Rename internal::AudioReceiveStream to AudioReceiveStreamImpl Bug: webrtc:7484 Change-Id: Id0836a7fdd6fabbdc9bdc3b15e9965d9102bffa5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262803 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36959}
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc index bc7dddd..36ab5f5 100644 --- a/audio/audio_receive_stream.cc +++ b/audio/audio_receive_stream.cc
@@ -65,7 +65,6 @@ return ss.str(); } -namespace internal { namespace { std::unique_ptr<voe::ChannelReceiveInterface> CreateChannelReceive( Clock* clock, @@ -87,25 +86,25 @@ } } // namespace -AudioReceiveStream::AudioReceiveStream( +AudioReceiveStreamImpl::AudioReceiveStreamImpl( Clock* clock, PacketRouter* packet_router, NetEqFactory* neteq_factory, const webrtc::AudioReceiveStream::Config& config, const rtc::scoped_refptr<webrtc::AudioState>& audio_state, webrtc::RtcEventLog* event_log) - : AudioReceiveStream(clock, - packet_router, - config, - audio_state, - event_log, - CreateChannelReceive(clock, - audio_state.get(), - neteq_factory, - config, - event_log)) {} + : AudioReceiveStreamImpl(clock, + packet_router, + config, + audio_state, + event_log, + CreateChannelReceive(clock, + audio_state.get(), + neteq_factory, + config, + event_log)) {} -AudioReceiveStream::AudioReceiveStream( +AudioReceiveStreamImpl::AudioReceiveStreamImpl( Clock* clock, PacketRouter* packet_router, const webrtc::AudioReceiveStream::Config& config, @@ -116,7 +115,7 @@ audio_state_(audio_state), source_tracker_(clock), channel_receive_(std::move(channel_receive)) { - RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc; + RTC_LOG(LS_INFO) << "AudioReceiveStreamImpl: " << config.rtp.remote_ssrc; RTC_DCHECK(config.decoder_factory); RTC_DCHECK(config.rtcp_send_transport); RTC_DCHECK(audio_state_); @@ -143,15 +142,15 @@ // `channel_receive_` already. } -AudioReceiveStream::~AudioReceiveStream() { +AudioReceiveStreamImpl::~AudioReceiveStreamImpl() { RTC_DCHECK_RUN_ON(&worker_thread_checker_); - RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << remote_ssrc(); + RTC_LOG(LS_INFO) << "~AudioReceiveStreamImpl: " << remote_ssrc(); Stop(); channel_receive_->SetAssociatedSendChannel(nullptr); channel_receive_->ResetReceiverCongestionControlObjects(); } -void AudioReceiveStream::RegisterWithTransport( +void AudioReceiveStreamImpl::RegisterWithTransport( RtpStreamReceiverControllerInterface* receiver_controller) { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); RTC_DCHECK(!rtp_stream_receiver_); @@ -159,12 +158,12 @@ remote_ssrc(), channel_receive_.get()); } -void AudioReceiveStream::UnregisterFromTransport() { +void AudioReceiveStreamImpl::UnregisterFromTransport() { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); rtp_stream_receiver_.reset(); } -void AudioReceiveStream::ReconfigureForTesting( +void AudioReceiveStreamImpl::ReconfigureForTesting( const webrtc::AudioReceiveStream::Config& config) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); @@ -190,7 +189,7 @@ config_ = config; } -void AudioReceiveStream::Start() { +void AudioReceiveStreamImpl::Start() { RTC_DCHECK_RUN_ON(&worker_thread_checker_); if (playing_) { return; @@ -200,7 +199,7 @@ audio_state()->AddReceivingStream(this); } -void AudioReceiveStream::Stop() { +void AudioReceiveStreamImpl::Stop() { RTC_DCHECK_RUN_ON(&worker_thread_checker_); if (!playing_) { return; @@ -210,32 +209,33 @@ audio_state()->RemoveReceivingStream(this); } -bool AudioReceiveStream::transport_cc() const { +bool AudioReceiveStreamImpl::transport_cc() const { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); return config_.rtp.transport_cc; } -bool AudioReceiveStream::IsRunning() const { +bool AudioReceiveStreamImpl::IsRunning() const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); return playing_; } -void AudioReceiveStream::SetDepacketizerToDecoderFrameTransformer( +void AudioReceiveStreamImpl::SetDepacketizerToDecoderFrameTransformer( rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); channel_receive_->SetDepacketizerToDecoderFrameTransformer( std::move(frame_transformer)); } -void AudioReceiveStream::SetDecoderMap( +void AudioReceiveStreamImpl::SetDecoderMap( std::map<int, SdpAudioFormat> decoder_map) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); config_.decoder_map = std::move(decoder_map); channel_receive_->SetReceiveCodecs(config_.decoder_map); } -void AudioReceiveStream::SetUseTransportCcAndNackHistory(bool use_transport_cc, - int history_ms) { +void AudioReceiveStreamImpl::SetUseTransportCcAndNackHistory( + bool use_transport_cc, + int history_ms) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); RTC_DCHECK_GE(history_ms, 0); config_.rtp.transport_cc = use_transport_cc; @@ -247,13 +247,13 @@ } } -void AudioReceiveStream::SetNonSenderRttMeasurement(bool enabled) { +void AudioReceiveStreamImpl::SetNonSenderRttMeasurement(bool enabled) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); config_.enable_non_sender_rtt = enabled; channel_receive_->SetNonSenderRttMeasurement(enabled); } -void AudioReceiveStream::SetFrameDecryptor( +void AudioReceiveStreamImpl::SetFrameDecryptor( rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) { // TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream, // expect to be called on the network thread. @@ -261,7 +261,7 @@ channel_receive_->SetFrameDecryptor(std::move(frame_decryptor)); } -void AudioReceiveStream::SetRtpExtensions( +void AudioReceiveStreamImpl::SetRtpExtensions( std::vector<RtpExtension> extensions) { // TODO(bugs.webrtc.org/11993): This is called via WebRtcAudioReceiveStream, // expect to be called on the network thread. @@ -269,16 +269,17 @@ config_.rtp.extensions = std::move(extensions); } -const std::vector<RtpExtension>& AudioReceiveStream::GetRtpExtensions() const { +const std::vector<RtpExtension>& AudioReceiveStreamImpl::GetRtpExtensions() + const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); return config_.rtp.extensions; } -RtpHeaderExtensionMap AudioReceiveStream::GetRtpExtensionMap() const { +RtpHeaderExtensionMap AudioReceiveStreamImpl::GetRtpExtensionMap() const { return RtpHeaderExtensionMap(config_.rtp.extensions); } -webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats( +webrtc::AudioReceiveStream::Stats AudioReceiveStreamImpl::GetStats( bool get_and_clear_legacy_stats) const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); webrtc::AudioReceiveStream::Stats stats; @@ -374,34 +375,34 @@ return stats; } -void AudioReceiveStream::SetSink(AudioSinkInterface* sink) { +void AudioReceiveStreamImpl::SetSink(AudioSinkInterface* sink) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); channel_receive_->SetSink(sink); } -void AudioReceiveStream::SetGain(float gain) { +void AudioReceiveStreamImpl::SetGain(float gain) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); channel_receive_->SetChannelOutputVolumeScaling(gain); } -bool AudioReceiveStream::SetBaseMinimumPlayoutDelayMs(int delay_ms) { +bool AudioReceiveStreamImpl::SetBaseMinimumPlayoutDelayMs(int delay_ms) { RTC_DCHECK_RUN_ON(&worker_thread_checker_); return channel_receive_->SetBaseMinimumPlayoutDelayMs(delay_ms); } -int AudioReceiveStream::GetBaseMinimumPlayoutDelayMs() const { +int AudioReceiveStreamImpl::GetBaseMinimumPlayoutDelayMs() const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); return channel_receive_->GetBaseMinimumPlayoutDelayMs(); } -std::vector<RtpSource> AudioReceiveStream::GetSources() const { +std::vector<RtpSource> AudioReceiveStreamImpl::GetSources() const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); return source_tracker_.GetSources(); } -AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo( - int sample_rate_hz, - AudioFrame* audio_frame) { +AudioMixer::Source::AudioFrameInfo +AudioReceiveStreamImpl::GetAudioFrameWithInfo(int sample_rate_hz, + AudioFrame* audio_frame) { AudioMixer::Source::AudioFrameInfo audio_frame_info = channel_receive_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame); if (audio_frame_info != AudioMixer::Source::AudioFrameInfo::kError) { @@ -410,33 +411,33 @@ return audio_frame_info; } -int AudioReceiveStream::Ssrc() const { +int AudioReceiveStreamImpl::Ssrc() const { return remote_ssrc(); } -int AudioReceiveStream::PreferredSampleRate() const { +int AudioReceiveStreamImpl::PreferredSampleRate() const { return channel_receive_->PreferredSampleRate(); } -uint32_t AudioReceiveStream::id() const { +uint32_t AudioReceiveStreamImpl::id() const { RTC_DCHECK_RUN_ON(&worker_thread_checker_); return remote_ssrc(); } -absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const { +absl::optional<Syncable::Info> AudioReceiveStreamImpl::GetInfo() const { // TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer, // expect to be called on the network thread. RTC_DCHECK_RUN_ON(&worker_thread_checker_); return channel_receive_->GetSyncInfo(); } -bool AudioReceiveStream::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp, - int64_t* time_ms) const { +bool AudioReceiveStreamImpl::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp, + int64_t* time_ms) const { // Called on video capture thread. return channel_receive_->GetPlayoutRtpTimestamp(rtp_timestamp, time_ms); } -void AudioReceiveStream::SetEstimatedPlayoutNtpTimestampMs( +void AudioReceiveStreamImpl::SetEstimatedPlayoutNtpTimestampMs( int64_t ntp_timestamp_ms, int64_t time_ms) { // Called on video capture thread. @@ -444,21 +445,22 @@ time_ms); } -bool AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) { +bool AudioReceiveStreamImpl::SetMinimumPlayoutDelay(int delay_ms) { // TODO(bugs.webrtc.org/11993): This is called via RtpStreamsSynchronizer, // expect to be called on the network thread. RTC_DCHECK_RUN_ON(&worker_thread_checker_); return channel_receive_->SetMinimumPlayoutDelay(delay_ms); } -void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) { +void AudioReceiveStreamImpl::AssociateSendStream( + internal::AudioSendStream* send_stream) { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); channel_receive_->SetAssociatedSendChannel( send_stream ? send_stream->GetChannel() : nullptr); associated_send_stream_ = send_stream; } -void AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { +void AudioReceiveStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) { // TODO(solenberg): Tests call this function on a network thread, libjingle // calls on the worker thread. We should move towards always using a network // thread. Then this check can be enabled. @@ -466,39 +468,38 @@ channel_receive_->ReceivedRTCPPacket(packet, length); } -void AudioReceiveStream::SetSyncGroup(absl::string_view sync_group) { +void AudioReceiveStreamImpl::SetSyncGroup(absl::string_view sync_group) { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); config_.sync_group = std::string(sync_group); } -void AudioReceiveStream::SetLocalSsrc(uint32_t local_ssrc) { +void AudioReceiveStreamImpl::SetLocalSsrc(uint32_t local_ssrc) { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); // TODO(tommi): Consider storing local_ssrc in one place. config_.rtp.local_ssrc = local_ssrc; channel_receive_->OnLocalSsrcChange(local_ssrc); } -uint32_t