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>
 
@@ skipping 18 matching lines @@
 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.
 class CONTENT_EXPORT WebRtcLocalAudioTrack

[perkj_chrome, 2015/12/11 12:31:38]

todo rename to match your MediaStreamRemoteAudioTrack? (again depending on if
you want to separate local or remote tracks in the end or not.

[tommi (sloooow) - chröme, 2015/12/11 15:36:51]

Done. (added TODO since the full separation is a little further out)

-    : 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_