Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index f46337afd139e8e1175693e90820058ba6d90c00..4f5580669534d533c1cf5d886653e27ecb3682cc 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -225,13 +225,50 @@ void AudioReceiveStream::SetGain(float gain) {
   channel_proxy_->SetChannelOutputVolumeScaling(gain);
 }
 
-const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
+    int sample_rate_hz,
+    AudioFrame* audio_frame) {
+  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
+}
+
+int AudioReceiveStream::Ssrc() const {
+  return config_.rtp.remote_ssrc;
+}
+
+int AudioReceiveStream::PreferredSampleRate() const {
+  return channel_proxy_->NeededFrequency();
+}
+
+void AudioReceiveStream::GetRtpRtcp(RtpRtcp** rtp_rtcp,
+                                    RtpReceiver** rtp_receiver) const {
   RTC_DCHECK_RUN_ON(&thread_checker_);
-  return config_;
+  channel_proxy_->GetRtpRtcp(rtp_rtcp, rtp_receiver);
+}
+
+void AudioReceiveStream::GetDelayEstimate(int* jitter_buffer_delay_ms,
+                                          int* playout_buffer_delay_ms) const {
+  // Called on Call's module_process_thread_.
+  channel_proxy_->GetDelayEstimate(jitter_buffer_delay_ms,
+                                   playout_buffer_delay_ms);
+}
+
+uint32_t AudioReceiveStream::GetPlayoutTimestamp() const {
+  // Called on video capture thread.
+  return channel_proxy_->GetPlayoutTimestamp();
+}
+
+void AudioReceiveStream::SetMinimumPlayoutDelay(int delay_ms) {
+  // Called on Call's module_process_thread_.
+  return channel_proxy_->SetMinimumPlayoutDelay(delay_ms);
+}
+
+int AudioReceiveStream::id() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return config_.rtp.remote_ssrc;
 }
 
 void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
   if (send_stream) {
     VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
     std::unique_ptr<voe::ChannelProxy> send_channel_proxy =
@@ -282,18 +319,15 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }
 
-AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
-    int sample_rate_hz,
-    AudioFrame* audio_frame) {
-  return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
-}
-
-int AudioReceiveStream::PreferredSampleRate() const {
-  return channel_proxy_->NeededFrequency();
+const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return config_;
 }
 
-int AudioReceiveStream::Ssrc() const {
-  return config_.rtp.remote_ssrc;
+VoiceEngine* AudioReceiveStream::voice_engine() const {
+  auto* voice_engine = audio_state()->voice_engine();
+  RTC_DCHECK(voice_engine);
+  return voice_engine;
 }
 
 internal::AudioState* AudioReceiveStream::audio_state() const {
@@ -302,12 +336,6 @@ internal::AudioState* AudioReceiveStream::audio_state() const {
   return audio_state;
 }
 
-VoiceEngine* AudioReceiveStream::voice_engine() const {
-  auto* voice_engine = audio_state()->voice_engine();
-  RTC_DCHECK(voice_engine);
-  return voice_engine;
-}
-
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
@@ -316,6 +344,5 @@ int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
-
 }  // namespace internal
 }  // namespace webrtc