BASELINE: Update Chromium to 85.0.4183.14085-based

Change-Id: Iaa42f4680837c57725b1344f108c0196741f6057
Reviewed-by: Allan Sandfeld Jensen <allan.jensen@qt.io>

96 files changed, 1133 insertions, 2475 deletions

diff --git a/chromium/third_party/webrtc/api/BUILD.gn b/chromium/third_party/webrtc/api/BUILD.gn
index c5629c0fcda..571b89aacbc 100644
--- a/chromium/third_party/webrtc/api/BUILD.gn
+++ b/chromium/third_party/webrtc/api/BUILD.gn
@@ -71,8 +71,8 @@ rtc_library("rtp_headers") {
     "..:webrtc_common",
     "units:timestamp",
     "video:video_rtp_headers",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("rtp_packet_info") {
@@ -90,8 +90,8 @@ rtc_library("rtp_packet_info") {
     "..:webrtc_common",
     "../rtc_base:rtc_base_approved",
     "../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("media_stream_interface") {
@@ -111,8 +111,8 @@ rtc_library("media_stream_interface") {
     "../rtc_base/system:rtc_export",
     "video:recordable_encoded_frame",
     "video:video_frame",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("libjingle_peerconnection_api") {
@@ -166,12 +166,14 @@ rtc_library("libjingle_peerconnection_api") {
     ":media_stream_interface",
     ":network_state_predictor_api",
     ":packet_socket_factory",
+    ":priority",
     ":rtc_error",
     ":rtc_stats_api",
     ":rtp_packet_info",
     ":rtp_parameters",
     ":rtp_transceiver_direction",
     ":scoped_refptr",
+    "adaptation:resource_adaptation_api",
     "audio:audio_mixer_api",
     "audio_codecs:audio_codecs_api",
     "crypto:frame_decryptor_interface",
@@ -181,23 +183,15 @@ rtc_library("libjingle_peerconnection_api") {
     "rtc_event_log",
     "task_queue",
     "transport:bitrate_settings",
-    "transport:datagram_transport_interface",
     "transport:enums",
     "transport:network_control",
     "transport:webrtc_key_value_config",
-    "transport/media:audio_interfaces",
-    "transport/media:media_transport_interface",
-    "transport/media:video_interfaces",
     "transport/rtp:rtp_source",
     "units:data_rate",
     "units:timestamp",
     "video:encoded_image",
     "video:video_frame",
     "video:video_rtp_headers",
-    "//third_party/abseil-cpp/absl/algorithm:container",
-    "//third_party/abseil-cpp/absl/memory",
-    "//third_party/abseil-cpp/absl/strings",
-    "//third_party/abseil-cpp/absl/types:optional",
 
     # Basically, don't add stuff here. You might break sensitive downstream
     # targets like pnacl. API should not depend on anything outside of this
@@ -212,6 +206,12 @@ rtc_library("libjingle_peerconnection_api") {
     "../rtc_base:rtc_base_approved",
     "../rtc_base/system:rtc_export",
   ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
 }
 
 rtc_source_set("frame_transformer_interface") {
@@ -221,6 +221,7 @@ rtc_source_set("frame_transformer_interface") {
     ":scoped_refptr",
     "../rtc_base:refcount",
     "video:encoded_frame",
+    "video:video_frame_metadata",
   ]
 }
 
@@ -235,8 +236,8 @@ rtc_library("rtc_error") {
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("packet_socket_factory") {
@@ -272,7 +273,6 @@ rtc_source_set("video_quality_test_fixture_api") {
     "../test:video_test_common",
     "transport:bitrate_settings",
     "transport:network_control",
-    "transport/media:media_transport_interface",
     "video_codecs:video_codecs_api",
   ]
 }
@@ -283,11 +283,15 @@ rtc_source_set("video_quality_analyzer_api") {
   sources = [ "test/video_quality_analyzer_interface.h" ]
 
   deps = [
+    ":array_view",
     ":stats_observer_interface",
     "video:encoded_image",
     "video:video_frame",
     "video:video_rtp_headers",
     "video_codecs:video_codecs_api",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -302,6 +306,10 @@ rtc_source_set("rtp_transceiver_direction") {
   sources = [ "rtp_transceiver_direction.h" ]
 }
 
+rtc_source_set("priority") {
+  sources = [ "priority.h" ]
+}
+
 rtc_library("rtp_parameters") {
   visibility = [ "*" ]
   sources = [
@@ -312,18 +320,21 @@ rtc_library("rtp_parameters") {
   ]
   deps = [
     ":array_view",
+    ":priority",
     ":rtp_transceiver_direction",
     "../rtc_base:checks",
     "../rtc_base:stringutils",
     "../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
 
 if (is_android) {
-  java_cpp_enum("rtp_parameters_enums") {
-    sources = [ "rtp_parameters.h" ]
+  java_cpp_enum("priority_enums") {
+    sources = [ "priority.h" ]
   }
 }
 
@@ -343,11 +354,9 @@ rtc_source_set("stats_observer_interface") {
   testonly = true
   sources = [ "test/stats_observer_interface.h" ]
 
-  deps = [
-    # For api/stats_types.h
-    ":libjingle_peerconnection_api",
-    ":rtp_parameters",
-  ]
+  deps = [ ":rtc_stats_api" ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
 rtc_source_set("peer_connection_quality_test_fixture_api") {
@@ -370,14 +379,16 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
     ":stats_observer_interface",
     ":video_quality_analyzer_api",
     "../media:rtc_media_base",
+    "../rtc_base:deprecation",
     "../rtc_base:rtc_base",
     "rtc_event_log",
     "task_queue",
     "transport:network_control",
-    "transport/media:media_transport_interface",
     "units:time_delta",
     "video:video_frame",
     "video_codecs:video_codecs_api",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
@@ -392,8 +403,8 @@ rtc_source_set("frame_generator_api") {
   deps = [
     ":scoped_refptr",
     "video:video_frame",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("test_dependency_factory") {
@@ -406,7 +417,7 @@ rtc_library("test_dependency_factory") {
   deps = [
     ":video_quality_test_fixture_api",
     "../rtc_base:checks",
-    "../rtc_base:thread_checker",
+    "../rtc_base:platform_thread_types",
   ]
 }
 
@@ -470,8 +481,8 @@ rtc_library("create_frame_generator") {
     "../rtc_base:checks",
     "../system_wrappers",
     "../test:frame_generator_impl",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("create_peer_connection_quality_test_frame_generator") {
@@ -487,8 +498,8 @@ rtc_library("create_peer_connection_quality_test_frame_generator") {
     ":peer_connection_quality_test_fixture_api",
     "../rtc_base:checks",
     "../test:fileutils",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("libjingle_logging_api") {
@@ -541,8 +552,8 @@ rtc_library("audio_options_api") {
     ":array_view",
     "../rtc_base:stringutils",
     "../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("transport_api") {
@@ -569,8 +580,8 @@ rtc_source_set("simulated_network_api") {
   deps = [
     "../rtc_base",
     "../rtc_base:criticalsection",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 # TODO(srte): Move to network_emulation sub directory.
@@ -703,6 +714,8 @@ if (rtc_include_tests) {
         "../modules/audio_coding:neteq_test_factory",
         "../rtc_base:checks",
         "neteq:neteq_api",
+      ]
+      absl_deps = [
         "//third_party/abseil-cpp/absl/flags:flag",
         "//third_party/abseil-cpp/absl/flags:parse",
         "//third_party/abseil-cpp/absl/strings",
@@ -853,6 +866,7 @@ if (rtc_include_tests) {
   }
 
   rtc_source_set("mock_peerconnectioninterface") {
+    visibility = [ "*" ]
     testonly = true
     sources = [ "test/mock_peerconnectioninterface.h" ]
 
@@ -862,6 +876,17 @@ if (rtc_include_tests) {
     ]
   }
 
+  rtc_source_set("mock_peer_connection_factory_interface") {
+    visibility = [ "*" ]
+    testonly = true
+    sources = [ "test/mock_peer_connection_factory_interface.h" ]
+
+    deps = [
+      ":libjingle_peerconnection_api",
+      "../test:test_support",
+    ]
+  }
+
   rtc_source_set("mock_rtp") {
     testonly = true
     sources = [
@@ -875,6 +900,16 @@ if (rtc_include_tests) {
     ]
   }
 
+  rtc_source_set("mock_transformable_video_frame") {
+    testonly = true
+    sources = [ "test/mock_transformable_video_frame.h" ]
+
+    deps = [
+      ":frame_transformer_interface",
+      "../test:test_support",
+    ]
+  }
+
   rtc_source_set("mock_video_bitrate_allocator") {
     testonly = true
     sources = [ "test/mock_video_bitrate_allocator.h" ]
@@ -932,39 +967,6 @@ if (rtc_include_tests) {
     ]
   }
 
-  rtc_source_set("fake_media_transport") {
-    testonly = true
-
-    sources = [
-      "test/fake_datagram_transport.h",
-      "test/fake_media_transport.h",
-    ]
-
-    deps = [
-      "../rtc_base:checks",
-      "transport:datagram_transport_interface",
-      "transport/media:media_transport_interface",
-      "//third_party/abseil-cpp/absl/algorithm:container",
-    ]
-  }
-
-  rtc_library("loopback_media_transport") {
-    testonly = true
-
-    sources = [
-      "test/loopback_media_transport.cc",
-      "test/loopback_media_transport.h",
-    ]
-
-    deps = [
-      "../rtc_base",
-      "../rtc_base:checks",
-      "transport:datagram_transport_interface",
-      "transport/media:media_transport_interface",
-      "//third_party/abseil-cpp/absl/algorithm:container",
-    ]
-  }
-
   rtc_library("create_time_controller") {
     visibility = [ "*" ]
     testonly = true
@@ -995,7 +997,6 @@ if (rtc_include_tests) {
       "rtp_parameters_unittest.cc",
       "scoped_refptr_unittest.cc",
       "test/create_time_controller_unittest.cc",
-      "test/loopback_media_transport_unittest.cc",
     ]
 
     deps = [
@@ -1003,7 +1004,6 @@ if (rtc_include_tests) {
       ":create_time_controller",
       ":function_view",
       ":libjingle_peerconnection_api",
-      ":loopback_media_transport",
       ":rtc_error",
       ":rtc_event_log_output_file",
       ":rtp_packet_info",
@@ -1034,13 +1034,13 @@ if (rtc_include_tests) {
       ":dummy_peer_connection",
       ":fake_frame_decryptor",
       ":fake_frame_encryptor",
-      ":fake_media_transport",
-      ":loopback_media_transport",
       ":mock_audio_mixer",
       ":mock_frame_decryptor",
       ":mock_frame_encryptor",
+      ":mock_peer_connection_factory_interface",
       ":mock_peerconnectioninterface",
       ":mock_rtp",
+      ":mock_transformable_video_frame",
       ":mock_video_bitrate_allocator",
       ":mock_video_bitrate_allocator_factory",
       ":mock_video_codec_factory",
diff --git a/chromium/third_party/webrtc/api/DEPS b/chromium/third_party/webrtc/api/DEPS
index 1212b43be8a..995664e93ec 100644
--- a/chromium/third_party/webrtc/api/DEPS
+++ b/chromium/third_party/webrtc/api/DEPS
@@ -115,11 +115,6 @@ specific_include_rules = {
     "+rtc_base/ref_count.h",
   ],
 
-  "media_transport_interface\.h": [
-    "+rtc_base/copy_on_write_buffer.h",  # As used by datachannelinterface.h
-    "+rtc_base/network_route.h",
-  ],
-
   "packet_socket_factory\.h": [
     "+rtc_base/proxy_info.h",
     "+rtc_base/async_packet_socket.h",
diff --git a/chromium/third_party/webrtc/api/adaptation/BUILD.gn b/chromium/third_party/webrtc/api/adaptation/BUILD.gn
new file mode 100644
index 00000000000..dc4c73711ec
--- /dev/null
+++ b/chromium/third_party/webrtc/api/adaptation/BUILD.gn
@@ -0,0 +1,23 @@
+# Copyright(c) 2020 The WebRTC project authors.All Rights Reserved.
+#
+# Use of this source code is governed by a BSD - style license
+# that can be found in the LICENSE file in the root of the source
+# tree.An additional intellectual property rights grant can be found
+# in the file PATENTS.All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../webrtc.gni")
+
+rtc_source_set("resource_adaptation_api") {
+  visibility = [ "*" ]
+  sources = [
+    "resource.cc",
+    "resource.h",
+  ]
+  deps = [
+    "../../api:scoped_refptr",
+    "../../rtc_base:refcount",
+    "../../rtc_base:rtc_base_approved",
+    "../../rtc_base/system:rtc_export",
+  ]
+}
diff --git a/chromium/third_party/webrtc/api/adaptation/DEPS b/chromium/third_party/webrtc/api/adaptation/DEPS
new file mode 100644
index 00000000000..cab7fb8e140
--- /dev/null
+++ b/chromium/third_party/webrtc/api/adaptation/DEPS
@@ -0,0 +1,7 @@
+specific_include_rules = {
+  "resource\.h": [
+    # ref_count.h is a public_deps of rtc_base_approved. Necessary because of
+    # rtc::RefCountInterface.
+    "+rtc_base/ref_count.h",
+  ],
+}
\ No newline at end of file
diff --git a/chromium/third_party/webrtc/api/adaptation/resource.cc b/chromium/third_party/webrtc/api/adaptation/resource.cc
new file mode 100644
index 00000000000..0a9c83a3112
--- /dev/null
+++ b/chromium/third_party/webrtc/api/adaptation/resource.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/adaptation/resource.h"
+
+namespace webrtc {
+
+const char* ResourceUsageStateToString(ResourceUsageState usage_state) {
+  switch (usage_state) {
+    case ResourceUsageState::kOveruse:
+      return "kOveruse";
+    case ResourceUsageState::kUnderuse:
+      return "kUnderuse";
+  }
+}
+
+ResourceListener::~ResourceListener() {}
+
+Resource::Resource() {}
+
+Resource::~Resource() {}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/adaptation/resource.h b/chromium/third_party/webrtc/api/adaptation/resource.h
new file mode 100644
index 00000000000..9b3968055f6
--- /dev/null
+++ b/chromium/third_party/webrtc/api/adaptation/resource.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_ADAPTATION_RESOURCE_H_
+#define API_ADAPTATION_RESOURCE_H_
+
+#include <string>
+
+#include "api/scoped_refptr.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class Resource;
+
+enum class ResourceUsageState {
+  // Action is needed to minimze the load on this resource.
+  kOveruse,
+  // Increasing the load on this resource is desired, if possible.
+  kUnderuse,
+};
+
+RTC_EXPORT const char* ResourceUsageStateToString(
+    ResourceUsageState usage_state);
+
+class RTC_EXPORT ResourceListener {
+ public:
+  virtual ~ResourceListener();
+
+  virtual void OnResourceUsageStateMeasured(
+      rtc::scoped_refptr<Resource> resource,
+      ResourceUsageState usage_state) = 0;
+};
+
+// A Resource monitors an implementation-specific resource. It may report
+// kOveruse or kUnderuse when resource usage is high or low enough that we
+// should perform some sort of mitigation to fulfil the resource's constraints.
+//
+// The methods on this interface are invoked on the adaptation task queue.
+// Resource usage measurements may be performed on an any task queue.
+//
+// The Resource is reference counted to prevent use-after-free when posting
+// between task queues. As such, the implementation MUST NOT make any
+// assumptions about which task queue Resource is destructed on.
+class RTC_EXPORT Resource : public rtc::RefCountInterface {
+ public:
+  Resource();
+  // Destruction may happen on any task queue.
+  ~Resource() override;
+
+  virtual std::string Name() const = 0;
+  // The |listener| may be informed of resource usage measurements on any task
+  // queue, but not after this method is invoked with the null argument.
+  virtual void SetResourceListener(ResourceListener* listener) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_ADAPTATION_RESOURCE_H_
diff --git a/chromium/third_party/webrtc/api/array_view_unittest.cc b/chromium/third_party/webrtc/api/array_view_unittest.cc
index 8aa858805f0..0357f68aa2e 100644
--- a/chromium/third_party/webrtc/api/array_view_unittest.cc
+++ b/chromium/third_party/webrtc/api/array_view_unittest.cc
@@ -38,7 +38,7 @@ void CallFixed(ArrayView<T, N> av) {}
 
 }  // namespace
 
-TEST(ArrayViewTest, TestConstructFromPtrAndArray) {
+TEST(ArrayViewDeathTest, TestConstructFromPtrAndArray) {
   char arr[] = "Arrr!";
   const char carr[] = "Carrr!";
   EXPECT_EQ(6u, Call<const char>(arr));
@@ -409,7 +409,7 @@ TEST(FixArrayViewTest, TestSwapFixed) {
   // swap(x, w);  // Compile error, because different sizes.
 }
 
-TEST(ArrayViewTest, TestIndexing) {
+TEST(ArrayViewDeathTest, TestIndexing) {
   char arr[] = "abcdefg";
   ArrayView<char> x(arr);
   const ArrayView<char> y(arr);
diff --git a/chromium/third_party/webrtc/api/audio/BUILD.gn b/chromium/third_party/webrtc/api/audio/BUILD.gn
index 4c8004ed2db..117e5cc0abe 100644
--- a/chromium/third_party/webrtc/api/audio/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio/BUILD.gn
@@ -61,8 +61,8 @@ rtc_library("aec3_config_json") {
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:rtc_json",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/strings",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
 rtc_library("aec3_factory") {
diff --git a/chromium/third_party/webrtc/api/audio/audio_frame.cc b/chromium/third_party/webrtc/api/audio/audio_frame.cc
index 47459ac333b..c6e5cf4dd61 100644
--- a/chromium/third_party/webrtc/api/audio/audio_frame.cc
+++ b/chromium/third_party/webrtc/api/audio/audio_frame.cc
@@ -11,6 +11,8 @@
 #include "api/audio/audio_frame.h"
 
 #include <string.h>
+#include <algorithm>
+#include <utility>
 
 #include "rtc_base/checks.h"
 #include "rtc_base/time_utils.h"
@@ -22,6 +24,28 @@ AudioFrame::AudioFrame() {
   static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
 }
 
+void swap(AudioFrame& a, AudioFrame& b) {
+  using std::swap;
+  swap(a.timestamp_, b.timestamp_);
+  swap(a.elapsed_time_ms_, b.elapsed_time_ms_);
+  swap(a.ntp_time_ms_, b.ntp_time_ms_);
+  swap(a.samples_per_channel_, b.samples_per_channel_);
+  swap(a.sample_rate_hz_, b.sample_rate_hz_);
+  swap(a.num_channels_, b.num_channels_);
+  swap(a.channel_layout_, b.channel_layout_);
+  swap(a.speech_type_, b.speech_type_);
+  swap(a.vad_activity_, b.vad_activity_);
+  swap(a.profile_timestamp_ms_, b.profile_timestamp_ms_);
+  swap(a.packet_infos_, b.packet_infos_);
+  const size_t length_a = a.samples_per_channel_ * a.num_channels_;
+  const size_t length_b = b.samples_per_channel_ * b.num_channels_;
+  RTC_DCHECK_LE(length_a, AudioFrame::kMaxDataSizeSamples);
+  RTC_DCHECK_LE(length_b, AudioFrame::kMaxDataSizeSamples);
+  std::swap_ranges(a.data_, a.data_ + std::max(length_a, length_b), b.data_);
+  swap(a.muted_, b.muted_);
+  swap(a.absolute_capture_timestamp_ms_, b.absolute_capture_timestamp_ms_);
+}
+
 void AudioFrame::Reset() {
   ResetWithoutMuting();
   muted_ = true;
diff --git a/chromium/third_party/webrtc/api/audio/audio_frame.h b/chromium/third_party/webrtc/api/audio/audio_frame.h
index 06b0b28b38f..78539f57eb5 100644
--- a/chromium/third_party/webrtc/api/audio/audio_frame.h
+++ b/chromium/third_party/webrtc/api/audio/audio_frame.h
@@ -14,6 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include <utility>
+
 #include "api/audio/channel_layout.h"
 #include "api/rtp_packet_infos.h"
 #include "rtc_base/constructor_magic.h"
@@ -58,6 +60,8 @@ class AudioFrame {
 
   AudioFrame();
 
+  friend void swap(AudioFrame& a, AudioFrame& b);
+
   // Resets all members to their default state.
   void Reset();
   // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
diff --git a/chromium/third_party/webrtc/api/audio/test/audio_frame_unittest.cc b/chromium/third_party/webrtc/api/audio/test/audio_frame_unittest.cc
index dbf45ceabc8..f8d33182749 100644
--- a/chromium/third_party/webrtc/api/audio/test/audio_frame_unittest.cc
+++ b/chromium/third_party/webrtc/api/audio/test/audio_frame_unittest.cc
@@ -133,4 +133,54 @@ TEST(AudioFrameTest, CopyFrom) {
   EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
 }
 
+TEST(AudioFrameTest, SwapFrames) {
+  AudioFrame frame1, frame2;
+  int16_t samples1[kNumChannelsMono * kSamplesPerChannel];
+  for (size_t i = 0; i < kNumChannelsMono * kSamplesPerChannel; ++i) {
+    samples1[i] = i;
+  }
+  frame1.UpdateFrame(kTimestamp, samples1, kSamplesPerChannel, kSampleRateHz,
+                     AudioFrame::kPLC, AudioFrame::kVadActive,
+                     kNumChannelsMono);
+  frame1.set_absolute_capture_timestamp_ms(12345678);
+  const auto frame1_channel_layout = frame1.channel_layout();
+
+  int16_t samples2[(kNumChannelsMono + 1) * (kSamplesPerChannel + 1)];
+  for (size_t i = 0; i < (kNumChannelsMono + 1) * (kSamplesPerChannel + 1);
+       ++i) {
+    samples2[i] = 1000 + i;
+  }
+  frame2.UpdateFrame(kTimestamp + 1, samples2, kSamplesPerChannel + 1,
+                     kSampleRateHz + 1, AudioFrame::kNormalSpeech,
+                     AudioFrame::kVadPassive, kNumChannelsMono + 1);
+  const auto frame2_channel_layout = frame2.channel_layout();
+
+  swap(frame1, frame2);
+
+  EXPECT_EQ(kTimestamp + 1, frame1.timestamp_);
+  ASSERT_EQ(kSamplesPerChannel + 1, frame1.samples_per_channel_);
+  EXPECT_EQ(kSampleRateHz + 1, frame1.sample_rate_hz_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, frame1.speech_type_);
+  EXPECT_EQ(AudioFrame::kVadPassive, frame1.vad_activity_);
+  ASSERT_EQ(kNumChannelsMono + 1, frame1.num_channels_);
+  for (size_t i = 0; i < (kNumChannelsMono + 1) * (kSamplesPerChannel + 1);
+       ++i) {
+    EXPECT_EQ(samples2[i], frame1.data()[i]);
+  }
+  EXPECT_FALSE(frame1.absolute_capture_timestamp_ms());
+  EXPECT_EQ(frame2_channel_layout, frame1.channel_layout());
+
+  EXPECT_EQ(kTimestamp, frame2.timestamp_);
+  ASSERT_EQ(kSamplesPerChannel, frame2.samples_per_channel_);
+  EXPECT_EQ(kSampleRateHz, frame2.sample_rate_hz_);
+  EXPECT_EQ(AudioFrame::kPLC, frame2.speech_type_);
+  EXPECT_EQ(AudioFrame::kVadActive, frame2.vad_activity_);
+  ASSERT_EQ(kNumChannelsMono, frame2.num_channels_);
+  for (size_t i = 0; i < kNumChannelsMono * kSamplesPerChannel; ++i) {
+    EXPECT_EQ(samples1[i], frame2.data()[i]);
+  }
+  EXPECT_EQ(12345678, frame2.absolute_capture_timestamp_ms());
+  EXPECT_EQ(frame1_channel_layout, frame2.channel_layout());
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/audio_codecs/BUILD.gn b/chromium/third_party/webrtc/api/audio_codecs/BUILD.gn
index 987e20f178b..b6292de570d 100644
--- a/chromium/third_party/webrtc/api/audio_codecs/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio_codecs/BUILD.gn
@@ -38,6 +38,8 @@ rtc_library("audio_codecs_api") {
     "../../rtc_base:sanitizer",
     "../../rtc_base/system:rtc_export",
     "../units:time_delta",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
diff --git a/chromium/third_party/webrtc/api/audio_codecs/L16/BUILD.gn b/chromium/third_party/webrtc/api/audio_codecs/L16/BUILD.gn
index bef671237e2..1f7a1e5a0b6 100644
--- a/chromium/third_party/webrtc/api/audio_codecs/L16/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio_codecs/L16/BUILD.gn
@@ -25,6 +25,8 @@ rtc_library("audio_encoder_L16") {
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -42,6 +44,8 @@ rtc_library("audio_decoder_L16") {
     "../../../modules/audio_coding:pcm16b",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
diff --git a/chromium/third_party/webrtc/api/audio_codecs/g711/BUILD.gn b/chromium/third_party/webrtc/api/audio_codecs/g711/BUILD.gn
index ba0586b9010..92d77bed9f8 100644
--- a/chromium/third_party/webrtc/api/audio_codecs/g711/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio_codecs/g711/BUILD.gn
@@ -25,6 +25,8 @@ rtc_library("audio_encoder_g711") {
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -42,6 +44,8 @@ rtc_library("audio_decoder_g711") {
     "../../../modules/audio_coding:g711",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
diff --git a/chromium/third_party/webrtc/api/audio_codecs/g722/BUILD.gn b/chromium/third_party/webrtc/api/audio_codecs/g722/BUILD.gn
index 8738ef889aa..a186eabbb7a 100644
--- a/chromium/third_party/webrtc/api/audio_codecs/g722/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio_codecs/g722/BUILD.gn
@@ -31,6 +31,8 @@ rtc_library("audio_encoder_g722") {
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -48,6 +50,8 @@ rtc_library("audio_decoder_g722") {
     "../../../modules/audio_coding:g722",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
diff --git a/chromium/third_party/webrtc/api/audio_codecs/ilbc/BUILD.gn b/chromium/third_party/webrtc/api/audio_codecs/ilbc/BUILD.gn
index 066a73cff22..b6a5045eaf9 100644
--- a/chromium/third_party/webrtc/api/audio_codecs/ilbc/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio_codecs/ilbc/BUILD.gn
@@ -30,6 +30,8 @@ rtc_library("audio_encoder_ilbc") {
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -46,6 +48,8 @@ rtc_library("audio_decoder_ilbc") {
     "..:audio_codecs_api",
     "../../../modules/audio_coding:ilbc",
     "../../../rtc_base:rtc_base_approved",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
diff --git a/chromium/third_party/webrtc/api/audio_codecs/isac/BUILD.gn b/chromium/third_party/webrtc/api/audio_codecs/isac/BUILD.gn
index 9eb32147e14..6ff6e5f0923 100644
--- a/chromium/third_party/webrtc/api/audio_codecs/isac/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio_codecs/isac/BUILD.gn
@@ -68,6 +68,8 @@ rtc_library("audio_encoder_isac_fix") {
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -85,6 +87,8 @@ rtc_library("audio_decoder_isac_fix") {
     "../../../modules/audio_coding:isac_fix",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -102,6 +106,8 @@ rtc_library("audio_encoder_isac_float") {
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -119,6 +125,8 @@ rtc_library("audio_decoder_isac_float") {
     "../../../modules/audio_coding:isac",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
diff --git a/chromium/third_party/webrtc/api/audio_codecs/opus/BUILD.gn b/chromium/third_party/webrtc/api/audio_codecs/opus/BUILD.gn
index 5fb626d9904..586e9b3dd88 100644
--- a/chromium/third_party/webrtc/api/audio_codecs/opus/BUILD.gn
+++ b/chromium/third_party/webrtc/api/audio_codecs/opus/BUILD.gn
@@ -23,8 +23,8 @@ rtc_library("audio_encoder_opus_config") {
   deps = [
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   defines = []
   if (rtc_opus_variable_complexity) {
     defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
@@ -49,6 +49,8 @@ rtc_library("audio_encoder_opus") {
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -66,6 +68,8 @@ rtc_library("audio_decoder_opus") {
     "../../../modules/audio_coding:webrtc_opus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -82,8 +86,8 @@ rtc_library("audio_encoder_multiopus") {
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
     "../opus:audio_encoder_opus_config",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("audio_decoder_multiopus") {
@@ -99,6 +103,8 @@ rtc_library("audio_decoder_multiopus") {
     "../../../modules/audio_coding:webrtc_multiopus",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
diff --git a/chromium/third_party/webrtc/api/audio_options.h b/chromium/third_party/webrtc/api/audio_options.h
index b714998c6b7..1b0d1ad0bd1 100644
--- a/chromium/third_party/webrtc/api/audio_options.h
+++ b/chromium/third_party/webrtc/api/audio_options.h
@@ -75,6 +75,8 @@ struct RTC_EXPORT AudioOptions {
   // and check if any other AudioOptions members are unused.
   absl::optional<bool> combined_audio_video_bwe;
   // Enable audio network adaptor.
+  // TODO(webrtc:11717): Remove this API in favor of adaptivePtime in
+  // RtpEncodingParameters.
   absl::optional<bool> audio_network_adaptor;
   // Config string for audio network adaptor.
   absl::optional<std::string> audio_network_adaptor_config;
diff --git a/chromium/third_party/webrtc/api/data_channel_interface.h b/chromium/third_party/webrtc/api/data_channel_interface.h
index e08830feaf3..5b2b1263ab7 100644
--- a/chromium/third_party/webrtc/api/data_channel_interface.h
+++ b/chromium/third_party/webrtc/api/data_channel_interface.h
@@ -20,6 +20,7 @@
 #include <string>
 
 #include "absl/types/optional.h"
+#include "api/priority.h"
 #include "api/rtc_error.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
@@ -61,6 +62,9 @@ struct DataChannelInit {
 
   // The stream id, or SID, for SCTP data channels. -1 if unset (see above).
   int id = -1;
+
+  // https://w3c.github.io/webrtc-priority/#new-rtcdatachannelinit-member
+  absl::optional<Priority> priority;
 };
 
 // At the JavaScript level, data can be passed in as a string or a blob, so
@@ -154,6 +158,7 @@ class RTC_EXPORT DataChannelInterface : public rtc::RefCountInterface {
   // If negotiated in-band, this ID will be populated once the DTLS role is
   // determined, and until then this will return -1.
   virtual int id() const = 0;
+  virtual Priority priority() const { return Priority::kLow; }
   virtual DataState state() const = 0;
   // When state is kClosed, and the DataChannel was not closed using
   // the closing procedure, returns the error information about the closing.
diff --git a/chromium/third_party/webrtc/api/frame_transformer_interface.h b/chromium/third_party/webrtc/api/frame_transformer_interface.h
index e712b3c190a..2cfe6edb884 100644
--- a/chromium/third_party/webrtc/api/frame_transformer_interface.h
+++ b/chromium/third_party/webrtc/api/frame_transformer_interface.h
@@ -16,6 +16,7 @@
 
 #include "api/scoped_refptr.h"
 #include "api/video/encoded_frame.h"
+#include "api/video/video_frame_metadata.h"
 #include "rtc_base/ref_count.h"
 
 namespace webrtc {
@@ -48,6 +49,8 @@ class TransformableVideoFrameInterface : public TransformableFrameInterface {
   // TODO(bugs.webrtc.org/11380) remove from interface once
   // webrtc::RtpDescriptorAuthentication is exposed in api/.
   virtual std::vector<uint8_t> GetAdditionalData() const = 0;
+
+  virtual const VideoFrameMetadata& GetMetadata() const = 0;
 };
 
 // Extends the TransformableFrameInterface to expose audio-specific information.
diff --git a/chromium/third_party/webrtc/api/neteq/BUILD.gn b/chromium/third_party/webrtc/api/neteq/BUILD.gn
index 1ab02ec92b1..4e85c4d268c 100644
--- a/chromium/third_party/webrtc/api/neteq/BUILD.gn
+++ b/chromium/third_party/webrtc/api/neteq/BUILD.gn
@@ -23,8 +23,8 @@ rtc_source_set("neteq_api") {
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers:system_wrappers",
     "../audio_codecs:audio_codecs_api",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("custom_neteq_factory") {
@@ -56,8 +56,8 @@ rtc_source_set("neteq_controller_api") {
     ":tick_timer",
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers:system_wrappers",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("default_neteq_controller_factory") {
diff --git a/chromium/third_party/webrtc/api/neteq/neteq.cc b/chromium/third_party/webrtc/api/neteq/neteq.cc
index 155ddf2cf3d..e8ef4dbd39b 100644
--- a/chromium/third_party/webrtc/api/neteq/neteq.cc
+++ b/chromium/third_party/webrtc/api/neteq/neteq.cc
@@ -30,7 +30,8 @@ std::string NetEq::Config::ToString() const {
      << ", min_delay_ms=" << min_delay_ms << ", enable_fast_accelerate="
      << (enable_fast_accelerate ? "true" : "false")
      << ", enable_muted_state=" << (enable_muted_state ? "true" : "false")
-     << ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false");
+     << ", enable_rtx_handling=" << (enable_rtx_handling ? "true" : "false")
+     << ", extra_output_delay_ms=" << extra_output_delay_ms;
   return ss.str();
 }
 
diff --git a/chromium/third_party/webrtc/api/neteq/neteq.h b/chromium/third_party/webrtc/api/neteq/neteq.h
index f62d3795f05..15ad3aac0ec 100644
--- a/chromium/third_party/webrtc/api/neteq/neteq.h
+++ b/chromium/third_party/webrtc/api/neteq/neteq.h
@@ -138,6 +138,10 @@ class NetEq {
     bool enable_rtx_handling = false;
     absl::optional<AudioCodecPairId> codec_pair_id;
     bool for_test_no_time_stretching = false;  // Use only for testing.
+    // Adds extra delay to the output of NetEq, without affecting jitter or
+    // loss behavior. This is mainly for testing. Value must be a non-negative
+    // multiple of 10 ms.
+    int extra_output_delay_ms = 0;
   };
 
   enum ReturnCodes { kOK = 0, kFail = -1 };
diff --git a/chromium/third_party/webrtc/api/peer_connection_interface.h b/chromium/third_party/webrtc/api/peer_connection_interface.h
index 1d81de74d83..2664af4e5e6 100644
--- a/chromium/third_party/webrtc/api/peer_connection_interface.h
+++ b/chromium/third_party/webrtc/api/peer_connection_interface.h
@@ -73,6 +73,7 @@
 #include <string>
 #include <vector>
 
+#include "api/adaptation/resource.h"
 #include "api/async_resolver_factory.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
@@ -102,7 +103,6 @@
 #include "api/task_queue/task_queue_factory.h"
 #include "api/transport/bitrate_settings.h"
 #include "api/transport/enums.h"
-#include "api/transport/media/media_transport_interface.h"
 #include "api/transport/network_control.h"
 #include "api/transport/webrtc_key_value_config.h"
 #include "api/turn_customizer.h"
@@ -613,34 +613,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // correctly. This flag will be deprecated soon. Do not rely on it.
     bool active_reset_srtp_params = false;
 
-    // DEPRECATED.  Do not use.  This option is ignored by peer connection.
-    // TODO(webrtc:9719):  Delete this option.
-    bool use_media_transport = false;
-
-    // DEPRECATED.  Do not use.  This option is ignored by peer connection.
-    // TODO(webrtc:9719):  Delete this option.
-    bool use_media_transport_for_data_channels = false;
-
-    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
-    // informs PeerConnection that it should use the DatagramTransportInterface
-    // for packets instead DTLS. It's invalid to set it to |true| if the
-    // MediaTransportFactory wasn't provided.
-    absl::optional<bool> use_datagram_transport;
-
-    // If MediaTransportFactory is provided in PeerConnectionFactory, this flag
-    // informs PeerConnection that it should use the DatagramTransport's
-    // implementation of DataChannelTransportInterface for data channels instead
-    // of SCTP-DTLS.
-    absl::optional<bool> use_datagram_transport_for_data_channels;
-
-    // If true, this PeerConnection will only use datagram transport for data
-    // channels when receiving an incoming offer that includes datagram
-    // transport parameters.  It will not request use of a datagram transport
-    // when it creates the initial, outgoing offer.
-    // This setting only applies when |use_datagram_transport_for_data_channels|
-    // is true.
-    absl::optional<bool> use_datagram_transport_for_data_channels_receive_only;
-
     // Defines advanced optional cryptographic settings related to SRTP and
     // frame encryption for native WebRTC. Setting this will overwrite any
     // settings set in PeerConnectionFactory (which is deprecated).
@@ -666,8 +638,6 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     // Whether network condition based codec switching is allowed.
     absl::optional<bool> allow_codec_switching;
 
-    bool enable_simulcast_stats = true;
-
     //
     // Don't forget to update operator== if adding something.
     //
@@ -1118,6 +1088,14 @@ class RTC_EXPORT PeerConnectionInterface : public rtc::RefCountInterface {
     return absl::nullopt;
   }
 
+  // When a resource is overused, the PeerConnection will try to reduce the load
+  // on the sysem, for example by reducing the resolution or frame rate of
+  // encoded streams. The Resource API allows injecting platform-specific usage
+  // measurements. The conditions to trigger kOveruse or kUnderuse are up to the
+  // implementation.
+  // TODO(hbos): Make pure virtual when implemented by downstream projects.
+  virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) {}
+
   // Start RtcEventLog using an existing output-sink. Takes ownership of
   // |output| and passes it on to Call, which will take the ownership. If the
   // operation fails the output will be closed and deallocated. The event log
@@ -1332,7 +1310,6 @@ struct RTC_EXPORT PeerConnectionFactoryDependencies final {
   std::unique_ptr<NetworkStatePredictorFactoryInterface>
       network_state_predictor_factory;
   std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
-  std::unique_ptr<MediaTransportFactory> media_transport_factory;
   std::unique_ptr<NetEqFactory> neteq_factory;
   std::unique_ptr<WebRtcKeyValueConfig> trials;
 };
diff --git a/chromium/third_party/webrtc/api/peer_connection_proxy.h b/chromium/third_party/webrtc/api/peer_connection_proxy.h
index c278308ccbe..23887e53dae 100644
--- a/chromium/third_party/webrtc/api/peer_connection_proxy.h
+++ b/chromium/third_party/webrtc/api/peer_connection_proxy.h
@@ -132,6 +132,7 @@ PROXY_METHOD0(IceConnectionState, standardized_ice_connection_state)
 PROXY_METHOD0(PeerConnectionState, peer_connection_state)
 PROXY_METHOD0(IceGatheringState, ice_gathering_state)
 PROXY_METHOD0(absl::optional<bool>, can_trickle_ice_candidates)
+PROXY_METHOD1(void, AddAdaptationResource, rtc::scoped_refptr<Resource>)
 PROXY_METHOD2(bool,
               StartRtcEventLog,
               std::unique_ptr<RtcEventLogOutput>,
diff --git a/chromium/third_party/webrtc/api/priority.h b/chromium/third_party/webrtc/api/priority.h
new file mode 100644
index 00000000000..4953e453a32
--- /dev/null
+++ b/chromium/third_party/webrtc/api/priority.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_PRIORITY_H_
+#define API_PRIORITY_H_
+
+namespace webrtc {
+
+// GENERATED_JAVA_ENUM_PACKAGE: org.webrtc
+enum class Priority {
+  kVeryLow,
+  kLow,
+  kMedium,
+  kHigh,
+};
+
+}  // namespace webrtc
+
+#endif  // API_PRIORITY_H_
diff --git a/chromium/third_party/webrtc/api/proxy.cc b/chromium/third_party/webrtc/api/proxy.cc
index e668285ba29..67318e7dab0 100644
--- a/chromium/third_party/webrtc/api/proxy.cc
+++ b/chromium/third_party/webrtc/api/proxy.cc
@@ -10,28 +10,3 @@
 
 #include "api/proxy.h"
 
-namespace webrtc {
-namespace internal {
-
-SynchronousMethodCall::SynchronousMethodCall(rtc::MessageHandler* proxy)
-    : proxy_(proxy) {}
-
-SynchronousMethodCall::~SynchronousMethodCall() = default;
-
-void SynchronousMethodCall::Invoke(const rtc::Location& posted_from,
-                                   rtc::Thread* t) {
-  if (t->IsCurrent()) {
-    proxy_->OnMessage(nullptr);
-  } else {
-    t->Post(posted_from, this, 0);
-    e_.Wait(rtc::Event::kForever);
-  }
-}
-
-void SynchronousMethodCall::OnMessage(rtc::Message*) {
-  proxy_->OnMessage(nullptr);
-  e_.Set();
-}
-
-}  // namespace internal
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/proxy.h b/chromium/third_party/webrtc/api/proxy.h
index 385992e6594..0253ba3dc47 100644
--- a/chromium/third_party/webrtc/api/proxy.h
+++ b/chromium/third_party/webrtc/api/proxy.h
@@ -55,9 +55,12 @@
 #include <memory>
 #include <string>
 #include <tuple>
+#include <type_traits>
 #include <utility>
 
 #include "api/scoped_refptr.h"
+#include "api/task_queue/queued_task.h"
+#include "api/task_queue/task_queue_base.h"
 #include "rtc_base/event.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/ref_counted_object.h"
@@ -95,27 +98,8 @@ class ReturnType<void> {
   void moved_result() {}
 };
 
-namespace internal {
-
-class RTC_EXPORT SynchronousMethodCall : public rtc::MessageData,
-                                         public rtc::MessageHandler {
- public:
-  explicit SynchronousMethodCall(rtc::MessageHandler* proxy);
-  ~SynchronousMethodCall() override;
-
-  void Invoke(const rtc::Location& posted_from, rtc::Thread* t);
-
- private:
-  void OnMessage(rtc::Message*) override;
-
-  rtc::Event e_;
-  rtc::MessageHandler* proxy_;
-};
-
-}  // namespace internal
-
 template <typename C, typename R, typename... Args>
-class MethodCall : public rtc::Message, public rtc::MessageHandler {
+class MethodCall : public QueuedTask {
  public:
   typedef R (C::*Method)(Args...);
   MethodCall(C* c, Method m, Args&&... args)
@@ -124,12 +108,21 @@ class MethodCall : public rtc::Message, public rtc::MessageHandler {
         args_(std::forward_as_tuple(std::forward<Args>(args)...)) {}
 
   R Marshal(const rtc::Location& posted_from, rtc::Thread* t) {
-    internal::SynchronousMethodCall(this).Invoke(posted_from, t);
+    if (t->IsCurrent()) {
+      Invoke(std::index_sequence_for<Args...>());
+    } else {
+      t->PostTask(std::unique_ptr<QueuedTask>(this));
+      event_.Wait(rtc::Event::kForever);
+    }
     return r_.moved_result();
   }
 
  private:
-  void OnMessage(rtc::Message*) { Invoke(std::index_sequence_for<Args...>()); }
+  bool Run() override {
+    Invoke(std::index_sequence_for<Args...>());
+    event_.Set();
+    return false;
+  }
 
   template <size_t... Is>
   void Invoke(std::index_sequence<Is...>) {
@@ -140,10 +133,11 @@ class MethodCall : public rtc::Message, public rtc::MessageHandler {
   Method m_;
   ReturnType<R> r_;
   std::tuple<Args&&...> args_;
+  rtc::Event event_;
 };
 
 template <typename C, typename R, typename... Args>
-class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
+class ConstMethodCall : public QueuedTask {
  public:
   typedef R (C::*Method)(Args...) const;
   ConstMethodCall(const C* c, Method m, Args&&... args)
@@ -152,12 +146,21 @@ class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
         args_(std::forward_as_tuple(std::forward<Args>(args)...)) {}
 
   R Marshal(const rtc::Location& posted_from, rtc::Thread* t) {
-    internal::SynchronousMethodCall(this).Invoke(posted_from, t);
+    if (t->IsCurrent()) {
+      Invoke(std::index_sequence_for<Args...>());
+    } else {
+      t->PostTask(std::unique_ptr<QueuedTask>(this));
+      event_.Wait(rtc::Event::kForever);
+    }
     return r_.moved_result();
   }
 
  private:
-  void OnMessage(rtc::Message*) { Invoke(std::index_sequence_for<Args...>()); }
+  bool Run() override {
+    Invoke(std::index_sequence_for<Args...>());
+    event_.Set();
+    return false;
+  }
 
   template <size_t... Is>
   void Invoke(std::index_sequence<Is...>) {
@@ -168,6 +171,7 @@ class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
   Method m_;
   ReturnType<R> r_;
   std::tuple<Args&&...> args_;
+  rtc::Event event_;
 };
 
 // Helper macros to reduce code duplication.
@@ -396,6 +400,16 @@ class ConstMethodCall : public rtc::Message, public rtc::MessageHandler {
     return call.Marshal(RTC_FROM_HERE, worker_thread_);                   \
   }
 
+// For use when returning purely const state (set during construction).
+// Use with caution. This method should only be used when the return value will
+// always be the same.
+#define BYPASS_PROXY_CONSTMETHOD0(r, method)                            \
+  r method() const override {                                           \
+    static_assert(!std::is_pointer<r>::value, "Type is a pointer");     \
+    static_assert(!std::is_reference<r>::value, "Type is a reference"); \
+    return c_->method();                                                \
+  }
+
 }  // namespace webrtc
 
 #endif  //  API_PROXY_H_
diff --git a/chromium/third_party/webrtc/api/rtc_event_log_output_file_unittest.cc b/chromium/third_party/webrtc/api/rtc_event_log_output_file_unittest.cc
index 071909b2c54..4274215491f 100644
--- a/chromium/third_party/webrtc/api/rtc_event_log_output_file_unittest.cc
+++ b/chromium/third_party/webrtc/api/rtc_event_log_output_file_unittest.cc
@@ -141,14 +141,16 @@ TEST_F(RtcEventLogOutputFileTest, AllowReasonableFileSizeLimits) {
 }
 
 #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
-TEST_F(RtcEventLogOutputFileTest, WritingToInactiveFileForbidden) {
+class RtcEventLogOutputFileDeathTest : public RtcEventLogOutputFileTest {};
+
+TEST_F(RtcEventLogOutputFileDeathTest, WritingToInactiveFileForbidden) {
   RtcEventLogOutputFile output_file(output_file_name_, 2);
   ASSERT_FALSE(output_file.Write("abc"));
   ASSERT_FALSE(output_file.IsActive());
   EXPECT_DEATH(output_file.Write("abc"), "");
 }
 
-TEST_F(RtcEventLogOutputFileTest, DisallowUnreasonableFileSizeLimits) {
+TEST_F(RtcEventLogOutputFileDeathTest, DisallowUnreasonableFileSizeLimits) {
   // Keeping in a temporary unique_ptr to make it clearer that the death is
   // triggered by construction, not destruction.
   std::unique_ptr<RtcEventLogOutputFile> output_file;
diff --git a/chromium/third_party/webrtc/api/rtp_headers.cc b/chromium/third_party/webrtc/api/rtp_headers.cc
index bf973b6fe57..e0ad9eb26ef 100644
--- a/chromium/third_party/webrtc/api/rtp_headers.cc
+++ b/chromium/third_party/webrtc/api/rtp_headers.cc
@@ -26,9 +26,7 @@ RTPHeaderExtension::RTPHeaderExtension()
       videoRotation(kVideoRotation_0),
       hasVideoContentType(false),
       videoContentType(VideoContentType::UNSPECIFIED),
-      has_video_timing(false),
-      has_frame_marking(false),
-      frame_marking({false, false, false, false, false, 0xFF, 0, 0}) {}
+      has_video_timing(false) {}
 
 RTPHeaderExtension::RTPHeaderExtension(const RTPHeaderExtension& other) =
     default;
diff --git a/chromium/third_party/webrtc/api/rtp_headers.h b/chromium/third_party/webrtc/api/rtp_headers.h
index 163347f6758..454149ca6e7 100644
--- a/chromium/third_party/webrtc/api/rtp_headers.h
+++ b/chromium/third_party/webrtc/api/rtp_headers.h
@@ -21,10 +21,9 @@
 #include "api/units/timestamp.h"
 #include "api/video/color_space.h"
 #include "api/video/video_content_type.h"
-#include "api/video/video_frame_marking.h"
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "common_types.h"  // NOLINT (build/include)
 
 namespace webrtc {
 
@@ -143,9 +142,6 @@ struct RTPHeaderExtension {
   bool has_video_timing;
   VideoSendTiming video_timing;
 
