_times;
+  std::vector<int64_t> recv_times;
+  std::vector<size_t> packet_sizes;
+  const FieldTrialBasedConfig config;
+  TrendlineEstimator estimator;
+  size_t count;
+};
+}  // namespace
+
+TEST_F(TrendlineEstimatorTest, Normal) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          20 /*delivered at the same pace*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, Overusing) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          1.1 * 20 /*10% slower delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, Underusing) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          0.85 * 20 /*15% faster delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwUnderusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwUnderusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+TEST_F(TrendlineEstimatorTest, IncludesSmallPacketsByDefault) {
+  PacketTimeGenerator send_time_generator(123456789 /*initial clock*/,
+                                          20 /*20 ms between sent packets*/);
+  std::generate(send_times.begin(), send_times.end(), send_time_generator);
+
+  PacketTimeGenerator recv_time_generator(987654321 /*initial clock*/,
+                                          1.1 * 20 /*10% slower delivery*/);
+  std::generate(recv_times.begin(), recv_times.end(), recv_time_generator);
+
+  std::fill(packet_sizes.begin(), packet_sizes.end(), 100);
+
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwNormal);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  RunTestUntilStateChange();
+  EXPECT_EQ(estimator.State(), BandwidthUsage::kBwOverusing);
+  EXPECT_EQ(count, kPacketCount);  // All packets processed
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/include/receive_side_congestion_controller.h b/third_party/libwebrtc/modules/congestion_controller/include/receive_side_congestion_controller.h
new file mode 100644
index 0000000000..7696396016
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/include/receive_side_congestion_controller.h
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/transport/network_control.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "modules/congestion_controller/remb_throttler.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+class RemoteBitrateEstimator;
+
+// This class represents the congestion control state for receive
+// streams. For send side bandwidth estimation, this is simply
+// relaying for each received RTP packet back to the sender. While for
+// receive side bandwidth estimation, we do the estimation locally and
+// send our results back to the sender.
+class ReceiveSideCongestionController : public CallStatsObserver {
+ public:
+  ReceiveSideCongestionController(
+      Clock* clock,
+      RemoteEstimatorProxy::TransportFeedbackSender feedback_sender,
+      RembThrottler::RembSender remb_sender,
+      NetworkStateEstimator* network_state_estimator);
+
+  ~ReceiveSideCongestionController() override {}
+
+  virtual void OnReceivedPacket(int64_t arrival_time_ms,
+                                size_t payload_size,
+                                const RTPHeader& header);
+
+  void SetSendPeriodicFeedback(bool send_periodic_feedback);
+
+  // Implements CallStatsObserver.
+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
+
+  // This is send bitrate, used to control the rate of feedback messages.
+  void OnBitrateChanged(int bitrate_bps);
+
+  // Ensures the remote party is notified of the receive bitrate no larger than
+  // `bitrate` using RTCP REMB.
+  void SetMaxDesiredReceiveBitrate(DataRate bitrate);
+
+  void SetTransportOverhead(DataSize overhead_per_packet);
+
+  // Returns latest receive side bandwidth estimation.
+  // Returns zero if receive side bandwidth estimation is unavailable.
+  DataRate LatestReceiveSideEstimate() const;
+
+  // Removes stream from receive side bandwidth estimation.
+  // Noop if receive side bwe is not used or stream doesn't participate in it.
+  void RemoveStream(uint32_t ssrc);
+
+  // Runs periodic tasks if it is time to run them, returns time until next
+  // call to `MaybeProcess` should be non idle.
+  TimeDelta MaybeProcess();
+
+ private:
+  void PickEstimatorFromHeader(const RTPHeader& header)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+  void PickEstimator() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
+  Clock& clock_;
+  RembThrottler remb_throttler_;
+  RemoteEstimatorProxy remote_estimator_proxy_;
+
+  mutable Mutex mutex_;
+  std::unique_ptr<RemoteBitrateEstimator> rbe_ RTC_GUARDED_BY(mutex_);
+  bool using_absolute_send_time_ RTC_GUARDED_BY(mutex_);
+  uint32_t packets_since_absolute_send_time_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_INCLUDE_RECEIVE_SIDE_CONGESTION_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/pcc/BUILD.gn
new file mode 100644
index 0000000000..85b12b3771
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/BUILD.gn
@@ -0,0 +1,123 @@
+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_library("pcc") {
+  sources = [
+    "pcc_factory.cc",
+    "pcc_factory.h",
+  ]
+  deps = [
+    ":pcc_controller",
+    "../../../api/transport:network_control",
+    "../../../api/units:time_delta",
+  ]
+}
+
+rtc_library("pcc_controller") {
+  sources = [
+    "pcc_network_controller.cc",
+    "pcc_network_controller.h",
+  ]
+  deps = [
+    ":bitrate_controller",
+    ":monitor_interval",
+    ":rtt_tracker",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:random",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("monitor_interval") {
+  sources = [
+    "monitor_interval.cc",
+    "monitor_interval.h",
+  ]
+  deps = [
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:logging",
+  ]
+}
+
+rtc_library("rtt_tracker") {
+  sources = [
+    "rtt_tracker.cc",
+    "rtt_tracker.h",
+  ]
+  deps = [
+    "../../../api/transport:network_control",
+    "../../../api/units:time_delta",
+    "../../../api/units:timestamp",
+  ]
+}
+
+rtc_library("utility_function") {
+  sources = [
+    "utility_function.cc",
+    "utility_function.h",
+  ]
+  deps = [
+    ":monitor_interval",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../rtc_base:checks",
+  ]
+}
+
+rtc_library("bitrate_controller") {
+  sources = [
+    "bitrate_controller.cc",
+    "bitrate_controller.h",
+  ]
+  deps = [
+    ":monitor_interval",
+    ":utility_function",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+if (rtc_include_tests && !build_with_chromium) {
+  rtc_library("pcc_unittests") {
+    testonly = true
+    sources = [
+      "bitrate_controller_unittest.cc",
+      "monitor_interval_unittest.cc",
+      "pcc_network_controller_unittest.cc",
+      "rtt_tracker_unittest.cc",
+      "utility_function_unittest.cc",
+    ]
+    deps = [
+      ":bitrate_controller",
+      ":monitor_interval",
+      ":pcc",
+      ":pcc_controller",
+      ":rtt_tracker",
+      ":utility_function",
+      "../../../api/transport:network_control",
+      "../../../api/units:data_rate",
+      "../../../api/units:data_size",
+      "../../../api/units:time_delta",
+      "../../../api/units:timestamp",
+      "../../../test:test_support",
+      "../../../test/scenario",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.cc
new file mode 100644
index 0000000000..16b8e6966f
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.cc
@@ -0,0 +1,139 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/bitrate_controller.h"
+
+#include <algorithm>
+#include <cmath>
+#include <cstdlib>
+#include <memory>
+#include <utility>
+#include <vector>
+
+
+namespace webrtc {
+namespace pcc {
+
+PccBitrateController::PccBitrateController(double initial_conversion_factor,
+                                           double initial_dynamic_boundary,
+                                           double dynamic_boundary_increment,
+                                           double rtt_gradient_coefficient,
+                                           double loss_coefficient,
+                                           double throughput_coefficient,
+                                           double throughput_power,
+                                           double rtt_gradient_threshold,
+                                           double delay_gradient_negative_bound)
+    : PccBitrateController(initial_conversion_factor,
+                           initial_dynamic_boundary,
+                           dynamic_boundary_increment,
+                           std::make_unique<ModifiedVivaceUtilityFunction>(
+                               rtt_gradient_coefficient,
+                               loss_coefficient,
+                               throughput_coefficient,
+                               throughput_power,
+                               rtt_gradient_threshold,
+                               delay_gradient_negative_bound)) {}
+
+PccBitrateController::PccBitrateController(
+    double initial_conversion_factor,
+    double initial_dynamic_boundary,
+    double dynamic_boundary_increment,
+    std::unique_ptr<PccUtilityFunctionInterface> utility_function)
+    : consecutive_boundary_adjustments_number_(0),
+      initial_dynamic_boundary_(initial_dynamic_boundary),
+      dynamic_boundary_increment_(dynamic_boundary_increment),
+      utility_function_(std::move(utility_function)),
+      step_size_adjustments_number_(0),
+      initial_conversion_factor_(initial_conversion_factor) {}
+
+PccBitrateController::~PccBitrateController() = default;
+
+double PccBitrateController::ComputeStepSize(double utility_gradient) {
+  // Computes number of consecutive step size adjustments.
+  if (utility_gradient > 0) {
+    step_size_adjustments_number_ =
+        std::max<int64_t>(step_size_adjustments_number_ + 1, 1);
+  } else if (utility_gradient < 0) {
+    step_size_adjustments_number_ =
+        std::min<int64_t>(step_size_adjustments_number_ - 1, -1);
+  } else {
+    step_size_adjustments_number_ = 0;
+  }
+  // Computes step size amplifier.
+  int64_t step_size_amplifier = 1;
+  if (std::abs(step_size_adjustments_number_) <= 3) {
+    step_size_amplifier =
+        std::max<int64_t>(std::abs(step_size_adjustments_number_), 1);
+  } else {
+    step_size_amplifier = 2 * std::abs(step_size_adjustments_number_) - 3;
+  }
+  return step_size_amplifier * initial_conversion_factor_;
+}
+
+double PccBitrateController::ApplyDynamicBoundary(double rate_change,
+                                                  double bitrate) {
+  double rate_change_abs = std::abs(rate_change);
+  int64_t rate_change_sign = (rate_change > 0) ? 1 : -1;
+  if (consecutive_boundary_adjustments_number_ * rate_change_sign < 0) {
+    consecutive_boundary_adjustments_number_ = 0;
+  }
+  double dynamic_change_boundary =
+      initial_dynamic_boundary_ +
+      std::abs(consecutive_boundary_adjustments_number_) *
+          dynamic_boundary_increment_;
+  double boundary = bitrate * dynamic_change_boundary;
+  if (rate_change_abs > boundary) {
+    consecutive_boundary_adjustments_number_ += rate_change_sign;
+    return boundary * rate_change_sign;
+  }
+  // Rate change smaller than boundary. Reset boundary to the smallest possible
+  // that would allow the change.
+  while (rate_change_abs <= boundary &&
+         consecutive_boundary_adjustments_number_ * rate_change_sign > 0) {
+    consecutive_boundary_adjustments_number_ -= rate_change_sign;
+    dynamic_change_boundary =
+        initial_dynamic_boundary_ +
+        std::abs(consecutive_boundary_adjustments_number_) *
+            dynamic_boundary_increment_;
+    boundary = bitrate * dynamic_change_boundary;
+  }
+  consecutive_boundary_adjustments_number_ += rate_change_sign;
+  return rate_change;
+}
+
+absl::optional<DataRate>
+PccBitrateController::ComputeRateUpdateForSlowStartMode(
+    const PccMonitorInterval& monitor_interval) {
+  double utility_value = utility_function_->Compute(monitor_interval);
+  if (previous_utility_.has_value() && utility_value <= previous_utility_) {
+    return absl::nullopt;
+  }
+  previous_utility_ = utility_value;
+  return monitor_interval.GetTargetSendingRate();
+}
+
+DataRate PccBitrateController::ComputeRateUpdateForOnlineLearningMode(
+    const std::vector<PccMonitorInterval>& intervals,
+    DataRate bandwith_estimate) {
+  double first_utility = utility_function_->Compute(intervals[0]);
+  double second_utility = utility_function_->Compute(intervals[1]);
+  double first_bitrate_bps = intervals[0].GetTargetSendingRate().bps();
+  double second_bitrate_bps = intervals[1].GetTargetSendingRate().bps();
+  double gradient = (first_utility - second_utility) /
+                    (first_bitrate_bps - second_bitrate_bps);
+  double rate_change_bps = gradient * ComputeStepSize(gradient);  // delta_r
+  rate_change_bps =
+      ApplyDynamicBoundary(rate_change_bps, bandwith_estimate.bps());
+  return DataRate::BitsPerSec(
+      std::max(0.0, bandwith_estimate.bps() + rate_change_bps));
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.h b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.h
new file mode 100644
index 0000000000..fadeea1b55
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_BITRATE_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_BITRATE_CONTROLLER_H_
+
+#include <stdint.h>
+
+#include <memory>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+#include "modules/congestion_controller/pcc/utility_function.h"
+
+namespace webrtc {
+namespace pcc {
+
+class PccBitrateController {
+ public:
+  PccBitrateController(double initial_conversion_factor,
+                       double initial_dynamic_boundary,
+                       double dynamic_boundary_increment,
+                       double rtt_gradient_coefficient,
+                       double loss_coefficient,
+                       double throughput_coefficient,
+                       double throughput_power,
+                       double rtt_gradient_threshold,
+                       double delay_gradient_negative_bound);
+
+  PccBitrateController(
+      double initial_conversion_factor,
+      double initial_dynamic_boundary,
+      double dynamic_boundary_increment,
+      std::unique_ptr<PccUtilityFunctionInterface> utility_function);
+
+  absl::optional<DataRate> ComputeRateUpdateForSlowStartMode(
+      const PccMonitorInterval& monitor_interval);
+
+  DataRate ComputeRateUpdateForOnlineLearningMode(
+      const std::vector<PccMonitorInterval>& block,
+      DataRate bandwidth_estimate);
+
+  ~PccBitrateController();
+
+ private:
+  double ApplyDynamicBoundary(double rate_change, double bitrate);
+  double ComputeStepSize(double utility_gradient);
+
+  // Dynamic boundary variables:
+  int64_t consecutive_boundary_adjustments_number_;
+  const double initial_dynamic_boundary_;
+  const double dynamic_boundary_increment_;
+
+  const std::unique_ptr<PccUtilityFunctionInterface> utility_function_;
+  // Step Size variables:
+  int64_t step_size_adjustments_number_;
+  const double initial_conversion_factor_;
+
+  absl::optional<double> previous_utility_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_BITRATE_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller_unittest.cc
new file mode 100644
index 0000000000..957d99b1de
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/bitrate_controller_unittest.cc
@@ -0,0 +1,303 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/bitrate_controller.h"
+
+#include <memory>
+#include <utility>
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+constexpr double kInitialConversionFactor = 1;
+constexpr double kInitialDynamicBoundary = 0.05;
+constexpr double kDynamicBoundaryIncrement = 0.1;
+
+constexpr double kDelayGradientCoefficient = 900;
+constexpr double kLossCoefficient = 11.35;
+constexpr double kThroughputCoefficient = 500 * 1000;
+constexpr double kThroughputPower = 0.99;
+constexpr double kDelayGradientThreshold = 0.01;
+constexpr double kDelayGradientNegativeBound = 10;
+
+const DataRate kTargetSendingRate = DataRate::KilobitsPerSec(300);
+const double kEpsilon = 0.05;
+const Timestamp kStartTime = Timestamp::Micros(0);
+const TimeDelta kPacketsDelta = TimeDelta::Millis(1);
+const TimeDelta kIntervalDuration = TimeDelta::Millis(1000);
+const TimeDelta kDefaultRtt = TimeDelta::Millis(1000);
+const DataSize kDefaultDataSize = DataSize::Bytes(100);
+
+std::vector<PacketResult> CreatePacketResults(
+    const std::vector<Timestamp>& packets_send_times,
+    const std::vector<Timestamp>& packets_received_times = {},
+    const std::vector<DataSize>& packets_sizes = {}) {
+  std::vector<PacketResult> packet_results;
+  PacketResult packet_result;
+  SentPacket sent_packet;
+  for (size_t i = 0; i < packets_send_times.size(); ++i) {
+    sent_packet.send_time = packets_send_times[i];
+    if (packets_sizes.empty()) {
+      sent_packet.size = kDefaultDataSize;
+    } else {
+      sent_packet.size = packets_sizes[i];
+    }
+    packet_result.sent_packet = sent_packet;
+    if (packets_received_times.empty()) {
+      packet_result.receive_time = packets_send_times[i] + kDefaultRtt;
+    } else {
+      packet_result.receive_time = packets_received_times[i];
+    }
+    packet_results.push_back(packet_result);
+  }
+  return packet_results;
+}
+
+class MockUtilityFunction : public PccUtilityFunctionInterface {
+ public:
+  MOCK_METHOD(double,
+              Compute,
+              (const PccMonitorInterval& monitor_interval),
+              (const, override));
+};
+
+}  // namespace
+
+TEST(PccBitrateControllerTest, IncreaseRateWhenNoChangesForTestBitrates) {
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, kDelayGradientCoefficient, kLossCoefficient,
+      kThroughputCoefficient, kThroughputPower, kDelayGradientThreshold,
+      kDelayGradientNegativeBound);
+  VivaceUtilityFunction utility_function(
+      kDelayGradientCoefficient, kLossCoefficient, kThroughputCoefficient,
+      kThroughputPower, kDelayGradientThreshold, kDelayGradientNegativeBound);
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + kPacketsDelta,
+                           kStartTime + kIntervalDuration + kPacketsDelta,
+                           kStartTime + 3 * kIntervalDuration},
+                          {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + kPacketsDelta,
+                           kStartTime + kIntervalDuration + kPacketsDelta,
+                           kStartTime + 3 * kIntervalDuration},
+                          {}, {}));
+  // For both of the monitor intervals there were no change in rtt gradient
+  // and in packet loss. Since the only difference is in the sending rate,
+  // the higher sending rate should be chosen by congestion controller.
