MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

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 <stddef.h>
 
 #include <vector>
@@ skipping 17 matching lines @@
 class AudioParameters;
 }
 
 namespace blink {
 class WebAudioSourceProviderClient;
 }
 
 namespace content {
 
 // WebRtcLocalAudioSourceProvider provides a bridge between classes:
-//     WebRtcLocalAudioTrack ---> blink::WebAudioSourceProvider
+//     MediaStreamAudioTrack ---> blink::WebAudioSourceProvider
 //
-// WebRtcLocalAudioSourceProvider works as a sink to the WebRtcLocalAudioTrack
-// and store the capture data to a FIFO. When the media stream is connected to
+// WebRtcLocalAudioSourceProvider works as a sink to the MediaStreamAudioTrack
+// and stores the capture data to a FIFO. When the media stream is connected to
 // WebAudio MediaStreamAudioSourceNode as a source provider,
 // MediaStreamAudioSourceNode will periodically call provideInput() to get the
 // data from the FIFO.
 //
 // All calls are protected by a lock.
+//
+// TODO(miu): This class should be renamed to WebAudioMediaStreamSink since it
+// works with all media stream tracks and there is nothing specific to WebRTC
+// here.
 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;
 
   explicit WebRtcLocalAudioSourceProvider(
       const blink::WebMediaStreamTrack& track);
   ~WebRtcLocalAudioSourceProvider() override;
@@ skipping 47 matching lines @@
 
   // 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_