-  bool has_frame_marking;
-  FrameMarking frame_marking;
-
   PlayoutDelay playout_delay = {-1, -1};
 
   // For identification of a stream when ssrc is not signaled. See
diff --git a/chromium/third_party/webrtc/api/rtp_parameters.cc b/chromium/third_party/webrtc/api/rtp_parameters.cc
index a05b2bfa7bb..28acb68be6a 100644
--- a/chromium/third_party/webrtc/api/rtp_parameters.cc
+++ b/chromium/third_party/webrtc/api/rtp_parameters.cc
@@ -18,6 +18,20 @@
 
 namespace webrtc {
 
+const char* DegradationPreferenceToString(
+    DegradationPreference degradation_preference) {
+  switch (degradation_preference) {
+    case DegradationPreference::DISABLED:
+      return "disabled";
+    case DegradationPreference::MAINTAIN_FRAMERATE:
+      return "maintain-framerate";
+    case DegradationPreference::MAINTAIN_RESOLUTION:
+      return "maintain-resolution";
+    case DegradationPreference::BALANCED:
+      return "balanced";
+  }
+}
+
 const double kDefaultBitratePriority = 1.0;
 
 RtcpFeedback::RtcpFeedback() = default;
@@ -105,7 +119,6 @@ constexpr char RtpExtension::kAbsoluteCaptureTimeUri[];
 constexpr char RtpExtension::kVideoRotationUri[];
 constexpr char RtpExtension::kVideoContentTypeUri[];
 constexpr char RtpExtension::kVideoTimingUri[];
-constexpr char RtpExtension::kFrameMarkingUri[];
 constexpr char RtpExtension::kGenericFrameDescriptorUri00[];
 constexpr char RtpExtension::kDependencyDescriptorUri[];
 constexpr char RtpExtension::kTransportSequenceNumberUri[];
@@ -144,7 +157,6 @@ bool RtpExtension::IsSupportedForVideo(absl::string_view uri) {
          uri == webrtc::RtpExtension::kVideoContentTypeUri ||
          uri == webrtc::RtpExtension::kVideoTimingUri ||
          uri == webrtc::RtpExtension::kMidUri ||
-         uri == webrtc::RtpExtension::kFrameMarkingUri ||
          uri == webrtc::RtpExtension::kGenericFrameDescriptorUri00 ||
          uri == webrtc::RtpExtension::kDependencyDescriptorUri ||
          uri == webrtc::RtpExtension::kColorSpaceUri ||
diff --git a/chromium/third_party/webrtc/api/rtp_parameters.h b/chromium/third_party/webrtc/api/rtp_parameters.h
index 49c1e0c8852..b667bf812ca 100644
--- a/chromium/third_party/webrtc/api/rtp_parameters.h
+++ b/chromium/third_party/webrtc/api/rtp_parameters.h
@@ -20,6 +20,7 @@
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/media_types.h"
+#include "api/priority.h"
 #include "api/rtp_transceiver_direction.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -91,15 +92,10 @@ enum class DegradationPreference {
   BALANCED,
 };
 
-RTC_EXPORT extern const double kDefaultBitratePriority;
+RTC_EXPORT const char* DegradationPreferenceToString(
+    DegradationPreference degradation_preference);
 
-// GENERATED_JAVA_ENUM_PACKAGE: org.webrtc
-enum class Priority {
-  kVeryLow,
-  kLow,
-  kMedium,
-  kHigh,
-};
+RTC_EXPORT extern const double kDefaultBitratePriority;
 
 struct RTC_EXPORT RtcpFeedback {
   RtcpFeedbackType type = RtcpFeedbackType::CCM;
@@ -226,7 +222,7 @@ struct RTC_EXPORT RtpHeaderExtensionCapability {
   bool preferred_encrypt = false;
 
   // The direction of the extension. The kStopped value is only used with
-  // RtpTransceiverInterface::header_extensions_offered() and
+  // RtpTransceiverInterface::HeaderExtensionsToOffer() and
   // SetOfferedRtpHeaderExtensions().
   RtpTransceiverDirection direction = RtpTransceiverDirection::kSendRecv;
 
@@ -314,10 +310,6 @@ struct RTC_EXPORT RtpExtension {
   static constexpr char kVideoTimingUri[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
 
-  // Header extension for video frame marking.
-  static constexpr char kFrameMarkingUri[] =
-      "http://tools.ietf.org/html/draft-ietf-avtext-framemarking-07";
-
   // Experimental codec agnostic frame descriptor.
   static constexpr char kGenericFrameDescriptorUri00[] =
       "http://www.webrtc.org/experiments/rtp-hdrext/"
@@ -481,6 +473,10 @@ struct RTC_EXPORT RtpEncodingParameters {
   // Called "encodingId" in ORTC.
   std::string rid;
 
+  // Allow dynamic frame length changes for audio:
+  // https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime
+  bool adaptive_ptime = false;
+
   bool operator==(const RtpEncodingParameters& o) const {
     return ssrc == o.ssrc && bitrate_priority == o.bitrate_priority &&
            network_priority == o.network_priority &&
@@ -489,7 +485,8 @@ struct RTC_EXPORT RtpEncodingParameters {
            max_framerate == o.max_framerate &&
            num_temporal_layers == o.num_temporal_layers &&
            scale_resolution_down_by == o.scale_resolution_down_by &&
-           active == o.active && rid == o.rid;
+           active == o.active && rid == o.rid &&
+           adaptive_ptime == o.adaptive_ptime;
   }
   bool operator!=(const RtpEncodingParameters& o) const {
     return !(*this == o);
diff --git a/chromium/third_party/webrtc/api/rtp_transceiver_interface.cc b/chromium/third_party/webrtc/api/rtp_transceiver_interface.cc
index d4e2b26e333..e795e51dfb1 100644
--- a/chromium/third_party/webrtc/api/rtp_transceiver_interface.cc
+++ b/chromium/third_party/webrtc/api/rtp_transceiver_interface.cc
@@ -41,4 +41,10 @@ RtpTransceiverInterface::HeaderExtensionsToOffer() const {
   return {};
 }
 
+webrtc::RTCError RtpTransceiverInterface::SetOfferedRtpHeaderExtensions(
+    rtc::ArrayView<const RtpHeaderExtensionCapability>
+        header_extensions_to_offer) {
+  return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_OPERATION);
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/rtp_transceiver_interface.h b/chromium/third_party/webrtc/api/rtp_transceiver_interface.h
index 9dbafd46ecd..13277d9a502 100644
--- a/chromium/third_party/webrtc/api/rtp_transceiver_interface.h
+++ b/chromium/third_party/webrtc/api/rtp_transceiver_interface.h
@@ -133,6 +133,13 @@ class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
   virtual std::vector<RtpHeaderExtensionCapability> HeaderExtensionsToOffer()
       const;
 
+  // The SetOfferedRtpHeaderExtensions method modifies the next SDP negotiation
+  // so that it negotiates use of header extensions which are not kStopped.
+  // https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface
+  virtual webrtc::RTCError SetOfferedRtpHeaderExtensions(
+      rtc::ArrayView<const RtpHeaderExtensionCapability>
+          header_extensions_to_offer);
+
  protected:
   ~RtpTransceiverInterface() override = default;
 };
diff --git a/chromium/third_party/webrtc/api/stats/rtc_stats.h b/chromium/third_party/webrtc/api/stats/rtc_stats.h
index d45902e0a5c..5de5b7fbb04 100644
--- a/chromium/third_party/webrtc/api/stats/rtc_stats.h
+++ b/chromium/third_party/webrtc/api/stats/rtc_stats.h
@@ -319,6 +319,14 @@ class RTCStatsMember : public RTCStatsMemberInterface {
   std::string ValueToString() const override;
   std::string ValueToJson() const override;
 
+  template <typename U>
+  inline T ValueOrDefault(U default_value) const {
+    if (is_defined()) {
+      return *(*this);
+    }
+    return default_value;
+  }
+
   // Assignment operators.
   T& operator=(const T& value) {
     value_ = value;
diff --git a/chromium/third_party/webrtc/api/stats/rtcstats_objects.h b/chromium/third_party/webrtc/api/stats/rtcstats_objects.h
index 28d841db09b..3458d6fef7c 100644
--- a/chromium/third_party/webrtc/api/stats/rtcstats_objects.h
+++ b/chromium/third_party/webrtc/api/stats/rtcstats_objects.h
@@ -134,7 +134,7 @@ class RTC_EXPORT RTCDataChannelStats final : public RTCStats {
 
   RTCStatsMember<std::string> label;
   RTCStatsMember<std::string> protocol;
-  RTCStatsMember<int32_t> datachannelid;
+  RTCStatsMember<int32_t> data_channel_identifier;
   // TODO(hbos): Support enum types? "RTCStatsMember<RTCDataChannelState>"?
   RTCStatsMember<std::string> state;
   RTCStatsMember<uint32_t> messages_sent;
diff --git a/chromium/third_party/webrtc/api/task_queue/BUILD.gn b/chromium/third_party/webrtc/api/task_queue/BUILD.gn
index 4c9f591ec1d..1072057e3fc 100644
--- a/chromium/third_party/webrtc/api/task_queue/BUILD.gn
+++ b/chromium/third_party/webrtc/api/task_queue/BUILD.gn
@@ -21,6 +21,8 @@ rtc_library("task_queue") {
     "../../rtc_base:checks",
     "../../rtc_base:macromagic",
     "../../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/base:config",
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/strings",
@@ -51,6 +53,8 @@ rtc_library("task_queue_test") {
     deps = [
       "../../../webrtc_overrides:webrtc_component",
       "../../test:test_support",
+    ]
+    absl_deps = [
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/strings",
     ]
@@ -62,6 +66,8 @@ rtc_library("task_queue_test") {
       "../../rtc_base:timeutils",
       "../../rtc_base/task_utils:to_queued_task",
       "../../test:test_support",
+    ]
+    absl_deps = [
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/strings",
     ]
diff --git a/chromium/third_party/webrtc/api/task_queue/task_queue_test.cc b/chromium/third_party/webrtc/api/task_queue/task_queue_test.cc
index a8a799f11b7..0d411d2d9cb 100644
--- a/chromium/third_party/webrtc/api/task_queue/task_queue_test.cc
+++ b/chromium/third_party/webrtc/api/task_queue/task_queue_test.cc
@@ -37,9 +37,11 @@ TEST_P(TaskQueueTest, PostAndCheckCurrent) {
   rtc::Event event;
   auto queue = CreateTaskQueue(factory, "PostAndCheckCurrent");
 
-  // We're not running a task, so there shouldn't be a current queue.
+  // We're not running a task, so |queue| shouldn't be current.
+  // Note that because rtc::Thread also supports the TQ interface and
+  // TestMainImpl::Init wraps the main test thread (bugs.webrtc.org/9714), that
+  // means that TaskQueueBase::Current() will still return a valid value.
   EXPECT_FALSE(queue->IsCurrent());
-  EXPECT_FALSE(TaskQueueBase::Current());
 
   queue->PostTask(ToQueuedTask([&event, &queue] {
     EXPECT_TRUE(queue->IsCurrent());
@@ -269,5 +271,10 @@ TEST_P(TaskQueueTest, PostTwoWithSharedUnprotectedState) {
   EXPECT_TRUE(done.Wait(1000));
 }
 
+// TaskQueueTest is a set of tests for any implementation of the TaskQueueBase.
+// Tests are instantiated next to the concrete implementation(s).
+// https://github.com/google/googletest/blob/master/googletest/docs/advanced.md#creating-value-parameterized-abstract-tests
+GTEST_ALLOW_UNINSTANTIATED_PARAMETERIZED_TEST(TaskQueueTest);
+
 }  // namespace
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/test/audioproc_float.cc b/chromium/third_party/webrtc/api/test/audioproc_float.cc
index bba9c622a12..c8d7ff71939 100644
--- a/chromium/third_party/webrtc/api/test/audioproc_float.cc
+++ b/chromium/third_party/webrtc/api/test/audioproc_float.cc
@@ -17,6 +17,12 @@
 namespace webrtc {
 namespace test {
 
+int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
+                   int argc,
+                   char* argv[]) {
+  return AudioprocFloatImpl(std::move(audio_processing), argc, argv);
+}
+
 int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
                    int argc,
                    char* argv[]) {
diff --git a/chromium/third_party/webrtc/api/test/audioproc_float.h b/chromium/third_party/webrtc/api/test/audioproc_float.h
index 2625e6ad9ac..fec2ad11fa1 100644
--- a/chromium/third_party/webrtc/api/test/audioproc_float.h
+++ b/chromium/third_party/webrtc/api/test/audioproc_float.h
@@ -22,6 +22,22 @@ namespace test {
 // This is an interface for the audio processing simulation utility. This
 // utility can be used to simulate the audioprocessing module using a recording
 // (either an AEC dump or wav files), and generate the output as a wav file.
+// Any audio_processing object specified in the input is used for the
+// simulation. The optional |audio_processing| object provides the
+// AudioProcessing instance that is used during the simulation. Note that when
+// the audio_processing object is specified all functionality that relies on
+// using the AudioProcessingBuilder is deactivated, since the AudioProcessing
+// object is already created and the builder is not used in the simulation. It
+// is needed to pass the command line flags as |argc| and |argv|, so these can
+// be interpreted properly by the utility. To see a list of all supported
+// command line flags, run the executable with the '--help' flag.
+int AudioprocFloat(rtc::scoped_refptr<AudioProcessing> audio_processing,
+                   int argc,
+                   char* argv[]);
+
+// This is an interface for the audio processing simulation utility. This
+// utility can be used to simulate the audioprocessing module using a recording
+// (either an AEC dump or wav files), and generate the output as a wav file.
 // The |ap_builder| object will be used to create the AudioProcessing instance
 // that is used during the simulation. The |ap_builder| supports setting of
 // injectable components, which will be passed on to the created AudioProcessing
diff --git a/chromium/third_party/webrtc/api/test/compile_all_headers.cc b/chromium/third_party/webrtc/api/test/compile_all_headers.cc
index 47c5c6ec841..44b9b1fe687 100644
--- a/chromium/third_party/webrtc/api/test/compile_all_headers.cc
+++ b/chromium/third_party/webrtc/api/test/compile_all_headers.cc
@@ -29,14 +29,14 @@
 
 #include "api/test/fake_frame_decryptor.h"
 #include "api/test/fake_frame_encryptor.h"
-#include "api/test/fake_media_transport.h"
-#include "api/test/loopback_media_transport.h"
 #include "api/test/mock_audio_mixer.h"
 #include "api/test/mock_frame_decryptor.h"
 #include "api/test/mock_frame_encryptor.h"
+#include "api/test/mock_peer_connection_factory_interface.h"
 #include "api/test/mock_peerconnectioninterface.h"
 #include "api/test/mock_rtpreceiver.h"
 #include "api/test/mock_rtpsender.h"
+#include "api/test/mock_transformable_video_frame.h"
 #include "api/test/mock_video_bitrate_allocator.h"
 #include "api/test/mock_video_bitrate_allocator_factory.h"
 #include "api/test/mock_video_decoder.h"
diff --git a/chromium/third_party/webrtc/api/test/create_network_emulation_manager.h b/chromium/third_party/webrtc/api/test/create_network_emulation_manager.h
index c57c34874c3..f4447437865 100644
--- a/chromium/third_party/webrtc/api/test/create_network_emulation_manager.h
+++ b/chromium/third_party/webrtc/api/test/create_network_emulation_manager.h
@@ -1,4 +1,3 @@
-
 /*
  *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
  *
@@ -18,6 +17,7 @@
 
 namespace webrtc {
 
+// Returns a non-null NetworkEmulationManager instance.
 std::unique_ptr<NetworkEmulationManager> CreateNetworkEmulationManager(
     TimeMode mode = TimeMode::kRealTime);
 
diff --git a/chromium/third_party/webrtc/api/test/create_peerconnection_quality_test_fixture.h b/chromium/third_party/webrtc/api/test/create_peerconnection_quality_test_fixture.h
index 330d86de02a..9d9d0ef5e4a 100644
--- a/chromium/third_party/webrtc/api/test/create_peerconnection_quality_test_fixture.h
+++ b/chromium/third_party/webrtc/api/test/create_peerconnection_quality_test_fixture.h
@@ -25,6 +25,7 @@ namespace webrtc_pc_e2e {
 // During the test Alice will be caller and Bob will answer the call.
 // |test_case_name| is a name of test case, that will be used for all metrics
 // reporting.
+// Returns a non-null PeerConnectionE2EQualityTestFixture instance.
 std::unique_ptr<PeerConnectionE2EQualityTestFixture>
 CreatePeerConnectionE2EQualityTestFixture(
     std::string test_case_name,
diff --git a/chromium/third_party/webrtc/api/test/create_time_controller.cc b/chromium/third_party/webrtc/api/test/create_time_controller.cc
index d3b046bd61e..1a49020aa43 100644
--- a/chromium/third_party/webrtc/api/test/create_time_controller.cc
+++ b/chromium/third_party/webrtc/api/test/create_time_controller.cc
@@ -35,13 +35,17 @@ std::unique_ptr<CallFactoryInterface> CreateTimeControllerBasedCallFactory(
     explicit TimeControllerBasedCallFactory(TimeController* time_controller)
         : time_controller_(time_controller) {}
     Call* CreateCall(const Call::Config& config) override {
-      return Call::Create(config, time_controller_->GetClock(),
-                          time_controller_->CreateProcessThread("CallModules"),
+      if (!module_thread_) {
+        module_thread_ = SharedModuleThread::Create(
+            "CallModules", [this]() { module_thread_ = nullptr; });
+      }
+      return Call::Create(config, time_controller_->GetClock(), module_thread_,
                           time_controller_->CreateProcessThread("Pacer"));
     }
 