AudioReceiveStream::local_ssrc() const { +uint32_t AudioReceiveStreamImpl::local_ssrc() const { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); RTC_DCHECK_EQ(config_.rtp.local_ssrc, channel_receive_->GetLocalSsrc()); return config_.rtp.local_ssrc; } -const std::string& AudioReceiveStream::sync_group() const { +const std::string& AudioReceiveStreamImpl::sync_group() const { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); return config_.sync_group; } -const AudioSendStream* AudioReceiveStream::GetAssociatedSendStreamForTesting() - const { +const AudioSendStream* +AudioReceiveStreamImpl::GetAssociatedSendStreamForTesting() const { RTC_DCHECK_RUN_ON(&packet_sequence_checker_); return associated_send_stream_; } -internal::AudioState* AudioReceiveStream::audio_state() const { +internal::AudioState* AudioReceiveStreamImpl::audio_state() const { auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get()); RTC_DCHECK(audio_state); return audio_state; } -} // namespace internal } // namespace webrtc
diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h index 7fe9420..4b8ef9f 100644 --- a/audio/audio_receive_stream.h +++ b/audio/audio_receive_stream.h
@@ -40,19 +40,21 @@ namespace internal { class AudioSendStream; +} // namespace internal -class AudioReceiveStream final : public webrtc::AudioReceiveStream, - public AudioMixer::Source, - public Syncable { +class AudioReceiveStreamImpl final : public webrtc::AudioReceiveStream, + public AudioMixer::Source, + public Syncable { public: - AudioReceiveStream(Clock* clock, - PacketRouter* packet_router, - NetEqFactory* neteq_factory, - const webrtc::AudioReceiveStream::Config& config, - const rtc::scoped_refptr<webrtc::AudioState>& audio_state, - webrtc::RtcEventLog* event_log); + AudioReceiveStreamImpl( + Clock* clock, + PacketRouter* packet_router, + NetEqFactory* neteq_factory, + const webrtc::AudioReceiveStream::Config& config, + const rtc::scoped_refptr<webrtc::AudioState>& audio_state, + webrtc::RtcEventLog* event_log); // For unit tests, which need to supply a mock channel receive. - AudioReceiveStream( + AudioReceiveStreamImpl( Clock* clock, PacketRouter* packet_router, const webrtc::AudioReceiveStream::Config& config, @@ -60,16 +62,16 @@ webrtc::RtcEventLog* event_log, std::unique_ptr<voe::ChannelReceiveInterface> channel_receive); - AudioReceiveStream() = delete; - AudioReceiveStream(const AudioReceiveStream&) = delete; - AudioReceiveStream& operator=(const AudioReceiveStream&) = delete; + AudioReceiveStreamImpl() = delete; + AudioReceiveStreamImpl(const AudioReceiveStreamImpl&) = delete; + AudioReceiveStreamImpl& operator=(const AudioReceiveStreamImpl&) = delete; // Destruction happens on the worker thread. Prior to destruction the caller // must ensure that a registration with the transport has been cleared. See // `RegisterWithTransport` for details. // TODO(tommi): As a further improvement to this, performing the full // destruction on the network thread could be made the default. - ~AudioReceiveStream() override; + ~AudioReceiveStreamImpl() override; // Called on the network thread to register/unregister with the network // transport. @@ -121,7 +123,7 @@ int64_t time_ms) override; bool SetMinimumPlayoutDelay(int delay_ms) override; - void AssociateSendStream(AudioSendStream* send_stream); + void AssociateSendStream(internal::AudioSendStream* send_stream); void DeliverRtcp(const uint8_t* packet, size_t length); void SetSyncGroup(absl::string_view sync_group); @@ -146,7 +148,7 @@ void ReconfigureForTesting(const webrtc::AudioReceiveStream::Config& config); private: - AudioState* audio_state() const; + internal::AudioState* audio_state() const; RTC_NO_UNIQUE_ADDRESS SequenceChecker worker_thread_checker_; // TODO(bugs.webrtc.org/11993): This checker conceptually represents @@ -169,7 +171,6 @@ std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_ RTC_GUARDED_BY(packet_sequence_checker_); }; -} // namespace internal } // namespace webrtc #endif // AUDIO_AUDIO_RECEIVE_STREAM_H_