+  EXPECT_GT(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps());
+}
+
+TEST(PccBitrateControllerTest, NoChangesWhenUtilityFunctionDoesntChange) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(2)
+      .WillOnce(::testing::Return(100))
+      .WillOnce(::testing::Return(100));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  // Because we don't have any packets inside of monitor intervals, utility
+  // function should be zero for both of them and the sending rate should not
+  // change.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps());
+}
+
+TEST(PccBitrateControllerTest, NoBoundaryWhenSmallGradient) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstMonitorIntervalUtility = 0;
+  const double kSecondMonitorIntervalUtility =
+      2 * kTargetSendingRate.bps() * kEpsilon;
+
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(2)
+      .WillOnce(::testing::Return(kFirstMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kSecondMonitorIntervalUtility));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+
+  double gradient =
+      (kFirstMonitorIntervalUtility - kSecondMonitorIntervalUtility) /
+      (kTargetSendingRate.bps() * 2 * kEpsilon);
+  // When the gradient is small we don't hit the dynamic boundary.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() + kInitialConversionFactor * gradient);
+}
+
+TEST(PccBitrateControllerTest, FaceBoundaryWhenLargeGradient) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstMonitorIntervalUtility = 0;
+  const double kSecondMonitorIntervalUtility =
+      10 * kInitialDynamicBoundary * kTargetSendingRate.bps() * 2 *
+      kTargetSendingRate.bps() * kEpsilon;
+
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(4)
+      .WillOnce(::testing::Return(kFirstMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kSecondMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kFirstMonitorIntervalUtility))
+      .WillOnce(::testing::Return(kSecondMonitorIntervalUtility));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  // The utility function gradient is too big and we hit the dynamic boundary.
+  EXPECT_EQ(bitrate_controller.ComputeRateUpdateForOnlineLearningMode(
+                monitor_block, kTargetSendingRate),
+            kTargetSendingRate * (1 - kInitialDynamicBoundary));
+  // For the second time we hit the dynamic boundary in the same direction, the
+  // boundary should increase.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() *
+                (1 - kInitialDynamicBoundary - kDynamicBoundaryIncrement));
+}
+
+TEST(PccBitrateControllerTest, SlowStartMode) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstUtilityFunction = 1000;
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(4)
+      // For first 3 calls we expect to stay in the SLOW_START mode and double
+      // the sending rate since the utility function increases its value. For
+      // the last call utility function decreases its value, this means that
+      // we should not double the sending rate and exit SLOW_START mode.
+      .WillOnce(::testing::Return(kFirstUtilityFunction))
+      .WillOnce(::testing::Return(kFirstUtilityFunction + 1))
+      .WillOnce(::testing::Return(kFirstUtilityFunction + 2))
+      .WillOnce(::testing::Return(kFirstUtilityFunction + 1));
+
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  std::vector<PccMonitorInterval> monitor_block{PccMonitorInterval(
+      2 * kTargetSendingRate, kStartTime, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      kTargetSendingRate * 2);
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      kTargetSendingRate * 2);
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      kTargetSendingRate * 2);
+  EXPECT_EQ(
+      bitrate_controller.ComputeRateUpdateForSlowStartMode(monitor_block[0]),
+      absl::nullopt);
+}
+
+TEST(PccBitrateControllerTest, StepSizeIncrease) {
+  std::unique_ptr<MockUtilityFunction> mock_utility_function =
+      std::make_unique<MockUtilityFunction>();
+  constexpr double kFirstMiUtilityFunction = 0;
+  const double kSecondMiUtilityFunction =
+      2 * kTargetSendingRate.bps() * kEpsilon;
+
+  EXPECT_CALL(*mock_utility_function, Compute(::testing::_))
+      .Times(4)
+      .WillOnce(::testing::Return(kFirstMiUtilityFunction))
+      .WillOnce(::testing::Return(kSecondMiUtilityFunction))
+      .WillOnce(::testing::Return(kFirstMiUtilityFunction))
+      .WillOnce(::testing::Return(kSecondMiUtilityFunction));
+  std::vector<PccMonitorInterval> monitor_block{
+      PccMonitorInterval(kTargetSendingRate * (1 + kEpsilon), kStartTime,
+                         kIntervalDuration),
+      PccMonitorInterval(kTargetSendingRate * (1 - kEpsilon),
+                         kStartTime + kIntervalDuration, kIntervalDuration)};
+  // To complete collecting feedback within monitor intervals.
+  monitor_block[0].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+  monitor_block[1].OnPacketsFeedback(
+      CreatePacketResults({kStartTime + 3 * kIntervalDuration}, {}, {}));
+
+  double gradient = (kFirstMiUtilityFunction - kSecondMiUtilityFunction) /
+                    (kTargetSendingRate.bps() * 2 * kEpsilon);
+  PccBitrateController bitrate_controller(
+      kInitialConversionFactor, kInitialDynamicBoundary,
+      kDynamicBoundaryIncrement, std::move(mock_utility_function));
+  // If we are moving in the same direction - the step size should increase.
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() + kInitialConversionFactor * gradient);
+  EXPECT_EQ(bitrate_controller
+                .ComputeRateUpdateForOnlineLearningMode(monitor_block,
+                                                        kTargetSendingRate)
+                .bps(),
+            kTargetSendingRate.bps() + 2 * kInitialConversionFactor * gradient);
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.cc
new file mode 100644
index 0000000000..de1e2d5e69
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.cc
@@ -0,0 +1,135 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+
+#include <stddef.h>
+
+#include <cmath>
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace pcc {
+
+PccMonitorInterval::PccMonitorInterval(DataRate target_sending_rate,
+                                       Timestamp start_time,
+                                       TimeDelta duration)
+    : target_sending_rate_(target_sending_rate),
+      start_time_(start_time),
+      interval_duration_(duration),
+      received_packets_size_(DataSize::Zero()),
+      feedback_collection_done_(false) {}
+
+PccMonitorInterval::~PccMonitorInterval() = default;
+
+PccMonitorInterval::PccMonitorInterval(const PccMonitorInterval& other) =
+    default;
+
+void PccMonitorInterval::OnPacketsFeedback(
+    const std::vector<PacketResult>& packets_results) {
+  for (const PacketResult& packet_result : packets_results) {
+    if (packet_result.sent_packet.send_time <= start_time_) {
+      continue;
+    }
+    // Here we assume that if some packets are reordered with packets sent
+    // after the end of the monitor interval, then they are lost. (Otherwise
+    // it is not clear how long should we wait for packets feedback to arrive).
+    if (packet_result.sent_packet.send_time >
+        start_time_ + interval_duration_) {
+      feedback_collection_done_ = true;
+      return;
+    }
+    if (!packet_result.IsReceived()) {
+      lost_packets_sent_time_.push_back(packet_result.sent_packet.send_time);
+    } else {
+      received_packets_.push_back(
+          {packet_result.receive_time - packet_result.sent_packet.send_time,
+           packet_result.sent_packet.send_time});
+      received_packets_size_ += packet_result.sent_packet.size;
+    }
+  }
+}
+
+// For the formula used in computations see formula for "slope" in the second
+// method:
+// https://www.johndcook.com/blog/2008/10/20/comparing-two-ways-to-fit-a-line-to-data/
+double PccMonitorInterval::ComputeDelayGradient(
+    double delay_gradient_threshold) const {
+  // Early return to prevent division by 0 in case all packets are sent at the
+  // same time.
+  if (received_packets_.empty() || received_packets_.front().sent_time ==
+                                       received_packets_.back().sent_time) {
+    return 0;
+  }
+  double sum_times = 0;
+  for (const ReceivedPacket& packet : received_packets_) {
+    double time_delta_us =
+        (packet.sent_time - received_packets_[0].sent_time).us();
+    sum_times += time_delta_us;
+  }
+  double sum_squared_scaled_time_deltas = 0;
+  double sum_scaled_time_delta_dot_delay = 0;
+  for (const ReceivedPacket& packet : received_packets_) {
+    double time_delta_us =
+        (packet.sent_time - received_packets_[0].sent_time).us();
+    double delay = packet.delay.us();
+    double scaled_time_delta_us =
+        time_delta_us - sum_times / received_packets_.size();
+    sum_squared_scaled_time_deltas +=
+        scaled_time_delta_us * scaled_time_delta_us;
+    sum_scaled_time_delta_dot_delay += scaled_time_delta_us * delay;
+  }
+  double rtt_gradient =
+      sum_scaled_time_delta_dot_delay / sum_squared_scaled_time_deltas;
+  if (std::abs(rtt_gradient) < delay_gradient_threshold)
+    rtt_gradient = 0;
+  return rtt_gradient;
+}
+
+bool PccMonitorInterval::IsFeedbackCollectionDone() const {
+  return feedback_collection_done_;
+}
+
+Timestamp PccMonitorInterval::GetEndTime() const {
+  return start_time_ + interval_duration_;
+}
+
+double PccMonitorInterval::GetLossRate() const {
+  size_t packets_lost = lost_packets_sent_time_.size();
+  size_t packets_received = received_packets_.size();
+  if (packets_lost == 0)
+    return 0;
+  return static_cast<double>(packets_lost) / (packets_lost + packets_received);
+}
+
+DataRate PccMonitorInterval::GetTargetSendingRate() const {
+  return target_sending_rate_;
+}
+
+DataRate PccMonitorInterval::GetTransmittedPacketsRate() const {
+  if (received_packets_.empty()) {
+    return target_sending_rate_;
+  }
+  Timestamp receive_time_of_first_packet =
+      received_packets_.front().sent_time + received_packets_.front().delay;
+  Timestamp receive_time_of_last_packet =
+      received_packets_.back().sent_time + received_packets_.back().delay;
+  if (receive_time_of_first_packet == receive_time_of_last_packet) {
+    RTC_LOG(LS_WARNING)
+        << "All packets in monitor interval were received at the same time.";
+    return target_sending_rate_;
+  }
+  return received_packets_size_ /
+         (receive_time_of_last_packet - receive_time_of_first_packet);
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.h b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.h
new file mode 100644
index 0000000000..51bd0f068a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_MONITOR_INTERVAL_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_MONITOR_INTERVAL_H_
+
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+namespace pcc {
+
+// PCC divides time into consecutive monitor intervals which are used to test
+// consequences for performance of sending at a certain rate.
+class PccMonitorInterval {
+ public:
+  PccMonitorInterval(DataRate target_sending_rate,
+                     Timestamp start_time,
+                     TimeDelta duration);
+  ~PccMonitorInterval();
+  PccMonitorInterval(const PccMonitorInterval& other);
+  void OnPacketsFeedback(const std::vector<PacketResult>& packets_results);
+  // Returns true if got complete information about packets.
+  // Notice, this only happens when received feedback about the first packet
+  // which were sent after the end of the monitor interval. If such event
+  // doesn't occur, we don't mind anyway and stay in the same state.
+  bool IsFeedbackCollectionDone() const;
+  Timestamp GetEndTime() const;
+
+  double GetLossRate() const;
+  // Estimates the gradient using linear regression on the 2-dimensional
+  // dataset (sampled packets delay, time of sampling).
+  double ComputeDelayGradient(double delay_gradient_threshold) const;
+  DataRate GetTargetSendingRate() const;
+  // How fast receiving side gets packets.
+  DataRate GetTransmittedPacketsRate() const;
+
+ private:
+  struct ReceivedPacket {
+    TimeDelta delay;
+    Timestamp sent_time;
+  };
+  // Target bitrate used to generate and pace the outgoing packets.
+  // Actually sent bitrate might not match the target exactly.
+  DataRate target_sending_rate_;
+  // Start time is not included into interval while end time is included.
+  Timestamp start_time_;
+  TimeDelta interval_duration_;
+  // Vectors below updates while receiving feedback.
+  std::vector<ReceivedPacket> received_packets_;
+  std::vector<Timestamp> lost_packets_sent_time_;
+  DataSize received_packets_size_;
+  bool feedback_collection_done_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_MONITOR_INTERVAL_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval_unittest.cc
new file mode 100644
index 0000000000..aaff57bd2a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/monitor_interval_unittest.cc
@@ -0,0 +1,190 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+
+#include <stddef.h>
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+const DataRate kTargetSendingRate = DataRate::KilobitsPerSec(300);
+const Timestamp kStartTime = Timestamp::Micros(0);
+const TimeDelta kPacketsDelta = TimeDelta::Millis(1);
+const TimeDelta kIntervalDuration = TimeDelta::Millis(100);
+const TimeDelta kDefaultDelay = TimeDelta::Millis(100);
+const DataSize kDefaultPacketSize = DataSize::Bytes(100);
+constexpr double kDelayGradientThreshold = 0.01;
+
+std::vector<PacketResult> CreatePacketResults(
+    const std::vector<Timestamp>& packets_send_times,
+    const std::vector<Timestamp>& packets_received_times = {},
+    const std::vector<DataSize>& packets_sizes = {}) {
+  std::vector<PacketResult> packet_results;
+  for (size_t i = 0; i < packets_send_times.size(); ++i) {
+    SentPacket sent_packet;
+    sent_packet.send_time = packets_send_times[i];
+    if (packets_sizes.empty()) {
+      sent_packet.size = kDefaultPacketSize;
+    } else {
+      sent_packet.size = packets_sizes[i];
+    }
+    PacketResult packet_result;
+    packet_result.sent_packet = sent_packet;
+    if (packets_received_times.empty()) {
+      packet_result.receive_time = packets_send_times[i] + kDefaultDelay;
+    } else {
+      packet_result.receive_time = packets_received_times[i];
+    }
+    packet_results.push_back(packet_result);
+  }
+  return packet_results;
+}
+
+}  // namespace
+
+TEST(PccMonitorIntervalTest, InitialValuesAreEqualToOnesSetInConstructor) {
+  PccMonitorInterval interval{kTargetSendingRate, kStartTime,
+                              kIntervalDuration};
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), false);
+  EXPECT_EQ(interval.GetEndTime(), kStartTime + kIntervalDuration);
+  EXPECT_EQ(interval.GetTargetSendingRate(), kTargetSendingRate);
+}
+
+TEST(PccMonitorIntervalTest, IndicatesDoneWhenFeedbackReceivedAfterInterval) {
+  PccMonitorInterval interval{kTargetSendingRate, kStartTime,
+                              kIntervalDuration};
+  interval.OnPacketsFeedback(CreatePacketResults({kStartTime}));
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), false);
+  interval.OnPacketsFeedback(
+      CreatePacketResults({kStartTime, kStartTime + kIntervalDuration}));
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), false);
+  interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kIntervalDuration, kStartTime + 2 * kIntervalDuration}));
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), true);
+}
+
+TEST(PccMonitorIntervalTest, LossRateIsOneThirdIfLostOnePacketOutOfThree) {
+  PccMonitorInterval interval{kTargetSendingRate, kStartTime,
+                              kIntervalDuration};
+  std::vector<Timestamp> start_times = {
+      kStartTime, kStartTime + 0.1 * kIntervalDuration,
+      kStartTime + 0.5 * kIntervalDuration, kStartTime + kIntervalDuration,
+      kStartTime + 2 * kIntervalDuration};
+  std::vector<Timestamp> end_times = {
+      kStartTime + 2 * kIntervalDuration, kStartTime + 2 * kIntervalDuration,
+      Timestamp::PlusInfinity(), kStartTime + 2 * kIntervalDuration,
+      kStartTime + 4 * kIntervalDuration};
+  std::vector<DataSize> packet_sizes = {
+      kDefaultPacketSize, 2 * kDefaultPacketSize, 3 * kDefaultPacketSize,
+      4 * kDefaultPacketSize, 5 * kDefaultPacketSize};
+  std::vector<PacketResult> packet_results =
+      CreatePacketResults(start_times, end_times, packet_sizes);
+  interval.OnPacketsFeedback(packet_results);
+  EXPECT_EQ(interval.IsFeedbackCollectionDone(), true);
+
+  EXPECT_DOUBLE_EQ(interval.GetLossRate(), 1. / 3);
+}
+
+TEST(PccMonitorIntervalTest, DelayGradientIsZeroIfNoChangeInPacketDelay) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 3 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + 2 * kPacketsDelta,
+       Timestamp::PlusInfinity()},
+      {}));
+  // Delay gradient should be zero, because both received packets have the
+  // same one way delay.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+TEST(PccMonitorIntervalTest,
+     DelayGradientIsZeroWhenOnePacketSentInMonitorInterval) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, kStartTime + 3 * kIntervalDuration}, {}));
+  // Only one received packet belongs to the monitor_interval, delay gradient
+  // should be zero in this case.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+TEST(PccMonitorIntervalTest, DelayGradientIsOne) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 3 * kPacketsDelta, kStartTime + 3 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + 4 * kPacketsDelta + kDefaultDelay,
+       kStartTime + 3 * kIntervalDuration},
+      {}));
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 1);
+}
+
+TEST(PccMonitorIntervalTest, DelayGradientIsMinusOne) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 5 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay, kStartTime + 3 * kIntervalDuration},
+      {}));
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), -1);
+}
+
+TEST(PccMonitorIntervalTest,
+     DelayGradientIsZeroIfItSmallerWhenGradientThreshold) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + kPacketsDelta,
+       kStartTime + 102 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + kPacketsDelta,
+       kStartTime + 3 * kIntervalDuration},
+      {}));
+  // Delay gradient is less than 0.01 hence should be treated as zero.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+TEST(PccMonitorIntervalTest,
+     DelayGradientIsZeroWhenAllPacketsSentAtTheSameTime) {
+  PccMonitorInterval monitor_interval(kTargetSendingRate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + kPacketsDelta,
+       kStartTime + kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + kPacketsDelta,
+       kStartTime + 3 * kIntervalDuration},
+      {}));
+  // If all packets were sent at the same time, then delay gradient should be
+  // zero.