    private:
     TimeController* time_controller_;
+    rtc::scoped_refptr<SharedModuleThread> module_thread_;
   };
   return std::make_unique<TimeControllerBasedCallFactory>(time_controller);
 }
diff --git a/chromium/third_party/webrtc/api/test/fake_datagram_transport.h b/chromium/third_party/webrtc/api/test/fake_datagram_transport.h
deleted file mode 100644
index 847b4d842ab..00000000000
--- a/chromium/third_party/webrtc/api/test/fake_datagram_transport.h
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- *  Copyright 2019 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef API_TEST_FAKE_DATAGRAM_TRANSPORT_H_
-#define API_TEST_FAKE_DATAGRAM_TRANSPORT_H_
-
-#include <cstddef>
-#include <string>
-
-#include "api/transport/datagram_transport_interface.h"
-#include "api/transport/media/media_transport_interface.h"
-
-namespace webrtc {
-
-// Maxmum size of datagrams sent by |FakeDatagramTransport|.
-constexpr size_t kMaxFakeDatagramSize = 1000;
-
-// Fake datagram transport.  Does not support making an actual connection
-// or sending data.  Only used for tests that need to stub out a transport.
-class FakeDatagramTransport : public DatagramTransportInterface {
- public:
-  FakeDatagramTransport(
-      const MediaTransportSettings& settings,
-      std::string transport_parameters,
-      const std::function<bool(absl::string_view, absl::string_view)>&
-          are_parameters_compatible)
-      : settings_(settings),
-        transport_parameters_(transport_parameters),
-        are_parameters_compatible_(are_parameters_compatible) {}
-
-  ~FakeDatagramTransport() override { RTC_DCHECK(!state_callback_); }
-
-  void Connect(rtc::PacketTransportInternal* packet_transport) override {
-    packet_transport_ = packet_transport;
-  }
-
-  CongestionControlInterface* congestion_control() override {
-    return nullptr;  // Datagram interface doesn't provide this yet.
-  }
-
-  void SetTransportStateCallback(
-      MediaTransportStateCallback* callback) override {
-    state_callback_ = callback;
-  }
-
-  RTCError SendDatagram(rtc::ArrayView<const uint8_t> data,
-                        DatagramId datagram_id) override {
-    return RTCError::OK();
-  }
-
-  size_t GetLargestDatagramSize() const override {
-    return kMaxFakeDatagramSize;
-  }
-
-  void SetDatagramSink(DatagramSinkInterface* sink) override {}
-
-  std::string GetTransportParameters() const override {
-    if (settings_.remote_transport_parameters) {
-      return *settings_.remote_transport_parameters;
-    }
-    return transport_parameters_;
-  }
-
-  RTCError SetRemoteTransportParameters(
-      absl::string_view remote_parameters) override {
-    if (are_parameters_compatible_(GetTransportParameters(),
-                                   remote_parameters)) {
-      return RTCError::OK();
-    }
-    return RTCError(RTCErrorType::UNSUPPORTED_PARAMETER,
-                    "Incompatible remote transport parameters");
-  }
-
-  RTCError OpenChannel(int channel_id) override {
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
-  }
-
-  RTCError SendData(int channel_id,
-                    const SendDataParams& params,
-                    const rtc::CopyOnWriteBuffer& buffer) override {
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
-  }
-
-  RTCError CloseChannel(int channel_id) override {
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
-  }
-
-  void SetDataSink(DataChannelSink* /*sink*/) override {}
-
-  bool IsReadyToSend() const override { return false; }
-
-  rtc::PacketTransportInternal* packet_transport() { return packet_transport_; }
-
-  void set_state(webrtc::MediaTransportState state) {
-    if (state_callback_) {
-      state_callback_->OnStateChanged(state);
-    }
-  }
-
-  const MediaTransportSettings& settings() { return settings_; }
-
- private:
-  const MediaTransportSettings settings_;
-  const std::string transport_parameters_;
-  const std::function<bool(absl::string_view, absl::string_view)>
-      are_parameters_compatible_;
-
-  rtc::PacketTransportInternal* packet_transport_ = nullptr;
-  MediaTransportStateCallback* state_callback_ = nullptr;
-};
-
-}  // namespace webrtc
-
-#endif  // API_TEST_FAKE_DATAGRAM_TRANSPORT_H_
diff --git a/chromium/third_party/webrtc/api/test/fake_media_transport.h b/chromium/third_party/webrtc/api/test/fake_media_transport.h
deleted file mode 100644
index 530394710ab..00000000000
--- a/chromium/third_party/webrtc/api/test/fake_media_transport.h
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- *  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 API_TEST_FAKE_MEDIA_TRANSPORT_H_
-#define API_TEST_FAKE_MEDIA_TRANSPORT_H_
-
-#include <memory>
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "absl/algorithm/container.h"
-#include "api/test/fake_datagram_transport.h"
-#include "api/transport/media/media_transport_interface.h"
-
-namespace webrtc {
-
-// Fake media transport factory creates fake media transport.
-// Also creates fake datagram transport, since both media and datagram
-// transports are created by |MediaTransportFactory|.
-class FakeMediaTransportFactory : public MediaTransportFactory {
- public:
-  explicit FakeMediaTransportFactory(
-      const absl::optional<std::string>& transport_offer = "")
-      : transport_offer_(transport_offer) {}
-  ~FakeMediaTransportFactory() = default;
-
-  std::string GetTransportName() const override { return "fake"; }
-
-  RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
-      rtc::PacketTransportInternal* packet_transport,
-      rtc::Thread* network_thread,
-      const MediaTransportSettings& settings) override {
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
-  }
-
-  RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
-      rtc::Thread* network_thread,
-      const MediaTransportSettings& settings) override {
-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
-  }
-
-  RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
-  CreateDatagramTransport(rtc::Thread* network_thread,
-                          const MediaTransportSettings& settings) override {
-    return std::unique_ptr<DatagramTransportInterface>(
-        new FakeDatagramTransport(settings, transport_offer_.value_or(""),
-                                  transport_parameters_comparison_));
-  }
-
-  void set_transport_parameters_comparison(
-      std::function<bool(absl::string_view, absl::string_view)> comparison) {
-    transport_parameters_comparison_ = std::move(comparison);
-  }
-
- private:
-  const absl::optional<std::string> transport_offer_;
-  std::function<bool(absl::string_view, absl::string_view)>
-      transport_parameters_comparison_ =
-          [](absl::string_view local, absl::string_view remote) {
-            return local == remote;
-          };
-};
-
-}  // namespace webrtc
-
-#endif  // API_TEST_FAKE_MEDIA_TRANSPORT_H_
diff --git a/chromium/third_party/webrtc/api/test/loopback_media_transport.cc b/chromium/third_party/webrtc/api/test/loopback_media_transport.cc
deleted file mode 100644
index 18ce93cd7ec..00000000000
--- a/chromium/third_party/webrtc/api/test/loopback_media_transport.cc
+++ /dev/null
@@ -1,373 +0,0 @@
-/*
- *  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 "api/test/loopback_media_transport.h"
-
-#include <memory>
-
-#include "absl/algorithm/container.h"
-#include "rtc_base/time_utils.h"
-
-namespace webrtc {
-
-namespace {
-
-constexpr size_t kLoopbackMaxDatagramSize = 1200;
-
-class WrapperDatagramTransport : public DatagramTransportInterface {
- public:
-  explicit WrapperDatagramTransport(DatagramTransportInterface* wrapped)
-      : wrapped_(wrapped) {}
-
-  // Datagram transport overrides.
-  void Connect(rtc::PacketTransportInternal* packet_transport) override {
-    return wrapped_->Connect(packet_transport);
-  }
-
-  CongestionControlInterface* congestion_control() override {
-    return wrapped_->congestion_control();
-  }
-
-  void SetTransportStateCallback(
-      MediaTransportStateCallback* callback) override {
-    return wrapped_->SetTransportStateCallback(callback);
-  }
-
-  RTCError SendDatagram(rtc::ArrayView<const uint8_t> data,
-                        DatagramId datagram_id) override {
-    return wrapped_->SendDatagram(data, datagram_id);
-  }
-
-  size_t GetLargestDatagramSize() const override {
-    return wrapped_->GetLargestDatagramSize();
-  }
-
-  void SetDatagramSink(DatagramSinkInterface* sink) override {
-    return wrapped_->SetDatagramSink(sink);
-  }
-
-  std::string GetTransportParameters() const override {
-    return wrapped_->GetTransportParameters();
-  }
-
-  RTCError SetRemoteTransportParameters(absl::string_view parameters) override {
-    return wrapped_->SetRemoteTransportParameters(parameters);
-  }
-
-  // Data channel overrides.
-  RTCError OpenChannel(int channel_id) override {
-    return wrapped_->OpenChannel(channel_id);
-  }
-
-  RTCError SendData(int channel_id,
-                    const SendDataParams& params,
-                    const rtc::CopyOnWriteBuffer& buffer) override {
-    return wrapped_->SendData(channel_id, params, buffer);
-  }
-
-  RTCError CloseChannel(int channel_id) override {
-    return wrapped_->CloseChannel(channel_id);
-  }
-
-  void SetDataSink(DataChannelSink* sink) override {
-    wrapped_->SetDataSink(sink);
-  }
-
-  bool IsReadyToSend() const override { return wrapped_->IsReadyToSend(); }
-
- private:
-  DatagramTransportInterface* wrapped_;
-};
-
-}  // namespace
-
-WrapperMediaTransportFactory::WrapperMediaTransportFactory(
-    DatagramTransportInterface* wrapped_datagram_transport)
-    : wrapped_datagram_transport_(wrapped_datagram_transport) {}
-
-WrapperMediaTransportFactory::WrapperMediaTransportFactory(
-    MediaTransportFactory* wrapped)
-    : wrapped_factory_(wrapped) {}
-
-RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
-WrapperMediaTransportFactory::CreateMediaTransport(
-    rtc::PacketTransportInternal* packet_transport,
-    rtc::Thread* network_thread,
-    const MediaTransportSettings& settings) {
-  return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
-}
-
-RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
-WrapperMediaTransportFactory::CreateDatagramTransport(
-    rtc::Thread* network_thread,
-    const MediaTransportSettings& settings) {
-  created_transport_count_++;
-  if (wrapped_factory_) {
-    return wrapped_factory_->CreateDatagramTransport(network_thread, settings);
-  }
-  return {
-      std::make_unique<WrapperDatagramTransport>(wrapped_datagram_transport_)};
-}
-
-std::string WrapperMediaTransportFactory::GetTransportName() const {
-  if (wrapped_factory_) {
-    return wrapped_factory_->GetTransportName();
-  }
-  return "wrapped-transport";
-}
-
-int WrapperMediaTransportFactory::created_transport_count() const {
-  return created_transport_count_;
-}
-
-RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
-WrapperMediaTransportFactory::CreateMediaTransport(
-    rtc::Thread* network_thread,
-    const MediaTransportSettings& settings) {
-  return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
-}
-
-MediaTransportPair::MediaTransportPair(rtc::Thread* thread)
-    : first_datagram_transport_(thread),
-      second_datagram_transport_(thread),
-      first_factory_(&first_datagram_transport_),
-      second_factory_(&second_datagram_transport_) {
-  first_datagram_transport_.Connect(&second_datagram_transport_);
-  second_datagram_transport_.Connect(&first_datagram_transport_);
-}
-
-MediaTransportPair::~MediaTransportPair() = default;
-
-MediaTransportPair::LoopbackDataChannelTransport::LoopbackDataChannelTransport(
-    rtc::Thread* thread)
-    : thread_(thread) {}
-
-MediaTransportPair::LoopbackDataChannelTransport::
-    ~LoopbackDataChannelTransport() {
-  RTC_CHECK(data_sink_ == nullptr);
-}
-
-void MediaTransportPair::LoopbackDataChannelTransport::Connect(
-    LoopbackDataChannelTransport* other) {
-  other_ = other;
-}
-
-RTCError MediaTransportPair::LoopbackDataChannelTransport::OpenChannel(
-    int channel_id) {
-  // No-op.  No need to open channels for the loopback.
-  return RTCError::OK();
-}
-
-RTCError MediaTransportPair::LoopbackDataChannelTransport::SendData(
-    int channel_id,
-    const SendDataParams& params,
-    const rtc::CopyOnWriteBuffer& buffer) {
-  invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_,
-                             [this, channel_id, params, buffer] {
-                               other_->OnData(channel_id, params.type, buffer);
-                             });
-  return RTCError::OK();
-}
-
-RTCError MediaTransportPair::LoopbackDataChannelTransport::CloseChannel(
-    int channel_id) {
-  invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, channel_id] {
-    other_->OnRemoteCloseChannel(channel_id);
-    rtc::CritScope lock(&sink_lock_);
-    if (data_sink_) {
-      data_sink_->OnChannelClosed(channel_id);
-    }
-  });
-  return RTCError::OK();
-}
-
-void MediaTransportPair::LoopbackDataChannelTransport::SetDataSink(
-    DataChannelSink* sink) {
-  rtc::CritScope lock(&sink_lock_);
-  data_sink_ = sink;
-  if (data_sink_ && ready_to_send_) {
-    data_sink_->OnReadyToSend();
-  }
-}
-
-bool MediaTransportPair::LoopbackDataChannelTransport::IsReadyToSend() const {
-  rtc::CritScope lock(&sink_lock_);
-  return ready_to_send_;
-}
-
-void MediaTransportPair::LoopbackDataChannelTransport::FlushAsyncInvokes() {
-  invoker_.Flush(thread_);
-}
-
-void MediaTransportPair::LoopbackDataChannelTransport::OnData(
-    int channel_id,
-    DataMessageType type,
-    const rtc::CopyOnWriteBuffer& buffer) {
-  rtc::CritScope lock(&sink_lock_);
-  if (data_sink_) {
-    data_sink_->OnDataReceived(channel_id, type, buffer);
-  }
-}
-
-void MediaTransportPair::LoopbackDataChannelTransport::OnRemoteCloseChannel(
-    int channel_id) {
-  rtc::CritScope lock(&sink_lock_);
-  if (data_sink_) {
-    data_sink_->OnChannelClosing(channel_id);
-    data_sink_->OnChannelClosed(channel_id);
-  }
-}
-
-void MediaTransportPair::LoopbackDataChannelTransport::OnReadyToSend(
-    bool ready_to_send) {
-  invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, ready_to_send] {
-    rtc::CritScope lock(&sink_lock_);
-    ready_to_send_ = ready_to_send;
-    // Propagate state to data channel sink, if present.
-    if (data_sink_ && ready_to_send_) {
-      data_sink_->OnReadyToSend();
-    }
-  });
-}
-
-MediaTransportPair::LoopbackDatagramTransport::LoopbackDatagramTransport(
-    rtc::Thread* thread)
-    : thread_(thread), dc_transport_(thread) {}
-
-void MediaTransportPair::LoopbackDatagramTransport::Connect(
-    LoopbackDatagramTransport* other) {
-  other_ = other;
-  dc_transport_.Connect(&other->dc_transport_);
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::Connect(
-    rtc::PacketTransportInternal* packet_transport) {
-  if (state_after_connect_) {
-    SetState(*state_after_connect_);
-  }
-}
-
-CongestionControlInterface*
-MediaTransportPair::LoopbackDatagramTransport::congestion_control() {
-  return nullptr;
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::SetTransportStateCallback(
-    MediaTransportStateCallback* callback) {
-  RTC_DCHECK_RUN_ON(thread_);
-  state_callback_ = callback;
-  if (state_callback_) {
-    state_callback_->OnStateChanged(state_);
-  }
-}
-
-RTCError MediaTransportPair::LoopbackDatagramTransport::SendDatagram(
-    rtc::ArrayView<const uint8_t> data,
-    DatagramId datagram_id) {
-  rtc::CopyOnWriteBuffer buffer;
-  buffer.SetData(data.data(), data.size());
-  invoker_.AsyncInvoke<void>(
-      RTC_FROM_HERE, thread_, [this, datagram_id, buffer = std::move(buffer)] {
-        RTC_DCHECK_RUN_ON(thread_);
-        other_->DeliverDatagram(std::move(buffer));
-        if (sink_) {
-          DatagramAck ack;
-          ack.datagram_id = datagram_id;
-          ack.receive_timestamp = Timestamp::Micros(rtc::TimeMicros());
-          sink_->OnDatagramAcked(ack);
-        }
-      });
-  return RTCError::OK();
-}
-
-size_t MediaTransportPair::LoopbackDatagramTransport::GetLargestDatagramSize()
-    const {
-  return kLoopbackMaxDatagramSize;
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::SetDatagramSink(
-    DatagramSinkInterface* sink) {
-  RTC_DCHECK_RUN_ON(thread_);
-  sink_ = sink;
-}
-
-std::string
-MediaTransportPair::LoopbackDatagramTransport::GetTransportParameters() const {
-  return transport_parameters_;
-}
-
-RTCError
-MediaTransportPair::LoopbackDatagramTransport::SetRemoteTransportParameters(
-    absl::string_view remote_parameters) {
-  RTC_DCHECK_RUN_ON(thread_);
-  if (transport_parameters_comparison_(GetTransportParameters(),
-                                       remote_parameters)) {
-    return RTCError::OK();
-  }
-  return RTCError(RTCErrorType::UNSUPPORTED_PARAMETER,
-                  "Incompatible remote transport parameters");
-}
-
-RTCError MediaTransportPair::LoopbackDatagramTransport::OpenChannel(
-    int channel_id) {
-  return dc_transport_.OpenChannel(channel_id);
-}
-
-RTCError MediaTransportPair::LoopbackDatagramTransport::SendData(
-    int channel_id,
-    const SendDataParams& params,
-    const rtc::CopyOnWriteBuffer& buffer) {
-  return dc_transport_.SendData(channel_id, params, buffer);
-}
-
-RTCError MediaTransportPair::LoopbackDatagramTransport::CloseChannel(
-    int channel_id) {
-  return dc_transport_.CloseChannel(channel_id);
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::SetDataSink(
-    DataChannelSink* sink) {
-  dc_transport_.SetDataSink(sink);
-}
-
-bool MediaTransportPair::LoopbackDatagramTransport::IsReadyToSend() const {
-  return dc_transport_.IsReadyToSend();
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::SetState(
-    MediaTransportState state) {
-  invoker_.AsyncInvoke<void>(RTC_FROM_HERE, thread_, [this, state] {
-    RTC_DCHECK_RUN_ON(thread_);
-    state_ = state;
-    if (state_callback_) {
-      state_callback_->OnStateChanged(state_);
-    }
-  });
-  dc_transport_.OnReadyToSend(state == MediaTransportState::kWritable);
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::SetStateAfterConnect(
-    MediaTransportState state) {
-  state_after_connect_ = state;
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::FlushAsyncInvokes() {
-  dc_transport_.FlushAsyncInvokes();
-}
-
-void MediaTransportPair::LoopbackDatagramTransport::DeliverDatagram(
-    rtc::CopyOnWriteBuffer buffer) {
-  RTC_DCHECK_RUN_ON(thread_);
-  if (sink_) {
-    sink_->OnDatagramReceived(buffer);
-  }
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/test/loopback_media_transport.h b/chromium/third_party/webrtc/api/test/loopback_media_transport.h
deleted file mode 100644
index 468965ba311..00000000000
--- a/chromium/third_party/webrtc/api/test/loopback_media_transport.h
+++ /dev/null
@@ -1,269 +0,0 @@
-/*
- *  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 API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
-#define API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
-
-#include <memory>
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "api/transport/datagram_transport_interface.h"
-#include "api/transport/media/media_transport_interface.h"
-#include "rtc_base/async_invoker.h"
-#include "rtc_base/critical_section.h"
-#include "rtc_base/thread.h"
-#include "rtc_base/thread_checker.h"
-
-namespace webrtc {
-
-// Wrapper used to hand out unique_ptrs to loopback media
-// transport without ownership changes to the underlying
-// transport.
-// It works in two modes:
-// It can either wrap a factory, or it can wrap an existing interface.
-// In the former mode, it delegates the work to the wrapped factory.
-// In the latter mode, it always returns static instance of the transport
-// interface.
-//
-// Example use:
-// Factory wrap_static_interface = Wrapper(media_transport_interface);
-// Factory wrap_factory = Wrapper(wrap_static_interface);
-// The second factory may be created multiple times, and ownership may be passed
-// to the client. The first factory counts the number of invocations of
-// CreateMediaTransport();
-class WrapperMediaTransportFactory : public MediaTransportFactory {
- public:
-  explicit WrapperMediaTransportFactory(
-      DatagramTransportInterface* wrapped_datagram_transport);
-  explicit WrapperMediaTransportFactory(MediaTransportFactory* wrapped);
-
-  RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
-      rtc::PacketTransportInternal* packet_transport,
-      rtc::Thread* network_thread,
-      const MediaTransportSettings& settings) override;
-
-  RTCErrorOr<std::unique_ptr<MediaTransportInterface>> CreateMediaTransport(
-      rtc::Thread* network_thread,
-      const MediaTransportSettings& settings) override;
-
-  RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
-  CreateDatagramTransport(rtc::Thread* network_thread,
-                          const MediaTransportSettings& settings) override;
-
-  std::string GetTransportName() const override;
-
-  int created_transport_count() const;
-
- private:
-  DatagramTransportInterface* wrapped_datagram_transport_ = nullptr;
-  MediaTransportFactory* wrapped_factory_ = nullptr;
-  int created_transport_count_ = 0;
-};
-
-// Contains two MediaTransportsInterfaces that are connected to each other.
-// Currently supports audio only.
-class MediaTransportPair {
- public:
-  struct Stats {
-    int sent_audio_frames = 0;
-    int received_audio_frames = 0;
-    int sent_video_frames = 0;
-    int received_video_frames = 0;
-  };
-
-  explicit MediaTransportPair(rtc::Thread* thread);
-  ~MediaTransportPair();
-
-  DatagramTransportInterface* first_datagram_transport() {
-    return &first_datagram_transport_;
-  }
-  DatagramTransportInterface* second_datagram_transport() {
-    return &second_datagram_transport_;
-  }
-
-  std::unique_ptr<MediaTransportFactory> first_factory() {
-    return std::make_unique<WrapperMediaTransportFactory>(&first_factory_);
-  }
-
-  std::unique_ptr<MediaTransportFactory> second_factory() {
-    return std::make_unique<WrapperMediaTransportFactory>(&second_factory_);
-  }
-
-  void SetState(MediaTransportState state) {
-    first_datagram_transport_.SetState(state);
-    second_datagram_transport_.SetState(state);
-  }
-
-  void SetFirstState(MediaTransportState state) {
-    first_datagram_transport_.SetState(state);
-  }
-
-  void SetSecondStateAfterConnect(MediaTransportState state) {
-    second_datagram_transport_.SetState(state);
-  }
-
-  void SetFirstDatagramTransportParameters(const std::string& params) {
-    first_datagram_transport_.set_transport_parameters(params);
-  }
-
-  void SetSecondDatagramTransportParameters(const std::string& params) {
-    second_datagram_transport_.set_transport_parameters(params);
-  }
-
-  void SetFirstDatagramTransportParametersComparison(
-      std::function<bool(absl::string_view, absl::string_view)> comparison) {
-    first_datagram_transport_.set_transport_parameters_comparison(
-        std::move(comparison));
-  }
-
-  void SetSecondDatagramTransportParametersComparison(
-      std::function<bool(absl::string_view, absl::string_view)> comparison) {
-    second_datagram_transport_.set_transport_parameters_comparison(
-        std::move(comparison));
-  }
-
-  void FlushAsyncInvokes() {
-    first_datagram_transport_.FlushAsyncInvokes();
-    second_datagram_transport_.FlushAsyncInvokes();
-  }
-
-  int first_factory_transport_count() const {
-    return first_factory_.created_transport_count();
-  }
-
-  int second_factory_transport_count() const {
-    return second_factory_.created_transport_count();
-  }
-
- private:
-  class LoopbackDataChannelTransport : public DataChannelTransportInterface {
-   public:
-    explicit LoopbackDataChannelTransport(rtc::Thread* thread);
-    ~LoopbackDataChannelTransport() override;
-
-    void Connect(LoopbackDataChannelTransport* other);
-
-    RTCError OpenChannel(int channel_id) override;
-
-    RTCError SendData(int channel_id,
-                      const SendDataParams& params,
-                      const rtc::CopyOnWriteBuffer& buffer) override;
-
-    RTCError CloseChannel(int channel_id) override;
-
-    bool IsReadyToSend() const override;
-
-    void SetDataSink(DataChannelSink* sink) override;
-
-    void OnReadyToSend(bool ready_to_send);
-
-    void FlushAsyncInvokes();
-
-   private:
-    void OnData(int channel_id,
-                DataMessageType type,
-                const rtc::CopyOnWriteBuffer& buffer);
-
-    void OnRemoteCloseChannel(int channel_id);
-
-    rtc::Thread* const thread_;
-    rtc::CriticalSection sink_lock_;
-    DataChannelSink* data_sink_ RTC_GUARDED_BY(sink_lock_) = nullptr;
-
-    bool ready_to_send_ RTC_GUARDED_BY(sink_lock_) = false;
-
-    LoopbackDataChannelTransport* other_;
-
-    rtc::AsyncInvoker invoker_;
-  };
-
-  class LoopbackDatagramTransport : public DatagramTransportInterface {
-   public:
-    explicit LoopbackDatagramTransport(rtc::Thread* thread);
-
-    void Connect(LoopbackDatagramTransport* other);
-
-    // Datagram transport overrides.
-    void Connect(rtc::PacketTransportInternal* packet_transport) override;
-    CongestionControlInterface* congestion_control() override;
-    void SetTransportStateCallback(
-        MediaTransportStateCallback* callback) override;
-    RTCError SendDatagram(rtc::ArrayView<const uint8_t> data,
-                          DatagramId datagram_id) override;
-    size_t GetLargestDatagramSize() const override;
-    void SetDatagramSink(DatagramSinkInterface* sink) override;
-    std::string GetTransportParameters() const override;
-    RTCError SetRemoteTransportParameters(
-        absl::string_view remote_parameters) override;
-
-    // Data channel overrides.
-    RTCError OpenChannel(int channel_id) override;
-    RTCError SendData(int channel_id,
-                      const SendDataParams& params,
-                      const rtc::CopyOnWriteBuffer& buffer) override;
-    RTCError CloseChannel(int channel_id) override;
-    void SetDataSink(DataChannelSink* sink) override;
-    bool IsReadyToSend() const override;
-
-    // Loopback-specific functionality.
-    void SetState(MediaTransportState state);
-
-    // When Connect() is called, the datagram transport will enter this state.
-    // This is useful for mimicking zero-RTT connectivity, for example.
-    void SetStateAfterConnect(MediaTransportState state);
-    void FlushAsyncInvokes();
-
-    void set_transport_parameters(const std::string& value) {
-      transport_parameters_ = value;
-    }
-
-    void set_transport_parameters_comparison(
-        std::function<bool(absl::string_view, absl::string_view)> comparison) {
-      thread_->Invoke<void>(
-          RTC_FROM_HERE, [this, comparison = std::move(comparison)] {
-            RTC_DCHECK_RUN_ON(thread_);
-            transport_parameters_comparison_ = std::move(comparison);
-          });
-    }
-
-   private:
-    void DeliverDatagram(rtc::CopyOnWriteBuffer buffer);
-
-    rtc::Thread* thread_;
-    LoopbackDataChannelTransport dc_transport_;
-
-    MediaTransportState state_ RTC_GUARDED_BY(thread_) =
-        MediaTransportState::kPending;
-    DatagramSinkInterface* sink_ RTC_GUARDED_BY(thread_) = nullptr;
-    MediaTransportStateCallback* state_callback_ RTC_GUARDED_BY(thread_) =
-        nullptr;
-    LoopbackDatagramTransport* other_;
-
-    std::string transport_parameters_;
-    std::function<bool(absl::string_view, absl::string_view)>
-        transport_parameters_comparison_ RTC_GUARDED_BY(thread_) =
-            [](absl::string_view a, absl::string_view b) { return a == b; };
-
-    absl::optional<MediaTransportState> state_after_connect_;
-
-    rtc::AsyncInvoker invoker_;
-  };
-
-  LoopbackDatagramTransport first_datagram_transport_;
-  LoopbackDatagramTransport second_datagram_transport_;
-  WrapperMediaTransportFactory first_factory_;
-  WrapperMediaTransportFactory second_factory_;
-};
-
-}  // namespace webrtc
-
-#endif  // API_TEST_LOOPBACK_MEDIA_TRANSPORT_H_
diff --git a/chromium/third_party/webrtc/api/test/loopback_media_transport_unittest.cc b/chromium/third_party/webrtc/api/test/loopback_media_transport_unittest.cc
deleted file mode 100644
index f036de3eae0..00000000000
--- a/chromium/third_party/webrtc/api/test/loopback_media_transport_unittest.cc
+++ /dev/null
@@ -1,201 +0,0 @@
-/*
- *  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 "api/test/loopback_media_transport.h"
-
-#include <algorithm>
-#include <memory>
-#include <vector>
-
-#include "test/gmock.h"
-
-namespace webrtc {
-
-namespace {
-
-class MockMediaTransportAudioSinkInterface
-    : public MediaTransportAudioSinkInterface {
- public:
-  MOCK_METHOD2(OnData, void(uint64_t, MediaTransportEncodedAudioFrame));
-};
-
-class MockMediaTransportVideoSinkInterface
-    : public MediaTransportVideoSinkInterface {
- public:
-  MOCK_METHOD2(OnData, void(uint64_t, MediaTransportEncodedVideoFrame));
-};
-
-class MockMediaTransportKeyFrameRequestCallback
-    : public MediaTransportKeyFrameRequestCallback {
- public:
-  MOCK_METHOD1(OnKeyFrameRequested, void(uint64_t));
-};
-
-class MockDataChannelSink : public DataChannelSink {
- public:
-  MOCK_METHOD3(OnDataReceived,
-               void(int, DataMessageType, const rtc::CopyOnWriteBuffer&));
-  MOCK_METHOD1(OnChannelClosing, void(int));
-  MOCK_METHOD1(OnChannelClosed, void(int));
-  MOCK_METHOD0(OnReadyToSend, void());
-};
-
-class MockStateCallback : public MediaTransportStateCallback {
- public:
-  MOCK_METHOD1(OnStateChanged, void(MediaTransportState));
-};
-
-}  // namespace
-
-TEST(LoopbackMediaTransport, DataDeliveredToSink) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  MockDataChannelSink sink;
-  transport_pair.first_datagram_transport()->SetDataSink(&sink);
-
-  const int channel_id = 1;
-  EXPECT_CALL(
-      sink, OnDataReceived(
-                channel_id, DataMessageType::kText,
-                ::testing::Property<rtc::CopyOnWriteBuffer, const char*>(
-                    &rtc::CopyOnWriteBuffer::cdata, ::testing::StrEq("foo"))));
-
-  SendDataParams params;
-  params.type = DataMessageType::kText;
-  rtc::CopyOnWriteBuffer buffer("foo");
-  transport_pair.second_datagram_transport()->SendData(channel_id, params,
-                                                       buffer);
-
-  transport_pair.FlushAsyncInvokes();
-  transport_pair.first_datagram_transport()->SetDataSink(nullptr);
-}
-
-TEST(LoopbackMediaTransport, CloseDeliveredToSink) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  MockDataChannelSink first_sink;
-  transport_pair.first_datagram_transport()->SetDataSink(&first_sink);
-
-  MockDataChannelSink second_sink;
-  transport_pair.second_datagram_transport()->SetDataSink(&second_sink);
-
-  const int channel_id = 1;
-  {
-    ::testing::InSequence s;
-    EXPECT_CALL(second_sink, OnChannelClosing(channel_id));
-    EXPECT_CALL(second_sink, OnChannelClosed(channel_id));
-    EXPECT_CALL(first_sink, OnChannelClosed(channel_id));
-  }
-
-  transport_pair.first_datagram_transport()->CloseChannel(channel_id);
-
-  transport_pair.FlushAsyncInvokes();
-  transport_pair.first_datagram_transport()->SetDataSink(nullptr);
-  transport_pair.second_datagram_transport()->SetDataSink(nullptr);
-}
-
-TEST(LoopbackMediaTransport, InitialStateDeliveredWhenCallbackSet) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  MockStateCallback state_callback;
-  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kPending));
-
-  thread->Invoke<void>(RTC_FROM_HERE, [&transport_pair, &state_callback] {
-    transport_pair.first_datagram_transport()->SetTransportStateCallback(
-        &state_callback);
-  });
-  transport_pair.FlushAsyncInvokes();
-}
-
-TEST(LoopbackMediaTransport, ChangedStateDeliveredWhenCallbackSet) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  transport_pair.SetState(MediaTransportState::kWritable);
-  transport_pair.FlushAsyncInvokes();
-
-  MockStateCallback state_callback;
-
-  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kWritable));
-  thread->Invoke<void>(RTC_FROM_HERE, [&transport_pair, &state_callback] {
-    transport_pair.first_datagram_transport()->SetTransportStateCallback(
-        &state_callback);
-  });
-  transport_pair.FlushAsyncInvokes();
-}
-
-TEST(LoopbackMediaTransport, StateChangeDeliveredToCallback) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  MockStateCallback state_callback;
-
-  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kPending));
-  EXPECT_CALL(state_callback, OnStateChanged(MediaTransportState::kWritable));
-  thread->Invoke<void>(RTC_FROM_HERE, [&transport_pair, &state_callback] {
-    transport_pair.first_datagram_transport()->SetTransportStateCallback(
-        &state_callback);
-  });
-  transport_pair.SetState(MediaTransportState::kWritable);
-  transport_pair.FlushAsyncInvokes();
-}
-
-TEST(LoopbackMediaTransport, NotReadyToSendWhenDataSinkSet) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  MockDataChannelSink data_channel_sink;
-  EXPECT_CALL(data_channel_sink, OnReadyToSend()).Times(0);
-
-  transport_pair.first_datagram_transport()->SetDataSink(&data_channel_sink);
-  transport_pair.FlushAsyncInvokes();
-  transport_pair.first_datagram_transport()->SetDataSink(nullptr);
-}
-
-TEST(LoopbackMediaTransport, ReadyToSendWhenDataSinkSet) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  transport_pair.SetState(MediaTransportState::kWritable);
-  transport_pair.FlushAsyncInvokes();
-
-  MockDataChannelSink data_channel_sink;
-  EXPECT_CALL(data_channel_sink, OnReadyToSend());
-
-  transport_pair.first_datagram_transport()->SetDataSink(&data_channel_sink);
-  transport_pair.FlushAsyncInvokes();
-  transport_pair.first_datagram_transport()->SetDataSink(nullptr);
-}
-
-TEST(LoopbackMediaTransport, StateChangeDeliveredToDataSink) {
-  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
-  thread->Start();
-  MediaTransportPair transport_pair(thread.get());
-
-  MockDataChannelSink data_channel_sink;
-  EXPECT_CALL(data_channel_sink, OnReadyToSend());
-
-  transport_pair.first_datagram_transport()->SetDataSink(&data_channel_sink);
-  transport_pair.SetState(MediaTransportState::kWritable);
-  transport_pair.FlushAsyncInvokes();
-  transport_pair.first_datagram_transport()->SetDataSink(nullptr);
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/test/mock_audio_mixer.h b/chromium/third_party/webrtc/api/test/mock_audio_mixer.h
index aee717bebf4..88dc108ca3c 100644
--- a/chromium/third_party/webrtc/api/test/mock_audio_mixer.h
+++ b/chromium/third_party/webrtc/api/test/mock_audio_mixer.h
@@ -19,12 +19,9 @@ namespace test {
 
 class MockAudioMixer : public AudioMixer {
  public:
-  MOCK_METHOD(bool, AddSource, (Source * audio_source), (override));
-  MOCK_METHOD(void, RemoveSource, (Source * audio_source), (override));
-  MOCK_METHOD(void,
-              Mix,
-              (size_t number_of_channels, AudioFrame* audio_frame_for_mixing),
-              (override));
+  MOCK_METHOD(bool, AddSource, (Source*), (override));
+  MOCK_METHOD(void, RemoveSource, (Source*), (override));
+  MOCK_METHOD(void, Mix, (size_t number_of_channels, AudioFrame*), (override));
 };
 }  // namespace test
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/test/mock_peer_connection_factory_interface.h b/chromium/third_party/webrtc/api/test/mock_peer_connection_factory_interface.h
new file mode 100644
index 00000000000..19c3f4063e9
--- /dev/null
+++ b/chromium/third_party/webrtc/api/test/mock_peer_connection_factory_interface.h
@@ -0,0 +1,75 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_TEST_MOCK_PEER_CONNECTION_FACTORY_INTERFACE_H_
+#define API_TEST_MOCK_PEER_CONNECTION_FACTORY_INTERFACE_H_
+
+#include <memory>
+#include <string>
+
+#include "api/peer_connection_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockPeerConnectionFactoryInterface final
+    : public rtc::RefCountedObject<webrtc::PeerConnectionFactoryInterface> {
+ public:
+  rtc::scoped_refptr<MockPeerConnectionFactoryInterface> Create() {
+    return new MockPeerConnectionFactoryInterface();
+  }
+
+  MOCK_METHOD(void, SetOptions, (const Options&), (override));
+  MOCK_METHOD(rtc::scoped_refptr<PeerConnectionInterface>,
+              CreatePeerConnection,
+              (const PeerConnectionInterface::RTCConfiguration&,
+               PeerConnectionDependencies),
+              (override));
+  MOCK_METHOD(rtc::scoped_refptr<PeerConnectionInterface>,
+              CreatePeerConnection,
+              (const PeerConnectionInterface::RTCConfiguration&,
+               std::unique_ptr<cricket::PortAllocator>,
+               std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
+               PeerConnectionObserver*),
+              (override));
+  MOCK_METHOD(RtpCapabilities,
+              GetRtpSenderCapabilities,
+              (cricket::MediaType),
+              (const override));
+  MOCK_METHOD(RtpCapabilities,
+              GetRtpReceiverCapabilities,
+              (cricket::MediaType),
+              (const override));
+  MOCK_METHOD(rtc::scoped_refptr<MediaStreamInterface>,
+              CreateLocalMediaStream,
+              (const std::string&),
+              (override));
+  MOCK_METHOD(rtc::scoped_refptr<AudioSourceInterface>,
+              CreateAudioSource,
+              (const cricket::AudioOptions&),
+              (override));
+  MOCK_METHOD(rtc::scoped_refptr<VideoTrackInterface>,
+              CreateVideoTrack,
+              (const std::string&, VideoTrackSourceInterface*),
+              (override));
+  MOCK_METHOD(rtc::scoped_refptr<AudioTrackInterface>,
+              CreateAudioTrack,
+              (const std::string&, AudioSourceInterface*),
+              (override));
+  MOCK_METHOD(bool, StartAecDump, (FILE*, int64_t), (override));
+  MOCK_METHOD(void, StopAecDump, (), (override));
+
+ protected:
+  MockPeerConnectionFactoryInterface() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_PEER_CONNECTION_FACTORY_INTERFACE_H_
diff --git a/chromium/third_party/webrtc/api/test/mock_transformable_video_frame.h b/chromium/third_party/webrtc/api/test/mock_transformable_video_frame.h
new file mode 100644
index 00000000000..36798b5d730
--- /dev/null
+++ b/chromium/third_party/webrtc/api/test/mock_transformable_video_frame.h
@@ -0,0 +1,38 @@
+/*
+ *  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 API_TEST_MOCK_TRANSFORMABLE_VIDEO_FRAME_H_
+#define API_TEST_MOCK_TRANSFORMABLE_VIDEO_FRAME_H_
+
+#include <vector>
+
+#include "api/frame_transformer_interface.h"
+#include "test/gmock.h"
+
+namespace webrtc {
+
+class MockTransformableVideoFrame
+    : public webrtc::TransformableVideoFrameInterface {
+ public:
+  MOCK_METHOD(rtc::ArrayView<const uint8_t>, GetData, (), (const override));
+  MOCK_METHOD(void, SetData, (rtc::ArrayView<const uint8_t> data), (override));
+  MOCK_METHOD(uint32_t, GetTimestamp, (), (const override));
+  MOCK_METHOD(uint32_t, GetSsrc, (), (const, override));
+  MOCK_METHOD(bool, IsKeyFrame, (), (const, override));
+  MOCK_METHOD(std::vector<uint8_t>, GetAdditionalData, (), (const, override));
+  MOCK_METHOD(const webrtc::VideoFrameMetadata&,
+              GetMetadata,
+              (),
+              (const, override));
+};
+
+}  // namespace webrtc
+
+#endif  // API_TEST_MOCK_TRANSFORMABLE_VIDEO_FRAME_H_
diff --git a/chromium/third_party/webrtc/api/test/mock_video_bitrate_allocator_factory.h b/chromium/third_party/webrtc/api/test/mock_video_bitrate_allocator_factory.h
index c7d883a5b9c..16af191970e 100644
--- a/chromium/third_party/webrtc/api/test/mock_video_bitrate_allocator_factory.h
+++ b/chromium/third_party/webrtc/api/test/mock_video_bitrate_allocator_factory.h
@@ -26,7 +26,7 @@ class MockVideoBitrateAllocatorFactory
               CreateVideoBitrateAllocator,
               (const VideoCodec&),
               (override));
-  MOCK_METHOD(void, Die, (), ());
+  MOCK_METHOD(void, Die, ());
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/test/mock_video_decoder_factory.h b/chromium/third_party/webrtc/api/test/mock_video_decoder_factory.h
index 1f832a156bd..98a5d40eb6f 100644
--- a/chromium/third_party/webrtc/api/test/mock_video_decoder_factory.h
+++ b/chromium/third_party/webrtc/api/test/mock_video_decoder_factory.h
@@ -27,12 +27,12 @@ class MockVideoDecoderFactory : public webrtc::VideoDecoderFactory {
   MOCK_METHOD(std::vector<webrtc::SdpVideoFormat>,
               GetSupportedFormats,
               (),
-              (const override));
+              (const, override));
   MOCK_METHOD(std::unique_ptr<webrtc::VideoDecoder>,
               CreateVideoDecoder,
               (const webrtc::SdpVideoFormat&),
               (override));
-  MOCK_METHOD(void, Die, (), ());
+  MOCK_METHOD(void, Die, ());
 };
 }  // namespace webrtc
 
diff --git a/chromium/third_party/webrtc/api/test/mock_video_encoder.h b/chromium/third_party/webrtc/api/test/mock_video_encoder.h
index c4b6b3e50c1..26d758fd6a3 100644
--- a/chromium/third_party/webrtc/api/test/mock_video_encoder.h
+++ b/chromium/third_party/webrtc/api/test/mock_video_encoder.h
@@ -23,8 +23,8 @@ class MockEncodedImageCallback : public EncodedImageCallback {
   MOCK_METHOD(Result,
               OnEncodedImage,
               (const EncodedImage& encodedImage,
-               const CodecSpecificInfo* codecSpecificInfo,
-               const RTPFragmentationHeader* fragmentation),
+               const CodecSpecificInfo*,
+               const RTPFragmentationHeader*),
               (override));
   MOCK_METHOD(void, OnDroppedFrame, (DropReason reason), (override));
 };
@@ -33,31 +33,41 @@ class MockVideoEncoder : public VideoEncoder {
  public:
   MOCK_METHOD(void,
               SetFecControllerOverride,
-              (FecControllerOverride * fec_controller_override),
+              (FecControllerOverride*),
               (override));
   MOCK_METHOD(int32_t,
               InitEncode,
-              (const VideoCodec* codecSettings,
-               int32_t numberOfCores,
-               size_t maxPayloadSize),
+              (const VideoCodec*, int32_t numberOfCores, size_t maxPayloadSize),
+              (override));
+  MOCK_METHOD(int32_t,
+              InitEncode,
+              (const VideoCodec*, const VideoEncoder::Settings& settings),
               (override));
-  MOCK_METHOD2(InitEncode,
-               int32_t(const VideoCodec* codecSettings,
-                       const VideoEncoder::Settings& settings));
 
-  MOCK_METHOD2(Encode,
-               int32_t(const VideoFrame& inputImage,
-                       const std::vector<VideoFrameType>* frame_types));
-  MOCK_METHOD1(RegisterEncodeCompleteCallback,
-               int32_t(EncodedImageCallback* callback));
-  MOCK_METHOD0(Release, int32_t());
-  MOCK_METHOD0(Reset, int32_t());
-  MOCK_METHOD1(SetRates, void(const RateControlParameters& parameters));
-  MOCK_METHOD1(OnPacketLossRateUpdate, void(float packet_loss_rate));
-  MOCK_METHOD1(OnRttUpdate, void(int64_t rtt_ms));
-  MOCK_METHOD1(OnLossNotification,
-               void(const LossNotification& loss_notification));
-  MOCK_CONST_METHOD0(GetEncoderInfo, EncoderInfo(void));
+  MOCK_METHOD(int32_t,
+              Encode,
+              (const VideoFrame& inputImage,
+               const std::vector<VideoFrameType>*),
+              (override));
+  MOCK_METHOD(int32_t,
+              RegisterEncodeCompleteCallback,
+              (EncodedImageCallback*),
+              (override));
+  MOCK_METHOD(int32_t, Release, (), (override));
+  MOCK_METHOD(void,
+              SetRates,
+              (const RateControlParameters& parameters),
+              (override));
+  MOCK_METHOD(void,
+              OnPacketLossRateUpdate,
+              (float packet_loss_rate),
+              (override));
+  MOCK_METHOD(void, OnRttUpdate, (int64_t rtt_ms), (override));
+  MOCK_METHOD(void,
+              OnLossNotification,
+              (const LossNotification& loss_notification),
+              (override));
+  MOCK_METHOD(EncoderInfo, GetEncoderInfo, (), (const, override));
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/test/mock_video_encoder_factory.h b/chromium/third_party/webrtc/api/test/mock_video_encoder_factory.h
index fa08dbd6bc2..1aa14631bed 100644
--- a/chromium/third_party/webrtc/api/test/mock_video_encoder_factory.h
+++ b/chromium/third_party/webrtc/api/test/mock_video_encoder_factory.h
@@ -27,17 +27,17 @@ class MockVideoEncoderFactory : public webrtc::VideoEncoderFactory {
   MOCK_METHOD(std::vector<SdpVideoFormat>,
               GetSupportedFormats,
               (),
-              (const override));
+              (const, override));
   MOCK_METHOD(CodecInfo,
               QueryVideoEncoder,
               (const SdpVideoFormat&),
-              (const override));
+              (const, override));
   MOCK_METHOD(std::unique_ptr<VideoEncoder>,
               CreateVideoEncoder,
               (const SdpVideoFormat&),
               (override));
 
-  MOCK_METHOD(void, Die, (), ());
+  MOCK_METHOD(void, Die, ());
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/test/network_emulation/BUILD.gn b/chromium/third_party/webrtc/api/test/network_emulation/BUILD.gn
index 5fda1e288a1..a3dd961c811 100644
--- a/chromium/third_party/webrtc/api/test/network_emulation/BUILD.gn
+++ b/chromium/third_party/webrtc/api/test/network_emulation/BUILD.gn
@@ -23,6 +23,6 @@ rtc_library("network_emulation") {
     "../../units:data_rate",
     "../../units:data_size",
     "../../units:timestamp",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
diff --git a/chromium/third_party/webrtc/api/test/peerconnection_quality_test_fixture.h b/chromium/third_party/webrtc/api/test/peerconnection_quality_test_fixture.h
index 8165443d36e..35736876cce 100644
--- a/chromium/third_party/webrtc/api/test/peerconnection_quality_test_fixture.h
+++ b/chromium/third_party/webrtc/api/test/peerconnection_quality_test_fixture.h
@@ -33,13 +33,13 @@
 #include "api/test/simulated_network.h"
 #include "api/test/stats_observer_interface.h"
 #include "api/test/video_quality_analyzer_interface.h"
-#include "api/transport/media/media_transport_interface.h"
 #include "api/transport/network_control.h"
 #include "api/units/time_delta.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "media/base/media_constants.h"
+#include "rtc_base/deprecation.h"
 #include "rtc_base/network.h"
 #include "rtc_base/rtc_certificate_generator.h"
 #include "rtc_base/ssl_certificate.h"
@@ -287,8 +287,6 @@ class PeerConnectionE2EQualityTestFixture {
     virtual PeerConfigurer* SetNetworkControllerFactory(
         std::unique_ptr<NetworkControllerFactoryInterface>
             network_controller_factory) = 0;
-    virtual PeerConfigurer* SetMediaTransportFactory(
-        std::unique_ptr<MediaTransportFactory> media_transport_factory) = 0;
     virtual PeerConfigurer* SetVideoEncoderFactory(
         std::unique_ptr<VideoEncoderFactory> video_encoder_factory) = 0;
     virtual PeerConfigurer* SetVideoDecoderFactory(
@@ -337,6 +335,9 @@ class PeerConnectionE2EQualityTestFixture {
         PeerConnectionInterface::RTCConfiguration configuration) = 0;
     // Set bitrate parameters on PeerConnection. This constraints will be
     // applied to all summed RTP streams for this peer.
+    virtual PeerConfigurer* SetBitrateSettings(
+        BitrateSettings bitrate_settings) = 0;
+    RTC_DEPRECATED
     virtual PeerConfigurer* SetBitrateParameters(
         PeerConnectionInterface::BitrateParameters bitrate_params) = 0;
   };
@@ -448,6 +449,12 @@ class PeerConnectionE2EQualityTestFixture {
   virtual void AddPeer(rtc::Thread* network_thread,
                        rtc::NetworkManager* network_manager,
                        rtc::FunctionView<void(PeerConfigurer*)> configurer) = 0;
+  // Runs the media quality test, which includes setting up the call with
+  // configured participants, running it according to provided |run_params| and
+  // terminating it properly at the end. During call duration media quality
+  // metrics are gathered, which are then reported to stdout and (if configured)
+  // to the json/protobuf output file through the WebRTC perf test results
+  // reporting system.
   virtual void Run(RunParams run_params) = 0;
 
   // Returns real test duration - the time of test execution measured during
diff --git a/chromium/third_party/webrtc/api/test/stats_observer_interface.h b/chromium/third_party/webrtc/api/test/stats_observer_interface.h
index 98c8dd937f2..ea4d6c23db9 100644
--- a/chromium/third_party/webrtc/api/test/stats_observer_interface.h
+++ b/chromium/third_party/webrtc/api/test/stats_observer_interface.h
@@ -11,9 +11,8 @@
 #ifndef API_TEST_STATS_OBSERVER_INTERFACE_H_
 #define API_TEST_STATS_OBSERVER_INTERFACE_H_
 
-#include <string>
-
-#include "api/stats_types.h"
+#include "absl/strings/string_view.h"
+#include "api/stats/rtc_stats_report.h"
 
 namespace webrtc {
 namespace webrtc_pc_e2e {
@@ -25,8 +24,9 @@ class StatsObserverInterface {
 
   // Method called when stats reports are available for the PeerConnection
   // identified by |pc_label|.
-  virtual void OnStatsReports(const std::string& pc_label,
-                              const StatsReports& reports) = 0;
+  virtual void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;
 };
 
 }  // namespace webrtc_pc_e2e
diff --git a/chromium/third_party/webrtc/api/test/test_dependency_factory.cc b/chromium/third_party/webrtc/api/test/test_dependency_factory.cc
index e72f55aab57..41ad70cc3f7 100644
--- a/chromium/third_party/webrtc/api/test/test_dependency_factory.cc
+++ b/chromium/third_party/webrtc/api/test/test_dependency_factory.cc
@@ -14,22 +14,24 @@
 #include <utility>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/thread_checker.h"
+#include "rtc_base/platform_thread_types.h"
 
 namespace webrtc {
 
+namespace {
 // This checks everything in this file gets called on the same thread. It's
 // static because it needs to look at the static methods too.
-rtc::ThreadChecker* GetThreadChecker() {
-  static rtc::ThreadChecker checker;
-  return &checker;
+bool IsValidTestDependencyFactoryThread() {
+  const rtc::PlatformThreadRef main_thread = rtc::CurrentThreadRef();
+  return rtc::IsThreadRefEqual(main_thread, rtc::CurrentThreadRef());
 }
+}  // namespace
 
 std::unique_ptr<TestDependencyFactory> TestDependencyFactory::instance_ =
     nullptr;
 
 const TestDependencyFactory& TestDependencyFactory::GetInstance() {
-  RTC_DCHECK(GetThreadChecker()->IsCurrent());
+  RTC_DCHECK(IsValidTestDependencyFactoryThread());
   if (instance_ == nullptr) {
     instance_ = std::make_unique<TestDependencyFactory>();
   }
@@ -38,14 +40,14 @@ const TestDependencyFactory& TestDependencyFactory::GetInstance() {
 
 void TestDependencyFactory::SetInstance(
     std::unique_ptr<TestDependencyFactory> instance) {
-  RTC_DCHECK(GetThreadChecker()->IsCurrent());
+  RTC_DCHECK(IsValidTestDependencyFactoryThread());
   RTC_CHECK(instance_ == nullptr);
   instance_ = std::move(instance);
 }
 
 std::unique_ptr<VideoQualityTestFixtureInterface::InjectionComponents>
 TestDependencyFactory::CreateComponents() const {
-  RTC_DCHECK(GetThreadChecker()->IsCurrent());
+  RTC_DCHECK(IsValidTestDependencyFactoryThread());
   return nullptr;
 }
 
diff --git a/chromium/third_party/webrtc/api/test/video_quality_analyzer_interface.h b/chromium/third_party/webrtc/api/test/video_quality_analyzer_interface.h
index 0d3f4415348..c5370a7089e 100644
--- a/chromium/third_party/webrtc/api/test/video_quality_analyzer_interface.h
+++ b/chromium/third_party/webrtc/api/test/video_quality_analyzer_interface.h
@@ -14,7 +14,9 @@
 #include <memory>
 #include <string>
 
+#include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
+#include "api/array_view.h"
 #include "api/test/stats_observer_interface.h"
 #include "api/video/encoded_image.h"
 #include "api/video/video_frame.h"
@@ -76,42 +78,65 @@ class VideoQualityAnalyzerInterface : public StatsObserverInterface {
   // calculations. Analyzer can perform simple calculations on the calling
   // thread in each method, but should remember, that it is the same thread,
   // that is used in video pipeline.
-  virtual void Start(std::string test_case_name, int max_threads_count) {}
+  virtual void Start(std::string test_case_name,
+                     rtc::ArrayView<const std::string> peer_names,
+                     int max_threads_count) {}
 
