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-rw-r--r--third_party/libwebrtc/media/BUILD.gn32
-rw-r--r--third_party/libwebrtc/media/base/codec.cc3
-rw-r--r--third_party/libwebrtc/media/base/codec.h4
-rw-r--r--third_party/libwebrtc/media/codec_gn/moz.build5
-rw-r--r--third_party/libwebrtc/media/engine/null_webrtc_video_engine_unittest.cc2
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_media_engine.cc1
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_media_engine.h8
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.cc43
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_media_engine_defaults.h24
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_media_engine_unittest.cc13
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_video_engine_unittest.cc61
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_voice_engine.cc29
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_voice_engine.h8
-rw-r--r--third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc89
-rw-r--r--third_party/libwebrtc/media/media_channel_gn/moz.build7
-rw-r--r--third_party/libwebrtc/media/media_channel_impl_gn/moz.build7
-rw-r--r--third_party/libwebrtc/media/media_constants_gn/moz.build5
-rw-r--r--third_party/libwebrtc/media/rid_description_gn/moz.build7
-rw-r--r--third_party/libwebrtc/media/rtc_media_base_gn/moz.build5
-rw-r--r--third_party/libwebrtc/media/rtc_media_config_gn/moz.build7
-rw-r--r--third_party/libwebrtc/media/rtc_simulcast_encoder_adapter_gn/moz.build5
-rw-r--r--third_party/libwebrtc/media/rtp_utils_gn/moz.build7
-rw-r--r--third_party/libwebrtc/media/stream_params_gn/moz.build7
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