Merging upstream version 125.0.1.

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

23 files changed, 70 insertions, 309 deletions

diff --git a/third_party/libwebrtc/media/BUILD.gn b/third_party/libwebrtc/media/BUILD.gn
index 97ad4a889a..055bf75a19 100644
--- a/third_party/libwebrtc/media/BUILD.gn
+++ b/third_party/libwebrtc/media/BUILD.gn
@@ -262,7 +262,6 @@ rtc_library("codec") {
   ]
   deps = [
     ":media_constants",
-    "../api:field_trials_view",
     "../api:rtp_parameters",
     "../api/audio_codecs:audio_codecs_api",
     "../api/video_codecs:video_codecs_api",
@@ -546,7 +545,6 @@ rtc_library("rtc_audio_video") {
     "../rtc_base:copy_on_write_buffer",
     "../rtc_base:dscp",
     "../rtc_base:event_tracer",
-    "../rtc_base:ignore_wundef",
     "../rtc_base:logging",
     "../rtc_base:macromagic",
     "../rtc_base:network_route",
@@ -611,33 +609,6 @@ rtc_library("rtc_audio_video") {
   }
 }
 
-# Heavy but optional helper for unittests and webrtc users who prefer to use
-# defaults factories or do not worry about extra dependencies and binary size.
-rtc_library("rtc_media_engine_defaults") {
-  visibility = [ "*" ]
-  allow_poison = [
-    "audio_codecs",
-    "default_task_queue",
-    "software_video_codecs",
-  ]
-  sources = [
-    "engine/webrtc_media_engine_defaults.cc",
-    "engine/webrtc_media_engine_defaults.h",
-  ]
-  deps = [
-    ":rtc_audio_video",
-    "../api/audio_codecs:builtin_audio_decoder_factory",
-    "../api/audio_codecs:builtin_audio_encoder_factory",
-    "../api/task_queue:default_task_queue_factory",
-    "../api/video:builtin_video_bitrate_allocator_factory",
-    "../api/video_codecs:builtin_video_decoder_factory",
-    "../api/video_codecs:builtin_video_encoder_factory",
-    "../modules/audio_processing:api",
-    "../rtc_base:checks",
-    "../rtc_base/system:rtc_export",
-  ]
-}
-
 rtc_source_set("rtc_data_sctp_transport_internal") {
   sources = [ "sctp/sctp_transport_internal.h" ]
   deps = [
@@ -840,7 +811,6 @@ if (rtc_include_tests) {
         ":rtc_internal_video_codecs",
         ":rtc_media",
         ":rtc_media_base",
-        ":rtc_media_engine_defaults",
         ":rtc_media_tests_utils",
         ":rtc_sdp_video_format_utils",
         ":rtc_simulcast_encoder_adapter",
@@ -860,6 +830,8 @@ if (rtc_include_tests) {
         "../api:simulcast_test_fixture_api",
         "../api/audio_codecs:builtin_audio_decoder_factory",
         "../api/audio_codecs:builtin_audio_encoder_factory",
+        "../api/environment",
+        "../api/environment:environment_factory",
         "../api/rtc_event_log",
         "../api/task_queue",
         "../api/task_queue:default_task_queue_factory",
diff --git a/third_party/libwebrtc/media/base/codec.cc b/third_party/libwebrtc/media/base/codec.cc
index b819707702..c4e1c6f1f3 100644
--- a/third_party/libwebrtc/media/base/codec.cc
+++ b/third_party/libwebrtc/media/base/codec.cc
@@ -158,8 +158,7 @@ bool Codec::operator==(const Codec& c) const {
               : (packetization == c.packetization));
 }
 
-bool Codec::Matches(const Codec& codec,
-                    const webrtc::FieldTrialsView* field_trials) const {
+bool Codec::Matches(const Codec& codec) const {
   // Match the codec id/name based on the typical static/dynamic name rules.
   // Matching is case-insensitive.
 
diff --git a/third_party/libwebrtc/media/base/codec.h b/third_party/libwebrtc/media/base/codec.h
index 228acad07a..bd4239b251 100644
--- a/third_party/libwebrtc/media/base/codec.h
+++ b/third_party/libwebrtc/media/base/codec.h
@@ -20,7 +20,6 @@
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/audio_codecs/audio_format.h"
-#include "api/field_trials_view.h"
 #include "api/rtp_parameters.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "media/base/media_constants.h"
@@ -112,8 +111,7 @@ struct RTC_EXPORT Codec {
   // Indicates if this codec is compatible with the specified codec by
   // checking the assigned id and profile values for the relevant video codecs.
   // H264 levels are not compared.
-  bool Matches(const Codec& codec,
-               const webrtc::FieldTrialsView* field_trials = nullptr) const;
+  bool Matches(const Codec& codec) const;
   bool MatchesRtpCodec(const webrtc::RtpCodec& capability) const;
 