   // Will be called when frame was generated from the input stream.
+  // |peer_name| is name of the peer on which side frame was captured.
   // Returns frame id, that will be set by framework to the frame.
-  virtual uint16_t OnFrameCaptured(const std::string& stream_label,
+  virtual uint16_t OnFrameCaptured(absl::string_view peer_name,
+                                   const std::string& stream_label,
                                    const VideoFrame& frame) = 0;
   // Will be called before calling the encoder.
-  virtual void OnFramePreEncode(const VideoFrame& frame) {}
+  // |peer_name| is name of the peer on which side frame came to encoder.
+  virtual void OnFramePreEncode(absl::string_view peer_name,
+                                const VideoFrame& frame) {}
   // Will be called for each EncodedImage received from encoder. Single
   // VideoFrame can produce multiple EncodedImages. Each encoded image will
   // have id from VideoFrame.
-  virtual void OnFrameEncoded(uint16_t frame_id,
+  // |peer_name| is name of the peer on which side frame was encoded.
+  virtual void OnFrameEncoded(absl::string_view peer_name,
+                              uint16_t frame_id,
                               const EncodedImage& encoded_image,
                               const EncoderStats& stats) {}
   // Will be called for each frame dropped by encoder.
-  virtual void OnFrameDropped(EncodedImageCallback::DropReason reason) {}
+  // |peer_name| is name of the peer on which side frame drop was detected.
+  virtual void OnFrameDropped(absl::string_view peer_name,
+                              EncodedImageCallback::DropReason reason) {}
   // Will be called before calling the decoder.
-  virtual void OnFramePreDecode(uint16_t frame_id,
+  // |peer_name| is name of the peer on which side frame was received.
+  virtual void OnFramePreDecode(absl::string_view peer_name,
+                                uint16_t frame_id,
                                 const EncodedImage& encoded_image) {}
   // Will be called after decoding the frame.
-  virtual void OnFrameDecoded(const VideoFrame& frame,
+  // |peer_name| is name of the peer on which side frame was decoded.
+  virtual void OnFrameDecoded(absl::string_view peer_name,
+                              const VideoFrame& frame,
                               const DecoderStats& stats) {}
   // Will be called when frame will be obtained from PeerConnection stack.
-  virtual void OnFrameRendered(const VideoFrame& frame) {}
+  // |peer_name| is name of the peer on which side frame was rendered.
+  virtual void OnFrameRendered(absl::string_view peer_name,
+                               const VideoFrame& frame) {}
   // Will be called if encoder return not WEBRTC_VIDEO_CODEC_OK.
   // All available codes are listed in
   // modules/video_coding/include/video_error_codes.h
-  virtual void OnEncoderError(const VideoFrame& frame, int32_t error_code) {}
+  // |peer_name| is name of the peer on which side error acquired.
+  virtual void OnEncoderError(absl::string_view peer_name,
+                              const VideoFrame& frame,
+                              int32_t error_code) {}
   // Will be called if decoder return not WEBRTC_VIDEO_CODEC_OK.
   // All available codes are listed in
   // modules/video_coding/include/video_error_codes.h
-  virtual void OnDecoderError(uint16_t frame_id, int32_t error_code) {}
+  // |peer_name| is name of the peer on which side error acquired.
+  virtual void OnDecoderError(absl::string_view peer_name,
+                              uint16_t frame_id,
+                              int32_t error_code) {}
   // Will be called every time new stats reports are available for the
   // Peer Connection identified by |pc_label|.
-  void OnStatsReports(const std::string& pc_label,
-                      const StatsReports& stats_reports) override {}
+  void OnStatsReports(
+      absl::string_view pc_label,
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override {}
 
   // Tells analyzer that analysis complete and it should calculate final
   // statistics.
diff --git a/chromium/third_party/webrtc/api/transport/BUILD.gn b/chromium/third_party/webrtc/api/transport/BUILD.gn
index 0f07301fe41..a4ada07108b 100644
--- a/chromium/third_party/webrtc/api/transport/BUILD.gn
+++ b/chromium/third_party/webrtc/api/transport/BUILD.gn
@@ -14,10 +14,8 @@ rtc_library("bitrate_settings") {
     "bitrate_settings.cc",
     "bitrate_settings.h",
   ]
-  deps = [
-    "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
-  ]
+  deps = [ "../../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("enums") {
@@ -41,6 +39,8 @@ rtc_library("network_control") {
     "../units:data_size",
     "../units:time_delta",
     "../units:timestamp",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -49,10 +49,8 @@ rtc_library("network_control") {
 rtc_source_set("webrtc_key_value_config") {
   visibility = [ "*" ]
   sources = [ "webrtc_key_value_config.h" ]
-  deps = [
-    "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/strings",
-  ]
+  deps = [ "../../rtc_base/system:rtc_export" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
 rtc_library("field_trial_based_config") {
@@ -64,26 +62,20 @@ rtc_library("field_trial_based_config") {
   deps = [
     ":webrtc_key_value_config",
     "../../system_wrappers:field_trial",
-    "//third_party/abseil-cpp/absl/strings",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
+# TODO(nisse): Rename?
 rtc_source_set("datagram_transport_interface") {
   visibility = [ "*" ]
-  sources = [
-    "congestion_control_interface.h",
-    "data_channel_transport_interface.h",
-    "datagram_transport_interface.h",
-  ]
+  sources = [ "data_channel_transport_interface.h" ]
   deps = [
-    ":network_control",
     "..:array_view",
     "..:rtc_error",
     "../../rtc_base:rtc_base_approved",
-    "../units:data_rate",
-    "../units:timestamp",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("goog_cc") {
diff --git a/chromium/third_party/webrtc/api/transport/congestion_control_interface.h b/chromium/third_party/webrtc/api/transport/congestion_control_interface.h
deleted file mode 100644
index 40552cb4ffd..00000000000
--- a/chromium/third_party/webrtc/api/transport/congestion_control_interface.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/* 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.
- */
-
-// This is EXPERIMENTAL interface for media and datagram transports.
-
-#ifndef API_TRANSPORT_CONGESTION_CONTROL_INTERFACE_H_
-#define API_TRANSPORT_CONGESTION_CONTROL_INTERFACE_H_
-
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "api/transport/network_control.h"
-#include "api/units/data_rate.h"
-
-namespace webrtc {
-
-// TODO(nisse): Defined together with MediaTransportInterface. But we should use
-// types that aren't tied to media, so that MediaTransportInterface can depend
-// on CongestionControlInterface, but not the other way around.
-// api/transport/network_control.h may be a reasonable place.
-class MediaTransportRttObserver;
-struct MediaTransportAllocatedBitrateLimits;
-struct MediaTransportTargetRateConstraints;
-
-// Defines congestion control feedback interface for media and datagram
-// transports.
-class CongestionControlInterface {
- public:
-  virtual ~CongestionControlInterface() = default;
-
-  // Updates allocation limits.
-  virtual void SetAllocatedBitrateLimits(
-      const MediaTransportAllocatedBitrateLimits& limits) = 0;
-
-  // Sets starting rate.
-  virtual void SetTargetBitrateLimits(
-      const MediaTransportTargetRateConstraints& target_rate_constraints) = 0;
-
-  // Intended for receive side. AddRttObserver registers an observer to be
-  // called for each RTT measurement, typically once per ACK. Before media
-  // transport is destructed the observer must be unregistered.
-  //
-  // TODO(sukhanov): Looks like AddRttObserver and RemoveRttObserver were
-  // never implemented for media transport, so keeping noop implementation.
-  virtual void AddRttObserver(MediaTransportRttObserver* observer) {}
-  virtual void RemoveRttObserver(MediaTransportRttObserver* observer) {}
-
-  // Adds a target bitrate observer. Before media transport is destructed
-  // the observer must be unregistered (by calling
-  // RemoveTargetTransferRateObserver).
-  // A newly registered observer will be called back with the latest recorded
-  // target rate, if available.
-  virtual void AddTargetTransferRateObserver(
-      TargetTransferRateObserver* observer) = 0;
-
-  // Removes an existing |observer| from observers. If observer was never
-  // registered, an error is logged and method does nothing.
-  virtual void RemoveTargetTransferRateObserver(
-      TargetTransferRateObserver* observer) = 0;
-
-  // Returns the last known target transfer rate as reported to the above
-  // observers.
-  virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate() = 0;
-};
-
-}  // namespace webrtc
-
-#endif  // API_TRANSPORT_CONGESTION_CONTROL_INTERFACE_H_
diff --git a/chromium/third_party/webrtc/api/transport/data_channel_transport_interface.h b/chromium/third_party/webrtc/api/transport/data_channel_transport_interface.h
index 671deffc6ee..7b8c653c390 100644
--- a/chromium/third_party/webrtc/api/transport/data_channel_transport_interface.h
+++ b/chromium/third_party/webrtc/api/transport/data_channel_transport_interface.h
@@ -35,8 +35,8 @@ enum class DataMessageType {
 // sent reliably and in-order, even if the data channel is configured for
 // unreliable delivery.
 struct SendDataParams {
-  SendDataParams();
-  SendDataParams(const SendDataParams&);
+  SendDataParams() = default;
+  SendDataParams(const SendDataParams&) = default;
 
   DataMessageType type = DataMessageType::kText;
 
