Notify the track before source provider goes away.

content/renderer/media/webrtc_local_audio_source_provider.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_
 
 #include <vector>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
 #include "content/public/renderer/media_stream_audio_sink.h"
 #include "media/base/audio_converter.h"
 #include "third_party/WebKit/public/platform/WebAudioSourceProvider.h"
+#include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebVector.h"
 
 namespace media {
 class AudioBus;
 class AudioConverter;
 class AudioFifo;
 class AudioParameters;
 }
 
 namespace blink {
 class WebAudioSourceProviderClient;
 }
 
 namespace content {
 
 // WebRtcLocalAudioSourceProvider provides a bridge between classes:
-//     WebRtcAudioCapturer ---> blink::WebAudioSourceProvider
+//     WebRtcLocalAudioTrack ---> blink::WebAudioSourceProvider
 //
-// WebRtcLocalAudioSourceProvider works as a sink to the WebRtcAudiocapturer
+// WebRtcLocalAudioSourceProvider works as a sink to the WebRtcLocalAudioTrack
 // and store the capture data to a FIFO. When the media stream is connected to
-// WebAudio as a source provider, WebAudio will periodically call
-// provideInput() to get the data from the FIFO.
+// WebAudio MediaStreamAudioSourceNode as a source provider,
+// MediaStreamAudioSourceNode will periodically call provideInput() to get the

[perkj_chrome, 2014/04/03 09:42:06]

Nothing to do now but isn't provideInput() a weird name for something that fetch
audio data from a FIFO? 

[no longer working on chromium, 2014/04/03 09:48:20]

It is an API designed by WebAudio to allow WebAudio to get data from audio
element. And I reuse it for media stream.

I am afraid we can't change it.

+// data from the FIFO.
 //
 // All calls are protected by a lock.
 class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
     :  NON_EXPORTED_BASE(public blink::WebAudioSourceProvider),
        NON_EXPORTED_BASE(public media::AudioConverter::InputCallback),
        NON_EXPORTED_BASE(public MediaStreamAudioSink) {
  public:
   static const size_t kWebAudioRenderBufferSize;
 
-  WebRtcLocalAudioSourceProvider();
+  explicit WebRtcLocalAudioSourceProvider(
+      const blink::WebMediaStreamTrack& track);
   virtual ~WebRtcLocalAudioSourceProvider();
 
   // MediaStreamAudioSink implementation.
   virtual void OnData(const int16* audio_data,
                       int sample_rate,
                       int number_of_channels,
                       int number_of_frames) OVERRIDE;
   virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE;
+  virtual void OnReadyStateChanged(
+      blink::WebMediaStreamSource::ReadyState state) OVERRIDE;
 
   // blink::WebAudioSourceProvider implementation.
   virtual void setClient(blink::WebAudioSourceProviderClient* client) OVERRIDE;
   virtual void provideInput(const blink::WebVector<float*>& audio_data,
                             size_t number_of_frames) OVERRIDE;
 
   // media::AudioConverter::Inputcallback implementation.
   // This function is triggered by provideInput()on the WebAudio audio thread,
   // so it has been under the protection of |lock_|.
   virtual double ProvideInput(media::AudioBus* audio_bus,
@@ skipping 19 matching lines @@
   bool is_enabled_;
   media::AudioParameters source_params_;
   media::AudioParameters sink_params_;
 
   // Protects all the member variables above.
   base::Lock lock_;
 
   // Used to report the correct delay to |webaudio_source_|.
   base::TimeTicks last_fill_;
 
+  // The audio track that this source provider is connected to.
+  blink::WebMediaStreamTrack track_;
+
+  // Flag to tell if the track has been stopped or not.
+  bool track_stopped_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_