+  EXPECT_DOUBLE_EQ(
+      monitor_interval.ComputeDelayGradient(kDelayGradientThreshold), 0);
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.cc
new file mode 100644
index 0000000000..c35c6e8ab2
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/pcc_factory.h"
+
+#include <memory>
+
+#include "modules/congestion_controller/pcc/pcc_network_controller.h"
+
+namespace webrtc {
+
+PccNetworkControllerFactory::PccNetworkControllerFactory() {}
+
+std::unique_ptr<NetworkControllerInterface> PccNetworkControllerFactory::Create(
+    NetworkControllerConfig config) {
+  return std::make_unique<pcc::PccNetworkController>(config);
+}
+
+TimeDelta PccNetworkControllerFactory::GetProcessInterval() const {
+  return TimeDelta::PlusInfinity();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.h b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.h
new file mode 100644
index 0000000000..bb70d7a499
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_factory.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_PCC_FACTORY_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_PCC_FACTORY_H_
+
+#include <memory>
+
+#include "api/transport/network_control.h"
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+class PccNetworkControllerFactory : public NetworkControllerFactoryInterface {
+ public:
+  PccNetworkControllerFactory();
+  std::unique_ptr<NetworkControllerInterface> Create(
+      NetworkControllerConfig config) override;
+  TimeDelta GetProcessInterval() const override;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_PCC_FACTORY_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.cc
new file mode 100644
index 0000000000..8653470955
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.cc
@@ -0,0 +1,391 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/pcc_network_controller.h"
+
+#include <algorithm>
+
+#include "absl/types/optional.h"
+#include "api/units/data_size.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace pcc {
+namespace {
+constexpr int64_t kInitialRttMs = 200;
+constexpr int64_t kInitialBandwidthKbps = 300;
+constexpr double kMonitorIntervalDurationRatio = 1;
+constexpr double kDefaultSamplingStep = 0.05;
+constexpr double kTimeoutRatio = 2;
+constexpr double kAlphaForRtt = 0.9;
+constexpr double kSlowStartModeIncrease = 1.5;
+
+constexpr double kAlphaForPacketInterval = 0.9;
+constexpr int64_t kMinPacketsNumberPerInterval = 20;
+const TimeDelta kMinDurationOfMonitorInterval = TimeDelta::Millis(50);
+const TimeDelta kStartupDuration = TimeDelta::Millis(500);
+constexpr double kMinRateChangeBps = 4000;
+constexpr DataRate kMinRateHaveMultiplicativeRateChange = DataRate::BitsPerSec(
+    static_cast<int64_t>(kMinRateChangeBps / kDefaultSamplingStep));
+
+// Bitrate controller constants.
+constexpr double kInitialConversionFactor = 5;
+constexpr double kInitialDynamicBoundary = 0.1;
+constexpr double kDynamicBoundaryIncrement = 0.1;
+// Utility function parameters.
+constexpr double kRttGradientCoefficientBps = 0.005;
+constexpr double kLossCoefficientBps = 10;
+constexpr double kThroughputCoefficient = 0.001;
+constexpr double kThroughputPower = 0.9;
+constexpr double kRttGradientThreshold = 0.01;
+constexpr double kDelayGradientNegativeBound = 0.1;
+
+constexpr int64_t kNumberOfPacketsToKeep = 20;
+const uint64_t kRandomSeed = 100;
+}  // namespace
+
+PccNetworkController::PccNetworkController(NetworkControllerConfig config)
+    : start_time_(Timestamp::PlusInfinity()),
+      last_sent_packet_time_(Timestamp::PlusInfinity()),
+      smoothed_packets_sending_interval_(TimeDelta::Zero()),
+      mode_(Mode::kStartup),
+      default_bandwidth_(DataRate::KilobitsPerSec(kInitialBandwidthKbps)),
+      bandwidth_estimate_(default_bandwidth_),
+      rtt_tracker_(TimeDelta::Millis(kInitialRttMs), kAlphaForRtt),
+      monitor_interval_timeout_(TimeDelta::Millis(kInitialRttMs) *
+                                kTimeoutRatio),
+      monitor_interval_length_strategy_(MonitorIntervalLengthStrategy::kFixed),
+      monitor_interval_duration_ratio_(kMonitorIntervalDurationRatio),
+      sampling_step_(kDefaultSamplingStep),
+      monitor_interval_timeout_ratio_(kTimeoutRatio),
+      min_packets_number_per_interval_(kMinPacketsNumberPerInterval),
+      bitrate_controller_(kInitialConversionFactor,
+                          kInitialDynamicBoundary,
+                          kDynamicBoundaryIncrement,
+                          kRttGradientCoefficientBps,
+                          kLossCoefficientBps,
+                          kThroughputCoefficient,
+                          kThroughputPower,
+                          kRttGradientThreshold,
+                          kDelayGradientNegativeBound),
+      monitor_intervals_duration_(TimeDelta::Zero()),
+      complete_feedback_monitor_interval_number_(0),
+      random_generator_(kRandomSeed) {
+  if (config.constraints.starting_rate) {
+    default_bandwidth_ = *config.constraints.starting_rate;
+    bandwidth_estimate_ = default_bandwidth_;
+  }
+}
+
+PccNetworkController::~PccNetworkController() {}
+
+NetworkControlUpdate PccNetworkController::CreateRateUpdate(
+    Timestamp at_time) const {
+  DataRate sending_rate = DataRate::Zero();
+  if (monitor_intervals_.empty() ||
+      (monitor_intervals_.size() >= monitor_intervals_bitrates_.size() &&
+       at_time >= monitor_intervals_.back().GetEndTime())) {
+    sending_rate = bandwidth_estimate_;
+  } else {
+    sending_rate = monitor_intervals_.back().GetTargetSendingRate();
+  }
+  // Set up config when sending rate is computed.
+  NetworkControlUpdate update;
+
+  // Set up target rate to encoder.
+  TargetTransferRate target_rate_msg;
+  target_rate_msg.at_time = at_time;
+  target_rate_msg.network_estimate.at_time = at_time;
+  target_rate_msg.network_estimate.round_trip_time = rtt_tracker_.GetRtt();
+  // TODO(koloskova): Add correct estimate.
+  target_rate_msg.network_estimate.loss_rate_ratio = 0;
+  target_rate_msg.network_estimate.bwe_period =
+      monitor_interval_duration_ratio_ * rtt_tracker_.GetRtt();
+
+  target_rate_msg.target_rate = sending_rate;
+  update.target_rate = target_rate_msg;
+
+  // Set up pacing/padding target rate.
+  PacerConfig pacer_config;
+  pacer_config.at_time = at_time;
+  pacer_config.time_window = TimeDelta::Millis(1);
+  pacer_config.data_window = sending_rate * pacer_config.time_window;
+  pacer_config.pad_window = sending_rate * pacer_config.time_window;
+
+  update.pacer_config = pacer_config;
+  return update;
+}
+
+NetworkControlUpdate PccNetworkController::OnSentPacket(SentPacket msg) {
+  // Start new monitor interval if previous has finished.
+  // Monitor interval is initialized in OnProcessInterval function.
+  if (start_time_.IsInfinite()) {
+    start_time_ = msg.send_time;
+    monitor_intervals_duration_ = kStartupDuration;
+    monitor_intervals_bitrates_ = {bandwidth_estimate_};
+    monitor_intervals_.emplace_back(bandwidth_estimate_, msg.send_time,
+                                    monitor_intervals_duration_);
+    complete_feedback_monitor_interval_number_ = 0;
+  }
+  if (last_sent_packet_time_.IsFinite()) {
+    smoothed_packets_sending_interval_ =
+        (msg.send_time - last_sent_packet_time_) * kAlphaForPacketInterval +
+        (1 - kAlphaForPacketInterval) * smoothed_packets_sending_interval_;
+  }
+  last_sent_packet_time_ = msg.send_time;
+  if (!monitor_intervals_.empty() &&
+      msg.send_time >= monitor_intervals_.back().GetEndTime() &&
+      monitor_intervals_bitrates_.size() > monitor_intervals_.size()) {
+    // Start new monitor interval.
+    monitor_intervals_.emplace_back(
+        monitor_intervals_bitrates_[monitor_intervals_.size()], msg.send_time,
+        monitor_intervals_duration_);
+  }
+  if (IsTimeoutExpired(msg.send_time)) {
+    DataSize received_size = DataSize::Zero();
+    for (size_t i = 1; i < last_received_packets_.size(); ++i) {
+      received_size += last_received_packets_[i].sent_packet.size;
+    }
+    TimeDelta sending_time = TimeDelta::Zero();
+    if (last_received_packets_.size() > 0)
+      sending_time = last_received_packets_.back().receive_time -
+                     last_received_packets_.front().receive_time;
+    DataRate receiving_rate = bandwidth_estimate_;
+    if (sending_time > TimeDelta::Zero())
+      receiving_rate = received_size / sending_time;
+    bandwidth_estimate_ =
+        std::min<DataRate>(bandwidth_estimate_ * 0.5, receiving_rate);
+    if (mode_ == Mode::kSlowStart)
+      mode_ = Mode::kOnlineLearning;
+  }
+  if (mode_ == Mode::kStartup &&
+      msg.send_time - start_time_ >= kStartupDuration) {
+    DataSize received_size = DataSize::Zero();
+    for (size_t i = 1; i < last_received_packets_.size(); ++i) {
+      received_size += last_received_packets_[i].sent_packet.size;
+    }
+    TimeDelta sending_time = TimeDelta::Zero();
+    if (last_received_packets_.size() > 0)
+      sending_time = last_received_packets_.back().receive_time -
+                     last_received_packets_.front().receive_time;
+    DataRate receiving_rate = bandwidth_estimate_;
+    if (sending_time > TimeDelta::Zero())
+      receiving_rate = received_size / sending_time;
+    bandwidth_estimate_ = receiving_rate;
+    monitor_intervals_.clear();
+    mode_ = Mode::kSlowStart;
+    monitor_intervals_duration_ = ComputeMonitorIntervalsDuration();
+    monitor_intervals_bitrates_ = {bandwidth_estimate_};
+    monitor_intervals_.emplace_back(bandwidth_estimate_, msg.send_time,
+                                    monitor_intervals_duration_);
+    bandwidth_estimate_ = bandwidth_estimate_ * (1 / kSlowStartModeIncrease);
+    complete_feedback_monitor_interval_number_ = 0;
+    return CreateRateUpdate(msg.send_time);
+  }
+  if (IsFeedbackCollectionDone() || IsTimeoutExpired(msg.send_time)) {
+    // Creating new monitor intervals.
+    monitor_intervals_.clear();
+    monitor_interval_timeout_ =
+        rtt_tracker_.GetRtt() * monitor_interval_timeout_ratio_;
+    monitor_intervals_duration_ = ComputeMonitorIntervalsDuration();
+    complete_feedback_monitor_interval_number_ = 0;
+    // Compute bitrates and start first monitor interval.
+    if (mode_ == Mode::kSlowStart) {
+      monitor_intervals_bitrates_ = {kSlowStartModeIncrease *
+                                     bandwidth_estimate_};
+      monitor_intervals_.emplace_back(
+          kSlowStartModeIncrease * bandwidth_estimate_, msg.send_time,
+          monitor_intervals_duration_);
+    } else {
+      RTC_DCHECK(mode_ == Mode::kOnlineLearning || mode_ == Mode::kDoubleCheck);
+      monitor_intervals_.clear();
+      int64_t sign = 2 * (random_generator_.Rand(0, 1) % 2) - 1;
+      RTC_DCHECK_GE(sign, -1);
+      RTC_DCHECK_LE(sign, 1);
+      if (bandwidth_estimate_ >= kMinRateHaveMultiplicativeRateChange) {
+        monitor_intervals_bitrates_ = {
+            bandwidth_estimate_ * (1 + sign * sampling_step_),
+            bandwidth_estimate_ * (1 - sign * sampling_step_)};
+      } else {
+        monitor_intervals_bitrates_ = {
+            DataRate::BitsPerSec(std::max<double>(
+                bandwidth_estimate_.bps() + sign * kMinRateChangeBps, 0)),
+            DataRate::BitsPerSec(std::max<double>(
+                bandwidth_estimate_.bps() - sign * kMinRateChangeBps, 0))};
+      }
+      monitor_intervals_.emplace_back(monitor_intervals_bitrates_[0],
+                                      msg.send_time,
+                                      monitor_intervals_duration_);
+    }
+  }
+  return CreateRateUpdate(msg.send_time);
+}
+
+TimeDelta PccNetworkController::ComputeMonitorIntervalsDuration() const {
+  TimeDelta monitor_intervals_duration = TimeDelta::Zero();
+  if (monitor_interval_length_strategy_ ==
+      MonitorIntervalLengthStrategy::kAdaptive) {
+    monitor_intervals_duration = std::max(
+        rtt_tracker_.GetRtt() * monitor_interval_duration_ratio_,
+        smoothed_packets_sending_interval_ * min_packets_number_per_interval_);
+  } else {
+    RTC_DCHECK(monitor_interval_length_strategy_ ==
+               MonitorIntervalLengthStrategy::kFixed);
+    monitor_intervals_duration =
+        smoothed_packets_sending_interval_ * min_packets_number_per_interval_;
+  }
+  monitor_intervals_duration =
+      std::max(kMinDurationOfMonitorInterval, monitor_intervals_duration);
+  return monitor_intervals_duration;
+}
+
+bool PccNetworkController::IsTimeoutExpired(Timestamp current_time) const {
+  if (complete_feedback_monitor_interval_number_ >= monitor_intervals_.size()) {
+    return false;
+  }
+  return current_time -
+             monitor_intervals_[complete_feedback_monitor_interval_number_]
+                 .GetEndTime() >=
+         monitor_interval_timeout_;
+}
+
+bool PccNetworkController::IsFeedbackCollectionDone() const {
+  return complete_feedback_monitor_interval_number_ >=
+         monitor_intervals_bitrates_.size();
+}
+
+NetworkControlUpdate PccNetworkController::OnTransportPacketsFeedback(
+    TransportPacketsFeedback msg) {
+  if (msg.packet_feedbacks.empty())
+    return NetworkControlUpdate();
+  // Save packets to last_received_packets_ array.
+  for (const PacketResult& packet_result : msg.ReceivedWithSendInfo()) {
+    last_received_packets_.push_back(packet_result);
+  }
+  while (last_received_packets_.size() > kNumberOfPacketsToKeep) {
+    last_received_packets_.pop_front();
+  }
+  rtt_tracker_.OnPacketsFeedback(msg.PacketsWithFeedback(), msg.feedback_time);
+  // Skip rate update in case when online learning mode just started, but
+  // corresponding monitor intervals were not started yet.
+  if (mode_ == Mode::kOnlineLearning &&
+      monitor_intervals_bitrates_.size() < 2) {
+    return NetworkControlUpdate();
+  }
+  if (!IsFeedbackCollectionDone() && !monitor_intervals_.empty()) {
+    while (complete_feedback_monitor_interval_number_ <
+           monitor_intervals_.size()) {
+      monitor_intervals_[complete_feedback_monitor_interval_number_]
+          .OnPacketsFeedback(msg.PacketsWithFeedback());
+      if (!monitor_intervals_[complete_feedback_monitor_interval_number_]
+               .IsFeedbackCollectionDone())
+        break;
+      ++complete_feedback_monitor_interval_number_;
+    }
+  }
+  if (IsFeedbackCollectionDone()) {
+    if (mode_ == Mode::kDoubleCheck) {
+      mode_ = Mode::kOnlineLearning;
+    } else if (NeedDoubleCheckMeasurments()) {
+      mode_ = Mode::kDoubleCheck;
+    }
+    if (mode_ != Mode::kDoubleCheck)
+      UpdateSendingRateAndMode();
+  }
+  return NetworkControlUpdate();
+}
+
+bool PccNetworkController::NeedDoubleCheckMeasurments() const {
+  if (mode_ == Mode::kSlowStart) {
+    return false;
+  }
+  double first_loss_rate = monitor_intervals_[0].GetLossRate();
+  double second_loss_rate = monitor_intervals_[1].GetLossRate();
+  DataRate first_bitrate = monitor_intervals_[0].GetTargetSendingRate();
+  DataRate second_bitrate = monitor_intervals_[1].GetTargetSendingRate();
+  if ((first_bitrate.bps() - second_bitrate.bps()) *
+          (first_loss_rate - second_loss_rate) <
+      0) {
+    return true;
+  }
+  return false;
+}
+
+void PccNetworkController::UpdateSendingRateAndMode() {
+  if (monitor_intervals_.empty() || !IsFeedbackCollectionDone()) {
+    return;
+  }
+  if (mode_ == Mode::kSlowStart) {
+    DataRate old_bandwidth_estimate = bandwidth_estimate_;
+    bandwidth_estimate_ =
+        bitrate_controller_
+            .ComputeRateUpdateForSlowStartMode(monitor_intervals_[0])
+            .value_or(bandwidth_estimate_);
+    if (bandwidth_estimate_ <= old_bandwidth_estimate)
+      mode_ = Mode::kOnlineLearning;
+  } else {
+    RTC_DCHECK(mode_ == Mode::kOnlineLearning);
+    bandwidth_estimate_ =
+        bitrate_controller_.ComputeRateUpdateForOnlineLearningMode(
+            monitor_intervals_, bandwidth_estimate_);
+  }
+}
+
+NetworkControlUpdate PccNetworkController::OnNetworkAvailability(
+    NetworkAvailability msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnNetworkRouteChange(
+    NetworkRouteChange msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnProcessInterval(
+    ProcessInterval msg) {
+  return CreateRateUpdate(msg.at_time);
+}
+
+NetworkControlUpdate PccNetworkController::OnTargetRateConstraints(
+    TargetRateConstraints msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnRemoteBitrateReport(
+    RemoteBitrateReport) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnRoundTripTimeUpdate(
+    RoundTripTimeUpdate) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnTransportLossReport(
+    TransportLossReport) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnStreamsConfig(StreamsConfig msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnReceivedPacket(
+    ReceivedPacket msg) {
+  return NetworkControlUpdate();
+}
+
+NetworkControlUpdate PccNetworkController::OnNetworkStateEstimate(
+    NetworkStateEstimate msg) {
+  return NetworkControlUpdate();
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.h b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.h
new file mode 100644
index 0000000000..e5f65dd7d9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller.h
@@ -0,0 +1,125 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_PCC_NETWORK_CONTROLLER_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_PCC_NETWORK_CONTROLLER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <vector>
+
+#include "api/transport/network_control.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/pcc/bitrate_controller.h"
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+#include "modules/congestion_controller/pcc/rtt_tracker.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+namespace pcc {
+
+// PCC (Performance-oriented Congestion Control) Vivace is a congestion
+// control algorithm based on online (convex) optimization in machine learning.
+// It divides time into consecutive Monitor Intervals (MI) to test sending
+// rates r(1 + eps), r(1 - eps) for the current sending rate r.
+// At the end of each MI it computes utility function to transform the
+// performance statistics into a numerical value. Then it updates current
+// sending rate using gradient ascent to maximize utility function.
+class PccNetworkController : public NetworkControllerInterface {
+ public:
+  enum class Mode {
+    kStartup,
+    // Slow start phase of PCC doubles sending rate each monitor interval.
+    kSlowStart,
+    // After getting the first decrease in utility function PCC exits slow start
+    // and enters the online learning phase.
+    kOnlineLearning,
+    // If we got that sending with the lower rate resulted in higher packet
+    // loss, then the measurements are unreliable and we need to double check
+    // them.
+    kDoubleCheck
+  };
+
+  enum class MonitorIntervalLengthStrategy {
+    // Monitor interval length adaptive when it is proportional to packets RTT.
+    kAdaptive,
+    // Monitor interval length is fixed when it is equal to the time of sending
+    // predefined amount of packets (kMinPacketsNumberPerInterval).
+    kFixed
+  };
+
+  explicit PccNetworkController(NetworkControllerConfig config);
+  ~PccNetworkController() override;
+
+  // NetworkControllerInterface
+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
+  NetworkControlUpdate OnTargetRateConstraints(
+      TargetRateConstraints msg) override;
+  NetworkControlUpdate OnTransportPacketsFeedback(
+      TransportPacketsFeedback msg) override;
+
+  // Part of remote bitrate estimation api, not implemented for PCC
+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
+  NetworkControlUpdate OnReceivedPacket(ReceivedPacket msg) override;
+  NetworkControlUpdate OnNetworkStateEstimate(
+      NetworkStateEstimate msg) override;
+
+ private:
+  void UpdateSendingRateAndMode();
+  NetworkControlUpdate CreateRateUpdate(Timestamp at_time) const;
+  TimeDelta ComputeMonitorIntervalsDuration() const;
+  bool NeedDoubleCheckMeasurments() const;
+  bool IsTimeoutExpired(Timestamp current_time) const;
+  bool IsFeedbackCollectionDone() const;
+
+  Timestamp start_time_;
+  Timestamp last_sent_packet_time_;
+  TimeDelta smoothed_packets_sending_interval_;
+  Mode mode_;
+
+  // Default value used for initializing bandwidth.
+  DataRate default_bandwidth_;
+  // Current estimate r.
+  DataRate bandwidth_estimate_;
+
+  RttTracker rtt_tracker_;
+  TimeDelta monitor_interval_timeout_;
+  const MonitorIntervalLengthStrategy monitor_interval_length_strategy_;
+  const double monitor_interval_duration_ratio_;
+  const double sampling_step_;  // Epsilon.
+  const double monitor_interval_timeout_ratio_;
+  const int64_t min_packets_number_per_interval_;
+
+  PccBitrateController bitrate_controller_;
+
+  std::vector<PccMonitorInterval> monitor_intervals_;
+  std::vector<DataRate> monitor_intervals_bitrates_;
+  TimeDelta monitor_intervals_duration_;
+  size_t complete_feedback_monitor_interval_number_;
+
+  webrtc::Random random_generator_;
+  std::deque<PacketResult> last_received_packets_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_PCC_NETWORK_CONTROLLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
new file mode 100644
index 0000000000..c98680c785
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/pcc_network_controller_unittest.cc
@@ -0,0 +1,119 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/pcc_network_controller.h"
+
+#include <memory>
+
+#include "modules/congestion_controller/pcc/pcc_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+using ::testing::AllOf;
+using ::testing::Field;
+using ::testing::Ge;
+using ::testing::Le;
+using ::testing::Matcher;
+using ::testing::Property;
+
+namespace webrtc {
+namespace test {
+namespace {
+
+const DataRate kInitialBitrate = DataRate::KilobitsPerSec(60);
+const Timestamp kDefaultStartTime = Timestamp::Millis(10000000);
+
+constexpr double kDataRateMargin = 0.20;
+constexpr double kMinDataRateFactor = 1 - kDataRateMargin;
+constexpr double kMaxDataRateFactor = 1 + kDataRateMargin;
+inline Matcher<TargetTransferRate> TargetRateCloseTo(DataRate rate) {
+  DataRate min_data_rate = rate * kMinDataRateFactor;
+  DataRate max_data_rate = rate * kMaxDataRateFactor;
+  return Field(&TargetTransferRate::target_rate,
+               AllOf(Ge(min_data_rate), Le(max_data_rate)));
+}
+
+NetworkControllerConfig InitialConfig(
+    int starting_bandwidth_kbps = kInitialBitrate.kbps(),
+    int min_data_rate_kbps = 0,
+    int max_data_rate_kbps = 5 * kInitialBitrate.kbps()) {
+  NetworkControllerConfig config;
+  config.constraints.at_time = kDefaultStartTime;
+  config.constraints.min_data_rate =
+      DataRate::KilobitsPerSec(min_data_rate_kbps);
+  config.constraints.max_data_rate =
+      DataRate::KilobitsPerSec(max_data_rate_kbps);
+  config.constraints.starting_rate =
+      DataRate::KilobitsPerSec(starting_bandwidth_kbps);
+  return config;
+}
+
+ProcessInterval InitialProcessInterval() {
+  ProcessInterval process_interval;
+  process_interval.at_time = kDefaultStartTime;
+  return process_interval;
+}
+
+}  // namespace
+
+TEST(PccNetworkControllerTest, SendsConfigurationOnFirstProcess) {
+  std::unique_ptr<NetworkControllerInterface> controller_;
+  controller_.reset(new pcc::PccNetworkController(InitialConfig()));
+
+  NetworkControlUpdate update =
+      controller_->OnProcessInterval(InitialProcessInterval());
+  EXPECT_THAT(*update.target_rate, TargetRateCloseTo(kInitialBitrate));
+  EXPECT_THAT(*update.pacer_config,
+              Property(&PacerConfig::data_rate, Ge(kInitialBitrate)));
+}
+
+TEST(PccNetworkControllerTest, UpdatesTargetSendRate) {
+  PccNetworkControllerFactory factory;
+  Scenario s("pcc_unit/updates_rate", false);
+  CallClientConfig config;
+  config.transport.cc_factory = &factory;
+  config.transport.rates.min_rate = DataRate::KilobitsPerSec(10);
+  config.transport.rates.max_rate = DataRate::KilobitsPerSec(1500);
+  config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  auto send_net = s.CreateMutableSimulationNode([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(500);
+    c->delay = TimeDelta::Millis(100);
+  });
+  auto ret_net = s.CreateMutableSimulationNode(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(100); });
+
+  auto* client = s.CreateClient("send", config);
+  auto* route = s.CreateRoutes(client, {send_net->node()},
+                               s.CreateClient("return", CallClientConfig()),
+                               {ret_net->node()});
+  VideoStreamConfig video;
+  video.stream.use_rtx = false;
+  s.CreateVideoStream(route->forward(), video);
+  s.RunFor(TimeDelta::Seconds(30));
+  EXPECT_NEAR(client->target_rate().kbps(), 450, 100);
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(800);
+    c->delay = TimeDelta::Millis(100);
+  });
+  s.RunFor(TimeDelta::Seconds(20));
+  EXPECT_NEAR(client->target_rate().kbps(), 750, 150);
+  send_net->UpdateConfig([](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::KilobitsPerSec(200);
+    c->delay = TimeDelta::Millis(200);
+  });
+  ret_net->UpdateConfig(
+      [](NetworkSimulationConfig* c) { c->delay = TimeDelta::Millis(200); });
+  s.RunFor(TimeDelta::Seconds(35));
+  EXPECT_NEAR(client->target_rate().kbps(), 170, 50);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.cc
new file mode 100644
index 0000000000..af9dc8f11b
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.cc
@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/rtt_tracker.h"
+
+#include <algorithm>
+
+namespace webrtc {
+namespace pcc {
+
+RttTracker::RttTracker(TimeDelta initial_rtt, double alpha)
+    : rtt_estimate_(initial_rtt), alpha_(alpha) {}
+
+void RttTracker::OnPacketsFeedback(
+    const std::vector<PacketResult>& packet_feedbacks,
+    Timestamp feedback_received_time) {
+  TimeDelta packet_rtt = TimeDelta::MinusInfinity();
+  for (const PacketResult& packet_result : packet_feedbacks) {
+    if (!packet_result.IsReceived())
+      continue;
+    packet_rtt = std::max<TimeDelta>(
+        packet_rtt,
+        feedback_received_time - packet_result.sent_packet.send_time);
+  }
+  if (packet_rtt.IsFinite())
+    rtt_estimate_ = (1 - alpha_) * rtt_estimate_ + alpha_ * packet_rtt;
+}
+
+TimeDelta RttTracker::GetRtt() const {
+  return rtt_estimate_;
+}
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.h b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.h
new file mode 100644
index 0000000000..94033cd511
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_RTT_TRACKER_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_RTT_TRACKER_H_
+
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+
+namespace webrtc {
+namespace pcc {
+
+class RttTracker {
+ public:
+  RttTracker(TimeDelta initial_rtt, double alpha);
+  // Updates RTT estimate.
+  void OnPacketsFeedback(const std::vector<PacketResult>& packet_feedbacks,
+                         Timestamp feedback_received_time);
+  TimeDelta GetRtt() const;
+
+ private:
+  TimeDelta rtt_estimate_;
+  double alpha_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_RTT_TRACKER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker_unittest.cc
new file mode 100644
index 0000000000..7d90e86822
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/rtt_tracker_unittest.cc
@@ -0,0 +1,71 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/rtt_tracker.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+const TimeDelta kInitialRtt = TimeDelta::Micros(10);
+constexpr double kAlpha = 0.9;
+const Timestamp kStartTime = Timestamp::Seconds(0);
+
+PacketResult GetPacketWithRtt(TimeDelta rtt) {
+  SentPacket packet;
+  packet.send_time = kStartTime;
+  PacketResult packet_result;
+  packet_result.sent_packet = packet;
+  if (rtt.IsFinite()) {
+    packet_result.receive_time = kStartTime + rtt;
+  } else {
+    packet_result.receive_time = Timestamp::PlusInfinity();
+  }
+  return packet_result;
+}
+}  // namespace
+
+TEST(PccRttTrackerTest, InitialValue) {
+  RttTracker tracker{kInitialRtt, kAlpha};
+  EXPECT_EQ(kInitialRtt, tracker.GetRtt());
+  for (int i = 0; i < 100; ++i) {
+    tracker.OnPacketsFeedback({GetPacketWithRtt(kInitialRtt)},
+                              kStartTime + kInitialRtt);
+  }
+  EXPECT_EQ(kInitialRtt, tracker.GetRtt());
+}
+
+TEST(PccRttTrackerTest, DoNothingWhenPacketIsLost) {
+  RttTracker tracker{kInitialRtt, kAlpha};
+  tracker.OnPacketsFeedback({GetPacketWithRtt(TimeDelta::PlusInfinity())},
+                            kStartTime + kInitialRtt);
+  EXPECT_EQ(tracker.GetRtt(), kInitialRtt);
+}
+
+TEST(PccRttTrackerTest, ChangeInRtt) {
+  RttTracker tracker{kInitialRtt, kAlpha};
+  const TimeDelta kNewRtt = TimeDelta::Micros(100);
+  tracker.OnPacketsFeedback({GetPacketWithRtt(kNewRtt)}, kStartTime + kNewRtt);
+  EXPECT_GT(tracker.GetRtt(), kInitialRtt);
+  EXPECT_LE(tracker.GetRtt(), kNewRtt);
+  for (int i = 0; i < 100; ++i) {
+    tracker.OnPacketsFeedback({GetPacketWithRtt(kNewRtt)},
+                              kStartTime + kNewRtt);
+  }
+  const TimeDelta absolute_error = TimeDelta::Micros(1);
+  EXPECT_NEAR(tracker.GetRtt().us(), kNewRtt.us(), absolute_error.us());
+  EXPECT_LE(tracker.GetRtt(), kNewRtt);
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.cc
new file mode 100644
index 0000000000..006a2fccd9
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.cc
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/utility_function.h"
+
+#include <algorithm>
+#include <cmath>
+
+#include "api/units/data_rate.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace pcc {
+
+VivaceUtilityFunction::VivaceUtilityFunction(
+    double delay_gradient_coefficient,
+    double loss_coefficient,
+    double throughput_coefficient,
+    double throughput_power,
+    double delay_gradient_threshold,
+    double delay_gradient_negative_bound)
+    : delay_gradient_coefficient_(delay_gradient_coefficient),
+      loss_coefficient_(loss_coefficient),
+      throughput_power_(throughput_power),
+      throughput_coefficient_(throughput_coefficient),
+      delay_gradient_threshold_(delay_gradient_threshold),
+      delay_gradient_negative_bound_(delay_gradient_negative_bound) {
+  RTC_DCHECK_GE(delay_gradient_negative_bound_, 0);
+}
+
+double VivaceUtilityFunction::Compute(
+    const PccMonitorInterval& monitor_interval) const {
+  RTC_DCHECK(monitor_interval.IsFeedbackCollectionDone());
+  double bitrate = monitor_interval.GetTargetSendingRate().bps();
+  double loss_rate = monitor_interval.GetLossRate();
+  double rtt_gradient =
+      monitor_interval.ComputeDelayGradient(delay_gradient_threshold_);
+  rtt_gradient = std::max(rtt_gradient, -delay_gradient_negative_bound_);
+  return (throughput_coefficient_ * std::pow(bitrate, throughput_power_)) -
+         (delay_gradient_coefficient_ * bitrate * rtt_gradient) -
+         (loss_coefficient_ * bitrate * loss_rate);
+}
+
+VivaceUtilityFunction::~VivaceUtilityFunction() = default;
+
+ModifiedVivaceUtilityFunction::ModifiedVivaceUtilityFunction(
+    double delay_gradient_coefficient,
+    double loss_coefficient,
+    double throughput_coefficient,
+    double throughput_power,
+    double delay_gradient_threshold,
+    double delay_gradient_negative_bound)
+    : delay_gradient_coefficient_(delay_gradient_coefficient),
+      loss_coefficient_(loss_coefficient),
+      throughput_power_(throughput_power),
+      throughput_coefficient_(throughput_coefficient),
+      delay_gradient_threshold_(delay_gradient_threshold),
+      delay_gradient_negative_bound_(delay_gradient_negative_bound) {
+  RTC_DCHECK_GE(delay_gradient_negative_bound_, 0);
+}
+
+double ModifiedVivaceUtilityFunction::Compute(
+    const PccMonitorInterval& monitor_interval) const {
+  RTC_DCHECK(monitor_interval.IsFeedbackCollectionDone());
+  double bitrate = monitor_interval.GetTargetSendingRate().bps();
+  double loss_rate = monitor_interval.GetLossRate();
+  double rtt_gradient =
+      monitor_interval.ComputeDelayGradient(delay_gradient_threshold_);
+  rtt_gradient = std::max(rtt_gradient, -delay_gradient_negative_bound_);
+  return (throughput_coefficient_ * std::pow(bitrate, throughput_power_) *
+          bitrate) -
+         (delay_gradient_coefficient_ * bitrate * bitrate * rtt_gradient) -
+         (loss_coefficient_ * bitrate * bitrate * loss_rate);
+}
+
+ModifiedVivaceUtilityFunction::~ModifiedVivaceUtilityFunction() = default;
+
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.h b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.h
new file mode 100644
index 0000000000..98bb0744c1
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function.h
@@ -0,0 +1,78 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_PCC_UTILITY_FUNCTION_H_
+#define MODULES_CONGESTION_CONTROLLER_PCC_UTILITY_FUNCTION_H_
+
+#include "modules/congestion_controller/pcc/monitor_interval.h"
+
+namespace webrtc {
+namespace pcc {
+
+// Utility function is used by PCC to transform the performance statistics
+// (sending rate, loss rate, packets latency) gathered at one monitor interval
+// into a numerical value.
