Upstream version 7.35.139.0

[platform/framework/web/crosswalk.git] / src / content / renderer / media / webrtc_local_audio_source_provider.cc

diff --git a/src/content/renderer/media/webrtc_local_audio_source_provider.cc b/src/content/renderer/media/webrtc_local_audio_source_provider.cc
index b460404..10ee943 100644 (file)
--- a/src/content/renderer/media/webrtc_local_audio_source_provider.cc
+++ b/src/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;
+}
+
 void WebRtcLocalAudioSourceProvider::OnData(
     const int16* audio_data,
     int sample_rate,