diff options
Diffstat (limited to '')
10 files changed, 69 insertions, 111 deletions
diff --git a/third_party/libwebrtc/media/engine/internal_decoder_factory_unittest.cc b/third_party/libwebrtc/media/engine/internal_decoder_factory_unittest.cc index bb2e24d5d8..51d6a94dd6 100644 --- a/third_party/libwebrtc/media/engine/internal_decoder_factory_unittest.cc +++ b/third_party/libwebrtc/media/engine/internal_decoder_factory_unittest.cc @@ -43,6 +43,8 @@ constexpr bool kDav1dIsIncluded = true; #else constexpr bool kDav1dIsIncluded = false; #endif +constexpr bool kH265Enabled = false; + constexpr VideoDecoderFactory::CodecSupport kSupported = { /*is_supported=*/true, /*is_power_efficient=*/false}; constexpr VideoDecoderFactory::CodecSupport kUnsupported = { @@ -99,6 +101,14 @@ TEST(InternalDecoderFactoryTest, Av1Profile0) { } } +// At current stage since internal H.265 decoder is not implemented, +TEST(InternalDecoderFactoryTest, H265IsNotEnabled) { + InternalDecoderFactory factory; + std::unique_ptr<VideoDecoder> decoder = + factory.CreateVideoDecoder(SdpVideoFormat(cricket::kH265CodecName)); + EXPECT_EQ(static_cast<bool>(decoder), kH265Enabled); +} + #if defined(RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY) TEST(InternalDecoderFactoryTest, Av1) { InternalDecoderFactory factory; diff --git a/third_party/libwebrtc/media/engine/internal_encoder_factory_unittest.cc b/third_party/libwebrtc/media/engine/internal_encoder_factory_unittest.cc index a1c90b8cf4..b9ca6d88c2 100644 --- a/third_party/libwebrtc/media/engine/internal_encoder_factory_unittest.cc +++ b/third_party/libwebrtc/media/engine/internal_encoder_factory_unittest.cc @@ -33,6 +33,8 @@ constexpr bool kH264Enabled = true; #else constexpr bool kH264Enabled = false; #endif +constexpr bool kH265Enabled = false; + constexpr VideoEncoderFactory::CodecSupport kSupported = { /*is_supported=*/true, /*is_power_efficient=*/false}; constexpr VideoEncoderFactory::CodecSupport kUnsupported = { @@ -78,6 +80,17 @@ TEST(InternalEncoderFactoryTest, H264) { } } +// At current stage H.265 is not supported by internal encoder factory. +TEST(InternalEncoderFactoryTest, H265IsNotEnabled) { + InternalEncoderFactory factory; + std::unique_ptr<VideoEncoder> encoder = + factory.CreateVideoEncoder(SdpVideoFormat(cricket::kH265CodecName)); + EXPECT_EQ(static_cast<bool>(encoder), kH265Enabled); + EXPECT_THAT( + factory.GetSupportedFormats(), + Not(Contains(Field(&SdpVideoFormat::name, cricket::kH265CodecName)))); +} + TEST(InternalEncoderFactoryTest, QueryCodecSupportWithScalabilityMode) { InternalEncoderFactory factory; // VP8 and VP9 supported for singles spatial layers. diff --git a/third_party/libwebrtc/media/engine/simulcast_encoder_adapter_unittest.cc b/third_party/libwebrtc/media/engine/simulcast_encoder_adapter_unittest.cc index e2ac5ea390..3ee3465e13 100644 --- a/third_party/libwebrtc/media/engine/simulcast_encoder_adapter_unittest.cc +++ b/third_party/libwebrtc/media/engine/simulcast_encoder_adapter_unittest.cc @@ -586,7 +586,7 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test, absl::optional<int> last_encoded_image_simulcast_index_; std::unique_ptr<SimulcastRateAllocator> rate_allocator_; bool use_fallback_factory_; - SdpVideoFormat::Parameters sdp_video_parameters_; + CodecParameterMap sdp_video_parameters_; test::ScopedKeyValueConfig field_trials_; }; diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine.cc b/third_party/libwebrtc/media/engine/webrtc_media_engine.cc index 463ed29109..31769e05de 100644 --- a/third_party/libwebrtc/media/engine/webrtc_media_engine.cc +++ b/third_party/libwebrtc/media/engine/webrtc_media_engine.cc @@ -12,53 +12,16 @@ #include <algorithm> #include <map> -#include <memory> #include <string> #include <utility> #include "absl/algorithm/container.h" #include "absl/strings/match.h" -#include "api/transport/field_trial_based_config.h" #include "media/base/media_constants.h" -#include "media/engine/webrtc_voice_engine.