+// https://www.usenix.org/conference/nsdi18/presentation/dong
+class PccUtilityFunctionInterface {
+ public:
+  virtual double Compute(const PccMonitorInterval& monitor_interval) const = 0;
+  virtual ~PccUtilityFunctionInterface() = default;
+};
+
+// Vivace utility function were suggested in the paper "PCC Vivace:
+// Online-Learning Congestion Control", Mo Dong et all.
+class VivaceUtilityFunction : public PccUtilityFunctionInterface {
+ public:
+  VivaceUtilityFunction(double delay_gradient_coefficient,
+                        double loss_coefficient,
+                        double throughput_coefficient,
+                        double throughput_power,
+                        double delay_gradient_threshold,
+                        double delay_gradient_negative_bound);
+  double Compute(const PccMonitorInterval& monitor_interval) const override;
+  ~VivaceUtilityFunction() override;
+
+ private:
+  const double delay_gradient_coefficient_;
+  const double loss_coefficient_;
+  const double throughput_power_;
+  const double throughput_coefficient_;
+  const double delay_gradient_threshold_;
+  const double delay_gradient_negative_bound_;
+};
+
+// This utility function were obtained by tuning Vivace utility function.
+// The main difference is that gradient of modified utilify funtion (as well as
+// rate updates) scales proportionally to the sending rate which leads to
+// better performance in case of single sender.
+class ModifiedVivaceUtilityFunction : public PccUtilityFunctionInterface {
+ public:
+  ModifiedVivaceUtilityFunction(double delay_gradient_coefficient,
+                                double loss_coefficient,
+                                double throughput_coefficient,
+                                double throughput_power,
+                                double delay_gradient_threshold,
+                                double delay_gradient_negative_bound);
+  double Compute(const PccMonitorInterval& monitor_interval) const override;
+  ~ModifiedVivaceUtilityFunction() override;
+
+ private:
+  const double delay_gradient_coefficient_;
+  const double loss_coefficient_;
+  const double throughput_power_;
+  const double throughput_coefficient_;
+  const double delay_gradient_threshold_;
+  const double delay_gradient_negative_bound_;
+};
+
+}  // namespace pcc
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_PCC_UTILITY_FUNCTION_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function_unittest.cc
new file mode 100644
index 0000000000..19b2d15920
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/pcc/utility_function_unittest.cc
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/pcc/utility_function.h"
+
+#include <stddef.h>
+
+#include <cmath>
+#include <type_traits>
+#include <vector>
+
+#include "api/transport/network_types.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace pcc {
+namespace test {
+namespace {
+constexpr double kLossCoefficient = 11.35;
+constexpr double kThroughputPower = 0.9;
+constexpr double kThroughputCoefficient = 1;
+constexpr double kDelayGradientNegativeBound = 10;
+
+const Timestamp kStartTime = Timestamp::Micros(0);
+const TimeDelta kPacketsDelta = TimeDelta::Millis(1);
+const TimeDelta kIntervalDuration = TimeDelta::Millis(100);
+const DataRate kSendingBitrate = DataRate::BitsPerSec(1000);
+
+const DataSize kDefaultDataSize = DataSize::Bytes(100);
+const TimeDelta kDefaultDelay = TimeDelta::Millis(100);
+
+std::vector<PacketResult> CreatePacketResults(
+    const std::vector<Timestamp>& packets_send_times,
+    const std::vector<Timestamp>& packets_received_times = {},
+    const std::vector<DataSize>& packets_sizes = {}) {
+  std::vector<PacketResult> packet_results;
+  PacketResult packet_result;
+  SentPacket sent_packet;
+  for (size_t i = 0; i < packets_send_times.size(); ++i) {
+    sent_packet.send_time = packets_send_times[i];
+    if (packets_sizes.empty()) {
+      sent_packet.size = kDefaultDataSize;
+    } else {
+      sent_packet.size = packets_sizes[i];
+    }
+    packet_result.sent_packet = sent_packet;
+    if (packets_received_times.empty()) {
+      packet_result.receive_time = packets_send_times[i] + kDefaultDelay;
+    } else {
+      packet_result.receive_time = packets_received_times[i];
+    }
+    packet_results.push_back(packet_result);
+  }
+  return packet_results;
+}
+
+}  // namespace
+
+TEST(PccVivaceUtilityFunctionTest,
+     UtilityIsThroughputTermIfAllRestCoefficientsAreZero) {
+  VivaceUtilityFunction utility_function(0, 0, kThroughputCoefficient,
+                                         kThroughputPower, 0,
+                                         kDelayGradientNegativeBound);
+  PccMonitorInterval monitor_interval(kSendingBitrate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 3 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kPacketsDelta + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay + 3 * kPacketsDelta,
+       Timestamp::PlusInfinity()},
+      {kDefaultDataSize, kDefaultDataSize, kDefaultDataSize,
+       kDefaultDataSize}));
+  EXPECT_DOUBLE_EQ(utility_function.Compute(monitor_interval),
+                   kThroughputCoefficient *
+                       std::pow(kSendingBitrate.bps(), kThroughputPower));
+}
+
+TEST(PccVivaceUtilityFunctionTest,
+     LossTermIsNonZeroIfLossCoefficientIsNonZero) {
+  VivaceUtilityFunction utility_function(
+      0, kLossCoefficient, kThroughputCoefficient, kThroughputPower, 0,
+      kDelayGradientNegativeBound);
+  PccMonitorInterval monitor_interval(kSendingBitrate, kStartTime,
+                                      kIntervalDuration);
+  monitor_interval.OnPacketsFeedback(CreatePacketResults(
+      {kStartTime + kPacketsDelta, kStartTime + 2 * kPacketsDelta,
+       kStartTime + 5 * kPacketsDelta, kStartTime + 2 * kIntervalDuration},
+      {kStartTime + kDefaultDelay, Timestamp::PlusInfinity(),
+       kStartTime + kDefaultDelay, kStartTime + 3 * kIntervalDuration},
+      {}));
+  // The second packet was lost.
+  EXPECT_DOUBLE_EQ(utility_function.Compute(monitor_interval),
+                   kThroughputCoefficient *
+                           std::pow(kSendingBitrate.bps(), kThroughputPower) -
+                       kLossCoefficient * kSendingBitrate.bps() *
+                           monitor_interval.GetLossRate());
+}
+
+}  // namespace test
+}  // namespace pcc
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc
new file mode 100644
index 0000000000..e43b020f6e
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller.cc
@@ -0,0 +1,134 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+
+#include "api/units/data_rate.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
+#include "modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+namespace {
+static const uint32_t kTimeOffsetSwitchThreshold = 30;
+}  // namespace
+
+void ReceiveSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms,
+                                                  int64_t max_rtt_ms) {
+  MutexLock lock(&mutex_);
+  rbe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
+}
+
+void ReceiveSideCongestionController::RemoveStream(uint32_t ssrc) {
+  MutexLock lock(&mutex_);
+  rbe_->RemoveStream(ssrc);
+}
+
+DataRate ReceiveSideCongestionController::LatestReceiveSideEstimate() const {
+  MutexLock lock(&mutex_);
+  return rbe_->LatestEstimate();
+}
+
+void ReceiveSideCongestionController::PickEstimatorFromHeader(
+    const RTPHeader& header) {
+  if (header.extension.hasAbsoluteSendTime) {
+    // If we see AST in header, switch RBE strategy immediately.
+    if (!using_absolute_send_time_) {
+      RTC_LOG(LS_INFO)
+          << "WrappingBitrateEstimator: Switching to absolute send time RBE.";
+      using_absolute_send_time_ = true;
+      PickEstimator();
+    }
+    packets_since_absolute_send_time_ = 0;
+  } else {
+    // When we don't see AST, wait for a few packets before going back to TOF.
+    if (using_absolute_send_time_) {
+      ++packets_since_absolute_send_time_;
+      if (packets_since_absolute_send_time_ >= kTimeOffsetSwitchThreshold) {
+        RTC_LOG(LS_INFO)
+            << "WrappingBitrateEstimator: Switching to transmission "
+               "time offset RBE.";
+        using_absolute_send_time_ = false;
+        PickEstimator();
+      }
+    }
+  }
+}
+
+// Instantiate RBE for Time Offset or Absolute Send Time extensions.
+void ReceiveSideCongestionController::PickEstimator() {
+  if (using_absolute_send_time_) {
+    rbe_ = std::make_unique<RemoteBitrateEstimatorAbsSendTime>(&remb_throttler_,
+                                                               &clock_);
+  } else {
+    rbe_ = std::make_unique<RemoteBitrateEstimatorSingleStream>(
+        &remb_throttler_, &clock_);
+  }
+}
+
+ReceiveSideCongestionController::ReceiveSideCongestionController(
+    Clock* clock,
+    RemoteEstimatorProxy::TransportFeedbackSender feedback_sender,
+    RembThrottler::RembSender remb_sender,
+    NetworkStateEstimator* network_state_estimator)
+    : clock_(*clock),
+      remb_throttler_(std::move(remb_sender), clock),
+      remote_estimator_proxy_(std::move(feedback_sender),
+                              network_state_estimator),
+      rbe_(new RemoteBitrateEstimatorSingleStream(&remb_throttler_, clock)),
+      using_absolute_send_time_(false),
+      packets_since_absolute_send_time_(0) {}
+
+void ReceiveSideCongestionController::OnReceivedPacket(
+    int64_t arrival_time_ms,
+    size_t payload_size,
+    const RTPHeader& header) {
+  remote_estimator_proxy_.IncomingPacket(arrival_time_ms, payload_size, header);
+  if (!header.extension.hasTransportSequenceNumber) {
+    // Receive-side BWE.
+    MutexLock lock(&mutex_);
+    PickEstimatorFromHeader(header);
+    rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
+  }
+}
+
+void ReceiveSideCongestionController::SetSendPeriodicFeedback(
+    bool send_periodic_feedback) {
+  remote_estimator_proxy_.SetSendPeriodicFeedback(send_periodic_feedback);
+}
+
+void ReceiveSideCongestionController::OnBitrateChanged(int bitrate_bps) {
+  remote_estimator_proxy_.OnBitrateChanged(bitrate_bps);
+}
+
+TimeDelta ReceiveSideCongestionController::MaybeProcess() {
+  Timestamp now = clock_.CurrentTime();
+  mutex_.Lock();
+  TimeDelta time_until_rbe = rbe_->Process();
+  mutex_.Unlock();
+  TimeDelta time_until_rep = remote_estimator_proxy_.Process(now);
+  TimeDelta time_until = std::min(time_until_rbe, time_until_rep);
+  return std::max(time_until, TimeDelta::Zero());
+}
+
+void ReceiveSideCongestionController::SetMaxDesiredReceiveBitrate(
+    DataRate bitrate) {
+  remb_throttler_.SetMaxDesiredReceiveBitrate(bitrate);
+}
+
+void ReceiveSideCongestionController::SetTransportOverhead(
+    DataSize overhead_per_packet) {
+  remote_estimator_proxy_.SetTransportOverhead(overhead_per_packet);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller_unittest.cc
new file mode 100644
index 0000000000..f2fd6d11d7
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/receive_side_congestion_controller_unittest.cc
@@ -0,0 +1,126 @@
+/*
+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+
+#include "api/test/network_emulation/create_cross_traffic.h"
+#include "api/test/network_emulation/cross_traffic.h"
+#include "modules/pacing/packet_router.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+using ::testing::_;
+using ::testing::AtLeast;
+using ::testing::ElementsAre;
+using ::testing::MockFunction;
+
+namespace webrtc {
+
+namespace {
+
+// Helper to convert some time format to resolution used in absolute send time
+// header extension, rounded upwards. `t` is the time to convert, in some
+// resolution. `denom` is the value to divide `t` by to get whole seconds,
+// e.g. `denom` = 1000 if `t` is in milliseconds.
+uint32_t AbsSendTime(int64_t t, int64_t denom) {
+  return (((t << 18) + (denom >> 1)) / denom) & 0x00fffffful;
+}
+
+const uint32_t kInitialBitrateBps = 60000;
+
+}  // namespace
+
+namespace test {
+
+TEST(ReceiveSideCongestionControllerTest, SendsRembWithAbsSendTime) {
+  MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
+      feedback_sender;
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  SimulatedClock clock_(123456);
+
+  ReceiveSideCongestionController controller(
+      &clock_, feedback_sender.AsStdFunction(), remb_sender.AsStdFunction(),
+      nullptr);
+
+  size_t payload_size = 1000;
+  RTPHeader header;
+  header.ssrc = 0x11eb21c;
+  header.extension.hasAbsoluteSendTime = true;
+
+  EXPECT_CALL(remb_sender, Call(_, ElementsAre(header.ssrc))).Times(AtLeast(1));
+
+  for (int i = 0; i < 10; ++i) {
+    clock_.AdvanceTimeMilliseconds((1000 * payload_size) / kInitialBitrateBps);
+    int64_t now_ms = clock_.TimeInMilliseconds();
+    header.extension.absoluteSendTime = AbsSendTime(now_ms, 1000);
+    controller.OnReceivedPacket(now_ms, payload_size, header);
+  }
+}
+
+TEST(ReceiveSideCongestionControllerTest,
+     SendsRembAfterSetMaxDesiredReceiveBitrate) {
+  MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
+      feedback_sender;
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  SimulatedClock clock_(123456);
+
+  ReceiveSideCongestionController controller(
+      &clock_, feedback_sender.AsStdFunction(), remb_sender.AsStdFunction(),
+      nullptr);
+  EXPECT_CALL(remb_sender, Call(123, _));
+  controller.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(123));
+}
+
+TEST(ReceiveSideCongestionControllerTest, ConvergesToCapacity) {
+  Scenario s("receive_cc_unit/converge");
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(300);
+  });
+
+  auto* route = s.CreateRoutes(client, {s.CreateSimulationNode(net_conf)},
+                               s.CreateClient("return", CallClientConfig()),
+                               {s.CreateSimulationNode(net_conf)});
+  VideoStreamConfig video;
+  video.stream.packet_feedback = false;
+  s.CreateVideoStream(route->forward(), video);
+  s.RunFor(TimeDelta::Seconds(30));
+  EXPECT_NEAR(client->send_bandwidth().kbps(), 900, 150);
+}
+
+TEST(ReceiveSideCongestionControllerTest, IsFairToTCP) {
+  Scenario s("receive_cc_unit/tcp_fairness");
+  NetworkSimulationConfig net_conf;
+  net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
+  net_conf.delay = TimeDelta::Millis(50);
+  auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+    c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
+  });
+  auto send_net = {s.CreateSimulationNode(net_conf)};
+  auto ret_net = {s.CreateSimulationNode(net_conf)};
+  auto* route = s.CreateRoutes(
+      client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+  VideoStreamConfig video;
+  video.stream.packet_feedback = false;
+  s.CreateVideoStream(route->forward(), video);
+  s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+      s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+      FakeTcpConfig()));
+  s.RunFor(TimeDelta::Seconds(30));
+  // For some reason we get outcompeted by TCP here, this should probably be
+  // fixed and a lower bound should be added to the test.
