Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 207 matching lines @@
 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   channel_proxy_->SetSink(std::move(sink));
 }
 
 void AudioReceiveStream::SetGain(float gain) {
   RTC_DCHECK_RUN_ON(&thread_checker_);
   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 =
         voe_impl->GetChannelProxy(send_stream->config().voe_channel_id);
     channel_proxy_->AssociateSendChannel(*send_channel_proxy.get());
   } else {
     channel_proxy_->DisassociateSendChannel();
   }
 }
 
@@ skipping 30 matching lines @@
     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header);
   }
 
   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);
+const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return config_;
 }
 
-int AudioReceiveStream::PreferredSampleRate() const {
-  return channel_proxy_->NeededFrequency();
-}
-
-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 {
   auto* audio_state = static_cast<internal::AudioState*>(audio_state_.get());
   RTC_DCHECK(audio_state);
   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) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
-
 }  // namespace internal
 }  // namespace webrtc