Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.

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 136 matching lines @@
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }
 }
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  int error = base->StartPlayout(config_.voe_channel_id);
+  if (error != 0) {
+    LOG(LS_ERROR) << "AudioReceiveStream::Start failed with error: " << error;
+  }

[aleloi, 2016/08/03 13:09:58]

Similar to AudioSendStream::Start/Stop

 }
 
 void AudioReceiveStream::Stop() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  int error = base->StopPlayout(config_.voe_channel_id);
+  if (error != 0) {
+    LOG(LS_ERROR) << "AudioReceiveStream::Start failed with error: " << error;
+  }
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioReceiveStream::Stats stats;
   stats.remote_ssrc = config_.rtp.remote_ssrc;
   ScopedVoEInterface<VoECodec> codec(voice_engine());
 
   webrtc::CallStatistics call_stats = channel_proxy_->GetRTCPStatistics();
   webrtc::CodecInst codec_inst = {0};
@@ skipping 94 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc