Notify the track before source provider goes away.

content/renderer/media/webrtc_local_audio_source_provider.cc

Index: content/renderer/media/webrtc_local_audio_source_provider.cc
diff --git a/content/renderer/media/webrtc_local_audio_source_provider.cc b/content/renderer/media/webrtc_local_audio_source_provider.cc
index b460404d18cd8cf866986b804d43f265a30a8fdc..10ee943f00fc435dc1743805cdf21e132fddd587 100644
--- a/content/renderer/media/webrtc_local_audio_source_provider.cc
+++ b/content/renderer/media/webrtc_local_audio_source_provider.cc
@@ -22,8 +22,11 @@ static const size_t kMaxNumberOfBuffers = 10;
 // static
 const size_t WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize = 128;
 
-WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider()
-    : is_enabled_(false) {
+WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider(
+    const blink::WebMediaStreamTrack& track)
+    : is_enabled_(false),
+      track_(track),
+      track_stopped_(false) {
   // Get the native audio output hardware sample-rate for the sink.
   // We need to check if RenderThreadImpl is valid here since the unittests
   // do not have one and they will inject their own |sink_params_| for testing.
@@ -36,11 +39,19 @@ WebRtcLocalAudioSourceProvider::WebRtcLocalAudioSourceProvider()
         media::CHANNEL_LAYOUT_STEREO, 2, 0, sample_rate, 16,
         kWebAudioRenderBufferSize);
   }
+
+  // Connect the source provider to the track as a sink.
+  MediaStreamAudioSink::AddToAudioTrack(this, track_);
 }
 
 WebRtcLocalAudioSourceProvider::~WebRtcLocalAudioSourceProvider() {
   if (audio_converter_.get())
     audio_converter_->RemoveInput(this);
+
+  // If the track is still active, it is necessary to notify the track before
+  // the source provider goes away.
+  if (!track_stopped_)
+    MediaStreamAudioSink::RemoveFromAudioTrack(this, track_);
 }
 
 void WebRtcLocalAudioSourceProvider::OnSetFormat(
@@ -68,6 +79,12 @@ void WebRtcLocalAudioSourceProvider::OnSetFormat(
                                        params.frames_per_buffer());
 }
 
+void WebRtcLocalAudioSourceProvider::OnReadyStateChanged(
+      blink::WebMediaStreamSource::ReadyState state) {
+  if (state ==  blink::WebMediaStreamSource::ReadyStateEnded)
+    track_stopped_ = true;

[perkj_chrome, 2014/04/03 07:45:43]

Can' you instead always remove sinks from teh tracks when the sink is dying? Why
do the track cleanup the sinks when its stopped? Why not just stop providing
data to the sinks?

[no longer working on chromium, 2014/04/03 09:18:57]

As discussed, lets do this in a follow-up to change all the sink to hold a
reference of the blink track object.

+}
+
 void WebRtcLocalAudioSourceProvider::OnData(
     const int16* audio_data,
     int sample_rate,