diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index 36ab5f5196..ecbf9bbcb5 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc @@ -33,7 +33,7 @@ namespace webrtc { -std::string AudioReceiveStream::Config::Rtp::ToString() const { +std::string AudioReceiveStreamInterface::Config::Rtp::ToString() const { char ss_buf[1024]; rtc::SimpleStringBuilder ss(ss_buf); ss << "{remote_ssrc: " << remote_ssrc; @@ -52,7 +52,7 @@ std::string AudioReceiveStream::Config::Rtp::ToString() const { return ss.str(); } -std::string AudioReceiveStream::Config::ToString() const { +std::string AudioReceiveStreamInterface::Config::ToString() const { char ss_buf[1024]; rtc::SimpleStringBuilder ss(ss_buf); ss << "{rtp: " << rtp.ToString(); @@ -70,7 +70,7 @@ std::unique_ptr CreateChannelReceive( Clock* clock, webrtc::AudioState* audio_state, NetEqFactory* neteq_factory, - const webrtc::AudioReceiveStream::Config& config, + const webrtc::AudioReceiveStreamInterface::Config& config, RtcEventLog* event_log) { RTC_DCHECK(audio_state); internal::AudioState* internal_audio_state = @@ -90,7 +90,7 @@ AudioReceiveStreamImpl::AudioReceiveStreamImpl( Clock* clock, PacketRouter* packet_router, NetEqFactory* neteq_factory, - const webrtc::AudioReceiveStream::Config& config, + const webrtc::AudioReceiveStreamInterface::Config& config, const rtc::scoped_refptr& audio_state, webrtc::RtcEventLog* event_log) : AudioReceiveStreamImpl(clock, @@ -107,7 +107,7 @@ AudioReceiveStreamImpl::AudioReceiveStreamImpl( AudioReceiveStreamImpl::AudioReceiveStreamImpl( Clock* clock, PacketRouter* packet_router, - const webrtc::AudioReceiveStream::Config& config, + const webrtc::AudioReceiveStreamInterface::Config& config, const rtc::scoped_refptr& audio_state, webrtc::RtcEventLog* event_log, std::unique_ptr channel_receive) @@ -164,7 +164,7 @@ void AudioReceiveStreamImpl::UnregisterFromTransport() { } void AudioReceiveStreamImpl::ReconfigureForTesting( - const webrtc::AudioReceiveStream::Config& config) { + const webrtc::AudioReceiveStreamInterface::Config& config) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); // SSRC can't be changed mid-stream. @@ -279,10 +279,10 @@ RtpHeaderExtensionMap AudioReceiveStreamImpl::GetRtpExtensionMap() const { return RtpHeaderExtensionMap(config_.rtp.extensions); } -webrtc::AudioReceiveStream::Stats AudioReceiveStreamImpl::GetStats( +webrtc::AudioReceiveStreamInterface::Stats AudioReceiveStreamImpl::GetStats( bool get_and_clear_legacy_stats) const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); - webrtc::AudioReceiveStream::Stats stats; + webrtc::AudioReceiveStreamInterface::Stats stats; stats.remote_ssrc = remote_ssrc(); webrtc::CallReceiveStatistics call_stats = diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h index 4b8ef9f685..6debdec7dd 100644 --- a/audio/audio_receive_stream.h +++ b/audio/audio_receive_stream.h @@ -42,7 +42,7 @@ namespace internal { class AudioSendStream; } // namespace internal -class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStream, +class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStreamInterface, public AudioMixer::Source, public Syncable { public: @@ -50,14 +50,14 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStream, Clock* clock, PacketRouter* packet_router, NetEqFactory* neteq_factory, - const webrtc::AudioReceiveStream::Config& config, + const webrtc::AudioReceiveStreamInterface::Config& config, const rtc::scoped_refptr& audio_state, webrtc::RtcEventLog* event_log); // For unit tests, which need to supply a mock channel receive. AudioReceiveStreamImpl( Clock* clock, PacketRouter* packet_router, - const webrtc::AudioReceiveStream::Config& config, + const webrtc::AudioReceiveStreamInterface::Config& config, const rtc::scoped_refptr& audio_state, webrtc::RtcEventLog* event_log, std::unique_ptr channel_receive); @@ -82,7 +82,7 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStream, // network thread. void UnregisterFromTransport(); - // webrtc::AudioReceiveStream implementation. + // webrtc::AudioReceiveStreamInterface implementation. void Start() override; void Stop() override; bool transport_cc() const override; @@ -100,7 +100,7 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStream, const std::vector& GetRtpExtensions() const override; RtpHeaderExtensionMap GetRtpExtensionMap() const override; - webrtc::AudioReceiveStream::Stats GetStats( + webrtc::AudioReceiveStreamInterface::Stats GetStats( bool get_and_clear_legacy_stats) const override; void SetSink(AudioSinkInterface* sink) override; void SetGain(float gain) override; @@ -145,7 +145,8 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStream, const AudioSendStream* GetAssociatedSendStreamForTesting() const; // TODO(tommi): Remove this method. - void ReconfigureForTesting(const webrtc::AudioReceiveStream::Config& config); + void ReconfigureForTesting( + const webrtc::AudioReceiveStreamInterface::Config& config); private: internal::AudioState* audio_state() const; @@ -159,7 +160,7 @@ class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStream, // that belong to the network thread. Once the packets are fully delivered // on the network thread, this comment will be deleted. RTC_NO_UNIQUE_ADDRESS SequenceChecker packet_sequence_checker_; - webrtc::AudioReceiveStream::Config config_; + webrtc::AudioReceiveStreamInterface::Config config_; rtc::scoped_refptr audio_state_; SourceTracker source_tracker_; const std::unique_ptr channel_receive_; diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc index ab80c4582b..62d7180805 100644 --- a/audio/audio_receive_stream_unittest.cc +++ b/audio/audio_receive_stream_unittest.cc @@ -155,7 +155,7 @@ struct ConfigHelper { return ret; } - AudioReceiveStream::Config& config() { return stream_config_; } + AudioReceiveStreamInterface::Config& config() { return stream_config_; } rtc::scoped_refptr audio_mixer() { return audio_mixer_; } MockChannelReceive* channel_receive() { return channel_receive_; } @@ -189,7 +189,7 @@ struct ConfigHelper { MockRtcEventLog event_log_; rtc::scoped_refptr audio_state_; rtc::scoped_refptr audio_mixer_; - AudioReceiveStream::Config stream_config_; + AudioReceiveStreamInterface::Config stream_config_; ::testing::StrictMock* channel_receive_ = nullptr; RtpStreamReceiverController rtp_stream_receiver_controller_; MockTransport rtcp_send_transport_; @@ -209,7 +209,7 @@ const std::vector CreateRtcpSenderReport() { } // namespace TEST(AudioReceiveStreamTest, ConfigToString) { - AudioReceiveStream::Config config; + AudioReceiveStreamInterface::Config config; config.rtp.remote_ssrc = kRemoteSsrc; config.rtp.local_ssrc = kLocalSsrc; config.rtp.extensions.push_back( @@ -249,7 +249,7 @@ TEST(AudioReceiveStreamTest, GetStats) { ConfigHelper helper(use_null_audio_processing); auto recv_stream = helper.CreateAudioReceiveStream(); helper.SetupMockForGetStats(); - AudioReceiveStream::Stats stats = + AudioReceiveStreamInterface::Stats stats = recv_stream->GetStats(/*get_and_clear_legacy_stats=*/true); EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc); EXPECT_EQ(kCallStats.payload_bytes_rcvd, stats.payload_bytes_rcvd); diff --git a/audio/audio_send_stream_tests.cc b/audio/audio_send_stream_tests.cc