   // Find the parameter for `name` and write the value to `out`.
diff --git a/third_party/libwebrtc/media/codec_gn/moz.build b/third_party/libwebrtc/media/codec_gn/moz.build
index a6fa3b4063..b5ebd454d3 100644
--- a/third_party/libwebrtc/media/codec_gn/moz.build
+++ b/third_party/libwebrtc/media/codec_gn/moz.build
@@ -195,7 +195,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
@@ -205,10 +204,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
         "-msse2"
     ]
 
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/engine/null_webrtc_video_engine_unittest.cc b/third_party/libwebrtc/media/engine/null_webrtc_video_engine_unittest.cc
index 31c442d53d..9515d44be9 100644
--- a/third_party/libwebrtc/media/engine/null_webrtc_video_engine_unittest.cc
+++ b/third_party/libwebrtc/media/engine/null_webrtc_video_engine_unittest.cc
@@ -37,7 +37,7 @@ TEST(NullWebRtcVideoEngineTest, CheckInterface) {
       task_queue_factory.get(), adm.get(),
       webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
       webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr,
-      webrtc::AudioProcessingBuilder().Create(), nullptr, nullptr, trials);
+      webrtc::AudioProcessingBuilder().Create(), nullptr, trials);
 
   CompositeMediaEngine engine(std::move(audio_engine),
                               std::make_unique<NullWebRtcVideoEngine>());
diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine.cc b/third_party/libwebrtc/media/engine/webrtc_media_engine.cc
index 99d7dd2704..463ed29109 100644
--- a/third_party/libwebrtc/media/engine/webrtc_media_engine.cc
+++ b/third_party/libwebrtc/media/engine/webrtc_media_engine.cc
@@ -46,7 +46,6 @@ std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
       std::move(dependencies.audio_decoder_factory),
       std::move(dependencies.audio_mixer),
       std::move(dependencies.audio_processing),
-      dependencies.audio_frame_processor,
       std::move(dependencies.owned_audio_frame_processor), trials);
 #ifdef HAVE_WEBRTC_VIDEO
   auto video_engine = std::make_unique<WebRtcVideoEngine>(
diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine.h b/third_party/libwebrtc/media/engine/webrtc_media_engine.h
index 0f6dce35b5..863db9f278 100644
--- a/third_party/libwebrtc/media/engine/webrtc_media_engine.h
+++ b/third_party/libwebrtc/media/engine/webrtc_media_engine.h
@@ -49,9 +49,6 @@ struct MediaEngineDependencies {
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory;
   rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer;
   rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
-  // TODO(bugs.webrtc.org/15111):
-  //   Remove the raw AudioFrameProcessor pointer in the follow-up.
-  webrtc::AudioFrameProcessor* audio_frame_processor = nullptr;
   std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor;
 
   std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory;
@@ -63,8 +60,9 @@ struct MediaEngineDependencies {
 // CreateMediaEngine may be called on any thread, though the engine is
 // only expected to be used on one thread, internally called the "worker
 // thread". This is the thread Init must be called on.
-RTC_EXPORT std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
-    MediaEngineDependencies dependencies);
+[[deprecated("bugs.webrtc.org/15574")]]  //
+RTC_EXPORT std::unique_ptr<MediaEngineInterface>
+CreateMediaEngine(MediaEngineDependencies dependencies);
 
 // Verify that extension IDs are within 1-byte extension range and are not
 // overlapping, and that they form a legal change from previously registerd
diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.cc b/third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.cc
deleted file mode 100644
index 1660873e8b..0000000000
--- a/third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.cc
+++ /dev/null
@@ -1,43 +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.
- */
-#include "media/engine/webrtc_media_engine_defaults.h"
-
-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "api/task_queue/default_task_queue_factory.h"
-#include "api/video/builtin_video_bitrate_allocator_factory.h"
-#include "api/video_codecs/builtin_video_decoder_factory.h"
-#include "api/video_codecs/builtin_video_encoder_factory.h"
-#include "modules/audio_processing/include/audio_processing.h"
-#include "rtc_base/checks.h"
-
-namespace webrtc {
-
-void SetMediaEngineDefaults(cricket::MediaEngineDependencies* deps) {
-  RTC_DCHECK(deps);
-  if (deps->task_queue_factory == nullptr) {
-    static TaskQueueFactory* const task_queue_factory =
-        CreateDefaultTaskQueueFactory().release();
-    deps->task_queue_factory = task_queue_factory;
-  }
-  if (deps->audio_encoder_factory == nullptr)
-    deps->audio_encoder_factory = CreateBuiltinAudioEncoderFactory();
-  if (deps->audio_decoder_factory == nullptr)
-    deps->audio_decoder_factory = CreateBuiltinAudioDecoderFactory();
-  if (deps->audio_processing == nullptr)
-    deps->audio_processing = AudioProcessingBuilder().Create();
-
-  if (deps->video_encoder_factory == nullptr)
-    deps->video_encoder_factory = CreateBuiltinVideoEncoderFactory();
-  if (deps->video_decoder_factory == nullptr)
-    deps->video_decoder_factory = CreateBuiltinVideoDecoderFactory();
-}
-
-}  // namespace webrtc
diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.h b/third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.h
deleted file mode 100644
index 16b1d462e3..0000000000
--- a/third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.h
+++ /dev/null
@@ -1,24 +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.
- */
-
-#ifndef MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_DEFAULTS_H_
-#define MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_DEFAULTS_H_
-
-#include "media/engine/webrtc_media_engine.h"
-#include "rtc_base/system/rtc_export.h"
-
-namespace webrtc {
-
-// Sets required but null dependencies with default factories.
-RTC_EXPORT void SetMediaEngineDefaults(cricket::MediaEngineDependencies* deps);
-
-}  // namespace webrtc
-
-#endif  // MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_DEFAULTS_H_
diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine_unittest.cc b/third_party/libwebrtc/media/engine/webrtc_media_engine_unittest.cc
index 4615f03deb..40cad75701 100644
--- a/third_party/libwebrtc/media/engine/webrtc_media_engine_unittest.cc
+++ b/third_party/libwebrtc/media/engine/webrtc_media_engine_unittest.cc
@@ -14,7 +14,6 @@
 #include <string>
 #include <utility>
 
-#include "media/engine/webrtc_media_engine_defaults.h"
 #include "test/gtest.h"
 #include "test/scoped_key_value_config.h"
 
@@ -322,16 +321,4 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensionsRemoveRedundantBwe3) {
   EXPECT_EQ(RtpExtension::kTimestampOffsetUri, filtered[0].uri);
 }
 
-TEST(WebRtcMediaEngineTest, Create) {
-  MediaEngineDependencies deps;
-  webrtc::SetMediaEngineDefaults(&deps);
-  webrtc::test::ScopedKeyValueConfig trials;
-  deps.trials = &trials;
-
-  std::unique_ptr<MediaEngineInterface> engine =
-      CreateMediaEngine(std::move(deps));
-
-  EXPECT_TRUE(engine);
-}
-
 }  // namespace cricket
diff --git a/third_party/libwebrtc/media/engine/webrtc_video_engine_unittest.cc b/third_party/libwebrtc/media/engine/webrtc_video_engine_unittest.cc
index e8b7ee4b2d..f5736679be 100644
--- a/third_party/libwebrtc/media/engine/webrtc_video_engine_unittest.cc
+++ b/third_party/libwebrtc/media/engine/webrtc_video_engine_unittest.cc
@@ -20,9 +20,9 @@
 
 #include "absl/algorithm/container.h"
 #include "absl/strings/match.h"
-#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
 #include "api/rtp_parameters.h"
-#include "api/task_queue/default_task_queue_factory.h"
 #include "api/test/mock_encoder_selector.h"
 #include "api/test/mock_video_bitrate_allocator.h"
 #include "api/test/mock_video_bitrate_allocator_factory.h"
@@ -101,6 +101,8 @@ using ::testing::WithArg;
 using ::webrtc::BitrateConstraints;
 using ::webrtc::Call;
 using ::webrtc::CallConfig;
+using ::webrtc::CreateEnvironment;
+using ::webrtc::Environment;
 using ::webrtc::kDefaultScalabilityModeStr;
 using ::webrtc::RtpExtension;
 using ::webrtc::RtpPacket;
@@ -355,13 +357,10 @@ class WebRtcVideoEngineTest : public ::testing::Test {
   explicit WebRtcVideoEngineTest(const std::string& field_trials)
       : field_trials_(field_trials),
         time_controller_(webrtc::Timestamp::Millis(4711)),
-        task_queue_factory_(time_controller_.CreateTaskQueueFactory()),
-        call_(Call::Create([&] {
-          CallConfig call_config(&event_log_);
-          call_config.task_queue_factory = task_queue_factory_.get();
-          call_config.trials = &field_trials_;
-          return call_config;
-        }())),
+        env_(CreateEnvironment(&field_trials_,
+                               time_controller_.CreateTaskQueueFactory(),
+                               time_controller_.GetClock())),
+        call_(Call::Create(CallConfig(env_))),
         encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
         decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
         video_bitrate_allocator_factory_(
@@ -398,8 +397,7 @@ class WebRtcVideoEngineTest : public ::testing::Test {
 
   webrtc::test::ScopedKeyValueConfig field_trials_;
   webrtc::GlobalSimulatedTimeController time_controller_;
-  webrtc::RtcEventLogNull event_log_;
-  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
+  Environment env_;
   // Used in WebRtcVideoEngineVoiceTest, but defined here so it's properly
   // initialized when the constructor is called.
   std::unique_ptr<Call> call_;
@@ -1479,14 +1477,10 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
   EXPECT_CALL(*decoder_factory, CreateVideoDecoder(format)).Times(0);
 
   // Create a call.
-  webrtc::RtcEventLogNull event_log;
   webrtc::GlobalSimulatedTimeController time_controller(
       webrtc::Timestamp::Millis(4711));
-  auto task_queue_factory = time_controller.CreateTaskQueueFactory();
-  CallConfig call_config(&event_log);
-  webrtc::FieldTrialBasedConfig field_trials;
-  call_config.trials = &field_trials;
-  call_config.task_queue_factory = task_queue_factory.get();
+  CallConfig call_config(CreateEnvironment(
+      time_controller.CreateTaskQueueFactory(), time_controller.GetClock()));
   const std::unique_ptr<Call> call = Call::Create(call_config);
 
   // Create send channel.
@@ -1615,18 +1609,11 @@ TEST_F(WebRtcVideoEngineTest, SetVideoRtxEnabled) {
 
 class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test {
  protected:
-  CallConfig GetCallConfig(webrtc::RtcEventLogNull* event_log,
-                           webrtc::TaskQueueFactory* task_queue_factory) {
-    CallConfig call_config(event_log);
-    call_config.task_queue_factory = task_queue_factory;
-    call_config.trials = &field_trials_;
-    return call_config;
-  }
-
   WebRtcVideoChannelEncodedFrameCallbackTest()
-      : task_queue_factory_(time_controller_.CreateTaskQueueFactory()),
-        call_(Call::Create(
-            GetCallConfig(&event_log_, task_queue_factory_.get()))),
+      : env_(CreateEnvironment(&field_trials_,
+                               time_controller_.CreateTaskQueueFactory(),
+                               time_controller_.GetClock())),
+        call_(Call::Create(CallConfig(env_))),
         video_bitrate_allocator_factory_(
             webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
         engine_(
@@ -1676,8 +1663,7 @@ class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test {
   webrtc::GlobalSimulatedTimeController time_controller_{
       Timestamp::Seconds(1000)};
   webrtc::test::ScopedKeyValueConfig field_trials_;
-  webrtc::RtcEventLogNull event_log_;
-  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
+  Environment env_;
   std::unique_ptr<Call> call_;
   std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
       video_bitrate_allocator_factory_;
@@ -1796,7 +1782,9 @@ TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest, DoesNotDecodeWhenDisabled) {
 class WebRtcVideoChannelBaseTest : public ::testing::Test {
  protected:
   WebRtcVideoChannelBaseTest()
-      : task_queue_factory_(time_controller_.CreateTaskQueueFactory()),
+      : env_(CreateEnvironment(&field_trials_,
+                               time_controller_.CreateTaskQueueFactory(),
+                               time_controller_.GetClock())),
         video_bitrate_allocator_factory_(
             webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
         engine_(std::make_unique<webrtc::VideoEncoderFactoryTemplate<
@@ -1814,10 +1802,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
   void SetUp() override {
     // One testcase calls SetUp in a loop, only create call_ once.
     if (!call_) {
-      CallConfig call_config(&event_log_);
-      call_config.task_queue_factory = task_queue_factory_.get();
-      call_config.trials = &field_trials_;
-      call_ = Call::Create(call_config);
+      call_ = Call::Create(CallConfig(env_));
     }
 
     cricket::MediaConfig media_config;
@@ -2016,10 +2001,9 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
   webrtc::GlobalSimulatedTimeController time_controller_{
       Timestamp::Seconds(1000)};
 
-  webrtc::RtcEventLogNull event_log_;
   webrtc::test::ScopedKeyValueConfig field_trials_;
   std::unique_ptr<webrtc::test::ScopedKeyValueConfig> override_field_trials_;
-  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
+  Environment env_;
   std::unique_ptr<Call> call_;
   std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
       video_bitrate_allocator_factory_;
@@ -4378,7 +4362,7 @@ TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) {
 
   absl::optional<VideoCodec> codec = send_channel_->GetSendCodec();
   ASSERT_TRUE(codec);
-  EXPECT_TRUE(codec->Matches(engine_.send_codecs()[0], &field_trials_));
+  EXPECT_TRUE(codec->Matches(engine_.send_codecs()[0]));
 
   // Using a RTX setup to verify that the default RTX payload type is good.
   const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs1);
@@ -9863,7 +9847,6 @@ class WebRtcVideoChannelSimulcastTest : public ::testing::Test {
   }
 
   webrtc::test::ScopedKeyValueConfig field_trials_;
-  webrtc::RtcEventLogNull event_log_;
   FakeCall fake_call_;
   cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
   cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc
index adf8b5c51d..adf662074d 100644
--- a/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc
+++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc
@@ -66,7 +66,6 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/dscp.h"
 #include "rtc_base/experiments/struct_parameters_parser.h"
-#include "rtc_base/ignore_wundef.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/string_encode.h"
@@ -79,13 +78,12 @@
 #include "system_wrappers/include/metrics.h"
 
 #if WEBRTC_ENABLE_PROTOBUF
-RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #else
 #include "modules/audio_coding/audio_network_adaptor/config.pb.h"
 #endif
-RTC_POP_IGNORING_WUNDEF()
+
 #endif
 
 namespace cricket {
@@ -147,12 +145,10 @@ bool IsCodec(const AudioCodec& codec, const char* ref_name) {
   return absl::EqualsIgnoreCase(codec.name, ref_name);
 }
 
-absl::optional<AudioCodec> FindCodec(
-    const std::vector<AudioCodec>& codecs,
-    const AudioCodec& codec,
-    const webrtc::FieldTrialsView* field_trials) {
+absl::optional<AudioCodec> FindCodec(const std::vector<AudioCodec>& codecs,
+                                     const AudioCodec& codec) {
   for (const AudioCodec& c : codecs) {
-    if (c.Matches(codec, field_trials)) {
+    if (c.Matches(codec)) {
       return c;
     }
   }
@@ -344,10 +340,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
     const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
     rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
     rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
-    // TODO(bugs.webrtc.org/15111):
-    //   Remove the raw AudioFrameProcessor pointer in the follow-up.
-    webrtc::AudioFrameProcessor* audio_frame_processor,
-    std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor,
+    std::unique_ptr<webrtc::AudioFrameProcessor> audio_frame_processor,
     const webrtc::FieldTrialsView& trials)
     : task_queue_factory_(task_queue_factory),
       adm_(adm),
@@ -355,8 +348,7 @@ WebRtcVoiceEngine::WebRtcVoiceEngine(
       decoder_factory_(decoder_factory),
       audio_mixer_(audio_mixer),
       apm_(audio_processing),
-      audio_frame_processor_(audio_frame_processor),
-      owned_audio_frame_processor_(std::move(owned_audio_frame_processor)),
+      audio_frame_processor_(std::move(audio_frame_processor)),
       minimized_remsampling_on_mobile_trial_enabled_(
           IsEnabled(trials, "WebRTC-Audio-MinimizeResamplingOnMobile")) {
   RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
@@ -425,11 +417,7 @@ void WebRtcVoiceEngine::Init() {
     if (audio_frame_processor_) {
       config.async_audio_processing_factory =
           rtc::make_ref_counted<webrtc::AsyncAudioProcessing::Factory>(
-              *audio_frame_processor_, *task_queue_factory_);
-    } else if (owned_audio_frame_processor_) {
-      config.async_audio_processing_factory =
-          rtc::make_ref_counted<webrtc::AsyncAudioProcessing::Factory>(
-              std::move(owned_audio_frame_processor_), *task_queue_factory_);
+              std::move(audio_frame_processor_), *task_queue_factory_);
     }
     audio_state_ = webrtc::AudioState::Create(config);
   }
@@ -2151,8 +2139,7 @@ bool WebRtcVoiceReceiveChannel::SetRecvCodecs(
   for (const AudioCodec& codec : codecs) {
     // Log a warning if a codec's payload type is changing. This used to be
     // treated as an error. It's abnormal, but not really illegal.
-    absl::optional<AudioCodec> old_codec =
-        FindCodec(recv_codecs_, codec, &call_->trials());
+    absl::optional<AudioCodec> old_codec = FindCodec(recv_codecs_, codec);
     if (old_codec && old_codec->id != codec.id) {
       RTC_LOG(LS_WARNING) << codec.name << " mapped to a second payload type ("
                           << codec.id << ", was already mapped to "
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.h b/third_party/libwebrtc/media/engine/webrtc_voice_engine.h
index a3e6d3acab..ed71667525 100644
--- a/third_party/libwebrtc/media/engine/webrtc_voice_engine.h
+++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine.h
@@ -90,9 +90,6 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
       rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
       rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing,
-      // TODO(bugs.webrtc.org/15111):
-      //   Remove the raw AudioFrameProcessor pointer in the follow-up.
-      webrtc::AudioFrameProcessor* audio_frame_processor,
       std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor,
       const webrtc::FieldTrialsView& trials);
 