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc index a1e8c1f..ab80c45 100644 --- a/audio/audio_receive_stream_unittest.cc +++ b/audio/audio_receive_stream_unittest.cc
@@ -146,8 +146,8 @@ rtc::make_ref_counted<MockAudioDecoderFactory>(); } - std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() { - auto ret = std::make_unique<internal::AudioReceiveStream>( + std::unique_ptr<AudioReceiveStreamImpl> CreateAudioReceiveStream() { + auto ret = std::make_unique<AudioReceiveStreamImpl>( Clock::GetRealTimeClock(), &packet_router_, stream_config_, audio_state_, &event_log_, std::unique_ptr<voe::ChannelReceiveInterface>(channel_receive_));
diff --git a/audio/audio_state.cc b/audio/audio_state.cc index fe71947..4215bcb 100644 --- a/audio/audio_state.cc +++ b/audio/audio_state.cc
@@ -55,7 +55,7 @@ RTC_DCHECK_EQ(0, receiving_streams_.count(stream)); receiving_streams_.insert(stream); if (!config_.audio_mixer->AddSource( - static_cast<internal::AudioReceiveStream*>(stream))) { + static_cast<AudioReceiveStreamImpl*>(stream))) { RTC_DLOG(LS_ERROR) << "Failed to add source to mixer."; } @@ -78,7 +78,7 @@ auto count = receiving_streams_.erase(stream); RTC_DCHECK_EQ(1, count); config_.audio_mixer->RemoveSource( - static_cast<internal::AudioReceiveStream*>(stream)); + static_cast<AudioReceiveStreamImpl*>(stream)); UpdateNullAudioPollerState(); if (receiving_streams_.empty()) { config_.audio_device_module->StopPlayout();
diff --git a/call/call.cc b/call/call.cc index 793f8d8..71fea06 100644 --- a/call/call.cc +++ b/call/call.cc
@@ -350,8 +350,8 @@ rtc::CopyOnWriteBuffer packet, int64_t packet_time_us) RTC_RUN_ON(worker_thread_); - AudioReceiveStream* FindAudioStreamForSyncGroup(absl::string_view sync_group) - RTC_RUN_ON(worker_thread_); + AudioReceiveStreamImpl* FindAudioStreamForSyncGroup( + absl::string_view sync_group) RTC_RUN_ON(worker_thread_); void ConfigureSync(absl::string_view sync_group) RTC_RUN_ON(worker_thread_); void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet, @@ -398,7 +398,7 @@ // creates them. // TODO(bugs.webrtc.org/11993): Move audio_receive_streams_, // video_receive_streams_ over to the network thread. - std::set<AudioReceiveStream*> audio_receive_streams_ + std::set<AudioReceiveStreamImpl*> audio_receive_streams_ RTC_GUARDED_BY(worker_thread_); std::set<VideoReceiveStream2*> video_receive_streams_ RTC_GUARDED_BY(worker_thread_); @@ -927,7 +927,7 @@ // TODO(bugs.webrtc.org/11993): call AssociateSendStream and // UpdateAggregateNetworkState asynchronously on the network thread. - for (AudioReceiveStream* stream : audio_receive_streams_) { + for (AudioReceiveStreamImpl* stream : audio_receive_streams_) { if (stream->local_ssrc() == config.rtp.ssrc) { stream->AssociateSendStream(send_stream); } @@ -955,7 +955,7 @@ // TODO(bugs.webrtc.org/11993): call AssociateSendStream and // UpdateAggregateNetworkState asynchronously on the network thread. - for (AudioReceiveStream* stream : audio_receive_streams_) { + for (AudioReceiveStreamImpl* stream : audio_receive_streams_) { if (stream->local_ssrc() == ssrc) { stream->AssociateSendStream(nullptr); } @@ -974,7 +974,7 @@ event_log_->Log(std::make_unique<RtcEventAudioReceiveStreamConfig>( CreateRtcLogStreamConfig(config))); - AudioReceiveStream* receive_stream = new AudioReceiveStream( + AudioReceiveStreamImpl* receive_stream = new AudioReceiveStreamImpl( clock_, transport_send_->packet_router(), config_.neteq_factory, config, config_.audio_state, event_log_); audio_receive_streams_.insert(receive_stream); @@ -1005,8 +1005,8 @@ TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream"); RTC_DCHECK_RUN_ON(worker_thread_); RTC_DCHECK(receive_stream != nullptr); - webrtc::internal::AudioReceiveStream* audio_receive_stream = - static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream); + webrtc::AudioReceiveStreamImpl* audio_receive_stream = + static_cast<webrtc::AudioReceiveStreamImpl*>(receive_stream); // TODO(bugs.webrtc.org/11993): Access the map, rtp config, call ConfigureSync // and UpdateAggregateNetworkState on the network thread. The call to @@ -1383,8 +1383,8 @@ void Call::OnLocalSsrcUpdated(webrtc::AudioReceiveStream& stream, uint32_t local_ssrc) { RTC_DCHECK_RUN_ON(worker_thread_); - webrtc::internal::AudioReceiveStream& receive_stream = - static_cast<webrtc::internal::AudioReceiveStream&>(stream); + webrtc::AudioReceiveStreamImpl& receive_stream = + static_cast<webrtc::AudioReceiveStreamImpl&>(stream); receive_stream.SetLocalSsrc(local_ssrc); auto it = audio_send_ssrcs_.find(local_ssrc); @@ -1407,8 +1407,8 @@ void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStream& stream, absl::string_view sync_group) { RTC_DCHECK_RUN_ON(worker_thread_); - webrtc::internal::AudioReceiveStream& receive_stream = - static_cast<webrtc::internal::AudioReceiveStream&>(stream); + webrtc::AudioReceiveStreamImpl& receive_stream = + static_cast<webrtc::AudioReceiveStreamImpl&>(stream); receive_stream.SetSyncGroup(sync_group); ConfigureSync(sync_group); } @@ -1477,11 +1477,11 @@ } // RTC_RUN_ON(worker_thread_) -AudioReceiveStream* Call::FindAudioStreamForSyncGroup( +AudioReceiveStreamImpl* Call::FindAudioStreamForSyncGroup( absl::string_view sync_group) { RTC_DCHECK_RUN_ON(&receive_11993_checker_); if (!sync_group.empty()) { - for (AudioReceiveStream* stream : audio_receive_streams_) { + for (AudioReceiveStreamImpl* stream : audio_receive_streams_) { if (stream->sync_group() == sync_group) return stream; } @@ -1494,7 +1494,8 @@ // RTC_RUN_ON(worker_thread_) void Call::ConfigureSync(absl::string_view sync_group) { // `audio_stream` may be nullptr when clearing the audio stream for a group. - AudioReceiveStream* audio_stream = FindAudioStreamForSyncGroup(sync_group); + AudioReceiveStreamImpl* audio_stream = + FindAudioStreamForSyncGroup(sync_group); size_t num_synced_streams = 0; for (VideoReceiveStream2* video_stream : video_receive_streams_) { @@ -1543,7 +1544,7 @@ rtcp_delivered = true; } - for (AudioReceiveStream* stream : audio_receive_streams_) { + for (AudioReceiveStreamImpl* stream : audio_receive_streams_) { stream->DeliverRtcp(packet.cdata(), packet.size()); rtcp_delivered = true; }
diff --git a/call/call_unittest.cc b/call/call_unittest.cc index e07e447..ee243ff 100644 --- a/call/call_unittest.cc +++ b/call/call_unittest.cc
@@ -199,8 +199,8 @@ AudioSendStream* send_stream = call->CreateAudioSendStream(send_config); EXPECT_NE(send_stream, nullptr); - internal::AudioReceiveStream* internal_recv_stream = - static_cast<internal::AudioReceiveStream*>(recv_stream); + AudioReceiveStreamImpl* internal_recv_stream = + static_cast<AudioReceiveStreamImpl*>(recv_stream); EXPECT_EQ(send_stream, internal_recv_stream->GetAssociatedSendStreamForTesting()); @@ -232,8 +232,8 @@ call->CreateAudioReceiveStream(recv_config); EXPECT_NE(recv_stream, nullptr); - internal::AudioReceiveStream* internal_recv_stream = - static_cast<internal::AudioReceiveStream*>(recv_stream); + AudioReceiveStreamImpl* internal_recv_stream = + static_cast<AudioReceiveStreamImpl*>(recv_stream); EXPECT_EQ(send_stream, internal_recv_stream->GetAssociatedSendStreamForTesting());