index e3895039d8..64eb5f610a 100644 --- a/audio/audio_send_stream_tests.cc +++ b/audio/audio_send_stream_tests.cc @@ -61,9 +61,9 @@ TEST_F(AudioSendStreamCallTest, SupportsCName) { return SEND_PACKET; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->rtp.c_name = kCName; } @@ -90,9 +90,9 @@ TEST_F(AudioSendStreamCallTest, NoExtensionsByDefault) { return SEND_PACKET; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->rtp.extensions.clear(); } @@ -129,9 +129,9 @@ TEST_F(AudioSendStreamCallTest, SupportsAudioLevel) { return SEND_PACKET; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->rtp.extensions.clear(); send_config->rtp.extensions.push_back( RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelExtensionId)); @@ -171,9 +171,9 @@ class TransportWideSequenceNumberObserver : public AudioSendTest { return SEND_PACKET; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->rtp.extensions.clear(); send_config->rtp.extensions.push_back( RtpExtension(RtpExtension::kTransportSequenceNumberUri, @@ -223,9 +223,9 @@ TEST_F(AudioSendStreamCallTest, SendDtmf) { return SEND_PACKET; } - void OnAudioStreamsCreated( - AudioSendStream* send_stream, - const std::vector& receive_streams) override { + void OnAudioStreamsCreated(AudioSendStream* send_stream, + const std::vector& + receive_streams) override { // Need to start stream here, else DTMF events are dropped. send_stream->Start(); for (int event = kDtmfEventFirst; event <= kDtmfEventLast; ++event) { diff --git a/audio/audio_state.cc b/audio/audio_state.cc index 4215bcb8dd..bd7cde77b2 100644 --- a/audio/audio_state.cc +++ b/audio/audio_state.cc @@ -50,7 +50,8 @@ AudioTransport* AudioState::audio_transport() { return &audio_transport_; } -void AudioState::AddReceivingStream(webrtc::AudioReceiveStream* stream) { +void AudioState::AddReceivingStream( + webrtc::AudioReceiveStreamInterface* stream) { RTC_DCHECK(thread_checker_.IsCurrent()); RTC_DCHECK_EQ(0, receiving_streams_.count(stream)); receiving_streams_.insert(stream); @@ -73,7 +74,8 @@ void AudioState::AddReceivingStream(webrtc::AudioReceiveStream* stream) { } } -void AudioState::RemoveReceivingStream(webrtc::AudioReceiveStream* stream) { +void AudioState::RemoveReceivingStream( + webrtc::AudioReceiveStreamInterface* stream) { RTC_DCHECK(thread_checker_.IsCurrent()); auto count = receiving_streams_.erase(stream); RTC_DCHECK_EQ(1, count); diff --git a/audio/audio_state.h b/audio/audio_state.h index b8ef4fd978..f9d9d72b89 100644 --- a/audio/audio_state.h +++ b/audio/audio_state.h @@ -24,7 +24,7 @@ namespace webrtc { class AudioSendStream; -class AudioReceiveStream; +class AudioReceiveStreamInterface; namespace internal { @@ -51,8 +51,8 @@ class AudioState : public webrtc::AudioState { return config_.audio_device_module.get(); } - void AddReceivingStream(webrtc::AudioReceiveStream* stream); - void RemoveReceivingStream(webrtc::AudioReceiveStream* stream); + void AddReceivingStream(webrtc::AudioReceiveStreamInterface* stream); + void RemoveReceivingStream(webrtc::AudioReceiveStreamInterface* stream); void AddSendingStream(webrtc::AudioSendStream* stream, int sample_rate_hz, @@ -78,7 +78,7 @@ class AudioState : public webrtc::AudioState { // stats are still updated. std::unique_ptr null_audio_poller_; - webrtc::flat_set receiving_streams_; + webrtc::flat_set receiving_streams_; struct StreamProperties { int sample_rate_hz = 0; size_t num_channels = 0; diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc index fcf3dc1dd2..0bf60c0a7e 100644 --- a/audio/channel_receive.cc +++ b/audio/channel_receive.cc @@ -317,13 +317,13 @@ void ChannelReceive::OnReceivedPayloadData( // packet as discarded. // If we have a source_tracker_, tell it that the frame has been - // "delivered". Normally, this happens in AudioReceiveStream when audio - // frames are pulled out, but when playout is muted, nothing is pulling - // frames. The downside of this approach is that frames delivered this way - // won't be delayed for playout, and therefore will be unsynchronized with - // (a) audio delay when playing and (b) any audio/video synchronization. But - // the alternative is that muting playout also stops the SourceTracker from - // updating RtpSource information. + // "delivered". Normally, this happens in AudioReceiveStreamInterface when + // audio frames are pulled out, but when playout is muted, nothing is + // pulling frames. The downside of this approach is that frames delivered + // this way won't be delayed for playout, and therefore will be + // unsynchronized with (a) audio delay when playing and (b) any audio/video + // synchronization. But the alternative is that muting playout also stops + // the SourceTracker from updating RtpSource information. if (source_tracker_) { RtpPacketInfos::vector_type packet_vector = { RtpPacketInfo(rtpHeader, clock_->CurrentTime())}; diff --git a/audio/channel_receive.h b/audio/channel_receive.h index 1c3b192a62..ae9cd59db8 100644 --- a/audio/channel_receive.h +++ b/audio/channel_receive.h @@ -83,7 +83,7 @@ namespace voe { class ChannelSendInterface; -// Interface class needed for AudioReceiveStream tests that use a +// Interface class needed for AudioReceiveStreamInterface tests that use a // MockChannelReceive. class ChannelReceiveInterface : public RtpPacketSinkInterface { diff --git a/audio/test/audio_bwe_integration_test.cc b/audio/test/audio_bwe_integration_test.cc index f9953955df..23b8183db4 100644 --- a/audio/test/audio_bwe_integration_test.cc +++ b/audio/test/audio_bwe_integration_test.cc @@ -105,9 +105,9 @@ class NoBandwidthDropAfterDtx : public AudioBweTest { NoBandwidthDropAfterDtx() : sender_call_(nullptr), stats_poller_("stats poller task queue") {} - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( test::CallTest::kAudioSendPayloadType, {"OPUS", @@ -120,7 +120,7 @@ class NoBandwidthDropAfterDtx : public AudioBweTest { send_config->rtp.extensions.push_back( RtpExtension(RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberExtensionId)); - for (AudioReceiveStream::Config& recv_config : *receive_configs) { + for (AudioReceiveStreamInterface::Config& recv_config : *receive_configs) { recv_config.rtp.transport_cc = true; recv_config.rtp.extensions = send_config->rtp.extensions; recv_config.rtp.remote_ssrc = send_config->rtp.ssrc; diff --git a/audio/test/audio_end_to_end_test.cc b/audio/test/audio_end_to_end_test.cc index 0d8529a913..b3e2cf1606 100644 --- a/audio/test/audio_end_to_end_test.cc +++ b/audio/test/audio_end_to_end_test.cc @@ -86,7 +86,7 @@ AudioEndToEndTest::CreateReceiveTransport(TaskQueueBase* task_queue) { void AudioEndToEndTest::ModifyAudioConfigs( AudioSendStream::Config* send_config, - std::vector* receive_configs) { + std::vector* receive_configs) { // Large bitrate by default. const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2, {{"stereo", "1"}}); @@ -98,7 +98,7 @@ void AudioEndToEndTest::ModifyAudioConfigs( void AudioEndToEndTest::OnAudioStreamsCreated( AudioSendStream* send_stream, - const std::vector& receive_streams) { + const std::vector& receive_streams) { ASSERT_NE(nullptr, send_stream); ASSERT_EQ(1u, receive_streams.size()); ASSERT_NE(nullptr, receive_streams[0]); diff --git