Base A/V synchronization on sync_labels.

talk/media/webrtc/fakewebrtccall.cc

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 226 matching lines @@
 webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   return nullptr;
 }
 
 void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
 }
 
 webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
     const webrtc::AudioReceiveStream::Config& config) {
+  DCHECK(config.channel_id != -1);

[the sun, 2015/06/11 11:52:04]

Validate config in object ctor instead.

[pbos-webrtc, 2015/06/11 14:48:53]

Done.

   audio_receive_streams_.push_back(new FakeAudioReceiveStream(config));
   ++num_created_receive_streams_;
   return audio_receive_streams_.back();
 }
 
 void FakeCall::DestroyAudioReceiveStream(
     webrtc::AudioReceiveStream* receive_stream) {
   auto it = std::find(audio_receive_streams_.begin(),
                       audio_receive_streams_.end(),
                       static_cast<FakeAudioReceiveStream*>(receive_stream));
@@ skipping 96 matching lines @@
 
 void FakeCall::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   config_.bitrate_config = bitrate_config;
 }
 
 void FakeCall::SignalNetworkState(webrtc::Call::NetworkState state) {
   network_state_ = state;
 }
 }  // namespace cricket