@@ -166,10 +163,7 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface {
   // The audio processing module.
   rtc::scoped_refptr<webrtc::AudioProcessing> apm_;
   // Asynchronous audio processing.
-  // TODO(bugs.webrtc.org/15111):
-  //   Remove the raw AudioFrameProcessor pointer in the follow-up.
-  webrtc::AudioFrameProcessor* const audio_frame_processor_;
-  std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor_;
+  std::unique_ptr<webrtc::AudioFrameProcessor> audio_frame_processor_;
   // The primary instance of WebRtc VoiceEngine.
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::vector<AudioCodec> send_codecs_;
diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc
index b1393eec74..4d6580631d 100644
--- a/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc
+++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc
@@ -18,8 +18,9 @@
 #include "absl/types/optional.h"
 #include "api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
 #include "api/media_types.h"
-#include "api/rtc_event_log/rtc_event_log.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
 #include "api/task_queue/default_task_queue_factory.h"
@@ -45,6 +46,7 @@
 #include "test/mock_audio_encoder_factory.h"
 #include "test/scoped_key_value_config.h"
 
+namespace {
 using ::testing::_;
 using ::testing::ContainerEq;
 using ::testing::Contains;
@@ -55,11 +57,11 @@ using ::testing::ReturnPointee;
 using ::testing::SaveArg;
 using ::testing::StrictMock;
 using ::testing::UnorderedElementsAreArray;
+using ::webrtc::BitrateConstraints;
 using ::webrtc::Call;
 using ::webrtc::CallConfig;
-
-namespace {
-using webrtc::BitrateConstraints;
+using ::webrtc::CreateEnvironment;
+using ::webrtc::Environment;
 
 constexpr uint32_t kMaxUnsignaledRecvStreams = 4;
 
@@ -174,7 +176,7 @@ TEST(WebRtcVoiceEngineTestStubLibrary, StartupShutdown) {
           task_queue_factory.get(), adm.get(),
           webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
           webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
-          nullptr, nullptr, trials);
+          nullptr, trials);
       engine.Init();
     }
   }
@@ -220,7 +222,7 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam<bool> {
     auto decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
     engine_.reset(new cricket::WebRtcVoiceEngine(
         task_queue_factory_.get(), adm_.get(), encoder_factory, decoder_factory,
-        nullptr, apm_, nullptr, nullptr, field_trials_));
+        nullptr, apm_, nullptr, field_trials_));
     engine_->Init();
     send_parameters_.codecs.push_back(kPcmuCodec);
     recv_parameters_.codecs.push_back(kPcmuCodec);
@@ -3678,24 +3680,18 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
   for (bool use_null_apm : {false, true}) {
     // If the VoiceEngine wants to gather available codecs early, that's fine
     // but we never want it to create a decoder at this stage.
-    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
-        webrtc::CreateDefaultTaskQueueFactory();
+    Environment env = CreateEnvironment();
     rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
         webrtc::test::MockAudioDeviceModule::CreateNice();
     rtc::scoped_refptr<webrtc::AudioProcessing> apm =
         use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
-    webrtc::FieldTrialBasedConfig field_trials;
     cricket::WebRtcVoiceEngine engine(
-        task_queue_factory.get(), adm.get(),
+        &env.task_queue_factory(), adm.get(),
         webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
         webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
-        nullptr, nullptr, field_trials);
+        nullptr, env.field_trials());
     engine.Init();
-    webrtc::RtcEventLogNull event_log;
-    CallConfig call_config(&event_log);
-    call_config.trials = &field_trials;
-    call_config.task_queue_factory = task_queue_factory.get();
-    std::unique_ptr<Call> call = Call::Create(call_config);
+    std::unique_ptr<Call> call = Call::Create(CallConfig(env));
     std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel =
         engine.CreateSendChannel(
             call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
@@ -3713,25 +3709,19 @@ TEST(WebRtcVoiceEngineTest, StartupShutdown) {
 TEST(WebRtcVoiceEngineTest, StartupShutdownWithExternalADM) {
   rtc::AutoThread main_thread;
   for (bool use_null_apm : {false, true}) {
-    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
-        webrtc::CreateDefaultTaskQueueFactory();
+    Environment env = CreateEnvironment();
     auto adm = rtc::make_ref_counted<
         ::testing::NiceMock<webrtc::test::MockAudioDeviceModule>>();
     {
       rtc::scoped_refptr<webrtc::AudioProcessing> apm =
           use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
-      webrtc::FieldTrialBasedConfig field_trials;
       cricket::WebRtcVoiceEngine engine(
-          task_queue_factory.get(), adm.get(),
+          &env.task_queue_factory(), adm.get(),
           webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
           webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
-          nullptr, nullptr, field_trials);
+          nullptr, env.field_trials());
       engine.Init();
-      webrtc::RtcEventLogNull event_log;
-      CallConfig call_config(&event_log);
-      call_config.trials = &field_trials;
-      call_config.task_queue_factory = task_queue_factory.get();
-      std::unique_ptr<Call> call = Call::Create(call_config);
+      std::unique_ptr<Call> call = Call::Create(CallConfig(env));
       std::unique_ptr<cricket::VoiceMediaSendChannelInterface> send_channel =
           engine.CreateSendChannel(
               call.get(), cricket::MediaConfig(), cricket::AudioOptions(),
@@ -3765,7 +3755,7 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
         task_queue_factory.get(), adm.get(),
         webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
         webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
-        nullptr, nullptr, field_trials);
+        nullptr, field_trials);
     engine.Init();
     for (const cricket::AudioCodec& codec : engine.send_codecs()) {
       auto is_codec = [&codec](const char* name, int clockrate = 0) {
@@ -3804,24 +3794,18 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
 TEST(WebRtcVoiceEngineTest, Has32Channels) {
   rtc::AutoThread main_thread;
   for (bool use_null_apm : {false, true}) {
-    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
-        webrtc::CreateDefaultTaskQueueFactory();
+    Environment env = CreateEnvironment();
     rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
         webrtc::test::MockAudioDeviceModule::CreateNice();
     rtc::scoped_refptr<webrtc::AudioProcessing> apm =
         use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
-    webrtc::FieldTrialBasedConfig field_trials;
     cricket::WebRtcVoiceEngine engine(
-        task_queue_factory.get(), adm.get(),
+        &env.task_queue_factory(), adm.get(),
         webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
         webrtc::MockAudioDecoderFactory::CreateUnusedFactory(), nullptr, apm,
-        nullptr, nullptr, field_trials);
+        nullptr, env.field_trials());
     engine.Init();
-    webrtc::RtcEventLogNull event_log;
-    CallConfig call_config(&event_log);
-    call_config.trials = &field_trials;
-    call_config.task_queue_factory = task_queue_factory.get();
-    std::unique_ptr<Call> call = Call::Create(call_config);
+    std::unique_ptr<Call> call = Call::Create(CallConfig(env));
 