a/audio/test/audio_end_to_end_test.h b/audio/test/audio_end_to_end_test.h index c47cb47076..6afa0baea3 100644 --- a/audio/test/audio_end_to_end_test.h +++ b/audio/test/audio_end_to_end_test.h @@ -28,7 +28,9 @@ class AudioEndToEndTest : public test::EndToEndTest { protected: TestAudioDeviceModule* send_audio_device() { return send_audio_device_; } const AudioSendStream* send_stream() const { return send_stream_; } - const AudioReceiveStream* receive_stream() const { return receive_stream_; } + const AudioReceiveStreamInterface* receive_stream() const { + return receive_stream_; + } virtual BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const; @@ -49,19 +51,19 @@ class AudioEndToEndTest : public test::EndToEndTest { std::unique_ptr CreateReceiveTransport( TaskQueueBase* task_queue) override; - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override; - void OnAudioStreamsCreated( - AudioSendStream* send_stream, - const std::vector& receive_streams) override; + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override; + void OnAudioStreamsCreated(AudioSendStream* send_stream, + const std::vector& + receive_streams) override; void PerformTest() override; private: TestAudioDeviceModule* send_audio_device_ = nullptr; AudioSendStream* send_stream_ = nullptr; - AudioReceiveStream* receive_stream_ = nullptr; + AudioReceiveStreamInterface* receive_stream_ = nullptr; }; } // namespace test diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc index 8f599b0213..febcb066fd 100644 --- a/audio/test/audio_stats_test.cc +++ b/audio/test/audio_stats_test.cc @@ -64,7 +64,7 @@ class NoLossTest : public AudioEndToEndTest { EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood); EXPECT_FALSE(send_stats.apm_statistics.residual_echo_likelihood_recent_max); - AudioReceiveStream::Stats recv_stats = + AudioReceiveStreamInterface::Stats recv_stats = receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true); EXPECT_PRED2(IsNear, kBytesSent, recv_stats.payload_bytes_rcvd); EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd); diff --git a/audio/test/low_bandwidth_audio_test.cc b/audio/test/low_bandwidth_audio_test.cc index 50cf499920..948dcbc8f2 100644 --- a/audio/test/low_bandwidth_audio_test.cc +++ b/audio/test/low_bandwidth_audio_test.cc @@ -73,9 +73,9 @@ class AudioQualityTest : public AudioEndToEndTest { }; class Mobile2GNetworkTest : public AudioQualityTest { - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec( test::CallTest::kAudioSendPayloadType, {"OPUS", diff --git a/audio/test/nack_test.cc b/audio/test/nack_test.cc index 13cfe74a28..f383627dbe 100644 --- a/audio/test/nack_test.cc +++ b/audio/test/nack_test.cc @@ -32,9 +32,9 @@ TEST_F(NackTest, ShouldNackInLossyNetwork) { return pipe_config; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { ASSERT_EQ(receive_configs->size(), 1U); (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackHistoryMs; AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs); @@ -43,7 +43,7 @@ TEST_F(NackTest, ShouldNackInLossyNetwork) { void PerformTest() override { SleepMs(kTestDurationMs); } void OnStreamsStopped() override { - AudioReceiveStream::Stats recv_stats = + AudioReceiveStreamInterface::Stats recv_stats = receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true); EXPECT_GT(recv_stats.nacks_sent, 0U); AudioSendStream::Stats send_stats = send_stream()->GetStats(); diff --git a/audio/test/non_sender_rtt_test.cc b/audio/test/non_sender_rtt_test.cc index 5c5b15eecf..07de99ac37 100644 --- a/audio/test/non_sender_rtt_test.cc +++ b/audio/test/non_sender_rtt_test.cc @@ -29,9 +29,9 @@ TEST_F(NonSenderRttTest, NonSenderRttStats) { return pipe_config; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { ASSERT_EQ(receive_configs->size(), 1U); (*receive_configs)[0].enable_non_sender_rtt = true; AudioEndToEndTest::ModifyAudioConfigs(send_config, receive_configs); @@ -41,7 +41,7 @@ TEST_F(NonSenderRttTest, NonSenderRttStats) { void PerformTest() override { SleepMs(kTestDurationMs); } void OnStreamsStopped() override { - AudioReceiveStream::Stats recv_stats = + AudioReceiveStreamInterface::Stats recv_stats = receive_stream()->GetStats(/*get_and_clear_legacy_stats=*/true); EXPECT_GT(recv_stats.round_trip_time_measurements, 0); ASSERT_TRUE(recv_stats.round_trip_time.has_value()); diff --git a/call/audio_receive_stream.cc b/call/audio_receive_stream.cc index c3c2ac77d0..0766eb6bbb 100644 --- a/call/audio_receive_stream.cc +++ b/call/audio_receive_stream.cc @@ -12,13 +12,13 @@ namespace webrtc { -AudioReceiveStream::Stats::Stats() = default; -AudioReceiveStream::Stats::~Stats() = default; +AudioReceiveStreamInterface::Stats::Stats() = default; +AudioReceiveStreamInterface::Stats::~Stats() = default; -AudioReceiveStream::Config::Config() = default; -AudioReceiveStream::Config::~Config() = default; +AudioReceiveStreamInterface::Config::Config() = default; +AudioReceiveStreamInterface::Config::~Config() = default; -AudioReceiveStream::Config::Rtp::Rtp() = default; -AudioReceiveStream::Config::Rtp::~Rtp() = default; +AudioReceiveStreamInterface::Config::Rtp::Rtp() = default; +AudioReceiveStreamInterface::Config::Rtp::~Rtp() = default; } // namespace webrtc diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h index 2a6ed6079e..3c521edbab 100644 --- a/call/audio_receive_stream.h +++ b/call/audio_receive_stream.h @@ -27,7 +27,7 @@ namespace webrtc { class AudioSinkInterface; -class AudioReceiveStream : public MediaReceiveStreamInterface { +class AudioReceiveStreamInterface : public MediaReceiveStreamInterface { public: struct Stats { Stats(); @@ -147,14 +147,14 @@ class AudioReceiveStream : public MediaReceiveStreamInterface { // decrypted in whatever way the caller choses. This is not required by // default. // TODO(tommi): Remove this member variable from the struct. It's not - // a part of the AudioReceiveStream state but rather a pass through + // a part of the AudioReceiveStreamInterface state but rather a pass through // variable. rtc::scoped_refptr frame_decryptor; // An optional frame transformer used by insertable streams to transform // encoded frames. // TODO(tommi): Remove this member variable from the struct. It's not - // a part of the AudioReceiveStream state but rather a pass through + // a part of the AudioReceiveStreamInterface state but rather a pass through // variable. rtc::scoped_refptr frame_transformer; }; @@ -205,8 +205,14 @@ class AudioReceiveStream : public MediaReceiveStreamInterface { virtual const std::vector& GetRtpExtensions() const = 0; protected: - virtual ~AudioReceiveStream() {} + virtual ~AudioReceiveStreamInterface() {} }; + +// TODO(bugs.webrtc.org/7484): Remove this once downstream usage of the +// deprecated name is gone. +using AudioReceiveStream [[deprecated("Use AudioReceiveStreamInterface")]] = + AudioReceiveStreamInterface; + } // namespace webrtc #endif // CALL_AUDIO_RECEIVE_STREAM_H_ diff --git a/call/call.cc b/call/call.cc index 71fea0644b..73efe6f7de 100644 --- a/call/call.cc +++ b/call/call.cc @@ -134,7 +134,7 @@ std::unique_ptr CreateRtcLogStreamConfig( } std::unique_ptr CreateRtcLogStreamConfig( - const AudioReceiveStream::Config& config) { + const