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" -#ifdef HAVE_WEBRTC_VIDEO -#include "media/engine/webrtc_video_engine.h" -#else -#include "media/engine/null_webrtc_video_engine.h" -#endif - namespace cricket { - -std::unique_ptr<MediaEngineInterface> CreateMediaEngine( - MediaEngineDependencies dependencies) { - // TODO(sprang): Make populating `dependencies.trials` mandatory and remove - // these fallbacks. - std::unique_ptr<webrtc::FieldTrialsView> fallback_trials( - dependencies.trials ? nullptr : new webrtc::FieldTrialBasedConfig()); - const webrtc::FieldTrialsView& trials = - dependencies.trials ? *dependencies.trials : *fallback_trials; - auto audio_engine = std::make_unique<WebRtcVoiceEngine>( - dependencies.task_queue_factory, dependencies.adm.get(), - std::move(dependencies.audio_encoder_factory), - std::move(dependencies.audio_decoder_factory), - std::move(dependencies.audio_mixer), - std::move(dependencies.audio_processing), - std::move(dependencies.owned_audio_frame_processor), trials); -#ifdef HAVE_WEBRTC_VIDEO - auto video_engine = std::make_unique<WebRtcVideoEngine>( - std::move(dependencies.video_encoder_factory), - std::move(dependencies.video_decoder_factory), trials); -#else - auto video_engine = std::make_unique<NullWebRtcVideoEngine>(); -#endif - return std::make_unique<CompositeMediaEngine>(std::move(fallback_trials), - std::move(audio_engine), - std::move(video_engine)); -} - namespace { // Remove mutually exclusive extensions with lower priority. void DiscardRedundantExtensions( diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine.h b/third_party/libwebrtc/media/engine/webrtc_media_engine.h index 863db9f278..5bd5a8bce5 100644 --- a/third_party/libwebrtc/media/engine/webrtc_media_engine.h +++ b/third_party/libwebrtc/media/engine/webrtc_media_engine.h @@ -11,59 +11,17 @@ #ifndef MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_ #define MEDIA_ENGINE_WEBRTC_MEDIA_ENGINE_H_ -#include <memory> #include <vector> #include "absl/strings/string_view.h" #include "api/array_view.h" -#include "api/audio/audio_frame_processor.h" -#include "api/audio/audio_mixer.h" -#include "api/audio_codecs/audio_decoder_factory.h" -#include "api/audio_codecs/audio_encoder_factory.h" #include "api/field_trials_view.h" #include "api/rtp_parameters.h" -#include "api/scoped_refptr.h" -#include "api/task_queue/task_queue_factory.h" #include "api/transport/bitrate_settings.h" -#include "api/video_codecs/video_decoder_factory.h" -#include "api/video_codecs/video_encoder_factory.h" #include "media/base/codec.h" -#include "media/base/media_engine.h" -#include "modules/audio_device/include/audio_device.h" -#include "modules/audio_processing/include/audio_processing.h" -#include "rtc_base/system/rtc_export.h" namespace cricket { -struct MediaEngineDependencies { - MediaEngineDependencies() = default; - MediaEngineDependencies(const MediaEngineDependencies&) = delete; - MediaEngineDependencies(MediaEngineDependencies&&) = default; - MediaEngineDependencies& operator=(const MediaEngineDependencies&) = delete; - MediaEngineDependencies& operator=(MediaEngineDependencies&&) = default; - ~MediaEngineDependencies() = default; - - webrtc::TaskQueueFactory* task_queue_factory = nullptr; - rtc::scoped_refptr<webrtc::AudioDeviceModule> adm; - rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory; - rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory; - rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer; - rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing; - std::unique_ptr<webrtc::AudioFrameProcessor> owned_audio_frame_processor; - - std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory; - std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory; - - const webrtc::FieldTrialsView* trials = nullptr; -}; - -// 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. -[[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 // extensions (if any). diff --git a/third_party/libwebrtc/media/engine/webrtc_video_engine.cc b/third_party/libwebrtc/media/engine/webrtc_video_engine.cc index 8a9d6ff95c..a5b46d3344 100644 --- a/third_party/libwebrtc/media/engine/webrtc_video_engine.cc +++ b/third_party/libwebrtc/media/engine/webrtc_video_engine.cc @@ -1038,13 +1038,19 @@ bool WebRtcVideoSendChannel::GetChangedSenderParameters( return false; } + std::vector<VideoCodecSettings> mapped_codecs = MapCodecs(params.codecs); + if (mapped_codecs.empty()) { + // This suggests a failure in MapCodecs, e.g. inconsistent RTX codecs. + return false; + } + std::vector<VideoCodecSettings> negotiated_codecs = - SelectSendVideoCodecs(MapCodecs(params.codecs)); + SelectSendVideoCodecs(mapped_codecs); - // We should only fail here if send direction is enabled. if (params.is_stream_active && negotiated_codecs.empty()) { - RTC_LOG(LS_ERROR) << "No video codecs supported."; - return false; + // This is not a failure but will lead to the answer being rejected. + RTC_LOG(LS_ERROR) << "No video codecs in common."; + return true; } // Never enable sending FlexFEC, unless we are in the experiment. 