Base A/V synchronization on sync_labels.

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 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 1606 matching lines @@
   return CreateVoiceChannel(voe_wrapper_.get());
 }
 
 class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
     : public AudioRenderer::Sink {
  public:
   WebRtcVoiceChannelRenderer(int ch,
                              webrtc::AudioTransport* voe_audio_transport)
       : channel_(ch),
         voe_audio_transport_(voe_audio_transport),
-        renderer_(NULL) {
-  }
+        renderer_(NULL) {}
   ~WebRtcVoiceChannelRenderer() override { Stop(); }
 
   // Starts the rendering by setting a sink to the renderer to get data
   // callback.
   // This method is called on the libjingle worker thread.
   // TODO(xians): Make sure Start() is called only once.
   void Start(AudioRenderer* renderer) {
     rtc::CritScope lock(&lock_);
     DCHECK(renderer != NULL);
     if (renderer_ != NULL) {
@@ skipping 833 matching lines @@
       return false;
     }
 
     ConfigureSendChannel(channel);
   }
 
   // Save the channel to send_channels_, so that RemoveSendStream() can still
   // delete the channel in case failure happens below.
   webrtc::AudioTransport* audio_transport =
       engine()->voe()->base()->audio_transport();
-  send_channels_.insert(std::make_pair(
-      sp.first_ssrc(),
-      new WebRtcVoiceChannelRenderer(channel, audio_transport)));
+  send_channels_.insert(
+      std::make_pair(sp.first_ssrc(),
+                     new WebRtcVoiceChannelRenderer(channel, audio_transport)));
 
   // Set the send (local) SSRC.
   // If there are multiple send SSRCs, we can only set the first one here, and
   // the rest of the SSRC(s) need to be set after SetSendCodec has been called
   // (with a codec requires multiple SSRC(s)).
   if (engine()->voe()->rtp()->SetLocalSSRC(channel, sp.first_ssrc()) == -1) {
     LOG_RTCERR2(SetSendSSRC, channel, sp.first_ssrc());
     return false;
   }
 
@@ skipping 72 matching lines @@
   if (ssrc == 0) {
     LOG(LS_WARNING) << "AddRecvStream with 0 ssrc is not supported.";
     return false;
   }
 
   if (receive_channels_.find(ssrc) != receive_channels_.end()) {
     LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
     return false;
   }
 
-  TryAddAudioRecvStream(ssrc);
+  DCHECK(receive_stream_params_.find(ssrc) == receive_stream_params_.end());

[stefan-webrtc, 2015/06/16 15:33:39]

Can someone give us a fake sdp and cause a crash here, or is this ssrc always
generated locally?

[the sun, 2015/06/18 08:38:37]

Notice that we look for the ssrc in receive_channels_ a few lines up, and return
gracefully if we find it. This assert is just to make sure receive_channels_ and
receive_sgtream_params_ are in sync.

[stefan-webrtc, 2015/06/22 14:19:42]

Acknowledged.

[pbos-webrtc, 2015/06/24 09:46:09]

I moved the storing of the StreamParams to prevent having to erase them, now
they are stored at the same place as receive_channels_.

+  receive_stream_params_[ssrc] = sp;
 
   // Reuse default channel for recv stream in non-conference mode call
   // when the default channel is not being used.
   webrtc::AudioTransport* audio_transport =
       engine()->voe()->base()->audio_transport();
   if (!InConferenceMode() && default_receive_ssrc_ == 0) {
     LOG(LS_INFO) << "Recv stream " << ssrc << " reuse default channel";
     default_receive_ssrc_ = ssrc;
-    receive_channels_.insert(std::make_pair(
-        default_receive_ssrc_,
-        new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport)));
+    WebRtcVoiceChannelRenderer* channel_renderer =
+        new WebRtcVoiceChannelRenderer(voe_channel(), audio_transport);
+    receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
+    TryAddAudioRecvStream(ssrc);
     return SetPlayout(voe_channel(), playout_);
   }
 
   // Create a new channel for receiving audio data.
   int channel = engine()->CreateMediaVoiceChannel();
   if (channel == -1) {
     LOG_RTCERR0(CreateChannel);
+    receive_stream_params_.erase(ssrc);
     return false;
   }
 
   if (!ConfigureRecvChannel(channel)) {
     DeleteChannel(channel);
+    receive_stream_params_.erase(ssrc);
     return false;
   }
 