AudioReceiveStreamInterface::Config& config) { auto rtclog_config = std::make_unique(); rtclog_config->remote_ssrc = config.rtp.remote_ssrc; rtclog_config->local_ssrc = config.rtp.local_ssrc; @@ -217,10 +217,10 @@ class Call final : public webrtc::Call, const webrtc::AudioSendStream::Config& config) override; void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; - webrtc::AudioReceiveStream* CreateAudioReceiveStream( - const webrtc::AudioReceiveStream::Config& config) override; + webrtc::AudioReceiveStreamInterface* CreateAudioReceiveStream( + const webrtc::AudioReceiveStreamInterface::Config& config) override; void DestroyAudioReceiveStream( - webrtc::AudioReceiveStream* receive_stream) override; + webrtc::AudioReceiveStreamInterface* receive_stream) override; webrtc::VideoSendStream* CreateVideoSendStream( webrtc::VideoSendStream::Config config, @@ -265,14 +265,14 @@ class Call final : public webrtc::Call, void OnAudioTransportOverheadChanged( int transport_overhead_per_packet) override; - void OnLocalSsrcUpdated(webrtc::AudioReceiveStream& stream, + void OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(FlexfecReceiveStream& stream, uint32_t local_ssrc) override; - void OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream, + void OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream, absl::string_view sync_group) override; void OnSentPacket(const rtc::SentPacket& sent_packet) override; @@ -966,8 +966,8 @@ void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { delete send_stream; } -webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( - const webrtc::AudioReceiveStream::Config& config) { +webrtc::AudioReceiveStreamInterface* Call::CreateAudioReceiveStream( + const webrtc::AudioReceiveStreamInterface::Config& config) { TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream"); RTC_DCHECK_RUN_ON(worker_thread_); EnsureStarted(); @@ -1001,7 +1001,7 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream( } void Call::DestroyAudioReceiveStream( - webrtc::AudioReceiveStream* receive_stream) { + webrtc::AudioReceiveStreamInterface* receive_stream) { TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream"); RTC_DCHECK_RUN_ON(worker_thread_); RTC_DCHECK(receive_stream != nullptr); @@ -1380,7 +1380,7 @@ void Call::UpdateAggregateNetworkState() { transport_send_->OnNetworkAvailability(aggregate_network_up); } -void Call::OnLocalSsrcUpdated(webrtc::AudioReceiveStream& stream, +void Call::OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream, uint32_t local_ssrc) { RTC_DCHECK_RUN_ON(worker_thread_); webrtc::AudioReceiveStreamImpl& receive_stream = @@ -1404,7 +1404,7 @@ void Call::OnLocalSsrcUpdated(FlexfecReceiveStream& stream, static_cast(stream).SetLocalSsrc(local_ssrc); } -void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream, +void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream, absl::string_view sync_group) { RTC_DCHECK_RUN_ON(worker_thread_); webrtc::AudioReceiveStreamImpl& receive_stream = diff --git a/call/call.h b/call/call.h index bdbf7bd29a..a4d5974898 100644 --- a/call/call.h +++ b/call/call.h @@ -105,10 +105,10 @@ class Call { virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0; - virtual AudioReceiveStream* CreateAudioReceiveStream( - const AudioReceiveStream::Config& config) = 0; + virtual AudioReceiveStreamInterface* CreateAudioReceiveStream( + const AudioReceiveStreamInterface::Config& config) = 0; virtual void DestroyAudioReceiveStream( - AudioReceiveStream* receive_stream) = 0; + AudioReceiveStreamInterface* receive_stream) = 0; virtual VideoSendStream* CreateVideoSendStream( VideoSendStream::Config config, @@ -164,14 +164,14 @@ class Call { // Called when a receive stream's local ssrc has changed and association with // send streams needs to be updated. - virtual void OnLocalSsrcUpdated(AudioReceiveStream& stream, + virtual void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream, uint32_t local_ssrc) = 0; virtual void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream, uint32_t local_ssrc) = 0; virtual void OnLocalSsrcUpdated(FlexfecReceiveStream& stream, uint32_t local_ssrc) = 0; - virtual void OnUpdateSyncGroup(AudioReceiveStream& stream, + virtual void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream, absl::string_view sync_group) = 0; virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0; diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc index 81e51dcb90..db384c189f 100644 --- a/call/call_perf_tests.cc +++ b/call/call_perf_tests.cc @@ -190,7 +190,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, std::unique_ptr receive_transport; AudioSendStream* audio_send_stream; - AudioReceiveStream* audio_receive_stream; + AudioReceiveStreamInterface* audio_receive_stream; std::unique_ptr drifting_clock; SendTask(RTC_FROM_HERE, task_queue(), [&]() { @@ -273,7 +273,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, video_receive_configs_[0].renderer = observer.get(); video_receive_configs_[0].sync_group = kSyncGroup; - AudioReceiveStream::Config audio_recv_config; + AudioReceiveStreamInterface::Config audio_recv_config; audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc; audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc; audio_recv_config.rtcp_send_transport = receive_transport.get(); @@ -1021,9 +1021,9 @@ void CallPerfTest::TestMinAudioVideoBitrate(int test_bitrate_from, size_t GetNumAudioStreams() const override { return 1; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->send_codec_spec->target_bitrate_bps = absl::optional(kOpusBitrateFbBps); } diff --git a/call/call_unittest.cc b/call/call_unittest.cc index ee243ff34c..5d30f45dc5 100644 --- a/call/call_unittest.cc +++ b/call/call_unittest.cc @@ -114,13 +114,14 @@ TEST(CallTest, CreateDestroy_AudioSendStream) { TEST(CallTest, CreateDestroy_AudioReceiveStream) { for (bool use_null_audio_processing : {false, true}) { CallHelper call(use_null_audio_processing); - AudioReceiveStream::Config config; + AudioReceiveStreamInterface::Config config; MockTransport rtcp_send_transport; config.rtp.remote_ssrc = 42; config.rtcp_send_transport = &rtcp_send_transport; config.decoder_factory = rtc::make_ref_counted(); - AudioReceiveStream* stream = call->CreateAudioReceiveStream(config); + AudioReceiveStreamInterface* stream = + call->CreateAudioReceiveStream(config); EXPECT_NE(stream, nullptr); call->DestroyAudioReceiveStream(stream); } @@ -154,16 +155,17 @@ TEST(CallTest, CreateDestroy_AudioSendStreams) { TEST(CallTest, CreateDestroy_AudioReceiveStreams) { for (bool use_null_audio_processing : {false, true}) { CallHelper call(use_null_audio_processing); - AudioReceiveStream::Config config; + AudioReceiveStreamInterface::Config config; MockTransport rtcp_send_transport; config.rtcp_send_transport = &rtcp_send_transport; config.decoder_factory = rtc::make_ref_counted(); - std::list streams; + std::list streams; for (int i = 0; i < 2; ++i) { for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) { config.rtp.remote_ssrc = ssrc; - AudioReceiveStream* stream = call->CreateAudioReceiveStream(config); + AudioReceiveStreamInterface* stream = + call->CreateAudioReceiveStream(config); EXPECT_NE(stream, nullptr); if (ssrc & 