     std::vector<std::unique_ptr<cricket::VoiceMediaSendChannelInterface>>
         channels;
@@ -3843,8 +3827,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
 TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
   rtc::AutoThread main_thread;
   for (bool use_null_apm : {false, true}) {
-    std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
-        webrtc::CreateDefaultTaskQueueFactory();
+    Environment env = CreateEnvironment();
     // TODO(ossu): I'm not sure of the intent of this test. It's either:
     // - Check that our builtin codecs are usable by Channel.
     // - The codecs provided by the engine is usable by Channel.
@@ -3856,18 +3839,13 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
         webrtc::test::MockAudioDeviceModule::CreateNice();
     rtc::scoped_refptr<webrtc::AudioProcessing> apm =
         use_null_apm ? nullptr : webrtc::AudioProcessingBuilder().Create();
-    webrtc::FieldTrialBasedConfig field_trials;
     cricket::WebRtcVoiceEngine engine(
-        task_queue_factory.get(), adm.get(),
+        &env.task_queue_factory(), adm.get(),
         webrtc::MockAudioEncoderFactory::CreateUnusedFactory(),
         webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, apm, nullptr,
-        nullptr, field_trials);
+        env.field_trials());
     engine.Init();
-    webrtc::RtcEventLogNull event_log;
-    CallConfig call_config(&event_log);
-    call_config.trials = &field_trials;
-    call_config.task_queue_factory = task_queue_factory.get();
-    std::unique_ptr<Call> call = Call::Create(call_config);
+    std::unique_ptr<Call> call = Call::Create(CallConfig(env));
     cricket::WebRtcVoiceReceiveChannel channel(
         &engine, cricket::MediaConfig(), cricket::AudioOptions(),
         webrtc::CryptoOptions(), call.get(),
@@ -3880,22 +3858,17 @@ TEST(WebRtcVoiceEngineTest, SetRecvCodecs) {
 
 TEST(WebRtcVoiceEngineTest, SetRtpSendParametersMaxBitrate) {
   rtc::AutoThread main_thread;
-  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
-      webrtc::CreateDefaultTaskQueueFactory();
+  Environment env = CreateEnvironment();
   rtc::scoped_refptr<webrtc::test::MockAudioDeviceModule> adm =
       webrtc::test::MockAudioDeviceModule::CreateNice();
-  webrtc::FieldTrialBasedConfig field_trials;
   FakeAudioSource source;
-  cricket::WebRtcVoiceEngine engine(task_queue_factory.get(), adm.get(),
+  cricket::WebRtcVoiceEngine engine(&env.task_queue_factory(), adm.get(),
                                     webrtc::CreateBuiltinAudioEncoderFactory(),
                                     webrtc::CreateBuiltinAudioDecoderFactory(),
-                                    nullptr, nullptr, nullptr, nullptr,
-                                    field_trials);
+                                    nullptr, nullptr, nullptr,
+                                    env.field_trials());
   engine.Init();
-  webrtc::RtcEventLogNull event_log;
-  CallConfig call_config(&event_log);
-  call_config.trials = &field_trials;
-  call_config.task_queue_factory = task_queue_factory.get();
+  CallConfig call_config(env);
   {
     webrtc::AudioState::Config config;
     config.audio_mixer = webrtc::AudioMixerImpl::Create();
@@ -3965,7 +3938,7 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
     webrtc::FieldTrialBasedConfig field_trials;
     cricket::WebRtcVoiceEngine engine(
         task_queue_factory.get(), adm.get(), unused_encoder_factory,
-        mock_decoder_factory, nullptr, apm, nullptr, nullptr, field_trials);
+        mock_decoder_factory, nullptr, apm, nullptr, field_trials);
     engine.Init();
     auto codecs = engine.recv_codecs();
     EXPECT_EQ(11u, codecs.size());
diff --git a/third_party/libwebrtc/media/media_channel_gn/moz.build b/third_party/libwebrtc/media/media_channel_gn/moz.build
index 1bedb41bf2..c665368568 100644
--- a/third_party/libwebrtc/media/media_channel_gn/moz.build
+++ b/third_party/libwebrtc/media/media_channel_gn/moz.build
@@ -192,16 +192,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
-if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
-
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/media_channel_impl_gn/moz.build b/third_party/libwebrtc/media/media_channel_impl_gn/moz.build
index 7d0a4bd650..27bfa53fff 100644
--- a/third_party/libwebrtc/media/media_channel_impl_gn/moz.build
+++ b/third_party/libwebrtc/media/media_channel_impl_gn/moz.build
@@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
-if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
-
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/media_constants_gn/moz.build b/third_party/libwebrtc/media/media_constants_gn/moz.build
index af4cd6b257..95a0c3a056 100644
--- a/third_party/libwebrtc/media/media_constants_gn/moz.build
+++ b/third_party/libwebrtc/media/media_constants_gn/moz.build
@@ -184,7 +184,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
@@ -194,10 +193,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
         "-msse2"
     ]
 
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/rid_description_gn/moz.build b/third_party/libwebrtc/media/rid_description_gn/moz.build
index 61afeec945..944901a1ca 100644
--- a/third_party/libwebrtc/media/rid_description_gn/moz.build
+++ b/third_party/libwebrtc/media/rid_description_gn/moz.build
@@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
-if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
-
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/rtc_media_base_gn/moz.build b/third_party/libwebrtc/media/rtc_media_base_gn/moz.build
index cfff6f3411..a5b3661adc 100644
--- a/third_party/libwebrtc/media/rtc_media_base_gn/moz.build
+++ b/third_party/libwebrtc/media/rtc_media_base_gn/moz.build
@@ -203,7 +203,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
@@ -213,10 +212,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
         "-msse2"
     ]
 
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/rtc_media_config_gn/moz.build b/third_party/libwebrtc/media/rtc_media_config_gn/moz.build
index 17afebe8da..8f3f81cc5b 100644
--- a/third_party/libwebrtc/media/rtc_media_config_gn/moz.build
+++ b/third_party/libwebrtc/media/rtc_media_config_gn/moz.build
@@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
-if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
-
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/rtc_simulcast_encoder_adapter_gn/moz.build b/third_party/libwebrtc/media/rtc_simulcast_encoder_adapter_gn/moz.build
index c09703ddd6..6b1032e1b0 100644
--- a/third_party/libwebrtc/media/rtc_simulcast_encoder_adapter_gn/moz.build
+++ b/third_party/libwebrtc/media/rtc_simulcast_encoder_adapter_gn/moz.build
@@ -200,7 +200,6 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
@@ -210,10 +209,6 @@ if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
         "-msse2"
     ]
 
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/rtp_utils_gn/moz.build b/third_party/libwebrtc/media/rtp_utils_gn/moz.build
index 1aaa347151..e2e5c11695 100644
--- a/third_party/libwebrtc/media/rtp_utils_gn/moz.build
+++ b/third_party/libwebrtc/media/rtp_utils_gn/moz.build
@@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
-if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
-
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True
diff --git a/third_party/libwebrtc/media/stream_params_gn/moz.build b/third_party/libwebrtc/media/stream_params_gn/moz.build
index 71875c4e01..1582a42c0d 100644
--- a/third_party/libwebrtc/media/stream_params_gn/moz.build
+++ b/third_party/libwebrtc/media/stream_params_gn/moz.build
@@ -176,16 +176,9 @@ if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
 if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
 
     OS_LIBS += [
-        "android_support",
         "unwind"
     ]
 
-if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
-
-    OS_LIBS += [
-        "android_support"
-    ]
-
 if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
 
     DEFINES["_GNU_SOURCE"] = True