diff --git a/chromium/third_party/webrtc/api/transport/datagram_transport_interface.h b/chromium/third_party/webrtc/api/transport/datagram_transport_interface.h
deleted file mode 100644
index 01736b978d0..00000000000
--- a/chromium/third_party/webrtc/api/transport/datagram_transport_interface.h
+++ /dev/null
@@ -1,151 +0,0 @@
-/* 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.
- */
-
-// This is EXPERIMENTAL interface for media and datagram transports.
-
-#ifndef API_TRANSPORT_DATAGRAM_TRANSPORT_INTERFACE_H_
-#define API_TRANSPORT_DATAGRAM_TRANSPORT_INTERFACE_H_
-
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "absl/types/optional.h"
-#include "api/array_view.h"
-#include "api/rtc_error.h"
-#include "api/transport/congestion_control_interface.h"
-#include "api/transport/data_channel_transport_interface.h"
-#include "api/units/data_rate.h"
-#include "api/units/timestamp.h"
-
-namespace rtc {
-class PacketTransportInternal;
-}  // namespace rtc
-
-namespace webrtc {
-
-class MediaTransportStateCallback;
-
-typedef int64_t DatagramId;
-
-struct DatagramAck {
-  // |datagram_id| is same as passed in
-  // DatagramTransportInterface::SendDatagram.
-  DatagramId datagram_id;
-
-  // The timestamp at which the remote peer received the identified datagram,
-  // according to that peer's clock.
-  Timestamp receive_timestamp = Timestamp::MinusInfinity();
-};
-
-// All sink methods are called on network thread.
-class DatagramSinkInterface {
- public:
-  virtual ~DatagramSinkInterface() {}
-
-  // Called when new packet is received.
-  virtual void OnDatagramReceived(rtc::ArrayView<const uint8_t> data) = 0;
-
-  // Called when datagram is actually sent (datragram can be delayed due
-  // to congestion control or fusing). |datagram_id| is same as passed in
-  // DatagramTransportInterface::SendDatagram.
-  virtual void OnDatagramSent(DatagramId datagram_id) = 0;
-
-  // Called when datagram is ACKed.
-  virtual void OnDatagramAcked(const DatagramAck& datagram_ack) = 0;
-
-  // Called when a datagram is lost.
-  virtual void OnDatagramLost(DatagramId datagram_id) = 0;
-};
-
-// Datagram transport allows to send and receive unreliable packets (datagrams)
-// and receive feedback from congestion control (via
-// CongestionControlInterface). The idea is to send RTP packets as datagrams and
-// have underlying implementation of datagram transport to use QUIC datagram
-// protocol.
-class DatagramTransportInterface : public DataChannelTransportInterface {
- public:
-  virtual ~DatagramTransportInterface() = default;
-
-  // Connect the datagram transport to the ICE transport.
-  // The implementation must be able to ignore incoming packets that don't
-  // belong to it.
-  virtual void Connect(rtc::PacketTransportInternal* packet_transport) = 0;
-
-  // Returns congestion control feedback interface or nullptr if datagram
-  // transport does not implement congestion control.
-  //
-  // Note that right now datagram transport is used without congestion control,
-  // but we plan to use it in the future.
-  virtual CongestionControlInterface* congestion_control() = 0;
-
-  // Sets a state observer callback. Before datagram transport is destroyed, the
-  // callback must be unregistered by setting it to nullptr.
-  // A newly registered callback will be called with the current state.
-  // Datagram transport does not invoke this callback concurrently.
-  virtual void SetTransportStateCallback(
-      MediaTransportStateCallback* callback) = 0;
-
-  // Start asynchronous send of datagram. The status returned by this method
-  // only pertains to the synchronous operations (e.g. serialization /
-  // packetization), not to the asynchronous operation.
-  //
-  // Datagrams larger than GetLargestDatagramSize() will fail and return error.
-  //
-  // Datagrams are sent in FIFO order.
-  //
-  // |datagram_id| is only used in ACK/LOST notifications in
-  // DatagramSinkInterface and does not need to be unique.
-  virtual RTCError SendDatagram(rtc::ArrayView<const uint8_t> data,
-                                DatagramId datagram_id) = 0;
-
-  // Returns maximum size of datagram message, does not change.
-  // TODO(sukhanov): Because value may be undefined before connection setup
-  // is complete, consider returning error when called before connection is
-  // established. Currently returns hardcoded const, because integration
-  // prototype may call before connection is established.
-  virtual size_t GetLargestDatagramSize() const = 0;
-
-  // Sets packet sink. Sink must be unset by calling
-  // SetDataTransportSink(nullptr) before the data transport is destroyed or
-  // before new sink is set.
-  virtual void SetDatagramSink(DatagramSinkInterface* sink) = 0;
-
-  // Retrieves transport parameters for this datagram transport.  May be called
-  // on either client- or server-perspective transports.
-  //
-  // For servers, the parameters represent what kind of connections and data the
-  // server is prepared to accept.  This is generally a superset of acceptable
-  // parameters.
-  //
-  // For clients, the parameters echo the server configuration used to create
-  // the client, possibly removing any fields or parameters which the client
-  // does not understand.
-  virtual std::string GetTransportParameters() const = 0;
-
-  // Sets remote transport parameters.  |remote_params| is a serialized string
-  // of opaque parameters, understood by the datagram transport implementation.
-  // Returns an error if |remote_params| are not compatible with this transport.
-  //
-  // TODO(mellem): Make pure virtual.  The default implementation maintains
-  // original negotiation behavior (negotiation falls back to RTP if the
-  // remote datagram transport fails to echo exactly the local parameters).
-  virtual RTCError SetRemoteTransportParameters(
-      absl::string_view remote_params) {
-    if (remote_params == GetTransportParameters()) {
-      return RTCError::OK();
-    }
-    return RTCError(RTCErrorType::UNSUPPORTED_PARAMETER,
-                    "Local and remote transport parameters do not match");
-  }
-};
-
-}  // namespace webrtc
-
-#endif  // API_TRANSPORT_DATAGRAM_TRANSPORT_INTERFACE_H_
diff --git a/chromium/third_party/webrtc/api/transport/media/BUILD.gn b/chromium/third_party/webrtc/api/transport/media/BUILD.gn
deleted file mode 100644
index 24a364c2e5d..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/BUILD.gn
+++ /dev/null
@@ -1,52 +0,0 @@
-# Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-import("../../../webrtc.gni")
-
-rtc_library("media_transport_interface") {
-  visibility = [ "*" ]
-  sources = [
-    "media_transport_config.cc",
-    "media_transport_config.h",
-    "media_transport_interface.cc",
-    "media_transport_interface.h",
-  ]
-  deps = [
-    ":audio_interfaces",
-    ":video_interfaces",
-    "..:datagram_transport_interface",
-    "..:network_control",
-    "../..:array_view",
-    "../..:rtc_error",
-    "../../..:webrtc_common",
-    "../../../rtc_base",
-    "../../../rtc_base:checks",
-    "../../../rtc_base:rtc_base_approved",
-    "../../../rtc_base:stringutils",
-    "../../units:data_rate",
-    "//third_party/abseil-cpp/absl/types:optional",
-  ]
-}
-
-rtc_library("audio_interfaces") {
-  visibility = [ "*" ]
-  sources = [
-    "audio_transport.cc",
-    "audio_transport.h",
-  ]
-  deps = [ "../..:array_view" ]
-}
-
-rtc_library("video_interfaces") {
-  visibility = [ "*" ]
-  sources = [
-    "video_transport.cc",
-    "video_transport.h",
-  ]
-  deps = [ "../../video:encoded_image" ]
-}
diff --git a/chromium/third_party/webrtc/api/transport/media/audio_transport.cc b/chromium/third_party/webrtc/api/transport/media/audio_transport.cc
deleted file mode 100644
index 0f5fe8bcf2e..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/audio_transport.cc
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This is EXPERIMENTAL interface for media transport.
-//
-// The goal is to refactor WebRTC code so that audio and video frames
-// are sent / received through the media transport interface. This will
-// enable different media transport implementations, including QUIC-based
-// media transport.
-
-#include "api/transport/media/audio_transport.h"
-
-#include <utility>
-
-namespace webrtc {
-
-MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}
-
-MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
-    int sampling_rate_hz,
-    int starting_sample_index,
-    int samples_per_channel,
-    int sequence_number,
-    FrameType frame_type,
-    int payload_type,
-    std::vector<uint8_t> encoded_data)
-    : sampling_rate_hz_(sampling_rate_hz),
-      starting_sample_index_(starting_sample_index),
-      samples_per_channel_(samples_per_channel),
-      sequence_number_(sequence_number),
-      frame_type_(frame_type),
-      payload_type_(payload_type),
-      encoded_data_(std::move(encoded_data)) {}
-
-MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
-    const MediaTransportEncodedAudioFrame&) = default;
-
-MediaTransportEncodedAudioFrame& MediaTransportEncodedAudioFrame::operator=(
-    MediaTransportEncodedAudioFrame&&) = default;
-
-MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
-    const MediaTransportEncodedAudioFrame&) = default;
-
-MediaTransportEncodedAudioFrame::MediaTransportEncodedAudioFrame(
-    MediaTransportEncodedAudioFrame&&) = default;
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/transport/media/audio_transport.h b/chromium/third_party/webrtc/api/transport/media/audio_transport.h
deleted file mode 100644
index dcbdcd7afe2..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/audio_transport.h
+++ /dev/null
@@ -1,120 +0,0 @@
-/* Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This is EXPERIMENTAL interface for media transport.
-//
-// The goal is to refactor WebRTC code so that audio and video frames
-// are sent / received through the media transport interface. This will
-// enable different media transport implementations, including QUIC-based
-// media transport.
-
-#ifndef API_TRANSPORT_MEDIA_AUDIO_TRANSPORT_H_
-#define API_TRANSPORT_MEDIA_AUDIO_TRANSPORT_H_
-
-#include <vector>
-
-#include "api/array_view.h"
-
-namespace webrtc {
-
-// Represents encoded audio frame in any encoding (type of encoding is opaque).
-// To avoid copying of encoded data use move semantics when passing by value.
-class MediaTransportEncodedAudioFrame final {
- public:
-  enum class FrameType {
-    // Normal audio frame (equivalent to webrtc::kAudioFrameSpeech).
-    kSpeech,
-
-    // DTX frame (equivalent to webrtc::kAudioFrameCN).
-    kDiscontinuousTransmission,
-    // TODO(nisse): Mis-spelled version, update users, then delete.
-    kDiscountinuousTransmission = kDiscontinuousTransmission,
-  };
-
-  MediaTransportEncodedAudioFrame(
-      // Audio sampling rate, for example 48000.
-      int sampling_rate_hz,
-
-      // Starting sample index of the frame, i.e. how many audio samples were
-      // before this frame since the beginning of the call or beginning of time
-      // in one channel (the starting point should not matter for NetEq). In
-      // WebRTC it is used as a timestamp of the frame.
-      // TODO(sukhanov): Starting_sample_index is currently adjusted on the
-      // receiver side in RTP path. Non-RTP implementations should preserve it.
-      // For NetEq initial offset should not matter so we should consider fixing
-      // RTP path.
-      int starting_sample_index,
-
-      // Number of audio samples in audio frame in 1 channel.
-      int samples_per_channel,
-
-      // Sequence number of the frame in the order sent, it is currently
-      // required by NetEq, but we can fix NetEq, because starting_sample_index
-      // should be enough.
-      int sequence_number,
-
-      // If audio frame is a speech or discontinued transmission.
-      FrameType frame_type,
-
-      // Opaque payload type. In RTP codepath payload type is stored in RTP
-      // header. In other implementations it should be simply passed through the
-      // wire -- it's needed for decoder.
-      int payload_type,
-
-      // Vector with opaque encoded data.
-      std::vector<uint8_t> encoded_data);
-
-  ~MediaTransportEncodedAudioFrame();
-  MediaTransportEncodedAudioFrame(const MediaTransportEncodedAudioFrame&);
-  MediaTransportEncodedAudioFrame& operator=(
-      const MediaTransportEncodedAudioFrame& other);
-  MediaTransportEncodedAudioFrame& operator=(
-      MediaTransportEncodedAudioFrame&& other);
-  MediaTransportEncodedAudioFrame(MediaTransportEncodedAudioFrame&&);
-
-  // Getters.
-  int sampling_rate_hz() const { return sampling_rate_hz_; }
-  int starting_sample_index() const { return starting_sample_index_; }
-  int samples_per_channel() const { return samples_per_channel_; }
-  int sequence_number() const { return sequence_number_; }
-
-  int payload_type() const { return payload_type_; }
-  FrameType frame_type() const { return frame_type_; }
-
-  rtc::ArrayView<const uint8_t> encoded_data() const { return encoded_data_; }
-
- private:
-  int sampling_rate_hz_;
-  int starting_sample_index_;
-  int samples_per_channel_;
-
-  // TODO(sukhanov): Refactor NetEq so we don't need sequence number.
-  // Having sample_index and samples_per_channel should be enough.
-  int sequence_number_;
-
-  FrameType frame_type_;
-
-  int payload_type_;
-
-  std::vector<uint8_t> encoded_data_;
-};
-
-// Interface for receiving encoded audio frames from MediaTransportInterface
-// implementations.
-class MediaTransportAudioSinkInterface {
- public:
-  virtual ~MediaTransportAudioSinkInterface() = default;
-
-  // Called when new encoded audio frame is received.
-  virtual void OnData(uint64_t channel_id,
-                      MediaTransportEncodedAudioFrame frame) = 0;
-};
-
-}  // namespace webrtc
-#endif  // API_TRANSPORT_MEDIA_AUDIO_TRANSPORT_H_
diff --git a/chromium/third_party/webrtc/api/transport/media/media_transport_config.cc b/chromium/third_party/webrtc/api/transport/media/media_transport_config.cc
deleted file mode 100644
index b9b19cb6f0f..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/media_transport_config.cc
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- *  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 "api/transport/media/media_transport_config.h"
-
-#include "rtc_base/checks.h"
-#include "rtc_base/strings/string_builder.h"
-
-namespace webrtc {
-
-MediaTransportConfig::MediaTransportConfig(size_t rtp_max_packet_size)
-    : rtp_max_packet_size(rtp_max_packet_size) {
-  RTC_DCHECK_GT(rtp_max_packet_size, 0);
-}
-
-std::string MediaTransportConfig::DebugString() const {
-  rtc::StringBuilder result;
-  result << "{rtp_max_packet_size: " << rtp_max_packet_size.value_or(0) << "}";
-  return result.Release();
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/transport/media/media_transport_config.h b/chromium/third_party/webrtc/api/transport/media/media_transport_config.h
deleted file mode 100644
index 7ef65453ae2..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/media_transport_config.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/* 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 API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
-#define API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
-
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "absl/types/optional.h"
-
-namespace webrtc {
-
-// Media transport config is made available to both transport and audio / video
-// layers, but access to individual interfaces should not be open without
-// necessity.
-struct MediaTransportConfig {
-  // Default constructor for no-media transport scenarios.
-  MediaTransportConfig() = default;
-
-  // Constructor for datagram transport scenarios.
-  explicit MediaTransportConfig(size_t rtp_max_packet_size);
-
-  std::string DebugString() const;
-
-  // If provided, limits RTP packet size (excludes ICE, IP or network overhead).
-  absl::optional<size_t> rtp_max_packet_size;
-};
-
-}  // namespace webrtc
-
-#endif  // API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_CONFIG_H_
diff --git a/chromium/third_party/webrtc/api/transport/media/media_transport_interface.cc b/chromium/third_party/webrtc/api/transport/media/media_transport_interface.cc
deleted file mode 100644
index 323ddca6895..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/media_transport_interface.cc
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- *  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.
- */
-
-// This is EXPERIMENTAL interface for media transport.
-//
-// The goal is to refactor WebRTC code so that audio and video frames
-// are sent / received through the media transport interface. This will
-// enable different media transport implementations, including QUIC-based
-// media transport.
-
-#include "api/transport/media/media_transport_interface.h"
-
-#include <cstdint>
-#include <utility>
-
-#include "api/transport/datagram_transport_interface.h"
-
-namespace webrtc {
-
-MediaTransportSettings::MediaTransportSettings() = default;
-MediaTransportSettings::MediaTransportSettings(const MediaTransportSettings&) =
-    default;
-MediaTransportSettings& MediaTransportSettings::operator=(
-    const MediaTransportSettings&) = default;
-MediaTransportSettings::~MediaTransportSettings() = default;
-
-SendDataParams::SendDataParams() = default;
-SendDataParams::SendDataParams(const SendDataParams&) = default;
-
-RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
-MediaTransportFactory::CreateMediaTransport(
-    rtc::PacketTransportInternal* packet_transport,
-    rtc::Thread* network_thread,
-    const MediaTransportSettings& settings) {
-  return std::unique_ptr<MediaTransportInterface>(nullptr);
-}
-
-RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
-MediaTransportFactory::CreateMediaTransport(
-    rtc::Thread* network_thread,
-    const MediaTransportSettings& settings) {
-  return std::unique_ptr<MediaTransportInterface>(nullptr);
-}
-
-RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
-MediaTransportFactory::CreateDatagramTransport(
-    rtc::Thread* network_thread,
-    const MediaTransportSettings& settings) {
-  return std::unique_ptr<DatagramTransportInterface>(nullptr);
-}
-
-std::string MediaTransportFactory::GetTransportName() const {
-  return "";
-}
-
-MediaTransportInterface::MediaTransportInterface() = default;
-MediaTransportInterface::~MediaTransportInterface() = default;
-
-absl::optional<std::string>
-MediaTransportInterface::GetTransportParametersOffer() const {
-  return absl::nullopt;
-}
-
-void MediaTransportInterface::Connect(
-    rtc::PacketTransportInternal* packet_transport) {}
-
-void MediaTransportInterface::SetKeyFrameRequestCallback(
-    MediaTransportKeyFrameRequestCallback* callback) {}
-
-absl::optional<TargetTransferRate>
-MediaTransportInterface::GetLatestTargetTransferRate() {
-  return absl::nullopt;
-}
-
-void MediaTransportInterface::AddNetworkChangeCallback(
-    MediaTransportNetworkChangeCallback* callback) {}
-
-void MediaTransportInterface::RemoveNetworkChangeCallback(
-    MediaTransportNetworkChangeCallback* callback) {}
-
-void MediaTransportInterface::SetFirstAudioPacketReceivedObserver(
-    AudioPacketReceivedObserver* observer) {}
-
-void MediaTransportInterface::AddTargetTransferRateObserver(
-    TargetTransferRateObserver* observer) {}
-void MediaTransportInterface::RemoveTargetTransferRateObserver(
-    TargetTransferRateObserver* observer) {}
-
-void MediaTransportInterface::AddRttObserver(
-    MediaTransportRttObserver* observer) {}
-void MediaTransportInterface::RemoveRttObserver(
-    MediaTransportRttObserver* observer) {}
-
-size_t MediaTransportInterface::GetAudioPacketOverhead() const {
-  return 0;
-}
-
-void MediaTransportInterface::SetAllocatedBitrateLimits(
-    const MediaTransportAllocatedBitrateLimits& limits) {}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/transport/media/media_transport_interface.h b/chromium/third_party/webrtc/api/transport/media/media_transport_interface.h
deleted file mode 100644
index dbe68d344b6..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/media_transport_interface.h
+++ /dev/null
@@ -1,320 +0,0 @@
-/* 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.
- */
-
-// This is EXPERIMENTAL interface for media transport.
-//
-// The goal is to refactor WebRTC code so that audio and video frames
-// are sent / received through the media transport interface. This will
-// enable different media transport implementations, including QUIC-based
-// media transport.
-
-#ifndef API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_INTERFACE_H_
-#define API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_INTERFACE_H_
-
-#include <memory>
-#include <string>
-#include <utility>
-
-#include "absl/types/optional.h"
-#include "api/array_view.h"
-#include "api/rtc_error.h"
-#include "api/transport/data_channel_transport_interface.h"
-#include "api/transport/media/audio_transport.h"
-#include "api/transport/media/video_transport.h"
-#include "api/transport/network_control.h"
-#include "api/units/data_rate.h"
-#include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/network_route.h"
-
-namespace rtc {
-class PacketTransportInternal;
-class Thread;
-}  // namespace rtc
-
-namespace webrtc {
-
-class DatagramTransportInterface;
-class RtcEventLog;
-
-class AudioPacketReceivedObserver {
- public:
-  virtual ~AudioPacketReceivedObserver() = default;
-
-  // Invoked for the first received audio packet on a given channel id.
-  // It will be invoked once for each channel id.
-  virtual void OnFirstAudioPacketReceived(int64_t channel_id) = 0;
-};
-
-// Used to configure stream allocations.
-struct MediaTransportAllocatedBitrateLimits {
-  DataRate min_pacing_rate = DataRate::Zero();
-  DataRate max_padding_bitrate = DataRate::Zero();
-  DataRate max_total_allocated_bitrate = DataRate::Zero();
-};
-
-// Used to configure target bitrate constraints.
-// If the value is provided, the constraint is updated.
-// If the value is omitted, the value is left unchanged.
-struct MediaTransportTargetRateConstraints {
-  absl::optional<DataRate> min_bitrate;
-  absl::optional<DataRate> max_bitrate;
-  absl::optional<DataRate> starting_bitrate;
-};
-
-// A collection of settings for creation of media transport.
-struct MediaTransportSettings final {
-  MediaTransportSettings();
-  MediaTransportSettings(const MediaTransportSettings&);
-  MediaTransportSettings& operator=(const MediaTransportSettings&);
-  ~MediaTransportSettings();
-
-  // Group calls are not currently supported, in 1:1 call one side must set
-  // is_caller = true and another is_caller = false.
-  bool is_caller;
-
-  // Must be set if a pre-shared key is used for the call.
-  // TODO(bugs.webrtc.org/9944): This should become zero buffer in the distant
-  // future.
-  absl::optional<std::string> pre_shared_key;
-
-  // If present, this is a config passed from the caller to the answerer in the
-  // offer. Each media transport knows how to understand its own parameters.
-  absl::optional<std::string> remote_transport_parameters;
-
-  // If present, provides the event log that media transport should use.
-  // Media transport does not own it. The lifetime of |event_log| will exceed
-  // the lifetime of the instance of MediaTransportInterface instance.
-  RtcEventLog* event_log = nullptr;
-};
-
-// Callback to notify about network route changes.
-class MediaTransportNetworkChangeCallback {
- public:
-  virtual ~MediaTransportNetworkChangeCallback() = default;
-
-  // Called when the network route is changed, with the new network route.
-  virtual void OnNetworkRouteChanged(
-      const rtc::NetworkRoute& new_network_route) = 0;
-};
-
-// State of the media transport.  Media transport begins in the pending state.
-// It transitions to writable when it is ready to send media.  It may transition
-// back to pending if the connection is blocked.  It may transition to closed at
-// any time.  Closed is terminal: a transport will never re-open once closed.
-enum class MediaTransportState {
-  kPending,
-  kWritable,
-  kClosed,
-};
-
-// Callback invoked whenever the state of the media transport changes.
-class MediaTransportStateCallback {
- public:
-  virtual ~MediaTransportStateCallback() = default;
-
-  // Invoked whenever the state of the media transport changes.
-  virtual void OnStateChanged(MediaTransportState state) = 0;
-};
-
-// Callback for RTT measurements on the receive side.
-// TODO(nisse): Related interfaces: CallStatsObserver and RtcpRttStats. It's
-// somewhat unclear what type of measurement is needed. It's used to configure
-// NACK generation and playout buffer. Either raw measurement values or recent
-// maximum would make sense for this use. Need consolidation of RTT signalling.
-class MediaTransportRttObserver {
- public:
-  virtual ~MediaTransportRttObserver() = default;
-
-  // Invoked when a new RTT measurement is available, typically once per ACK.
-  virtual void OnRttUpdated(int64_t rtt_ms) = 0;
-};
-
-// Media transport interface for sending / receiving encoded audio/video frames
-// and receiving bandwidth estimate update from congestion control.
-class MediaTransportInterface : public DataChannelTransportInterface {
- public:
-  MediaTransportInterface();
-  virtual ~MediaTransportInterface();
-
-  // Retrieves callers config (i.e. media transport offer) that should be passed
-  // to the callee, before the call is connected. Such config is opaque to SDP
-  // (sdp just passes it through). The config is a binary blob, so SDP may
-  // choose to use base64 to serialize it (or any other approach that guarantees
-  // that the binary blob goes through). This should only be called for the
-  // caller's perspective.
-  //
-  // This may return an unset optional, which means that the given media
-  // transport is not supported / disabled and shouldn't be reported in SDP.
-  //
-  // It may also return an empty string, in which case the media transport is
-  // supported, but without any extra settings.
-  // TODO(psla): Make abstract.
-  virtual absl::optional<std::string> GetTransportParametersOffer() const;
-
-  // Connect the media transport to the ICE transport.
-  // The implementation must be able to ignore incoming packets that don't
-  // belong to it.
-  // TODO(psla): Make abstract.
-  virtual void Connect(rtc::PacketTransportInternal* packet_transport);
-
-  // Start asynchronous send of audio frame. The status returned by this method
-  // only pertains to the synchronous operations (e.g.
-  // serialization/packetization), not to the asynchronous operation.
-
-  virtual RTCError SendAudioFrame(uint64_t channel_id,
-                                  MediaTransportEncodedAudioFrame frame) = 0;
-
-  // Start asynchronous send of video frame. The status returned by this method
-  // only pertains to the synchronous operations (e.g.
-  // serialization/packetization), not to the asynchronous operation.
-  virtual RTCError SendVideoFrame(
-      uint64_t channel_id,
-      const MediaTransportEncodedVideoFrame& frame) = 0;
-
-  // Used by video sender to be notified on key frame requests.
-  virtual void SetKeyFrameRequestCallback(
-      MediaTransportKeyFrameRequestCallback* callback);
-
-  // Requests a keyframe for the particular channel (stream). The caller should
-  // check that the keyframe is not present in a jitter buffer already (i.e.
-  // don't request a keyframe if there is one that you will get from the jitter
-  // buffer in a moment).
-  virtual RTCError RequestKeyFrame(uint64_t channel_id) = 0;
-
-  // Sets audio sink. Sink must be unset by calling SetReceiveAudioSink(nullptr)
-  // before the media transport is destroyed or before new sink is set.
-  virtual void SetReceiveAudioSink(MediaTransportAudioSinkInterface* sink) = 0;
-
-  // Registers a video sink. Before destruction of media transport, you must
-  // pass a nullptr.
-  virtual void SetReceiveVideoSink(MediaTransportVideoSinkInterface* sink) = 0;
-
-  // Adds a target bitrate observer. Before media transport is destructed
-  // the observer must be unregistered (by calling
-  // RemoveTargetTransferRateObserver).
-  // A newly registered observer will be called back with the latest recorded
-  // target rate, if available.
-  virtual void AddTargetTransferRateObserver(
-      TargetTransferRateObserver* observer);
-
-  // Removes an existing |observer| from observers. If observer was never
-  // registered, an error is logged and method does nothing.
-  virtual void RemoveTargetTransferRateObserver(
-      TargetTransferRateObserver* observer);
-
-  // Sets audio packets observer, which gets informed about incoming audio
-  // packets. Before destruction, the observer must be unregistered by setting
-  // nullptr.
-  //
-  // This method may be temporary, when the multiplexer is implemented (or
-  // multiplexer may use it to demultiplex channel ids).
-  virtual void SetFirstAudioPacketReceivedObserver(
-      AudioPacketReceivedObserver* observer);
-
-  // Intended for receive side. AddRttObserver registers an observer to be
-  // called for each RTT measurement, typically once per ACK. Before media
-  // transport is destructed the observer must be unregistered.
-  virtual void AddRttObserver(MediaTransportRttObserver* observer);
-  virtual void RemoveRttObserver(MediaTransportRttObserver* observer);
-
-  // Returns the last known target transfer rate as reported to the above
-  // observers.
-  virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate();
-
-  // Gets the audio packet overhead in bytes. Returned overhead does not include
-  // transport overhead (ipv4/6, turn channeldata, tcp/udp, etc.).
-  // If the transport is capable of fusing packets together, this overhead
-  // might not be a very accurate number.
-  // TODO(nisse): Deprecated.
-  virtual size_t GetAudioPacketOverhead() const;
-
-  // Corresponding observers for audio and video overhead. Before destruction,
-  // the observers must be unregistered by setting nullptr.
-
-  // Registers an observer for network change events. If the network route is
-  // already established when the callback is added, |callback| will be called
-  // immediately with the current network route. Before media transport is
-  // destroyed, the callback must be removed.
-  virtual void AddNetworkChangeCallback(
-      MediaTransportNetworkChangeCallback* callback);
-  virtual void RemoveNetworkChangeCallback(
-      MediaTransportNetworkChangeCallback* callback);
-
-  // Sets a state observer callback. Before media transport is destroyed, the
-  // callback must be unregistered by setting it to nullptr.
-  // A newly registered callback will be called with the current state.
-  // Media transport does not invoke this callback concurrently.
-  virtual void SetMediaTransportStateCallback(
-      MediaTransportStateCallback* callback) = 0;
-
-  // Updates allocation limits.
-  // TODO(psla): Make abstract when downstream implementation implement it.
-  virtual void SetAllocatedBitrateLimits(
-      const MediaTransportAllocatedBitrateLimits& limits);
-
-  // Sets starting rate.
-  // TODO(psla): Make abstract when downstream implementation implement it.
-  virtual void SetTargetBitrateLimits(
-      const MediaTransportTargetRateConstraints& target_rate_constraints) {}
-
-  // TODO(sukhanov): RtcEventLogs.
-};
-
-// If media transport factory is set in peer connection factory, it will be
-// used to create media transport for sending/receiving encoded frames and
-// this transport will be used instead of default RTP/SRTP transport.
-//
-// Currently Media Transport negotiation is not supported in SDP.
-// If application is using media transport, it must negotiate it before
-// setting media transport factory in peer connection.
-class MediaTransportFactory {
- public:
-  virtual ~MediaTransportFactory() = default;
-
-  // Creates media transport.
-  // - Does not take ownership of packet_transport or network_thread.
-  // - Does not support group calls, in 1:1 call one side must set
-  //   is_caller = true and another is_caller = false.
-  virtual RTCErrorOr<std::unique_ptr<MediaTransportInterface>>
-  CreateMediaTransport(rtc::PacketTransportInternal* packet_transport,
-                       rtc::Thread* network_thread,
-                       const MediaTransportSettings& settings);
-
-  // Creates a new Media Transport in a disconnected state. If the media
-  // transport for the caller is created, one can then call
-  // MediaTransportInterface::GetTransportParametersOffer on that new instance.
-  // TODO(psla): Make abstract.
-  virtual RTCErrorOr<std::unique_ptr<webrtc::MediaTransportInterface>>
-  CreateMediaTransport(rtc::Thread* network_thread,
-                       const MediaTransportSettings& settings);
-
-  // Creates a new Datagram Transport in a disconnected state. If the datagram
-  // transport for the caller is created, one can then call
-  // DatagramTransportInterface::GetTransportParametersOffer on that new
-  // instance.
-  //
-  // TODO(sukhanov): Consider separating media and datagram transport factories.
-  // TODO(sukhanov): Move factory to a separate .h file.
-  virtual RTCErrorOr<std::unique_ptr<DatagramTransportInterface>>
-  CreateDatagramTransport(rtc::Thread* network_thread,
-                          const MediaTransportSettings& settings);
-
-  // Gets a transport name which is supported by the implementation.
-  // Different factories should return different transport names, and at runtime
-  // it will be checked that different names were used.
-  // For example, "rtp" or "generic" may be returned by two different
-  // implementations.
-  // The value returned by this method must never change in the lifetime of the
-  // factory.
-  // TODO(psla): Make abstract.
-  virtual std::string GetTransportName() const;
-};
-
-}  // namespace webrtc
-#endif  // API_TRANSPORT_MEDIA_MEDIA_TRANSPORT_INTERFACE_H_
diff --git a/chromium/third_party/webrtc/api/transport/media/video_transport.cc b/chromium/third_party/webrtc/api/transport/media/video_transport.cc
deleted file mode 100644
index a6f53040487..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/video_transport.cc
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This is EXPERIMENTAL interface for media transport.
-//
-// The goal is to refactor WebRTC code so that audio and video frames
-// are sent / received through the media transport interface. This will
-// enable different media transport implementations, including QUIC-based
-// media transport.
-
-#include "api/transport/media/video_transport.h"
-
-#include <utility>
-
-namespace webrtc {
-
-MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame() = default;
-
-MediaTransportEncodedVideoFrame::~MediaTransportEncodedVideoFrame() = default;
-
-MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
-    int64_t frame_id,
-    std::vector<int64_t> referenced_frame_ids,
-    int payload_type,
-    const webrtc::EncodedImage& encoded_image)
-    : payload_type_(payload_type),
-      encoded_image_(encoded_image),
-      frame_id_(frame_id),
-      referenced_frame_ids_(std::move(referenced_frame_ids)) {}
-
-MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
-    const MediaTransportEncodedVideoFrame&) = default;
-
-MediaTransportEncodedVideoFrame& MediaTransportEncodedVideoFrame::operator=(
-    MediaTransportEncodedVideoFrame&&) = default;
-
-MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
-    const MediaTransportEncodedVideoFrame& o)
-    : MediaTransportEncodedVideoFrame() {
-  *this = o;
-}
-
-MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame(
-    MediaTransportEncodedVideoFrame&& o)
-    : MediaTransportEncodedVideoFrame() {
-  *this = std::move(o);
-}
-
-}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/transport/media/video_transport.h b/chromium/third_party/webrtc/api/transport/media/video_transport.h
deleted file mode 100644
index affd2e0d382..00000000000
--- a/chromium/third_party/webrtc/api/transport/media/video_transport.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/* Copyright 2019 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This is EXPERIMENTAL interface for media transport.
-//
-// The goal is to refactor WebRTC code so that audio and video frames
-// are sent / received through the media transport interface. This will
-// enable different media transport implementations, including QUIC-based
-// media transport.
-
-#ifndef API_TRANSPORT_MEDIA_VIDEO_TRANSPORT_H_
-#define API_TRANSPORT_MEDIA_VIDEO_TRANSPORT_H_
-
-#include <vector>
-
-#include "api/video/encoded_image.h"
-
-namespace webrtc {
-
-// Represents encoded video frame, along with the codec information.
-class MediaTransportEncodedVideoFrame final {
- public:
-  MediaTransportEncodedVideoFrame(int64_t frame_id,
-                                  std::vector<int64_t> referenced_frame_ids,
-                                  int payload_type,
-                                  const webrtc::EncodedImage& encoded_image);
-  ~MediaTransportEncodedVideoFrame();
-  MediaTransportEncodedVideoFrame(const MediaTransportEncodedVideoFrame&);
-  MediaTransportEncodedVideoFrame& operator=(
-      const MediaTransportEncodedVideoFrame& other);
-  MediaTransportEncodedVideoFrame& operator=(
-      MediaTransportEncodedVideoFrame&& other);
-  MediaTransportEncodedVideoFrame(MediaTransportEncodedVideoFrame&&);
-
-  int payload_type() const { return payload_type_; }
-  const webrtc::EncodedImage& encoded_image() const { return encoded_image_; }
-
-  int64_t frame_id() const { return frame_id_; }
-  const std::vector<int64_t>& referenced_frame_ids() const {
-    return referenced_frame_ids_;
-  }
-
-  // Hack to workaround lack of ownership of the EncodedImage buffer. If we
-  // don't already own the underlying data, make a copy.
-  void Retain() { encoded_image_.Retain(); }
-
- private:
-  MediaTransportEncodedVideoFrame();
-
-  int payload_type_;
-
-  // The buffer is not always owned by the encoded image. On the sender it means
-  // that it will need to make a copy using the Retain() method, if it wants to
-  // deliver it asynchronously.
-  webrtc::EncodedImage encoded_image_;
-
-  // Frame id uniquely identifies a frame in a stream. It needs to be unique in
-  // a given time window (i.e. technically unique identifier for the lifetime of
-  // the connection is not needed, but you need to guarantee that remote side
-  // got rid of the previous frame_id if you plan to reuse it).
-  //
-  // It is required by a remote jitter buffer, and is the same as
-  // EncodedFrame::id::picture_id.
-  //
-  // This data must be opaque to the media transport, and media transport should
-  // itself not make any assumptions about what it is and its uniqueness.
-  int64_t frame_id_;
-
-  // A single frame might depend on other frames. This is set of identifiers on
-  // which the current frame depends.
-  std::vector<int64_t> referenced_frame_ids_;
-};
-
-// Interface for receiving encoded video frames from MediaTransportInterface
-// implementations.
-class MediaTransportVideoSinkInterface {
- public:
-  virtual ~MediaTransportVideoSinkInterface() = default;
-
-  // Called when new encoded video frame is received.
-  virtual void OnData(uint64_t channel_id,
-                      MediaTransportEncodedVideoFrame frame) = 0;
-};
-
-// Interface for video sender to be notified of received key frame request.
-class MediaTransportKeyFrameRequestCallback {
- public:
-  virtual ~MediaTransportKeyFrameRequestCallback() = default;
-
-  // Called when a key frame request is received on the transport.
-  virtual void OnKeyFrameRequested(uint64_t channel_id) = 0;
-};
-
-}  // namespace webrtc
-#endif  // API_TRANSPORT_MEDIA_VIDEO_TRANSPORT_H_
diff --git a/chromium/third_party/webrtc/api/transport/rtp/BUILD.gn b/chromium/third_party/webrtc/api/transport/rtp/BUILD.gn
index b0849502c81..7b01169360c 100644
--- a/chromium/third_party/webrtc/api/transport/rtp/BUILD.gn
+++ b/chromium/third_party/webrtc/api/transport/rtp/BUILD.gn
@@ -14,15 +14,20 @@ rtc_source_set("rtp_source") {
   deps = [
     "../../../api:rtp_headers",
     "../../../rtc_base:checks",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("dependency_descriptor") {
   visibility = [ "*" ]
-  sources = [ "dependency_descriptor.h" ]
-  deps = [
+  sources = [
+    "dependency_descriptor.cc",
+    "dependency_descriptor.h",
+  ]
+  deps = [ "../../../rtc_base:checks" ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/container:inlined_vector",
+    "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
diff --git a/chromium/third_party/webrtc/api/transport/rtp/dependency_descriptor.cc b/chromium/third_party/webrtc/api/transport/rtp/dependency_descriptor.cc
new file mode 100644
index 00000000000..2a9b6d9a713
--- /dev/null
+++ b/chromium/third_party/webrtc/api/transport/rtp/dependency_descriptor.cc
@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/transport/rtp/dependency_descriptor.h"
+
+#include "absl/container/inlined_vector.h"
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+constexpr int DependencyDescriptor::kMaxSpatialIds;
+constexpr int DependencyDescriptor::kMaxTemporalIds;
+constexpr int DependencyDescriptor::kMaxTemplates;
+constexpr int DependencyDescriptor::kMaxDecodeTargets;
+
+namespace webrtc_impl {
+
+absl::InlinedVector<DecodeTargetIndication, 10> StringToDecodeTargetIndications(
+    absl::string_view symbols) {
+  absl::InlinedVector<DecodeTargetIndication, 10> dtis;
+  dtis.reserve(symbols.size());
+  for (char symbol : symbols) {
+    DecodeTargetIndication indication;
+    switch (symbol) {
+      case '-':
+        indication = DecodeTargetIndication::kNotPresent;
+        break;
+      case 'D':
+        indication = DecodeTargetIndication::kDiscardable;
+        break;
+      case 'R':
+        indication = DecodeTargetIndication::kRequired;
+        break;
+      case 'S':
+        indication = DecodeTargetIndication::kSwitch;
+        break;
+      default:
+        RTC_NOTREACHED();
+    }
+    dtis.push_back(indication);
+  }
+  return dtis;
+}
+
+}  // namespace webrtc_impl
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/transport/rtp/dependency_descriptor.h b/chromium/third_party/webrtc/api/transport/rtp/dependency_descriptor.h
index a488f56dfd0..a30608c1628 100644
--- a/chromium/third_party/webrtc/api/transport/rtp/dependency_descriptor.h
+++ b/chromium/third_party/webrtc/api/transport/rtp/dependency_descriptor.h
@@ -13,10 +13,12 @@
 
 #include <stdint.h>
 
+#include <initializer_list>
 #include <memory>
 #include <vector>
 
 #include "absl/container/inlined_vector.h"
+#include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 
 namespace webrtc {
@@ -52,6 +54,13 @@ enum class DecodeTargetIndication {
 };
 
 struct FrameDependencyTemplate {
+  // Setters are named briefly to chain them when building the template.
+  FrameDependencyTemplate& S(int spatial_layer);
+  FrameDependencyTemplate& T(int temporal_layer);
+  FrameDependencyTemplate& Dtis(absl::string_view dtis);
+  FrameDependencyTemplate& FrameDiffs(std::initializer_list<int> diffs);
+  FrameDependencyTemplate& ChainDiffs(std::initializer_list<int> diffs);
+
   friend bool operator==(const FrameDependencyTemplate& lhs,
                          const FrameDependencyTemplate& rhs) {
     return lhs.spatial_id == rhs.spatial_id &&
@@ -90,6 +99,11 @@ struct FrameDependencyStructure {
 };
 
 struct DependencyDescriptor {
+  static constexpr int kMaxSpatialIds = 4;
+  static constexpr int kMaxTemporalIds = 8;
+  static constexpr int kMaxDecodeTargets = 32;
+  static constexpr int kMaxTemplates = 64;
+
   bool first_packet_in_frame = true;
   bool last_packet_in_frame = true;
   int frame_number = 0;
@@ -99,6 +113,37 @@ struct DependencyDescriptor {
   std::unique_ptr<FrameDependencyStructure> attached_structure;
 };
 
+// Below are implementation details.
+namespace webrtc_impl {
+absl::InlinedVector<DecodeTargetIndication, 10> StringToDecodeTargetIndications(
+    absl::string_view indication_symbols);
+}  // namespace webrtc_impl
+
+inline FrameDependencyTemplate& FrameDependencyTemplate::S(int spatial_layer) {
+  this->spatial_id = spatial_layer;
+  return *this;
+}
+inline FrameDependencyTemplate& FrameDependencyTemplate::T(int temporal_layer) {
+  this->temporal_id = temporal_layer;
+  return *this;
+}
+inline FrameDependencyTemplate& FrameDependencyTemplate::Dtis(
+    absl::string_view dtis) {
+  this->decode_target_indications =
+      webrtc_impl::StringToDecodeTargetIndications(dtis);
+  return *this;
+}
+inline FrameDependencyTemplate& FrameDependencyTemplate::FrameDiffs(
+    std::initializer_list<int> diffs) {
+  this->frame_diffs.assign(diffs.begin(), diffs.end());
+  return *this;
+}
+inline FrameDependencyTemplate& FrameDependencyTemplate::ChainDiffs(
+    std::initializer_list<int> diffs) {
+  this->chain_diffs.assign(diffs.begin(), diffs.end());
+  return *this;
+}
+
 }  // namespace webrtc
 
 #endif  // API_TRANSPORT_RTP_DEPENDENCY_DESCRIPTOR_H_
diff --git a/chromium/third_party/webrtc/api/transport/test/mock_network_control.h b/chromium/third_party/webrtc/api/transport/test/mock_network_control.h
index 54a416cb770..f613004fb7a 100644
--- a/chromium/third_party/webrtc/api/transport/test/mock_network_control.h
+++ b/chromium/third_party/webrtc/api/transport/test/mock_network_control.h
@@ -18,11 +18,16 @@ namespace webrtc {
 
 class MockNetworkStateEstimator : public NetworkStateEstimator {
  public:
-  MOCK_METHOD0(GetCurrentEstimate, absl::optional<NetworkStateEstimate>());
-  MOCK_METHOD1(OnTransportPacketsFeedback,
-               void(const TransportPacketsFeedback&));
-  MOCK_METHOD1(OnReceivedPacket, void(const PacketResult&));
-  MOCK_METHOD1(OnRouteChange, void(const NetworkRouteChange&));
+  MOCK_METHOD(absl::optional<NetworkStateEstimate>,
+              GetCurrentEstimate,
+              (),
+              (override));
+  MOCK_METHOD(void,
+              OnTransportPacketsFeedback,
+              (const TransportPacketsFeedback&),
+              (override));
+  MOCK_METHOD(void, OnReceivedPacket, (const PacketResult&), (override));
+  MOCK_METHOD(void, OnRouteChange, (const NetworkRouteChange&), (override));
 };
 
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/units/data_rate_unittest.cc b/chromium/third_party/webrtc/api/units/data_rate_unittest.cc
index 4a6dd21af36..f77b3702d4d 100644
--- a/chromium/third_party/webrtc/api/units/data_rate_unittest.cc
+++ b/chromium/third_party/webrtc/api/units/data_rate_unittest.cc
@@ -175,7 +175,7 @@ TEST(UnitConversionTest, DataRateAndDataSizeAndFrequency) {
   EXPECT_EQ((rate_b / freq_a).bytes(), kBitsPerSecond / kHertz / 8);
 }
 
-TEST(UnitConversionTest, DivisionFailsOnLargeSize) {
+TEST(UnitConversionDeathTest, DivisionFailsOnLargeSize) {
   // Note that the failure is expected since the current implementation  is
   // implementated in a way that does not support division of large sizes. If
   // the implementation is changed, this test can safely be removed.
diff --git a/chromium/third_party/webrtc/api/video/BUILD.gn b/chromium/third_party/webrtc/api/video/BUILD.gn
index 290c2f2abba..ee7e51d4c42 100644
--- a/chromium/third_party/webrtc/api/video/BUILD.gn
+++ b/chromium/third_party/webrtc/api/video/BUILD.gn
@@ -21,7 +21,6 @@ rtc_library("video_rtp_headers") {
     "hdr_metadata.h",
     "video_content_type.cc",
     "video_content_type.h",
-    "video_frame_marking.h",
     "video_rotation.h",
     "video_timing.cc",
     "video_timing.h",
@@ -31,8 +30,8 @@ rtc_library("video_rtp_headers") {
     "..:array_view",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("video_frame") {
@@ -56,8 +55,8 @@ rtc_library("video_frame") {
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 if (is_android) {
@@ -141,8 +140,8 @@ rtc_library("encoded_image") {
     "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("encoded_frame") {
@@ -173,8 +172,8 @@ rtc_library("video_bitrate_allocation") {
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("video_bitrate_allocator") {
@@ -209,8 +208,8 @@ rtc_source_set("video_stream_decoder") {
     "../task_queue",
     "../units:time_delta",
     "../video_codecs:video_codecs_api",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("video_stream_decoder_create") {
@@ -237,7 +236,10 @@ rtc_library("video_adaptation") {
     "video_adaptation_reason.h",
   ]
 
-  deps = [ "../../rtc_base:checks" ]
+  deps = [
+    "../../rtc_base:checks",
+    "../../rtc_base:stringutils",
+  ]
 }
 
 rtc_source_set("video_stream_encoder") {
@@ -256,10 +258,29 @@ rtc_source_set("video_stream_encoder") {
     ":video_codec_constants",
     ":video_frame",
     "..:rtp_parameters",
+    "..:scoped_refptr",
     "../:fec_controller_api",
     "../:rtp_parameters",
+    "../adaptation:resource_adaptation_api",
     "../units:data_rate",
     "../video_codecs:video_codecs_api",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_source_set("video_frame_metadata") {
+  visibility = [ "*" ]
+  sources = [
+    "video_frame_metadata.cc",
+    "video_frame_metadata.h",
+  ]
+  deps = [
+    "..:array_view",
+    "../../modules/rtp_rtcp:rtp_video_header",
+    "../transport/rtp:dependency_descriptor",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
@@ -299,16 +320,21 @@ rtc_library("builtin_video_bitrate_allocator_factory") {
     "../../modules/video_coding:video_coding_utility",
     "../../modules/video_coding:webrtc_vp9_helpers",
     "../video_codecs:video_codecs_api",
-    "//third_party/abseil-cpp/absl/base:core_headers",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
 }
 
 if (rtc_include_tests) {
   rtc_library("video_unittests") {
     testonly = true
-    sources = [ "video_stream_decoder_create_unittest.cc" ]
+    sources = [
+      "video_frame_metadata_unittest.cc",
+      "video_stream_decoder_create_unittest.cc",
+    ]
     deps = [
+      ":video_frame_metadata",
       ":video_stream_decoder_create",
+      "../../modules/rtp_rtcp:rtp_video_header",
       "../../test:test_support",
       "../task_queue:default_task_queue_factory",
       "../video_codecs:builtin_video_decoder_factory",
diff --git a/chromium/third_party/webrtc/api/video/OWNERS b/chromium/third_party/webrtc/api/video/OWNERS
index 315f85e7d00..e4a16c360a1 100644
--- a/chromium/third_party/webrtc/api/video/OWNERS
+++ b/chromium/third_party/webrtc/api/video/OWNERS
@@ -1,3 +1,4 @@
+brandtr@webrtc.org
 magjed@webrtc.org
 nisse@webrtc.org
 
diff --git a/chromium/third_party/webrtc/api/video/test/BUILD.gn b/chromium/third_party/webrtc/api/video/test/BUILD.gn
index 5f697a081c8..5633371102b 100644
--- a/chromium/third_party/webrtc/api/video/test/BUILD.gn
+++ b/chromium/third_party/webrtc/api/video/test/BUILD.gn
@@ -21,8 +21,8 @@ rtc_library("rtc_api_video_unittests") {
     "..:video_frame",
     "..:video_rtp_headers",
     "../../../test:test_support",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_source_set("mock_recordable_encoded_frame") {
diff --git a/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h b/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h
index 1788a493c6d..2178932d2a6 100644
--- a/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h
+++ b/chromium/third_party/webrtc/api/video/test/mock_recordable_encoded_frame.h
@@ -17,13 +17,18 @@
 namespace webrtc {
 class MockRecordableEncodedFrame : public RecordableEncodedFrame {
  public:
-  MOCK_CONST_METHOD0(encoded_buffer,
-                     rtc::scoped_refptr<const EncodedImageBufferInterface>());
-  MOCK_CONST_METHOD0(color_space, absl::optional<webrtc::ColorSpace>());
-  MOCK_CONST_METHOD0(codec, VideoCodecType());
-  MOCK_CONST_METHOD0(is_key_frame, bool());
-  MOCK_CONST_METHOD0(resolution, EncodedResolution());
-  MOCK_CONST_METHOD0(render_time, Timestamp());
+  MOCK_METHOD(rtc::scoped_refptr<const EncodedImageBufferInterface>,
+              encoded_buffer,
+              (),
+              (const, override));
+  MOCK_METHOD(absl::optional<webrtc::ColorSpace>,
+              color_space,
+              (),
+              (const, override));
+  MOCK_METHOD(VideoCodecType, codec, (), (const, override));
+  MOCK_METHOD(bool, is_key_frame, (), (const, override));
+  MOCK_METHOD(EncodedResolution, resolution, (), (const, override));
+  MOCK_METHOD(Timestamp, render_time, (), (const, override));
 };
 }  // namespace webrtc
 #endif  // API_VIDEO_TEST_MOCK_RECORDABLE_ENCODED_FRAME_H_
diff --git a/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc b/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc
index 25e0bee1ff8..df1769d5d46 100644
--- a/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc
+++ b/chromium/third_party/webrtc/api/video/video_adaptation_counters.cc
@@ -10,6 +10,8 @@
 
 #include "api/video/video_adaptation_counters.h"
 
+#include "rtc_base/strings/string_builder.h"
+
 namespace webrtc {
 
 bool VideoAdaptationCounters::operator==(
@@ -30,4 +32,11 @@ VideoAdaptationCounters VideoAdaptationCounters::operator+(
       fps_adaptations + other.fps_adaptations);
 }
 
+std::string VideoAdaptationCounters::ToString() const {
+  rtc::StringBuilder ss;
+  ss << "{ res=" << resolution_adaptations << " fps=" << fps_adaptations
+     << " }";
+  return ss.Release();
+}
+
 }  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/video/video_adaptation_counters.h b/chromium/third_party/webrtc/api/video/video_adaptation_counters.h
index eff0baaa21f..2dea902f2f1 100644
--- a/chromium/third_party/webrtc/api/video/video_adaptation_counters.h
+++ b/chromium/third_party/webrtc/api/video/video_adaptation_counters.h
@@ -11,6 +11,8 @@
 #ifndef API_VIDEO_VIDEO_ADAPTATION_COUNTERS_H_
 #define API_VIDEO_VIDEO_ADAPTATION_COUNTERS_H_
 
+#include <string>
+
 #include "rtc_base/checks.h"
 
 namespace webrtc {
@@ -33,6 +35,8 @@ struct VideoAdaptationCounters {
 
   VideoAdaptationCounters operator+(const VideoAdaptationCounters& other) const;
 
+  std::string ToString() const;
+
   int resolution_adaptations;
   int fps_adaptations;
 };
diff --git a/chromium/third_party/webrtc/api/video/video_frame_marking.h b/chromium/third_party/webrtc/api/video/video_frame_marking.h
deleted file mode 100644
index 2a34852f1d6..00000000000
--- a/chromium/third_party/webrtc/api/video/video_frame_marking.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- *  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 API_VIDEO_VIDEO_FRAME_MARKING_H_
-#define API_VIDEO_VIDEO_FRAME_MARKING_H_
-
-namespace webrtc {
-
-struct FrameMarking {
-  bool start_of_frame;
-  bool end_of_frame;
-  bool independent_frame;
-  bool discardable_frame;
-  bool base_layer_sync;
-  uint8_t temporal_id;
-  uint8_t layer_id;
-  uint8_t tl0_pic_idx;
-};
-
-}  // namespace webrtc
-
-#endif  // API_VIDEO_VIDEO_FRAME_MARKING_H_
diff --git a/chromium/third_party/webrtc/api/video/video_frame_metadata.cc b/chromium/third_party/webrtc/api/video/video_frame_metadata.cc
new file mode 100644
index 00000000000..df82875eb95
--- /dev/null
+++ b/chromium/third_party/webrtc/api/video/video_frame_metadata.cc
@@ -0,0 +1,28 @@
+/*
+ *  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 "api/video/video_frame_metadata.h"
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+
+namespace webrtc {
+
+VideoFrameMetadata::VideoFrameMetadata(const RTPVideoHeader& header)
+    : width_(header.width), height_(header.height) {
+  if (header.generic) {
+    frame_id_ = header.generic->frame_id;
+    spatial_index_ = header.generic->spatial_index;
+    temporal_index_ = header.generic->temporal_index;
+    frame_dependencies_ = header.generic->dependencies;
+    decode_target_indications_ = header.generic->decode_target_indications;
+  }
+}
+
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/video/video_frame_metadata.h b/chromium/third_party/webrtc/api/video/video_frame_metadata.h
new file mode 100644
index 00000000000..2e9309841b0
--- /dev/null
+++ b/chromium/third_party/webrtc/api/video/video_frame_metadata.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_VIDEO_FRAME_METADATA_H_
+#define API_VIDEO_VIDEO_FRAME_METADATA_H_
+
+#include <cstdint>
+
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+
+namespace webrtc {
+
+struct RTPVideoHeader;
+
+// A subset of metadata from the RTP video header, exposed in insertable streams
+// API.
+class VideoFrameMetadata {
+ public:
+  explicit VideoFrameMetadata(const RTPVideoHeader& header);
+  VideoFrameMetadata(const VideoFrameMetadata&) = default;
+  VideoFrameMetadata& operator=(const VideoFrameMetadata&) = default;
+
+  uint16_t GetWidth() const { return width_; }
+  uint16_t GetHeight() const { return height_; }
+  absl::optional<int64_t> GetFrameId() const { return frame_id_; }
+  int GetSpatialIndex() const { return spatial_index_; }
+  int GetTemporalIndex() const { return temporal_index_; }
+
+  rtc::ArrayView<const int64_t> GetFrameDependencies() const {
+    return frame_dependencies_;
+  }
+
+  rtc::ArrayView<const DecodeTargetIndication> GetDecodeTargetIndications()
+      const {
+    return decode_target_indications_;
+  }
+
+ private:
+  int16_t width_;
+  int16_t height_;
+  absl::optional<int64_t> frame_id_;
+  int spatial_index_ = 0;
+  int temporal_index_ = 0;
+  absl::InlinedVector<int64_t, 5> frame_dependencies_;
+  absl::InlinedVector<DecodeTargetIndication, 10> decode_target_indications_;
+};
+}  // namespace webrtc
+
+#endif  // API_VIDEO_VIDEO_FRAME_METADATA_H_
diff --git a/chromium/third_party/webrtc/api/video/video_frame_metadata_unittest.cc b/chromium/third_party/webrtc/api/video/video_frame_metadata_unittest.cc
new file mode 100644
index 00000000000..7a808e1ea9b
--- /dev/null
+++ b/chromium/third_party/webrtc/api/video/video_frame_metadata_unittest.cc
@@ -0,0 +1,120 @@
+/*
+ *  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 "api/video/video_frame_metadata.h"
+
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::testing::ElementsAre;
+using ::testing::IsEmpty;
+
+TEST(VideoFrameMetadata, GetWidthReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  video_header.width = 1280u;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetWidth(), video_header.width);
+}
+
+TEST(VideoFrameMetadata, GetHeightReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  video_header.height = 720u;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetHeight(), video_header.height);
+}
+
+TEST(VideoFrameMetadata, GetFrameIdReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.frame_id = 10;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetFrameId().value(), 10);
+}
+
+TEST(VideoFrameMetadata, HasNoFrameIdForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetFrameId(), absl::nullopt);
+}
+
+TEST(VideoFrameMetadata, GetSpatialIndexReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.spatial_index = 2;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetSpatialIndex(), 2);
+}
+
+TEST(VideoFrameMetadata, SpatialIndexIsZeroForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetSpatialIndex(), 0);
+}
+
+TEST(VideoFrameMetadata, GetTemporalIndexReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.temporal_index = 3;
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_EQ(metadata.GetTemporalIndex(), 3);
+}
+
+TEST(VideoFrameMetadata, TemporalIndexIsZeroForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_EQ(metadata.GetTemporalIndex(), 0);
+}
+
+TEST(VideoFrameMetadata, GetFrameDependenciesReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.dependencies = {5, 6, 7};
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_THAT(metadata.GetFrameDependencies(), ElementsAre(5, 6, 7));
+}
+
+TEST(VideoFrameMetadata, FrameDependencyVectorIsEmptyForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_THAT(metadata.GetFrameDependencies(), IsEmpty());
+}
+
+TEST(VideoFrameMetadata, GetDecodeTargetIndicationsReturnsCorrectValue) {
+  RTPVideoHeader video_header;
+  RTPVideoHeader::GenericDescriptorInfo& generic =
+      video_header.generic.emplace();
+  generic.decode_target_indications = {DecodeTargetIndication::kSwitch};
+  VideoFrameMetadata metadata(video_header);
+  EXPECT_THAT(metadata.GetDecodeTargetIndications(),
+              ElementsAre(DecodeTargetIndication::kSwitch));
+}
+
+TEST(VideoFrameMetadata,
+     DecodeTargetIndicationsVectorIsEmptyForHeaderWithoutGeneric) {
+  RTPVideoHeader video_header;
+  VideoFrameMetadata metadata(video_header);
+  ASSERT_FALSE(video_header.generic);
+  EXPECT_THAT(metadata.GetDecodeTargetIndications(), IsEmpty());
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h b/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h
index 8e1df0f858a..d8dd8e15994 100644
--- a/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h
+++ b/chromium/third_party/webrtc/api/video/video_stream_encoder_interface.h
@@ -13,8 +13,10 @@
 
 #include <vector>
 
+#include "api/adaptation/resource.h"
 #include "api/fec_controller_override.h"
 #include "api/rtp_parameters.h"  // For DegradationPreference.
+#include "api/scoped_refptr.h"
 #include "api/units/data_rate.h"
 #include "api/video/video_bitrate_allocator.h"
 #include "api/video/video_sink_interface.h"
@@ -49,6 +51,15 @@ class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
         int min_transmit_bitrate_bps) = 0;
   };
 
+  // If the resource is overusing, the VideoStreamEncoder will try to reduce
+  // resolution or frame rate until no resource is overusing.
+  // TODO(https://crbug.com/webrtc/11565): When the ResourceAdaptationProcessor
+  // is moved to Call this method could be deleted altogether in favor of
+  // Call-level APIs only.
+  virtual void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) = 0;
+  virtual std::vector<rtc::scoped_refptr<Resource>>
+  GetAdaptationResources() = 0;
+
   // Sets the source that will provide video frames to the VideoStreamEncoder's
   // OnFrame method. |degradation_preference| control whether or not resolution
   // or frame rate may be reduced. The VideoStreamEncoder registers itself with
diff --git a/chromium/third_party/webrtc/api/video_codecs/BUILD.gn b/chromium/third_party/webrtc/api/video_codecs/BUILD.gn
index 21a5f6faa07..597478ba0a8 100644
--- a/chromium/third_party/webrtc/api/video_codecs/BUILD.gn
+++ b/chromium/third_party/webrtc/api/video_codecs/BUILD.gn
@@ -49,6 +49,8 @@ rtc_library("video_codecs_api") {
     "../video:video_codec_constants",
     "../video:video_frame",
     "../video:video_rtp_headers",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/algorithm:container",
     "//third_party/abseil-cpp/absl/container:inlined_vector",
     "//third_party/abseil-cpp/absl/strings",
@@ -100,8 +102,8 @@ rtc_library("builtin_video_encoder_factory") {
     "../../media:rtc_media_base",
     "../../rtc_base:checks",
     "../../rtc_base/system:rtc_export",
-    "//third_party/abseil-cpp/absl/strings",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
 
 rtc_library("vp8_temporal_layers_factory") {
@@ -148,6 +150,8 @@ rtc_library("rtc_software_fallback_wrappers") {
     "../video:video_bitrate_allocation",
     "../video:video_frame",
     "../video:video_rtp_headers",
+  ]
+  absl_deps = [
     "//third_party/abseil-cpp/absl/base:core_headers",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
diff --git a/chromium/third_party/webrtc/api/video_codecs/test/BUILD.gn b/chromium/third_party/webrtc/api/video_codecs/test/BUILD.gn
index 243b78267fc..10b18b6e5b5 100644
--- a/chromium/third_party/webrtc/api/video_codecs/test/BUILD.gn
+++ b/chromium/third_party/webrtc/api/video_codecs/test/BUILD.gn
@@ -40,5 +40,6 @@ if (rtc_include_tests) {
       "../../video:video_rtp_headers",
       "//testing/gtest",
     ]
+    absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
   }
 }
diff --git a/chromium/third_party/webrtc/api/video_codecs/video_codec.h b/chromium/third_party/webrtc/api/video_codecs/video_codec.h
index 330bbbce199..c07fae9b8bb 100644
--- a/chromium/third_party/webrtc/api/video_codecs/video_codec.h
+++ b/chromium/third_party/webrtc/api/video_codecs/video_codec.h
@@ -19,7 +19,7 @@
 #include "absl/types/optional.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "api/video/video_codec_type.h"
-#include "common_types.h"  // NOLINT(build/include)
+#include "common_types.h"  // NOLINT(build/include_directory)
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
diff --git a/chromium/third_party/webrtc/api/voip/BUILD.gn b/chromium/third_party/webrtc/api/voip/BUILD.gn
index 2c5f71c988e..6f92ed67f4e 100644
--- a/chromium/third_party/webrtc/api/voip/BUILD.gn
+++ b/chromium/third_party/webrtc/api/voip/BUILD.gn
@@ -19,8 +19,8 @@ rtc_source_set("voip_api") {
   deps = [
     "..:array_view",
     "../audio_codecs:audio_codecs_api",
-    "//third_party/abseil-cpp/absl/types:optional",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 
 rtc_library("voip_engine_factory") {