1) { streams.push_back(stream); @@ -182,14 +184,14 @@ TEST(CallTest, CreateDestroy_AudioReceiveStreams) { TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_RecvFirst) { for (bool use_null_audio_processing : {false, true}) { CallHelper call(use_null_audio_processing); - AudioReceiveStream::Config recv_config; + AudioReceiveStreamInterface::Config recv_config; MockTransport rtcp_send_transport; recv_config.rtp.remote_ssrc = 42; recv_config.rtp.local_ssrc = 777; recv_config.rtcp_send_transport = &rtcp_send_transport; recv_config.decoder_factory = rtc::make_ref_counted(); - AudioReceiveStream* recv_stream = + AudioReceiveStreamInterface* recv_stream = call->CreateAudioReceiveStream(recv_config); EXPECT_NE(recv_stream, nullptr); @@ -221,14 +223,14 @@ TEST(CallTest, CreateDestroy_AssociateAudioSendReceiveStreams_SendFirst) { AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); EXPECT_NE(send_stream, nullptr); - AudioReceiveStream::Config recv_config; + AudioReceiveStreamInterface::Config recv_config; MockTransport rtcp_send_transport; recv_config.rtp.remote_ssrc = 42; recv_config.rtp.local_ssrc = 777; recv_config.rtcp_send_transport = &rtcp_send_transport; recv_config.decoder_factory = rtc::make_ref_counted(); - AudioReceiveStream* recv_stream = + AudioReceiveStreamInterface* recv_stream = call->CreateAudioReceiveStream(recv_config); EXPECT_NE(recv_stream, nullptr); diff --git a/call/degraded_call.cc b/call/degraded_call.cc index ddb6d4d78d..2c01c997c4 100644 --- a/call/degraded_call.cc +++ b/call/degraded_call.cc @@ -187,13 +187,13 @@ void DegradedCall::DestroyAudioSendStream(AudioSendStream* send_stream) { audio_send_transport_adapters_.erase(send_stream); } -AudioReceiveStream* DegradedCall::CreateAudioReceiveStream( - const AudioReceiveStream::Config& config) { +AudioReceiveStreamInterface* DegradedCall::CreateAudioReceiveStream( + const AudioReceiveStreamInterface::Config& config) { return call_->CreateAudioReceiveStream(config); } void DegradedCall::DestroyAudioReceiveStream( - AudioReceiveStream* receive_stream) { + AudioReceiveStreamInterface* receive_stream) { call_->DestroyAudioReceiveStream(receive_stream); } @@ -300,7 +300,7 @@ void DegradedCall::OnAudioTransportOverheadChanged( call_->OnAudioTransportOverheadChanged(transport_overhead_per_packet); } -void DegradedCall::OnLocalSsrcUpdated(AudioReceiveStream& stream, +void DegradedCall::OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream, uint32_t local_ssrc) { call_->OnLocalSsrcUpdated(stream, local_ssrc); } @@ -315,7 +315,7 @@ void DegradedCall::OnLocalSsrcUpdated(FlexfecReceiveStream& stream, call_->OnLocalSsrcUpdated(stream, local_ssrc); } -void DegradedCall::OnUpdateSyncGroup(AudioReceiveStream& stream, +void DegradedCall::OnUpdateSyncGroup(AudioReceiveStreamInterface& stream, absl::string_view sync_group) { call_->OnUpdateSyncGroup(stream, sync_group); } diff --git a/call/degraded_call.h b/call/degraded_call.h index 84b331ed6c..a527677fc1 100644 --- a/call/degraded_call.h +++ b/call/degraded_call.h @@ -61,9 +61,10 @@ class DegradedCall : public Call, private PacketReceiver { const AudioSendStream::Config& config) override; void DestroyAudioSendStream(AudioSendStream* send_stream) override; - AudioReceiveStream* CreateAudioReceiveStream( - const AudioReceiveStream::Config& config) override; - void DestroyAudioReceiveStream(AudioReceiveStream* receive_stream) override; + AudioReceiveStreamInterface* CreateAudioReceiveStream( + const AudioReceiveStreamInterface::Config& config) override; + void DestroyAudioReceiveStream( + AudioReceiveStreamInterface* receive_stream) override; VideoSendStream* CreateVideoSendStream( VideoSendStream::Config config, @@ -100,13 +101,13 @@ class DegradedCall : public Call, private PacketReceiver { void SignalChannelNetworkState(MediaType media, NetworkState state) override; void OnAudioTransportOverheadChanged( int transport_overhead_per_packet) override; - void OnLocalSsrcUpdated(AudioReceiveStream& stream, + void OnLocalSsrcUpdated(AudioReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(FlexfecReceiveStream& stream, uint32_t local_ssrc) override; - void OnUpdateSyncGroup(AudioReceiveStream& stream, + void OnUpdateSyncGroup(AudioReceiveStreamInterface& stream, absl::string_view sync_group) override; void OnSentPacket(const rtc::SentPacket& sent_packet) override; diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc index e04229efe2..0ce03acf2e 100644 --- a/call/rampup_tests.cc +++ b/call/rampup_tests.cc @@ -256,7 +256,7 @@ void RampUpTester::ModifyVideoConfigs( void RampUpTester::ModifyAudioConfigs( AudioSendStream::Config* send_config, - std::vector* receive_configs) { + std::vector* receive_configs) { if (num_audio_streams_ == 0) return; @@ -278,7 +278,7 @@ void RampUpTester::ModifyAudioConfigs( extension_type_.c_str(), kTransportSequenceNumberExtensionId)); } - for (AudioReceiveStream::Config& recv_config : *receive_configs) { + for (AudioReceiveStreamInterface::Config& recv_config : *receive_configs) { recv_config.rtp.transport_cc = transport_cc; recv_config.rtp.extensions = send_config->rtp.extensions; recv_config.rtp.remote_ssrc = send_config->rtp.ssrc; diff --git a/call/rampup_tests.h b/call/rampup_tests.h index 25872088c4..4e9217fa58 100644 --- a/call/rampup_tests.h +++ b/call/rampup_tests.h @@ -100,9 +100,9 @@ class RampUpTester : public test::EndToEndTest { VideoSendStream::Config* send_config, std::vector* receive_configs, VideoEncoderConfig* encoder_config) override; - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override; + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override; void ModifyFlexfecConfigs( std::vector* receive_configs) override; void OnCallsCreated(Call* sender_call, Call* receiver_call) override; diff --git a/call/syncable.h b/call/syncable.h index 02cd4b5dc2..6817be9c55 100644 --- a/call/syncable.h +++ b/call/syncable.h @@ -9,7 +9,7 @@ */ // Syncable is used by RtpStreamsSynchronizer in VideoReceiveStreamInterface, -// and implemented by AudioReceiveStream. +// and implemented by AudioReceiveStreamInterface. #ifndef CALL_SYNCABLE_H_ #define CALL_SYNCABLE_H_ diff --git a/media/engine/fake_webrtc_call.cc b/media/engine/fake_webrtc_call.cc index cad48e4368..4558119f6b 100644 --- a/media/engine/fake_webrtc_call.cc +++ b/media/engine/fake_webrtc_call.cc @@ -74,16 +74,16 @@ webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats( FakeAudioReceiveStream::FakeAudioReceiveStream( int id, - const webrtc::AudioReceiveStream::Config& config) + const webrtc::AudioReceiveStreamInterface::Config& config) : id_(id), config_(config) {} -const webrtc::AudioReceiveStream::Config& FakeAudioReceiveStream::GetConfig() - const { +const webrtc::AudioReceiveStreamInterface::Config& +FakeAudioReceiveStream::GetConfig() const { return config_; } void FakeAudioReceiveStream::SetStats( - const webrtc::AudioReceiveStream::Stats& stats) { + const webrtc::AudioReceiveStreamInterface::Stats& stats) { stats_ = stats; } @@ -141,7 +141,7 @@ webrtc::RtpHeaderExtensionMap FakeAudioReceiveStream::GetRtpExtensionMap() return webrtc::RtpHeaderExtensionMap(config_.rtp.extensions); } -webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats( +webrtc::AudioReceiveStreamInterface::Stats