+  EXPECT_LT(client->send_bandwidth().kbps(), 750);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc
new file mode 100644
index 0000000000..fcc30af9a8
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.cc
@@ -0,0 +1,63 @@
+/*
+ *  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 "modules/congestion_controller/remb_throttler.h"
+
+#include <algorithm>
+#include <utility>
+
+namespace webrtc {
+
+namespace {
+constexpr TimeDelta kRembSendInterval = TimeDelta::Millis(200);
+}  // namespace
+
+RembThrottler::RembThrottler(RembSender remb_sender, Clock* clock)
+    : remb_sender_(std::move(remb_sender)),
+      clock_(clock),
+      last_remb_time_(Timestamp::MinusInfinity()),
+      last_send_remb_bitrate_(DataRate::PlusInfinity()),
+      max_remb_bitrate_(DataRate::PlusInfinity()) {}
+
+void RembThrottler::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                                            uint32_t bitrate_bps) {
+  DataRate receive_bitrate = DataRate::BitsPerSec(bitrate_bps);
+  Timestamp now = clock_->CurrentTime();
+  {
+    MutexLock lock(&mutex_);
+    // % threshold for if we should send a new REMB asap.
+    const int64_t kSendThresholdPercent = 103;
+    if (receive_bitrate * kSendThresholdPercent / 100 >
+            last_send_remb_bitrate_ &&
+        now < last_remb_time_ + kRembSendInterval) {
+      return;
+    }
+    last_remb_time_ = now;
+    last_send_remb_bitrate_ = receive_bitrate;
+    receive_bitrate = std::min(last_send_remb_bitrate_, max_remb_bitrate_);
+  }
+  remb_sender_(receive_bitrate.bps(), ssrcs);
+}
+
+void RembThrottler::SetMaxDesiredReceiveBitrate(DataRate bitrate) {
+  Timestamp now = clock_->CurrentTime();
+  {
+    MutexLock lock(&mutex_);
+    max_remb_bitrate_ = bitrate;
+    if (now - last_remb_time_ < kRembSendInterval &&
+        !last_send_remb_bitrate_.IsZero() &&
+        last_send_remb_bitrate_ <= max_remb_bitrate_) {
+      return;
+    }
+  }
+  remb_sender_(bitrate.bps(), /*ssrcs=*/{});
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/remb_throttler.h b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.h
new file mode 100644
index 0000000000..85292cbc09
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/remb_throttler.h
@@ -0,0 +1,54 @@
+/*
+ *  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 MODULES_CONGESTION_CONTROLLER_REMB_THROTTLER_H_
+#define MODULES_CONGESTION_CONTROLLER_REMB_THROTTLER_H_
+
+#include <functional>
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "rtc_base/synchronization/mutex.h"
+
+namespace webrtc {
+
+// RembThrottler is a helper class used for throttling RTCP REMB messages.
+// Throttles small changes to the received BWE within 200ms.
+class RembThrottler : public RemoteBitrateObserver {
+ public:
+  using RembSender =
+      std::function<void(int64_t bitrate_bps, std::vector<uint32_t> ssrcs)>;
+  RembThrottler(RembSender remb_sender, Clock* clock);
+
+  // Ensures the remote party is notified of the receive bitrate no larger than
+  // `bitrate` using RTCP REMB.
+  void SetMaxDesiredReceiveBitrate(DataRate bitrate);
+
+  // Implements RemoteBitrateObserver;
+  // Called every time there is a new bitrate estimate for a receive channel
+  // group. This call will trigger a new RTCP REMB packet if the bitrate
+  // estimate has decreased or if no RTCP REMB packet has been sent for
+  // a certain time interval.
+  void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
+                               uint32_t bitrate_bps) override;
+
+ private:
+  const RembSender remb_sender_;
+  Clock* const clock_;
+  mutable Mutex mutex_;
+  Timestamp last_remb_time_ RTC_GUARDED_BY(mutex_);
+  DataRate last_send_remb_bitrate_ RTC_GUARDED_BY(mutex_);
+  DataRate max_remb_bitrate_ RTC_GUARDED_BY(mutex_);
+};
+
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_REMB_THROTTLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/remb_throttler_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/remb_throttler_unittest.cc
new file mode 100644
index 0000000000..3f8df8a7bb
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/remb_throttler_unittest.cc
@@ -0,0 +1,100 @@
+/*
+ *  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 "modules/congestion_controller/remb_throttler.h"
+
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "api/units/time_delta.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+using ::testing::_;
+using ::testing::MockFunction;
+
+TEST(RembThrottlerTest, CallRembSenderOnFirstReceiveBitrateChange) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call(12345, std::vector<uint32_t>({1, 2, 3})));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12345);
+}
+
+TEST(RembThrottlerTest, ThrottlesSmallReceiveBitrateDecrease) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call);
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12346);
+  clock.AdvanceTime(TimeDelta::Millis(100));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12345);
+
+  EXPECT_CALL(remb_sender, Call(12345, _));
+  clock.AdvanceTime(TimeDelta::Millis(101));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/12345);
+}
+
+TEST(RembThrottlerTest, DoNotThrottleLargeReceiveBitrateDecrease) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call(2345, _));
+  EXPECT_CALL(remb_sender, Call(1234, _));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/2345);
+  clock.AdvanceTime(TimeDelta::Millis(1));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/1234);
+}
+
+TEST(RembThrottlerTest, ThrottlesReceiveBitrateIncrease) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call);
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/1234);
+  clock.AdvanceTime(TimeDelta::Millis(100));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/2345);
+
+  // Updates 200ms after previous callback is not throttled.
+  EXPECT_CALL(remb_sender, Call(2345, _));
+  clock.AdvanceTime(TimeDelta::Millis(101));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/2345);
+}
+
+TEST(RembThrottlerTest, CallRembSenderOnSetMaxDesiredReceiveBitrate) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+  EXPECT_CALL(remb_sender, Call(1234, _));
+  remb_throttler.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(1234));
+}
+
+TEST(RembThrottlerTest, CallRembSenderWithMinOfMaxDesiredAndOnReceivedBitrate) {
+  SimulatedClock clock(Timestamp::Zero());
+  MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
+  RembThrottler remb_throttler(remb_sender.AsStdFunction(), &clock);
+
+  EXPECT_CALL(remb_sender, Call(1234, _));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/1234);
+  clock.AdvanceTime(TimeDelta::Millis(1));
+  remb_throttler.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(4567));
+
+  clock.AdvanceTime(TimeDelta::Millis(200));
+  EXPECT_CALL(remb_sender, Call(4567, _));
+  remb_throttler.OnReceiveBitrateChanged({1, 2, 3}, /*bitrate_bps=*/5678);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn
new file mode 100644
index 0000000000..cd13332b7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn
@@ -0,0 +1,100 @@
+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+config("bwe_test_logging") {
+  if (rtc_enable_bwe_test_logging) {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
+  } else {
+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
+  }
+}
+
+rtc_library("control_handler") {
+  visibility = [ "*" ]
+  sources = [
+    "control_handler.cc",
+    "control_handler.h",
+  ]
+
+  deps = [
+    "../../../api:sequence_checker",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_rate",
+    "../../../api/units:data_size",
+    "../../../api/units:time_delta",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:safe_conversions",
+    "../../../rtc_base:safe_minmax",
+    "../../../rtc_base/system:no_unique_address",
+    "../../../system_wrappers:field_trial",
+    "../../pacing",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+rtc_library("transport_feedback") {
+  visibility = [ "*" ]
+  sources = [
+    "transport_feedback_adapter.cc",
+    "transport_feedback_adapter.h",
+    "transport_feedback_demuxer.cc",
+    "transport_feedback_demuxer.h",
+  ]
+
+  deps = [
+    "../..:module_api_public",
+    "../../../api:sequence_checker",
+    "../../../api/transport:network_control",
+    "../../../api/units:data_size",
+    "../../../api/units:timestamp",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:logging",
+    "../../../rtc_base:macromagic",
+    "../../../rtc_base:network_route",
+    "../../../rtc_base:rtc_numerics",
+    "../../../rtc_base/network:sent_packet",
+    "../../../rtc_base/synchronization:mutex",
+    "../../../rtc_base/system:no_unique_address",
+    "../../../system_wrappers",
+    "../../../system_wrappers:field_trial",
+    "../../rtp_rtcp:rtp_rtcp_format",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("congestion_controller_unittests") {
+    testonly = true
+
+    sources = [
+      "transport_feedback_adapter_unittest.cc",
+      "transport_feedback_demuxer_unittest.cc",
+    ]
+    deps = [
+      ":transport_feedback",
+      "../:congestion_controller",
+      "../../../api/transport:network_control",
+      "../../../logging:mocks",
+      "../../../rtc_base:checks",
+      "../../../rtc_base:safe_conversions",
+      "../../../rtc_base/network:sent_packet",
+      "../../../system_wrappers",
+      "../../../test:field_trial",
+      "../../../test:test_support",
+      "../../pacing",
+      "../../remote_bitrate_estimator",
+      "../../rtp_rtcp:rtp_rtcp_format",
+      "//testing/gmock",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc
new file mode 100644
index 0000000000..ffa373aeba
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc
@@ -0,0 +1,89 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/control_handler.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "modules/pacing/pacing_controller.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+namespace {
+
+// By default, pacer emergency stops encoder when buffer reaches a high level.
+bool IsPacerEmergencyStopDisabled() {
+  return field_trial::IsEnabled("WebRTC-DisablePacerEmergencyStop");
+}
+
+}  // namespace
+CongestionControlHandler::CongestionControlHandler()
+    : disable_pacer_emergency_stop_(IsPacerEmergencyStopDisabled()) {
+  sequenced_checker_.Detach();
+}
+
+CongestionControlHandler::~CongestionControlHandler() {}
+
+void CongestionControlHandler::SetTargetRate(
+    TargetTransferRate new_target_rate) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  RTC_CHECK(new_target_rate.at_time.IsFinite());
+  last_incoming_ = new_target_rate;
+}
+
+void CongestionControlHandler::SetNetworkAvailability(bool network_available) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  network_available_ = network_available;
+}
+
+void CongestionControlHandler::SetPacerQueue(TimeDelta expected_queue_time) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  pacer_expected_queue_ms_ = expected_queue_time.ms();
+}
+
+absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  if (!last_incoming_.has_value())
+    return absl::nullopt;
+  TargetTransferRate new_outgoing = *last_incoming_;
+  DataRate log_target_rate = new_outgoing.target_rate;
+  bool pause_encoding = false;
+  if (!network_available_) {
+    pause_encoding = true;
+  } else if (!disable_pacer_emergency_stop_ &&
+             pacer_expected_queue_ms_ >
+                 PacingController::kMaxExpectedQueueLength.ms()) {
+    pause_encoding = true;
+  }
+  if (pause_encoding)
+    new_outgoing.target_rate = DataRate::Zero();
+  if (!last_reported_ ||
+      last_reported_->target_rate != new_outgoing.target_rate ||
+      (!new_outgoing.target_rate.IsZero() &&
+       (last_reported_->network_estimate.loss_rate_ratio !=
+            new_outgoing.network_estimate.loss_rate_ratio ||
+        last_reported_->network_estimate.round_trip_time !=
+            new_outgoing.network_estimate.round_trip_time))) {
+    if (encoder_paused_in_last_report_ != pause_encoding)
+      RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
+                       << ToString(log_target_rate) << ".";
+    encoder_paused_in_last_report_ = pause_encoding;
+    last_reported_ = new_outgoing;
+    return new_outgoing;
+  }
+  return absl::nullopt;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h
new file mode 100644
index 0000000000..d8e7263a02
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+// This is used to observe the network controller state and route calls to
+// the proper handler. It also keeps cached values for safe asynchronous use.
+// This makes sure that things running on the worker queue can't access state
+// in RtpTransportControllerSend, which would risk causing data race on
+// destruction unless members are properly ordered.
+class CongestionControlHandler {
+ public:
+  CongestionControlHandler();
+  ~CongestionControlHandler();
+
+  CongestionControlHandler(const CongestionControlHandler&) = delete;
+  CongestionControlHandler& operator=(const CongestionControlHandler&) = delete;
+
+  void SetTargetRate(TargetTransferRate new_target_rate);
+  void SetNetworkAvailability(bool network_available);
+  void SetPacerQueue(TimeDelta expected_queue_time);
+  absl::optional<TargetTransferRate> GetUpdate();
+
+ private:
+  absl::optional<TargetTransferRate> last_incoming_;
+  absl::optional<TargetTransferRate> last_reported_;
+  bool network_available_ = true;
+  bool encoder_paused_in_last_report_ = false;
+
+  const bool disable_pacer_emergency_stop_;
+  int64_t pacer_expected_queue_ms_ = 0;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequenced_checker_;
+};
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build
new file mode 100644
index 0000000000..6ee156d2f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler_gn/moz.build
@@ -0,0 +1,232 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("control_handler_gn")
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
new file mode 100644
index 0000000000..e83d09d263
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <cmath>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr TimeDelta kSendTimeHistoryWindow = TimeDelta::Seconds(60);
+
+void InFlightBytesTracker::AddInFlightPacketBytes(
+    const PacketFeedback& packet) {
+  RTC_DCHECK(packet.sent.send_time.IsFinite());
+  auto it = in_flight_data_.find(packet.network_route);
+  if (it != in_flight_data_.end()) {
+    it->second += packet.sent.size;
+  } else {
+    in_flight_data_.insert({packet.network_route, packet.sent.size});
+  }
+}
+
+void InFlightBytesTracker::RemoveInFlightPacketBytes(
+    const PacketFeedback& packet) {
+  if (packet.sent.send_time.IsInfinite())
+    return;
+  auto it = in_flight_data_.find(packet.network_route);
+  if (it != in_flight_data_.end()) {
+    RTC_DCHECK_GE(it->second, packet.sent.size);
+    it->second -= packet.sent.size;
+    if (it->second.IsZero())
+      in_flight_data_.erase(it);
+  }
+}
+
+DataSize InFlightBytesTracker::GetOutstandingData(
+    const rtc::NetworkRoute& network_route) const {
+  auto it = in_flight_data_.find(network_route);
+  if (it != in_flight_data_.end()) {
+    return it->second;
+  } else {
+    return DataSize::Zero();
+  }
+}
+
+// Comparator for consistent map with NetworkRoute as key.
+bool InFlightBytesTracker::NetworkRouteComparator::operator()(
+    const rtc::NetworkRoute& a,
+    const rtc::NetworkRoute& b) const {
+  if (a.local.network_id() != b.local.network_id())
+    return a.local.network_id() < b.local.network_id();
+  if (a.remote.network_id() != b.remote.network_id())
+    return a.remote.network_id() < b.remote.network_id();
+
+  if (a.local.adapter_id() != b.local.adapter_id())
+    return a.local.adapter_id() < b.local.adapter_id();
+  if (a.remote.adapter_id() != b.remote.adapter_id())
+    return a.remote.adapter_id() < b.remote.adapter_id();
+
+  if (a.local.uses_turn() != b.local.uses_turn())
+    return a.local.uses_turn() < b.local.uses_turn();
+  if (a.remote.uses_turn() != b.remote.uses_turn())
+    return a.remote.uses_turn() < b.remote.uses_turn();
+
+  return a.connected < b.connected;
+}
+
+TransportFeedbackAdapter::TransportFeedbackAdapter() = default;
+
+
+void TransportFeedbackAdapter::AddPacket(const RtpPacketSendInfo& packet_info,
+                                         size_t overhead_bytes,
+                                         Timestamp creation_time) {
+  PacketFeedback packet;
+  packet.creation_time = creation_time;
+  packet.sent.sequence_number =
+      seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number);
+  packet.sent.size = DataSize::Bytes(packet_info.length + overhead_bytes);
+  packet.sent.audio = packet_info.packet_type == RtpPacketMediaType::kAudio;
+  packet.network_route = network_route_;
+  packet.sent.pacing_info = packet_info.pacing_info;
+
+  while (!history_.empty() &&
+         creation_time - history_.begin()->second.creation_time >
+             kSendTimeHistoryWindow) {
+    // TODO(sprang): Warn if erasing (too many) old items?
+    if (history_.begin()->second.sent.sequence_number > last_ack_seq_num_)
+      in_flight_.RemoveInFlightPacketBytes(history_.begin()->second);
+    history_.erase(history_.begin());
+  }
+  history_.insert(std::make_pair(packet.sent.sequence_number, packet));
+}
+
+absl::optional<SentPacket> TransportFeedbackAdapter::ProcessSentPacket(
+    const rtc::SentPacket& sent_packet) {
+  auto send_time = Timestamp::Millis(sent_packet.send_time_ms);
+  // TODO(srte): Only use one way to indicate that packet feedback is used.
+  if (sent_packet.info.included_in_feedback || sent_packet.packet_id != -1) {
+    int64_t unwrapped_seq_num =
+        seq_num_unwrapper_.Unwrap(sent_packet.packet_id);
+    auto it = history_.find(unwrapped_seq_num);
+    if (it != history_.end()) {
+      bool packet_retransmit = it->second.sent.send_time.IsFinite();
+      it->second.sent.send_time = send_time;
+      last_send_time_ = std::max(last_send_time_, send_time);
+      // TODO(srte): Don't do this on retransmit.
+      if (!pending_untracked_size_.IsZero()) {
+        if (send_time < last_untracked_send_time_)
+          RTC_LOG(LS_WARNING)
+              << "appending acknowledged data for out of order packet. (Diff: "
+              << ToString(last_untracked_send_time_ - send_time) << " ms.)";
+        it->second.sent.prior_unacked_data += pending_untracked_size_;
+        pending_untracked_size_ = DataSize::Zero();
+      }
+      if (!packet_retransmit) {
+        if (it->second.sent.sequence_number > last_ack_seq_num_)
+          in_flight_.AddInFlightPacketBytes(it->second);
+        it->second.sent.data_in_flight = GetOutstandingData();
+        return it->second.sent;
+      }
+    }
+  } else if (sent_packet.info.included_in_allocation) {
+    if (send_time < last_send_time_) {
+      RTC_LOG(LS_WARNING) << "ignoring untracked data for out of order packet.";
+    }
+    pending_untracked_size_ +=
+        DataSize::Bytes(sent_packet.info.packet_size_bytes);
+    last_untracked_send_time_ = std::max(last_untracked_send_time_, send_time);
+  }
+  return absl::nullopt;
+}
+
+absl::optional<TransportPacketsFeedback>
+TransportFeedbackAdapter::ProcessTransportFeedback(
+    const rtcp::TransportFeedback& feedback,
+    Timestamp feedback_receive_time) {
+  if (feedback.GetPacketStatusCount() == 0) {
+    RTC_LOG(LS_INFO) << "Empty transport feedback packet received.";
+    return absl::nullopt;
+  }
+
+  TransportPacketsFeedback msg;
+  msg.feedback_time = feedback_receive_time;
+
+  msg.prior_in_flight = in_flight_.GetOutstandingData(network_route_);
+  msg.packet_feedbacks =
+      ProcessTransportFeedbackInner(feedback, feedback_receive_time);
+  if (msg.packet_feedbacks.empty())
+    return absl::nullopt;
+
+  auto it = history_.find(last_ack_seq_num_);
+  if (it != history_.end()) {
+    msg.first_unacked_send_time = it->second.sent.send_time;
+  }
+  msg.data_in_flight = in_flight_.GetOutstandingData(network_route_);
+
+  return msg;
+}
+
+void TransportFeedbackAdapter::SetNetworkRoute(
+    const rtc::NetworkRoute& network_route) {
+  network_route_ = network_route;
+}
+
+DataSize TransportFeedbackAdapter::GetOutstandingData() const {
+  return in_flight_.GetOutstandingData(network_route_);
+}
+
+std::vector<PacketResult>
+TransportFeedbackAdapter::ProcessTransportFeedbackInner(
+    const rtcp::TransportFeedback& feedback,
+    Timestamp feedback_receive_time) {
+  // Add timestamp deltas to a local time base selected on first packet arrival.
+  // This won't be the true time base, but makes it easier to manually inspect
+  // time stamps.
+  if (last_timestamp_.IsInfinite()) {
+    current_offset_ = feedback_receive_time;
+  } else {
+    // TODO(srte): We shouldn't need to do rounding here.
+    const TimeDelta delta = feedback.GetBaseDelta(last_timestamp_)
+                                .RoundDownTo(TimeDelta::Millis(1));
+    // Protect against assigning current_offset_ negative value.
+    if (delta < Timestamp::Zero() - current_offset_) {
+      RTC_LOG(LS_WARNING) << "Unexpected feedback timestamp received.";
+      current_offset_ = feedback_receive_time;
+    } else {
+      current_offset_ += delta;
+    }
+  }
+  last_timestamp_ = feedback.BaseTime();
+
+  std::vector<PacketResult> packet_result_vector;
+  packet_result_vector.reserve(feedback.GetPacketStatusCount());
+
+  size_t failed_lookups = 0;
+  size_t ignored = 0;
+
+  feedback.ForAllPackets([&](uint16_t sequence_number,
+                             TimeDelta delta_since_base) {
+    int64_t seq_num = seq_num_unwrapper_.Unwrap(sequence_number);
+
+    if (seq_num > last_ack_seq_num_) {
+      // Starts at history_.begin() if last_ack_seq_num_ < 0, since any valid
+      // sequence number is >= 0.
+      for (auto it = history_.upper_bound(last_ack_seq_num_);
+           it != history_.upper_bound(seq_num); ++it) {
+        in_flight_.RemoveInFlightPacketBytes(it->second);
+      }
+      last_ack_seq_num_ = seq_num;
+    }
+
+    auto it = history_.find(seq_num);
+    if (it == history_.end()) {
+      ++failed_lookups;
+      return;
+    }
+
+    if (it->second.sent.send_time.IsInfinite()) {
+      // TODO(srte): Fix the tests that makes this happen and make this a
+      // DCHECK.
+      RTC_DLOG(LS_ERROR)
+          << "Received feedback before packet was indicated as sent";
+      return;
+    }
+
+    PacketFeedback packet_feedback = it->second;
+    if (delta_since_base.IsFinite()) {
+      packet_feedback.receive_time =
+          current_offset_ + delta_since_base.RoundDownTo(TimeDelta::Millis(1));
+      // Note: Lost packets are not removed from history because they might be
+      // reported as received by a later feedback.
+      history_.erase(it);
+    }
+    if (packet_feedback.network_route == network_route_) {
+      PacketResult result;
+      result.sent_packet = packet_feedback.sent;
+      result.receive_time = packet_feedback.receive_time;
+      packet_result_vector.push_back(result);
+    } else {
+      ++ignored;
+    }
+  });
+
+  if (failed_lookups > 0) {
+    RTC_LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
+                        << " packet" << (failed_lookups > 1 ? "s" : "")
+                        << ". Send time history too small?";
+  }
+  if (ignored > 0) {
+    RTC_LOG(LS_INFO) << "Ignoring " << ignored
+                     << " packets because they were sent on a different route.";
+  }
+
+  return packet_result_vector;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
new file mode 100644
index 0000000000..7b1243b64b
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
+
+#include <deque>
+#include <map>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+struct PacketFeedback {
+  PacketFeedback() = default;
+  // Time corresponding to when this object was created.
+  Timestamp creation_time = Timestamp::MinusInfinity();
+  SentPacket sent;
+  // Time corresponding to when the packet was received. Timestamped with the
+  // receiver's clock. For unreceived packet, Timestamp::PlusInfinity() is
+  // used.
+  Timestamp receive_time = Timestamp::PlusInfinity();
+
+  // The network route that this packet is associated with.
+  rtc::NetworkRoute network_route;
+};
+
+class InFlightBytesTracker {
+ public:
+  void AddInFlightPacketBytes(const PacketFeedback& packet);
+  void RemoveInFlightPacketBytes(const PacketFeedback& packet);
+  DataSize GetOutstandingData(const rtc::NetworkRoute& network_route) const;
+
+ private:
+  struct NetworkRouteComparator {
+    bool operator()(const rtc::NetworkRoute& a,
+                    const rtc::NetworkRoute& b) const;
+  };
+  std::map<rtc::NetworkRoute, DataSize, NetworkRouteComparator> in_flight_data_;
+};
+
+class TransportFeedbackAdapter {
+ public:
+  TransportFeedbackAdapter();
+
+  void AddPacket(const RtpPacketSendInfo& packet_info,
+                 size_t overhead_bytes,
+                 Timestamp creation_time);
+  absl::optional<SentPacket> ProcessSentPacket(
+      const rtc::SentPacket& sent_packet);
+
+  absl::optional<TransportPacketsFeedback> ProcessTransportFeedback(
+      const rtcp::TransportFeedback& feedback,
+      Timestamp feedback_receive_time);
+
+  void SetNetworkRoute(const rtc::NetworkRoute& network_route);
+
+  DataSize GetOutstandingData() const;
+
+ private:
+  enum class SendTimeHistoryStatus { kNotAdded, kOk, kDuplicate };
+
+  std::vector<PacketResult> ProcessTransportFeedbackInner(
+      const rtcp::TransportFeedback& feedback,
+      Timestamp feedback_receive_time);
+
+  DataSize pending_untracked_size_ = DataSize::Zero();
+  Timestamp last_send_time_ = Timestamp::MinusInfinity();
+  Timestamp last_untracked_send_time_ = Timestamp::MinusInfinity();
+  RtpSequenceNumberUnwrapper seq_num_unwrapper_;
+  std::map<int64_t, PacketFeedback> history_;
+
+  // Sequence numbers are never negative, using -1 as it always < a real
+  // sequence number.
+  int64_t last_ack_seq_num_ = -1;
+  InFlightBytesTracker in_flight_;
+
+  Timestamp current_offset_ = Timestamp::MinusInfinity();
+  Timestamp last_timestamp_ = Timestamp::MinusInfinity();
+
+  rtc::NetworkRoute network_route_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
new file mode 100644
index 0000000000..14a2b13831
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
@@ -0,0 +1,407 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+
+#include <limits>
+#include <memory>
+#include <vector>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+using ::testing::_;
+using ::testing::Invoke;
+
+namespace webrtc {
+
+namespace {
+constexpr uint32_t kSsrc = 8492;
+const PacedPacketInfo kPacingInfo0(0, 5, 2000);
+const PacedPacketInfo kPacingInfo1(1, 8, 4000);
+const PacedPacketInfo kPacingInfo2(2, 14, 7000);
+const PacedPacketInfo kPacingInfo3(3, 20, 10000);
+const PacedPacketInfo kPacingInfo4(4, 22, 10000);
+
+void ComparePacketFeedbackVectors(const std::vector<PacketResult>& truth,
+                                  const std::vector<PacketResult>& input) {
+  ASSERT_EQ(truth.size(), input.size());
+  size_t len = truth.size();
+  // truth contains the input data for the test, and input is what will be
+  // sent to the bandwidth estimator. truth.arrival_tims_ms is used to
+  // populate the transport feedback messages. As these times may be changed
+  // (because of resolution limits in the packets, and because of the time
+  // base adjustment performed by the TransportFeedbackAdapter at the first
+  // packet, the truth[x].arrival_time and input[x].arrival_time may not be
+  // equal. However, the difference must be the same for all x.
+  TimeDelta arrival_time_delta = truth[0].receive_time - input[0].receive_time;
+  for (size_t i = 0; i < len; ++i) {
+    RTC_CHECK(truth[i].IsReceived());
+    if (input[i].IsReceived()) {
+      EXPECT_EQ(truth[i].receive_time - input[i].receive_time,
+                arrival_time_delta);
+    }
+    EXPECT_EQ(truth[i].sent_packet.send_time, input[i].sent_packet.send_time);
+    EXPECT_EQ(truth[i].sent_packet.sequence_number,
+              input[i].sent_packet.sequence_number);
+    EXPECT_EQ(truth[i].sent_packet.size, input[i].sent_packet.size);
+    EXPECT_EQ(truth[i].sent_packet.pacing_info,
+              input[i].sent_packet.pacing_info);
+  }
+}
+
+PacketResult CreatePacket(int64_t receive_time_ms,
+                          int64_t send_time_ms,
+                          int64_t sequence_number,
+                          size_t payload_size,
+                          const PacedPacketInfo& pacing_info) {
+  PacketResult res;
+  res.receive_time = Timestamp::Millis(receive_time_ms);
+  res.sent_packet.send_time = Timestamp::Millis(send_time_ms);
+  res.sent_packet.sequence_number = sequence_number;
+  res.sent_packet.size = DataSize::Bytes(payload_size);
+  res.sent_packet.pacing_info = pacing_info;
+  return res;
+}
+
+class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver {
+ public:
+  MOCK_METHOD(void,
+              OnPacketFeedbackVector,
+              (std::vector<StreamPacketInfo> packet_feedback_vector),
+              (override));
+};
+
+}  // namespace
+
+class TransportFeedbackAdapterTest : public ::testing::Test {
+ public:
+  TransportFeedbackAdapterTest() : clock_(0) {}
+
+  virtual ~TransportFeedbackAdapterTest() {}
+
+  virtual void SetUp() { adapter_.reset(new TransportFeedbackAdapter()); }
+
+  virtual void TearDown() { adapter_.reset(); }
+
+ protected:
+  void OnReceivedEstimatedBitrate(uint32_t bitrate) {}
+
+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
+                                    int64_t rtt,
+                                    int64_t now_ms) {}
+
+  void OnSentPacket(const PacketResult& packet_feedback) {
+    RtpPacketSendInfo packet_info;
+    packet_info.media_ssrc = kSsrc;
+    packet_info.transport_sequence_number =
+        packet_feedback.sent_packet.sequence_number;
+    packet_info.rtp_sequence_number = 0;
+    packet_info.length = packet_feedback.sent_packet.size.bytes();
+    packet_info.pacing_info = packet_feedback.sent_packet.pacing_info;
+    packet_info.packet_type = RtpPacketMediaType::kVideo;
+    adapter_->AddPacket(RtpPacketSendInfo(packet_info), 0u,
+                        clock_.CurrentTime());
+    adapter_->ProcessSentPacket(rtc::SentPacket(
+        packet_feedback.sent_packet.sequence_number,
+        packet_feedback.sent_packet.send_time.ms(), rtc::PacketInfo()));
+  }
+
+  SimulatedClock clock_;
+  std::unique_ptr<TransportFeedbackAdapter> adapter_;
+};
+
+TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(100, 200, 0, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(110, 210, 1, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(120, 220, 2, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(130, 230, 3, 1500, kPacingInfo1));
+  packets.push_back(CreatePacket(140, 240, 4, 1500, kPacingInfo1));
+
+  for (const auto& packet : packets)
+    OnSentPacket(packet);
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(packets[0].sent_packet.sequence_number,
+                   packets[0].receive_time);
+
+  for (const auto& packet : packets) {
+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                           packet.receive_time));
+  }
+
+  feedback.Build();
+
+  auto result =
+      adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(packets, result->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
+  std::vector<PacketResult> sent_packets = {
+      CreatePacket(100, 220, 0, 1500, kPacingInfo0),
+      CreatePacket(110, 210, 1, 1500, kPacingInfo0),
+      CreatePacket(120, 220, 2, 1500, kPacingInfo0),
+      CreatePacket(130, 230, 3, 1500, kPacingInfo0),
+      CreatePacket(140, 240, 4, 1500, kPacingInfo0),
+      CreatePacket(150, 250, 5, 1500, kPacingInfo0),
+      CreatePacket(160, 260, 6, 1500, kPacingInfo0)};
+
+  for (const auto& packet : sent_packets)
+    OnSentPacket(packet);
+
+  // Note: Important to include the last packet, as only unreceived packets in
+  // between received packets can be inferred.
+  std::vector<PacketResult> received_packets = {
+      sent_packets[0], sent_packets[2], sent_packets[6]};
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(received_packets[0].sent_packet.sequence_number,
+                   received_packets[0].receive_time);
+
+  for (const auto& packet : received_packets) {
+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                           packet.receive_time));
+  }
+
+  feedback.Build();
+
+  auto res = adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(sent_packets, res->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(100, 200, 0, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(110, 210, 1, 1500, kPacingInfo1));
+  packets.push_back(CreatePacket(120, 220, 2, 1500, kPacingInfo2));
+  packets.push_back(CreatePacket(130, 230, 3, 1500, kPacingInfo3));
+  packets.push_back(CreatePacket(140, 240, 4, 1500, kPacingInfo4));
+
+  const uint16_t kSendSideDropBefore = 1;
+  const uint16_t kReceiveSideDropAfter = 3;
+
+  for (const auto& packet : packets) {
+    if (packet.sent_packet.sequence_number >= kSendSideDropBefore)
+      OnSentPacket(packet);
+  }
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(packets[0].sent_packet.sequence_number,
+                   packets[0].receive_time);
+
+  for (const auto& packet : packets) {
+    if (packet.sent_packet.sequence_number <= kReceiveSideDropAfter) {
+      EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                             packet.receive_time));
+    }
+  }
+
+  feedback.Build();
+
+  std::vector<PacketResult> expected_packets(
+      packets.begin() + kSendSideDropBefore,
+      packets.begin() + kReceiveSideDropAfter + 1);
+  // Packets that have timed out on the send-side have lost the
+  // information stored on the send-side. And they will not be reported to
+  // observers since we won't know that they come from the same networks.
