Add support for unmixed audio from remote WebRTC remote tracks.

content/renderer/media/webrtc_local_audio_track.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_TRACK_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_
 
 #include <list>
 #include <string>
 
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "content/renderer/media/media_stream_track.h"

[perkj_chrome, 2015/12/11 15:54:41]

MediaStreamAudioTrack

[tommi (sloooow) - chröme, 2015/12/11 16:03:01]

yup (doh!)

 #include "content/renderer/media/tagged_list.h"
 #include "media/audio/audio_parameters.h"
 
 namespace media {
 class AudioBus;
 }
 
 namespace content {
 
 class MediaStreamAudioLevelCalculator;
 class MediaStreamAudioProcessor;
 class MediaStreamAudioSink;
 class MediaStreamAudioSinkOwner;
 class MediaStreamAudioTrackSink;
 class WebAudioCapturerSource;
 class WebRtcAudioCapturer;
 class WebRtcLocalAudioTrackAdapter;
 
 // A WebRtcLocalAudioTrack instance contains the implementations of
 // MediaStreamTrackExtraData.
 // When an instance is created, it will register itself as a track to the
 // WebRtcAudioCapturer to get the captured data, and forward the data to
 // its |sinks_|. The data flow can be stopped by disabling the audio track.
+// TODO(tommi): Rename to MediaStreamLocalAudioTrack.
 class CONTENT_EXPORT WebRtcLocalAudioTrack
-    : NON_EXPORTED_BASE(public MediaStreamTrack) {
+    : NON_EXPORTED_BASE(public MediaStreamAudioTrack) {
  public:
   WebRtcLocalAudioTrack(WebRtcLocalAudioTrackAdapter* adapter,
                         const scoped_refptr<WebRtcAudioCapturer>& capturer,
                         WebAudioCapturerSource* webaudio_source);
 
   ~WebRtcLocalAudioTrack() override;
 
   // Add a sink to the track. This function will trigger a OnSetFormat()
   // call on the |sink|.
   // Called on the main render thread.
-  void AddSink(MediaStreamAudioSink* sink);
+  void AddSink(MediaStreamAudioSink* sink) override;
 
   // Remove a sink from the track.
   // Called on the main render thread.
-  void RemoveSink(MediaStreamAudioSink* sink);
+  void RemoveSink(MediaStreamAudioSink* sink) override;
 
   // Starts the local audio track. Called on the main render thread and
   // should be called only once when audio track is created.
   void Start();
 
   // Overrides for MediaStreamTrack.
 
   void SetEnabled(bool enabled) override;
 
   // Stops the local audio track. Called on the main render thread and
   // should be called only once when audio track going away.
   void Stop() override;
 
   webrtc::AudioTrackInterface* GetAudioAdapter() override;
 
   // Returns the output format of the capture source. May return an invalid
   // AudioParameters if the format is not yet available.
   // Called on the main render thread.
-  media::AudioParameters GetOutputFormat() const;
+  media::AudioParameters GetOutputFormat() const override;
 
   // Method called by the capturer to deliver the capture data.
   // Called on the capture audio thread.
   void Capture(const media::AudioBus& audio_bus,
                base::TimeTicks estimated_capture_time,
                bool force_report_nonzero_energy);
 
   // Method called by the capturer to set the audio parameters used by source
   // of the capture data..
   // Called on the capture audio thread.
@@ skipping 40 matching lines @@
   // Used to calculate the signal level that shows in the UI.
   // Accessed on only the audio thread.
   scoped_ptr<MediaStreamAudioLevelCalculator> level_calculator_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioTrack);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_TRACK_H_