FakeAudioReceiveStream::GetStats( bool get_and_clear_legacy_stats) const { return stats_; } @@ -543,8 +543,8 @@ void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) { } } -webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream( - const webrtc::AudioReceiveStream::Config& config) { +webrtc::AudioReceiveStreamInterface* FakeCall::CreateAudioReceiveStream( + const webrtc::AudioReceiveStreamInterface::Config& config) { audio_receive_streams_.push_back( new FakeAudioReceiveStream(next_stream_id_++, config)); ++num_created_receive_streams_; @@ -552,7 +552,7 @@ webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream( } void FakeCall::DestroyAudioReceiveStream( - webrtc::AudioReceiveStream* receive_stream) { + webrtc::AudioReceiveStreamInterface* receive_stream) { auto it = absl::c_find(audio_receive_streams_, static_cast(receive_stream)); if (it == audio_receive_streams_.end()) { @@ -709,7 +709,7 @@ void FakeCall::SignalChannelNetworkState(webrtc::MediaType media, void FakeCall::OnAudioTransportOverheadChanged( int transport_overhead_per_packet) {} -void FakeCall::OnLocalSsrcUpdated(webrtc::AudioReceiveStream& stream, +void FakeCall::OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream, uint32_t local_ssrc) { auto& fake_stream = static_cast(stream); fake_stream.SetLocalSsrc(local_ssrc); @@ -727,7 +727,7 @@ void FakeCall::OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream, fake_stream.SetLocalSsrc(local_ssrc); } -void FakeCall::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream, +void FakeCall::OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream, absl::string_view sync_group) { auto& fake_stream = static_cast(stream); fake_stream.SetSyncGroup(sync_group); diff --git a/media/engine/fake_webrtc_call.h b/media/engine/fake_webrtc_call.h index 11eedd68cf..3c458299d6 100644 --- a/media/engine/fake_webrtc_call.h +++ b/media/engine/fake_webrtc_call.h @@ -13,7 +13,7 @@ // // webrtc::Call // webrtc::AudioSendStream -// webrtc::AudioReceiveStream +// webrtc::AudioReceiveStreamInterface // webrtc::VideoSendStream // webrtc::VideoReceiveStreamInterface @@ -83,15 +83,16 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream { bool muted_ = false; }; -class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { +class FakeAudioReceiveStream final + : public webrtc::AudioReceiveStreamInterface { public: explicit FakeAudioReceiveStream( int id, - const webrtc::AudioReceiveStream::Config& config); + const webrtc::AudioReceiveStreamInterface::Config& config); int id() const { return id_; } - const webrtc::AudioReceiveStream::Config& GetConfig() const; - void SetStats(const webrtc::AudioReceiveStream::Stats& stats); + const webrtc::AudioReceiveStreamInterface::Config& GetConfig() const; + void SetStats(const webrtc::AudioReceiveStreamInterface::Stats& stats); int received_packets() const { return received_packets_; } bool VerifyLastPacket(const uint8_t* data, size_t length) const; const webrtc::AudioSinkInterface* sink() const { return sink_; } @@ -130,7 +131,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { const std::vector& GetRtpExtensions() const override; webrtc::RtpHeaderExtensionMap GetRtpExtensionMap() const override; - webrtc::AudioReceiveStream::Stats GetStats( + webrtc::AudioReceiveStreamInterface::Stats GetStats( bool get_and_clear_legacy_stats) const override; void SetSink(webrtc::AudioSinkInterface* sink) override; void SetGain(float gain) override; @@ -146,8 +147,8 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream { } int id_ = -1; - webrtc::AudioReceiveStream::Config config_; - webrtc::AudioReceiveStream::Stats stats_; + webrtc::AudioReceiveStreamInterface::Config config_; + webrtc::AudioReceiveStreamInterface::Stats stats_; int received_packets_ = 0; webrtc::AudioSinkInterface* sink_ = nullptr; float gain_ = 1.0f; @@ -377,10 +378,10 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { const webrtc::AudioSendStream::Config& config) override; void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; - webrtc::AudioReceiveStream* CreateAudioReceiveStream( - const webrtc::AudioReceiveStream::Config& config) override; + webrtc::AudioReceiveStreamInterface* CreateAudioReceiveStream( + const webrtc::AudioReceiveStreamInterface::Config& config) override; void DestroyAudioReceiveStream( - webrtc::AudioReceiveStream* receive_stream) override; + webrtc::AudioReceiveStreamInterface* receive_stream) override; webrtc::VideoSendStream* CreateVideoSendStream( webrtc::VideoSendStream::Config config, @@ -420,13 +421,13 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver { webrtc::NetworkState state) override; void OnAudioTransportOverheadChanged( int transport_overhead_per_packet) override; - void OnLocalSsrcUpdated(webrtc::AudioReceiveStream& stream, + void OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(webrtc::VideoReceiveStreamInterface& stream, uint32_t local_ssrc) override; void OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream, uint32_t local_ssrc) override; - void OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream, + void OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream, absl::string_view sync_group) override; void OnSentPacket(const rtc::SentPacket& sent_packet) override; diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc index 710823ac8f..064b72dd6e 100644 --- a/media/engine/webrtc_voice_engine.cc +++ b/media/engine/webrtc_voice_engine.cc @@ -245,7 +245,7 @@ struct AdaptivePtimeConfig { // TODO(tommi): Constructing a receive stream could be made simpler. // Move some of this boiler plate code into the config structs themselves. -webrtc::AudioReceiveStream::Config BuildReceiveStreamConfig( +webrtc::AudioReceiveStreamInterface::Config BuildReceiveStreamConfig( uint32_t remote_ssrc, uint32_t local_ssrc, bool use_transport_cc, @@ -263,7 +263,7 @@ webrtc::AudioReceiveStream::Config BuildReceiveStreamConfig( rtc::scoped_refptr frame_decryptor, const webrtc::CryptoOptions& crypto_options, rtc::scoped_refptr frame_transformer) { - webrtc::AudioReceiveStream::Config config; + webrtc::AudioReceiveStreamInterface::Config config; config.rtp.remote_ssrc = remote_ssrc; config.rtp.local_ssrc = local_ssrc; config.rtp.transport_cc = use_transport_cc; @@ -1151,7 +1151,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { public: - WebRtcAudioReceiveStream(webrtc::AudioReceiveStream::Config config, + WebRtcAudioReceiveStream(webrtc::AudioReceiveStreamInterface::Config config, webrtc::Call* call) : call_(call), stream_(call_->CreateAudioReceiveStream(config)) { RTC_DCHECK(call); @@ -1167,7 +1167,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { call_->DestroyAudioReceiveStream(stream_); } - webrtc::AudioReceiveStream& stream() { + webrtc::AudioReceiveStreamInterface& stream() { RTC_DCHECK(stream_); return *stream_; } @@ -1200,7 +1200,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { stream_->SetDecoderMap(decoder_map); } - webrtc::AudioReceiveStream::Stats GetStats( + webrtc::AudioReceiveStreamInterface::Stats GetStats( bool get_and_clear_legacy_stats) const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); return stream_->GetStats(get_and_clear_legacy_stats); @@ -1234,7 +1234,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { return true; RTC_LOG(LS_ERROR) << "Failed to SetBaseMinimumPlayoutDelayMs" - " on AudioReceiveStream on SSRC=" + " on AudioReceiveStreamInterface on SSRC=" << stream_->remote_ssrc() << " with delay_ms=" << delay_ms; return false; @@ -1267,7 +1267,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream { private: webrtc::SequenceChecker worker_thread_checker_; webrtc::Call* call_ = nullptr; - webrtc::AudioReceiveStream* const stream_ = nullptr; + webrtc::AudioReceiveStreamInterface* const stream_ = nullptr; std::unique_ptr raw_audio_sink_ RTC_GUARDED_BY(worker_thread_checker_); }; @@ -2341,7 +2341,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info, continue; } } - webrtc::AudioReceiveStream::Stats stats = + webrtc::AudioReceiveStreamInterface::Stats stats = stream.second->GetStats(get_and_clear_legacy_stats); VoiceReceiverInfo rinfo; rinfo.add_ssrc(stats.remote_ssrc); diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc index 59a1d58083..cb6e2ac556 100644 --- a/media/engine/webrtc_voice_engine_unittest.cc +++ b/media/engine/webrtc_voice_engine_unittest.cc @@ -294,7 +294,8 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam { return GetSendStream(ssrc).GetConfig(); } - const webrtc::AudioReceiveStream::Config& GetRecvStreamConfig(uint32_t ssrc) { + const webrtc::AudioReceiveStreamInterface::Config& GetRecvStreamConfig( + uint32_t ssrc) { return GetRecvStream(ssrc).GetConfig(); } @@ -656,8 +657,9 @@ class WebRtcVoiceEngineTestFake : public ::testing::TestWithParam { stats.ana_statistics.uplink_packet_loss_fraction); } - webrtc::AudioReceiveStream::Stats GetAudioReceiveStreamStats() const { - webrtc::AudioReceiveStream::Stats stats; + webrtc::AudioReceiveStreamInterface::Stats GetAudioReceiveStreamStats() + const { + webrtc::AudioReceiveStreamInterface::Stats stats; stats.remote_ssrc = 123; stats.payload_bytes_rcvd = 456; stats.header_and_padding_bytes_rcvd = 67; @@ -824,7 +826,7 @@ TEST_P(WebRtcVoiceEngineTestFake, CreateSendStream) { TEST_P(WebRtcVoiceEngineTestFake, CreateRecvStream) { EXPECT_TRUE(SetupChannel()); EXPECT_TRUE(AddRecvStream(kSsrcX)); - const webrtc::AudioReceiveStream::Config& config = + const webrtc::AudioReceiveStreamInterface::Config& config = GetRecvStreamConfig(kSsrcX); EXPECT_EQ(kSsrcX, config.rtp.remote_ssrc); EXPECT_EQ(0xFA17FA17, config.rtp.local_ssrc); @@ -3578,7 +3580,8 @@ TEST_P(WebRtcVoiceEngineTestFake, PreservePlayoutWhenRecreateRecvStream) { channel_->SetPlayout(true); EXPECT_TRUE(GetRecvStream(kSsrcX).started()); - // Changing RTP header extensions will recreate the AudioReceiveStream. + // Changing RTP header extensions will recreate the + // AudioReceiveStreamInterface. cricket::AudioRecvParameters parameters; parameters.extensions.push_back( webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri, 12)); diff --git a/test/call_test.cc b/test/call_test.cc index 5915e56d39..af49f5bdb5 100644 --- a/test/call_test.cc +++ b/test/call_test.cc @@ -420,12 +420,12 @@ void CallTest::CreateMatchingAudioConfigs(Transport* transport, audio_send_config_, audio_decoder_factory_, transport, sync_group)); } -AudioReceiveStream::Config CallTest::CreateMatchingAudioConfig( +AudioReceiveStreamInterface::Config CallTest::CreateMatchingAudioConfig( const AudioSendStream::Config& send_config, rtc::scoped_refptr audio_decoder_factory, Transport* transport, std::string sync_group) { - AudioReceiveStream::Config audio_config; + AudioReceiveStreamInterface::Config audio_config; audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; audio_config.rtcp_send_transport = transport; audio_config.rtp.remote_ssrc = send_config.rtp.ssrc; @@ -586,7 +586,7 @@ void CallTest::Start() { if (audio_send_stream_) { audio_send_stream_->Start(); } - for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) + for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_) audio_recv_stream->Start(); } @@ -598,7 +598,7 @@ void CallTest::StartVideoStreams() { } void CallTest::Stop() { - for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) + for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_) audio_recv_stream->Stop(); if (audio_send_stream_) { audio_send_stream_->Stop(); @@ -617,7 +617,7 @@ void CallTest::DestroyStreams() { if (audio_send_stream_) sender_call_->DestroyAudioSendStream(audio_send_stream_); audio_send_stream_ = nullptr; - for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) + for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_) receiver_call_->DestroyAudioReceiveStream(audio_recv_stream); DestroyVideoSendStreams(); @@ -798,11 +798,11 @@ void BaseTest::OnVideoStreamsCreated( void BaseTest::ModifyAudioConfigs( AudioSendStream::Config* send_config, - std::vector* receive_configs) {} + std::vector* receive_configs) {} void BaseTest::OnAudioStreamsCreated( AudioSendStream* send_stream, - const std::vector& receive_streams) {} + const std::vector& receive_streams) {} void BaseTest::ModifyFlexfecConfigs( std::vector* receive_configs) {} diff --git a/test/call_test.h b/test/call_test.h index ba7d123bc4..73860e35aa 100644 --- a/test/call_test.h +++ b/test/call_test.h @@ -129,7 +129,7 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface { void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport); void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group); - static AudioReceiveStream::Config CreateMatchingAudioConfig( + static AudioReceiveStreamInterface::Config CreateMatchingAudioConfig( const AudioSendStream::Config& send_config, rtc::scoped_refptr audio_decoder_factory, Transport* transport, @@ -198,8 +198,8 @@ class CallTest : public ::testing::Test, public RtpPacketSinkInterface { std::unique_ptr receive_transport_; std::vector video_receive_configs_; std::vector video_receive_streams_; - std::vector audio_receive_configs_; - std::vector audio_receive_streams_; + std::vector audio_receive_configs_; + std::vector audio_receive_streams_; std::vector flexfec_receive_configs_; std::vector flexfec_receive_streams_; @@ -289,10 +289,10 @@ class BaseTest : public RtpRtcpObserver { virtual void ModifyAudioConfigs( AudioSendStream::Config* send_config, - std::vector* receive_configs); + std::vector* receive_configs); virtual void OnAudioStreamsCreated( AudioSendStream* send_stream, - const std::vector& receive_streams); + const std::vector& receive_streams); virtual void ModifyFlexfecConfigs( std::vector* receive_configs); diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc index 63f78c8f71..ea170bc17c 100644 --- a/test/scenario/audio_stream.cc +++ b/test/scenario/audio_stream.cc @@ -175,7 +175,7 @@ ReceiveAudioStream::ReceiveAudioStream( rtc::scoped_refptr decoder_factory, Transport* feedback_transport) : receiver_(receiver), config_(config) { - AudioReceiveStream::Config recv_config; + AudioReceiveStreamInterface::Config recv_config; recv_config.rtp.local_ssrc = receiver_->GetNextAudioLocalSsrc(); recv_config.rtcp_send_transport = feedback_transport; recv_config.rtp.remote_ssrc = send_stream->ssrc_; @@ -209,8 +209,8 @@ void ReceiveAudioStream::Stop() { receiver_->SendTask([&] { receive_stream_->Stop(); }); } -AudioReceiveStream::Stats