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 f5736679be..148223f497 100644 --- a/third_party/libwebrtc/media/engine/webrtc_video_engine_unittest.cc +++ b/third_party/libwebrtc/media/engine/webrtc_video_engine_unittest.cc @@ -3781,7 +3781,7 @@ class Vp9SettingsTest : public WebRtcVideoChannelTest { TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) { encoder_factory_->AddSupportedVideoCodec( - webrtc::SdpVideoFormat("VP9", webrtc::SdpVideoFormat::Parameters(), + webrtc::SdpVideoFormat("VP9", webrtc::CodecParameterMap(), {ScalabilityMode::kL1T1, ScalabilityMode::kL2T1})); cricket::VideoSenderParameters parameters; @@ -8545,7 +8545,7 @@ TEST_F(WebRtcVideoChannelTest, FallbackForUnsetOrUnsupportedScalabilityMode) { ScalabilityMode::kL1T3}; encoder_factory_->AddSupportedVideoCodec(webrtc::SdpVideoFormat( - "VP8", webrtc::SdpVideoFormat::Parameters(), kSupportedModes)); + "VP8", webrtc::CodecParameterMap(), kSupportedModes)); FakeVideoSendStream* stream = SetUpSimulcast(true, /*with_rtx=*/false); @@ -8615,9 +8615,9 @@ TEST_F(WebRtcVideoChannelTest, kVp9SupportedModes = {ScalabilityMode::kL3T3}; encoder_factory_->AddSupportedVideoCodec(webrtc::SdpVideoFormat( - "VP8", webrtc::SdpVideoFormat::Parameters(), {ScalabilityMode::kL1T1})); + "VP8", webrtc::CodecParameterMap(), {ScalabilityMode::kL1T1})); encoder_factory_->AddSupportedVideoCodec(webrtc::SdpVideoFormat( - "VP9", webrtc::SdpVideoFormat::Parameters(), {ScalabilityMode::kL3T3})); + "VP9", webrtc::CodecParameterMap(), {ScalabilityMode::kL3T3})); cricket::VideoSenderParameters send_parameters; send_parameters.codecs.push_back(GetEngineCodec("VP9")); diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc b/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc index adf662074d..fcc2703f98 100644 --- a/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc +++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine.cc @@ -377,9 +377,8 @@ void WebRtcVoiceEngine::Init() { // TaskQueue expects to be created/destroyed on the same thread. RTC_DCHECK(!low_priority_worker_queue_); - low_priority_worker_queue_.reset( - new rtc::TaskQueue(task_queue_factory_->CreateTaskQueue( - "rtc-low-prio", webrtc::TaskQueueFactory::Priority::LOW))); + low_priority_worker_queue_ = task_queue_factory_->CreateTaskQueue( + "rtc-low-prio", webrtc::TaskQueueFactory::Priority::LOW); // Load our audio codec lists. RTC_LOG(LS_VERBOSE) << "Supported send codecs in order of preference:"; @@ -761,9 +760,9 @@ std::vector<AudioCodec> WebRtcVoiceEngine::CollectCodecs( out.push_back(codec); if (codec.name == kOpusCodecName) { - std::string redFmtp = + std::string red_fmtp = rtc::ToString(codec.id) + "/" + rtc::ToString(codec.id); - map_format({kRedCodecName, 48000, 2, {{"", redFmtp}}}, &out); + map_format({kRedCodecName, 48000, 2, {{"", red_fmtp}}}, &out); } } } @@ -1318,7 +1317,7 @@ bool WebRtcVoiceSendChannel::SetSenderParameters( } } - if (!SetMaxSendBitrate(params.max_bandwidth_bps)) { + if (send_codec_spec_ && !SetMaxSendBitrate(params.max_bandwidth_bps)) { return false; } return SetOptions(params.options); @@ -1402,7 +1401,8 @@ bool WebRtcVoiceSendChannel::SetSendCodecs( } if (!send_codec_spec) { - return false; + // No codecs in common, bail out early. + return true; } RTC_DCHECK(voice_codec_info); diff --git a/third_party/libwebrtc/media/engine/webrtc_voice_engine.h b/third_party/libwebrtc/media/engine/webrtc_voice_engine.h index ed71667525..b28b9652bb 100644 --- a/third_party/libwebrtc/media/engine/webrtc_voice_engine.h +++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine.h @@ -66,7 +66,6 @@ #include "rtc_base/network/sent_packet.h" #include "rtc_base/network_route.h" #include "rtc_base/system/file_wrapper.h" -#include "rtc_base/task_queue.h" namespace webrtc { class AudioFrameProcessor; @@ -141,7 +140,8 @@ class WebRtcVoiceEngine final : public VoiceEngineInterface { void ApplyOptions(const AudioOptions& options); webrtc::TaskQueueFactory* const task_queue_factory_; - std::unique_ptr<rtc::TaskQueue> low_priority_worker_queue_; + std::unique_ptr<webrtc::TaskQueueBase, webrtc::TaskQueueDeleter> + low_priority_worker_queue_; webrtc::AudioDeviceModule* adm(); webrtc::AudioProcessing* apm() const; 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 4d6580631d..8ae441bc69 100644 --- a/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc +++ b/third_party/libwebrtc/media/engine/webrtc_voice_engine_unittest.cc @@ -1702,27 +1702,29 @@ TEST_P(WebRtcVoiceEngineTestFake, DontRecreateSendStream) { // TODO(ossu): Revisit if these tests need to be here, now that these kinds of // tests should be available in AudioEncoderOpusTest. -// Test that if clockrate is not 48000 for opus, we fail. +// Test that if clockrate is not 48000 for opus, we do not have a send codec. TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBadClockrate) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSenderParameter parameters; parameters.codecs.push_back(kOpusCodec); parameters.codecs[0].bitrate = 0; parameters.codecs[0].clockrate = 50000; - EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); + EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt); } -// Test that if channels=0 for opus, we fail. +// Test that if channels=0 for opus, we do not have a send codec. TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0ChannelsNoStereo) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSenderParameter parameters; parameters.codecs.push_back(kOpusCodec); parameters.codecs[0].bitrate = 0; parameters.codecs[0].channels = 0; - EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); + EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt); } -// Test that if channels=0 for opus, we fail. +// Test that if channels=0 for opus, we do not have a send codec. TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSenderParameter parameters; @@ -1730,20 +1732,23 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad0Channels1Stereo) { parameters.codecs[0].bitrate = 0; parameters.codecs[0].channels = 0; parameters.codecs[0].params["stereo"] = "1"; - EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); + EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt); } -// Test that if channel is 1 for opus and there's no stereo, we fail. +// Test that if channel is 1 for opus and there's no stereo, we do not have a +// send codec. TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpus1ChannelNoStereo) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSenderParameter parameters; parameters.codecs.push_back(kOpusCodec); parameters.codecs[0].bitrate = 0; parameters.codecs[0].channels = 1; - EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); + EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt); } -// Test that if channel is 1 for opus and stereo=0, we fail. +// Test that if channel is 1 for opus and stereo=0, we do not have a send codec. TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSenderParameter parameters; @@ -1751,10 +1756,11 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel0Stereo) { parameters.codecs[0].bitrate = 0; parameters.codecs[0].channels = 1; parameters.codecs[0].params["stereo"] = "0"; - EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); + EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt); } -// Test that if channel is 1 for opus and stereo=1, we fail. +// Test that if channel is 1 for opus and stereo=1, we do not have a send codec. TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSenderParameter parameters; @@ -1762,7 +1768,8 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecOpusBad1Channel1Stereo) { parameters.codecs[0].bitrate = 0; parameters.codecs[0].channels = 1; parameters.codecs[0].params["stereo"] = "1"; - EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); + EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt); } // Test that with bitrate=0 and no stereo, bitrate is 32000. @@ -2035,11 +2042,12 @@ TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) { } } -// Test that we fail if no codecs are specified. +// Test that we do not fail if no codecs are specified. TEST_P(WebRtcVoiceEngineTestFake, SetSendCodecsNoCodecs) { EXPECT_TRUE(SetupSendStream()); cricket::AudioSenderParameter parameters; - EXPECT_FALSE(send_channel_->SetSenderParameters(parameters)); + EXPECT_TRUE(send_channel_->SetSenderParameters(parameters)); + EXPECT_EQ(send_channel_->GetSendCodec(), absl::nullopt); } // Test that we can set send codecs even with telephone-event codec as the first |