-  receive_channels_.insert(
-      std::make_pair(
-          ssrc, new WebRtcVoiceChannelRenderer(channel, audio_transport)));
+  WebRtcVoiceChannelRenderer* channel_renderer =
+      new WebRtcVoiceChannelRenderer(channel, audio_transport);
+  receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
+  TryAddAudioRecvStream(ssrc);
 
   LOG(LS_INFO) << "New audio stream " << ssrc
                << " registered to VoiceEngine channel #"
                << channel << ".";
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
   // Configure to use external transport, like our default channel.
   if (engine()->voe()->network()->RegisterExternalTransport(
@@ skipping 70 matching lines @@
   DCHECK(thread_checker_.CalledOnValidThread());
   rtc::CritScope lock(&receive_channels_cs_);
   ChannelMap::iterator it = receive_channels_.find(ssrc);
   if (it == receive_channels_.end()) {
     LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
                     << " which doesn't exist.";
     return false;
   }
 
   TryRemoveAudioRecvStream(ssrc);
+  receive_stream_params_.erase(ssrc);
 
   // Delete the WebRtcVoiceChannelRenderer object connected to the channel, this
   // will disconnect the audio renderer with the receive channel.
   // Cache the channel before the deletion.
   const int channel = it->second->channel();
   delete it->second;
   receive_channels_.erase(it);
 
   if (ssrc == default_receive_ssrc_) {
     DCHECK(IsDefaultChannel(channel));
@@ skipping 736 matching lines @@
   unsigned int ulevel;
   int ret =
       engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
   return (ret == 0) ? static_cast<int>(ulevel) : -1;
 }
 
 int WebRtcVoiceMediaChannel::GetReceiveChannelNum(uint32 ssrc) {
   ChannelMap::iterator it = receive_channels_.find(ssrc);
   if (it != receive_channels_.end())
     return it->second->channel();
-  return (ssrc == default_receive_ssrc_) ?  voe_channel() : -1;
+  return (ssrc == default_receive_ssrc_) ? voe_channel() : -1;
 }
 
 int WebRtcVoiceMediaChannel::GetSendChannelNum(uint32 ssrc) {
   ChannelMap::iterator it = send_channels_.find(ssrc);
   if (it != send_channels_.end())
     return it->second->channel();
 
   return -1;
 }
 
@@ skipping 152 matching lines @@
   }
   if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) {
     LOG_RTCERR4(*setter, uri, channel_id, enable, id);
     return false;
   }
   return true;
 }
 
 void WebRtcVoiceMediaChannel::TryAddAudioRecvStream(uint32 ssrc) {
   DCHECK(thread_checker_.CalledOnValidThread());
+  WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
+  DCHECK(channel != nullptr);
+  DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
   // If we are hooked up to a webrtc::Call, create an AudioReceiveStream too.
-  if (call_ && options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false)) {
-    DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
-    webrtc::AudioReceiveStream::Config config;
-    config.rtp.remote_ssrc = ssrc;
+  if (!call_) {
+    return;
+  }
+  webrtc::AudioReceiveStream::Config config;
+  config.rtp.remote_ssrc = ssrc;
+  // Only add RTP extensions if we support combined AV BWE.

[stefan-webrtc, 2015/06/16 15:33:39]

A/V BWE

[pbos-webrtc, 2015/06/24 09:46:09]

Done.

+  if (options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false)) {
     config.rtp.extensions = recv_rtp_extensions_;
-    webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
-    receive_streams_.insert(std::make_pair(ssrc, s));
   }
+  config.voe_channel_id = channel->channel();
+  config.sync_group = receive_stream_params_[ssrc].sync_label;

[stefan-webrtc, 2015/06/16 15:33:39]

Does this mean A/V BWE depends on sync labels? I would like some explanations on
this code.

[the sun, 2015/06/18 08:38:37]

No, we create the ARS regardless, but only set rtp extensions on it if we have
combined BWE enabled.

[stefan-webrtc, 2015/06/22 14:19:42]

Yes, I have no clue how I read this code. :)

+  webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
+  receive_streams_.insert(std::make_pair(ssrc, s));
 }
 
 void WebRtcVoiceMediaChannel::TryRemoveAudioRecvStream(uint32 ssrc) {
   DCHECK(thread_checker_.CalledOnValidThread());
   // If we are hooked up to a webrtc::Call, assume there is an
   // AudioReceiveStream to destroy too.
   if (call_) {
     auto stream_it = receive_streams_.find(ssrc);
     if (stream_it != receive_streams_.end()) {
       call_->DestroyAudioReceiveStream(stream_it->second);
@@ skipping 10 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE