Support external audio mixer.

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 294 matching lines @@
     int sample_rate_hz,
     AudioFrame* audio_frame) {
   return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
 }
 
 int AudioReceiveStream::PreferredSampleRate() const {
   return channel_proxy_->NeededFrequency();
 }
 
 int AudioReceiveStream::Ssrc() const {
-  return config_.rtp.local_ssrc;
+  return config_.rtp.remote_ssrc;

[aleloi, 2016/12/05 14:03:08]

Thanks, I think it is correct now.

 }
 
 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