ReceiveAudioStream::GetStats() const { - AudioReceiveStream::Stats result; +AudioReceiveStreamInterface::Stats ReceiveAudioStream::GetStats() const { + AudioReceiveStreamInterface::Stats result; receiver_->SendTask([&] { result = receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true); }); diff --git a/test/scenario/audio_stream.h b/test/scenario/audio_stream.h index 2110c1d5cb..ced3828e21 100644 --- a/test/scenario/audio_stream.h +++ b/test/scenario/audio_stream.h @@ -59,7 +59,7 @@ class ReceiveAudioStream { void Start(); void Stop(); - AudioReceiveStream::Stats GetStats() const; + AudioReceiveStreamInterface::Stats GetStats() const; private: friend class Scenario; @@ -69,7 +69,7 @@ class ReceiveAudioStream { SendAudioStream* send_stream, rtc::scoped_refptr decoder_factory, Transport* feedback_transport); - AudioReceiveStream* receive_stream_ = nullptr; + AudioReceiveStreamInterface* receive_stream_ = nullptr; CallClient* const receiver_; const AudioStreamConfig config_; }; diff --git a/test/scenario/stats_collection.cc b/test/scenario/stats_collection.cc index c6849985d7..e32696de71 100644 --- a/test/scenario/stats_collection.cc +++ b/test/scenario/stats_collection.cc @@ -142,7 +142,8 @@ void CallStatsCollector::AddStats(Call::Stats sample) { stats_.memory_usage.AddSample(rtc::GetProcessResidentSizeBytes()); } -void AudioReceiveStatsCollector::AddStats(AudioReceiveStream::Stats sample) { +void AudioReceiveStatsCollector::AddStats( + AudioReceiveStreamInterface::Stats sample) { stats_.expand_rate.AddSample(sample.expand_rate); stats_.accelerate_rate.AddSample(sample.accelerate_rate); stats_.jitter_buffer.AddSampleMs(sample.jitter_buffer_ms); diff --git a/test/scenario/stats_collection.h b/test/scenario/stats_collection.h index 0d2f487861..1f5d8daea7 100644 --- a/test/scenario/stats_collection.h +++ b/test/scenario/stats_collection.h @@ -72,7 +72,7 @@ class CallStatsCollector { }; class AudioReceiveStatsCollector { public: - void AddStats(AudioReceiveStream::Stats sample); + void AddStats(AudioReceiveStreamInterface::Stats sample); CollectedAudioReceiveStats& stats() { return stats_; } private: diff --git a/test/scenario/stats_collection_unittest.cc b/test/scenario/stats_collection_unittest.cc index 35f0ced349..3db1100a2a 100644 --- a/test/scenario/stats_collection_unittest.cc +++ b/test/scenario/stats_collection_unittest.cc @@ -38,7 +38,7 @@ void CreateAnalyzedStream(Scenario* s, caller->SendTask([&]() { send_stats = video->send()->GetStats(); }); collectors->video_send.AddStats(send_stats, s->Now()); - AudioReceiveStream::Stats receive_stats; + AudioReceiveStreamInterface::Stats receive_stats; caller->SendTask([&]() { receive_stats = audio->receive()->GetStats(); }); collectors->audio_receive.AddStats(receive_stats); diff --git a/video/end_to_end_tests/retransmission_tests.cc b/video/end_to_end_tests/retransmission_tests.cc index d6d0d41296..25fd69cd56 100644 --- a/video/end_to_end_tests/retransmission_tests.cc +++ b/video/end_to_end_tests/retransmission_tests.cc @@ -179,9 +179,9 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) { return SEND_PACKET; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc; remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc; diff --git a/video/end_to_end_tests/transport_feedback_tests.cc b/video/end_to_end_tests/transport_feedback_tests.cc index 499e5cd438..99a87a8e81 100644 --- a/video/end_to_end_tests/transport_feedback_tests.cc +++ b/video/end_to_end_tests/transport_feedback_tests.cc @@ -297,9 +297,9 @@ class TransportFeedbackTester : public test::EndToEndTest { (*receive_configs)[0].rtp.transport_cc = feedback_enabled_; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->rtp.extensions.clear(); send_config->rtp.extensions.push_back( RtpExtension(RtpExtension::kTransportSequenceNumberUri, @@ -448,9 +448,9 @@ TEST_F(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) { size_t GetNumVideoStreams() const override { return 1; } size_t GetNumAudioStreams() const override { return 1; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { send_config->rtp.extensions.clear(); send_config->rtp.extensions.push_back( RtpExtension(RtpExtension::kTransportSequenceNumberUri, diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc index 076ccc364b..f298f3302e 100644 --- a/video/video_analyzer.cc +++ b/video/video_analyzer.cc @@ -194,7 +194,8 @@ void VideoAnalyzer::SetReceiveStream(VideoReceiveStreamInterface* stream) { receive_stream_ = stream; } -void VideoAnalyzer::SetAudioReceiveStream(AudioReceiveStream* recv_stream) { +void VideoAnalyzer::SetAudioReceiveStream( + AudioReceiveStreamInterface* recv_stream) { MutexLock lock(&lock_); RTC_CHECK(!audio_receive_stream_); audio_receive_stream_ = recv_stream; @@ -526,7 +527,7 @@ void VideoAnalyzer::PollStats() { } if (audio_receive_stream_ != nullptr) { - AudioReceiveStream::Stats receive_stats = + AudioReceiveStreamInterface::Stats receive_stats = audio_receive_stream_->GetStats(/*get_and_clear_legacy_stats=*/true); audio_expand_rate_.AddSample(receive_stats.expand_rate); audio_accelerate_rate_.AddSample(receive_stats.accelerate_rate); diff --git a/video/video_analyzer.h b/video/video_analyzer.h index 3b44f3b142..725dacfdd8 100644 --- a/video/video_analyzer.h +++ b/video/video_analyzer.h @@ -62,7 +62,7 @@ class VideoAnalyzer : public PacketReceiver, void SetCall(Call* call); void SetSendStream(VideoSendStream* stream); void SetReceiveStream(VideoReceiveStreamInterface* stream); - void SetAudioReceiveStream(AudioReceiveStream* recv_stream); + void SetAudioReceiveStream(AudioReceiveStreamInterface* recv_stream); rtc::VideoSinkInterface* InputInterface(); rtc::VideoSourceInterface* OutputInterface(); @@ -222,7 +222,7 @@ class VideoAnalyzer : public PacketReceiver, Call* call_; VideoSendStream* send_stream_; VideoReceiveStreamInterface* receive_stream_; - AudioReceiveStream* audio_receive_stream_; + AudioReceiveStreamInterface* audio_receive_stream_; CapturedFrameForwarder captured_frame_forwarder_; const std::string test_label_; FILE* const graph_data_output_file_; diff --git a/video/video_quality_test.cc b/video/video_quality_test.cc index 956fcaebed..8c3906564e 100644 --- a/video/video_quality_test.cc +++ b/video/video_quality_test.cc @@ -1145,7 +1145,7 @@ void VideoQualityTest::CreateCapturers() { void VideoQualityTest::StartAudioStreams() { audio_send_stream_->Start(); - for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) + for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_) audio_recv_stream->Start(); } diff --git a/video/video_send_stream_tests.cc b/video/video_send_stream_tests.cc index 1c53835238..18b448a4b5 100644 --- a/video/video_send_stream_tests.cc +++ b/video/video_send_stream_tests.cc @@ -1605,9 +1605,9 @@ TEST_F(VideoSendStreamTest, ChangingNetworkRoute) { (*receive_configs)[0].rtp.transport_cc = true; } - void ModifyAudioConfigs( - AudioSendStream::Config* send_config, - std::vector* receive_configs) override { + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector* + receive_configs) override { RTC_DCHECK_RUN_ON(&task_queue_thread_); send_config->rtp.extensions.clear(); send_config->rtp.extensions.push_back(RtpExtension(