+
+  auto res = adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(expected_packets, res->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
+  TimeDelta kHighArrivalTime =
+      rtcp::TransportFeedback::kDeltaTick * (1 << 8) * ((1 << 23) - 1);
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(kHighArrivalTime.ms() + 64, 210, 0, 1500,
+                                 PacedPacketInfo()));
+  packets.push_back(CreatePacket(kHighArrivalTime.ms() - 64, 210, 1, 1500,
+                                 PacedPacketInfo()));
+  packets.push_back(
+      CreatePacket(kHighArrivalTime.ms(), 220, 2, 1500, PacedPacketInfo()));
+
+  for (const auto& packet : packets)
+    OnSentPacket(packet);
+
+  for (size_t i = 0; i < packets.size(); ++i) {
+    std::unique_ptr<rtcp::TransportFeedback> feedback(
+        new rtcp::TransportFeedback());
+    feedback->SetBase(packets[i].sent_packet.sequence_number,
+                      packets[i].receive_time);
+
+    EXPECT_TRUE(feedback->AddReceivedPacket(
+        packets[i].sent_packet.sequence_number, packets[i].receive_time));
+
+    rtc::Buffer raw_packet = feedback->Build();
+    feedback = rtcp::TransportFeedback::ParseFrom(raw_packet.data(),
+                                                  raw_packet.size());
+
+    std::vector<PacketResult> expected_packets;
+    expected_packets.push_back(packets[i]);
+
+    auto res = adapter_->ProcessTransportFeedback(*feedback.get(),
+                                                  clock_.CurrentTime());
+    ComparePacketFeedbackVectors(expected_packets, res->packet_feedbacks);
+  }
+}
+
+TEST_F(TransportFeedbackAdapterTest, HandlesArrivalReordering) {
+  std::vector<PacketResult> packets;
+  packets.push_back(CreatePacket(120, 200, 0, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(110, 210, 1, 1500, kPacingInfo0));
+  packets.push_back(CreatePacket(100, 220, 2, 1500, kPacingInfo0));
+
+  for (const auto& packet : packets)
+    OnSentPacket(packet);
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(packets[0].sent_packet.sequence_number,
+                   packets[0].receive_time);
+
+  for (const auto& packet : packets) {
+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sent_packet.sequence_number,
+                                           packet.receive_time));
+  }
+
+  feedback.Build();
+
+  // Adapter keeps the packets ordered by sequence number (which is itself
+  // assigned by the order of transmission). Reordering by some other criteria,
+  // eg. arrival time, is up to the observers.
+  auto res = adapter_->ProcessTransportFeedback(feedback, clock_.CurrentTime());
+  ComparePacketFeedbackVectors(packets, res->packet_feedbacks);
+}
+
+TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
+  std::vector<PacketResult> sent_packets;
+  // TODO(srte): Consider using us resolution in the constants.
+  const TimeDelta kSmallDelta = (rtcp::TransportFeedback::kDeltaTick * 0xFF)
+                                    .RoundDownTo(TimeDelta::Millis(1));
+  const TimeDelta kLargePositiveDelta = (rtcp::TransportFeedback::kDeltaTick *
+                                         std::numeric_limits<int16_t>::max())
+                                            .RoundDownTo(TimeDelta::Millis(1));
+  const TimeDelta kLargeNegativeDelta = (rtcp::TransportFeedback::kDeltaTick *
+                                         std::numeric_limits<int16_t>::min())
+                                            .RoundDownTo(TimeDelta::Millis(1));
+
+  PacketResult packet_feedback;
+  packet_feedback.sent_packet.sequence_number = 1;
+  packet_feedback.sent_packet.send_time = Timestamp::Millis(100);
+  packet_feedback.receive_time = Timestamp::Millis(200);
+  packet_feedback.sent_packet.size = DataSize::Bytes(1500);
+  sent_packets.push_back(packet_feedback);
+
+  // TODO(srte): This rounding maintains previous behavior, but should ot be
+  // required.
+  packet_feedback.sent_packet.send_time += kSmallDelta;
+  packet_feedback.receive_time += kSmallDelta;
+  ++packet_feedback.sent_packet.sequence_number;
+  sent_packets.push_back(packet_feedback);
+
+  packet_feedback.sent_packet.send_time += kLargePositiveDelta;
+  packet_feedback.receive_time += kLargePositiveDelta;
+  ++packet_feedback.sent_packet.sequence_number;
+  sent_packets.push_back(packet_feedback);
+
+  packet_feedback.sent_packet.send_time += kLargeNegativeDelta;
+  packet_feedback.receive_time += kLargeNegativeDelta;
+  ++packet_feedback.sent_packet.sequence_number;
+  sent_packets.push_back(packet_feedback);
+
+  // Too large, delta - will need two feedback messages.
+  packet_feedback.sent_packet.send_time +=
+      kLargePositiveDelta + TimeDelta::Millis(1);
+  packet_feedback.receive_time += kLargePositiveDelta + TimeDelta::Millis(1);
+  ++packet_feedback.sent_packet.sequence_number;
+
+  // Packets will be added to send history.
+  for (const auto& packet : sent_packets)
+    OnSentPacket(packet);
+  OnSentPacket(packet_feedback);
+
+  // Create expected feedback and send into adapter.
+  std::unique_ptr<rtcp::TransportFeedback> feedback(
+      new rtcp::TransportFeedback());
+  feedback->SetBase(sent_packets[0].sent_packet.sequence_number,
+                    sent_packets[0].receive_time);
+
+  for (const auto& packet : sent_packets) {
+    EXPECT_TRUE(feedback->AddReceivedPacket(packet.sent_packet.sequence_number,
+                                            packet.receive_time));
+  }
+  EXPECT_FALSE(
+      feedback->AddReceivedPacket(packet_feedback.sent_packet.sequence_number,
+                                  packet_feedback.receive_time));
+
+  rtc::Buffer raw_packet = feedback->Build();
+  feedback =
+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
+
+  std::vector<PacketResult> received_feedback;
+
+  EXPECT_TRUE(feedback.get() != nullptr);
+  auto res =
+      adapter_->ProcessTransportFeedback(*feedback.get(), clock_.CurrentTime());
+  ComparePacketFeedbackVectors(sent_packets, res->packet_feedbacks);
+
+  // Create a new feedback message and add the trailing item.
+  feedback.reset(new rtcp::TransportFeedback());
+  feedback->SetBase(packet_feedback.sent_packet.sequence_number,
+                    packet_feedback.receive_time);
+  EXPECT_TRUE(
+      feedback->AddReceivedPacket(packet_feedback.sent_packet.sequence_number,
+                                  packet_feedback.receive_time));
+  raw_packet = feedback->Build();
+  feedback =
+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
+
+  EXPECT_TRUE(feedback.get() != nullptr);
+  {
+    auto res = adapter_->ProcessTransportFeedback(*feedback.get(),
+                                                  clock_.CurrentTime());
+    std::vector<PacketResult> expected_packets;
+    expected_packets.push_back(packet_feedback);
+    ComparePacketFeedbackVectors(expected_packets, res->packet_feedbacks);
+  }
+}
+
+TEST_F(TransportFeedbackAdapterTest, IgnoreDuplicatePacketSentCalls) {
+  auto packet = CreatePacket(100, 200, 0, 1500, kPacingInfo0);
+
+  // Add a packet and then mark it as sent.
+  RtpPacketSendInfo packet_info;
+  packet_info.media_ssrc = kSsrc;
+  packet_info.transport_sequence_number = packet.sent_packet.sequence_number;
+  packet_info.length = packet.sent_packet.size.bytes();
+  packet_info.pacing_info = packet.sent_packet.pacing_info;
+  packet_info.packet_type = RtpPacketMediaType::kVideo;
+  adapter_->AddPacket(packet_info, 0u, clock_.CurrentTime());
+  absl::optional<SentPacket> sent_packet = adapter_->ProcessSentPacket(
+      rtc::SentPacket(packet.sent_packet.sequence_number,
+                      packet.sent_packet.send_time.ms(), rtc::PacketInfo()));
+  EXPECT_TRUE(sent_packet.has_value());
+
+  // Call ProcessSentPacket() again with the same sequence number. This packet
+  // has already been marked as sent and the call should be ignored.
+  absl::optional<SentPacket> duplicate_packet = adapter_->ProcessSentPacket(
+      rtc::SentPacket(packet.sent_packet.sequence_number,
+                      packet.sent_packet.send_time.ms(), rtc::PacketInfo()));
+  EXPECT_FALSE(duplicate_packet.has_value());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc
new file mode 100644
index 0000000000..469c21434a
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc
@@ -0,0 +1,94 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
+#include "absl/algorithm/container.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
+namespace webrtc {
+namespace {
+static const size_t kMaxPacketsInHistory = 5000;
+}
+
+TransportFeedbackDemuxer::TransportFeedbackDemuxer() {
+  // In case the construction thread is different from where the registration
+  // and callbacks occur, detach from the construction thread.
+  observer_checker_.Detach();
+}
+
+void TransportFeedbackDemuxer::RegisterStreamFeedbackObserver(
+    std::vector<uint32_t> ssrcs,
+    StreamFeedbackObserver* observer) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+  RTC_DCHECK(observer);
+  RTC_DCHECK(absl::c_find_if(observers_, [=](const auto& pair) {
+               return pair.second == observer;
+             }) == observers_.end());
+  observers_.push_back({ssrcs, observer});
+}
+
+void TransportFeedbackDemuxer::DeRegisterStreamFeedbackObserver(
+    StreamFeedbackObserver* observer) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+  RTC_DCHECK(observer);
+  const auto it = absl::c_find_if(
+      observers_, [=](const auto& pair) { return pair.second == observer; });
+  RTC_DCHECK(it != observers_.end());
+  observers_.erase(it);
+}
+
+void TransportFeedbackDemuxer::AddPacket(const RtpPacketSendInfo& packet_info) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+
+  StreamFeedbackObserver::StreamPacketInfo info;
+  info.ssrc = packet_info.media_ssrc;
+  info.rtp_sequence_number = packet_info.rtp_sequence_number;
+  info.received = false;
+  info.is_retransmission =
+      packet_info.packet_type == RtpPacketMediaType::kRetransmission;
+  history_.insert(
+      {seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number), info});
+
+  while (history_.size() > kMaxPacketsInHistory) {
+    history_.erase(history_.begin());
+  }
+}
+
+void TransportFeedbackDemuxer::OnTransportFeedback(
+    const rtcp::TransportFeedback& feedback) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+
+  std::vector<StreamFeedbackObserver::StreamPacketInfo> stream_feedbacks;
+  feedback.ForAllPackets(
+      [&](uint16_t sequence_number, TimeDelta delta_since_base) {
+        RTC_DCHECK_RUN_ON(&observer_checker_);
+        auto it = history_.find(seq_num_unwrapper_.PeekUnwrap(sequence_number));
+        if (it != history_.end()) {
+          auto packet_info = it->second;
+          packet_info.received = delta_since_base.IsFinite();
+          stream_feedbacks.push_back(std::move(packet_info));
+          if (delta_since_base.IsFinite())
+            history_.erase(it);
+        }
+      });
+
+  for (auto& observer : observers_) {
+    std::vector<StreamFeedbackObserver::StreamPacketInfo> selected_feedback;
+    for (const auto& packet_info : stream_feedbacks) {
+      if (absl::c_count(observer.first, packet_info.ssrc) > 0) {
+        selected_feedback.push_back(packet_info);
+      }
+    }
+    if (!selected_feedback.empty()) {
+      observer.second->OnPacketFeedbackVector(std::move(selected_feedback));
+    }
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h
new file mode 100644
index 0000000000..278c144b61
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h
@@ -0,0 +1,62 @@
+/*
+ *  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 MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
+
+#include <map>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+
+// Implementation of StreamFeedbackProvider that provides a way for
+// implementations of StreamFeedbackObserver to register for feedback callbacks
+// for a given set of SSRCs.
+// Registration methods need to be called from the same execution context
+// (thread or task queue) and callbacks to
+// StreamFeedbackObserver::OnPacketFeedbackVector will be made in that same
+// context.
+// TODO(tommi): This appears to be the only implementation of this interface.
+// Do we need the interface?
+class TransportFeedbackDemuxer final : public StreamFeedbackProvider {
+ public:
+  TransportFeedbackDemuxer();
+
+  // Implements StreamFeedbackProvider interface
+  void RegisterStreamFeedbackObserver(
+      std::vector<uint32_t> ssrcs,
+      StreamFeedbackObserver* observer) override;
+  void DeRegisterStreamFeedbackObserver(
+      StreamFeedbackObserver* observer) override;
+  void AddPacket(const RtpPacketSendInfo& packet_info);
+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker observer_checker_;
+  RtpSequenceNumberUnwrapper seq_num_unwrapper_
+      RTC_GUARDED_BY(&observer_checker_);
+  std::map<int64_t, StreamFeedbackObserver::StreamPacketInfo> history_
+      RTC_GUARDED_BY(&observer_checker_);
+
+  // Maps a set of ssrcs to corresponding observer. Vectors are used rather than
+  // set/map to ensure that the processing order is consistent independently of
+  // the randomized ssrcs.
+  std::vector<std::pair<std::vector<uint32_t>, StreamFeedbackObserver*>>
+      observers_ RTC_GUARDED_BY(&observer_checker_);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc
new file mode 100644
index 0000000000..52d8018bff
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc
@@ -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.
+ */
+#include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
+
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::AllOf;
+using ::testing::ElementsAre;
+using ::testing::Field;
+using PacketInfo = StreamFeedbackObserver::StreamPacketInfo;
+
+static constexpr uint32_t kSsrc = 8492;
+
+class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver {
+ public:
+  MOCK_METHOD(void,
+              OnPacketFeedbackVector,
+              (std::vector<StreamPacketInfo> packet_feedback_vector),
+              (override));
+};
+
+RtpPacketSendInfo CreatePacket(uint32_t ssrc,
+                               uint16_t rtp_sequence_number,
+                               int64_t transport_sequence_number,
+                               bool is_retransmission) {
+  RtpPacketSendInfo res;
+  res.media_ssrc = ssrc;
+  res.transport_sequence_number = transport_sequence_number;
+  res.rtp_sequence_number = rtp_sequence_number;
+  res.packet_type = is_retransmission ? RtpPacketMediaType::kRetransmission
+                                      : RtpPacketMediaType::kVideo;
+  return res;
+}
+}  // namespace
+
+TEST(TransportFeedbackDemuxerTest, ObserverSanity) {
+  TransportFeedbackDemuxer demuxer;
+  MockStreamFeedbackObserver mock;
+  demuxer.RegisterStreamFeedbackObserver({kSsrc}, &mock);
+
+  const uint16_t kRtpStartSeq = 55;
+  const int64_t kTransportStartSeq = 1;
+  demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq, kTransportStartSeq,
+                                 /*is_retransmit=*/false));
+  demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq + 1,
+                                 kTransportStartSeq + 1,
+                                 /*is_retransmit=*/false));
+  demuxer.AddPacket(CreatePacket(
+      kSsrc, kRtpStartSeq + 2, kTransportStartSeq + 2, /*is_retransmit=*/true));
+
+  rtcp::TransportFeedback feedback;
+  feedback.SetBase(kTransportStartSeq, Timestamp::Millis(1));
+  ASSERT_TRUE(
+      feedback.AddReceivedPacket(kTransportStartSeq, Timestamp::Millis(1)));
+  // Drop middle packet.
+  ASSERT_TRUE(
+      feedback.AddReceivedPacket(kTransportStartSeq + 2, Timestamp::Millis(3)));
+
+  EXPECT_CALL(
+      mock, OnPacketFeedbackVector(ElementsAre(
+                AllOf(Field(&PacketInfo::received, true),
+                      Field(&PacketInfo::ssrc, kSsrc),
+                      Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq),
+                      Field(&PacketInfo::is_retransmission, false)),
+                AllOf(Field(&PacketInfo::received, false),
+                      Field(&PacketInfo::ssrc, kSsrc),
+                      Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 1),
+                      Field(&PacketInfo::is_retransmission, false)),
+                AllOf(Field(&PacketInfo::received, true),
+                      Field(&PacketInfo::ssrc, kSsrc),
+                      Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 2),
+                      Field(&PacketInfo::is_retransmission, true)))));
+  demuxer.OnTransportFeedback(feedback);
+
+  demuxer.DeRegisterStreamFeedbackObserver(&mock);
+
+  demuxer.AddPacket(
+      CreatePacket(kSsrc, kRtpStartSeq + 3, kTransportStartSeq + 3, false));
+  rtcp::TransportFeedback second_feedback;
+  second_feedback.SetBase(kTransportStartSeq + 3, Timestamp::Millis(4));
+  ASSERT_TRUE(second_feedback.AddReceivedPacket(kTransportStartSeq + 3,
+                                                Timestamp::Millis(4)));
+
+  EXPECT_CALL(mock, OnPacketFeedbackVector).Times(0);
+  demuxer.OnTransportFeedback(second_feedback);
+}
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build
new file mode 100644
index 0000000000..eb5a2a87e0
--- /dev/null
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_gn/moz.build
@@ -0,0 +1,233 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc",
+    "/third_party/libwebrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+    DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "rt"
+    ]
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("transport_feedback_gn")