Add support for unmixed audio from remote WebRTC remote tracks.

content/renderer/media/webrtc_audio_renderer.cc

 // Copyright (c) 2012 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.
 
 #include "content/renderer/media/webrtc_audio_renderer.h"
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
 #include "content/renderer/media/audio_device_factory.h"
+#include "content/renderer/media/media_stream_audio_track.h"
 #include "content/renderer/media/media_stream_dispatcher.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_logging.h"
 #include "content/renderer/render_frame_impl.h"
 #include "media/audio/audio_output_device.h"
 #include "media/audio/audio_parameters.h"
 #include "media/audio/sample_rates.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
@@ skipping 525 matching lines @@
 }
 
 void WebRtcAudioRenderer::OnPlayStateChanged(
     const blink::WebMediaStream& media_stream,
     PlayingState* state) {
   DCHECK(thread_checker_.CalledOnValidThread());
   blink::WebVector<blink::WebMediaStreamTrack> web_tracks;
   media_stream.audioTracks(web_tracks);
 
   for (const blink::WebMediaStreamTrack& web_track : web_tracks) {
-    MediaStreamTrack* track = MediaStreamTrack::GetTrack(web_track);
+    MediaStreamAudioTrack* track = MediaStreamAudioTrack::GetTrack(web_track);
     // WebRtcAudioRenderer can only render audio tracks received from a remote
     // peer. Since the actual MediaStream is mutable from JavaScript, we need
     // to make sure |web_track| is actually a remote track.
     if (track->is_local_track())
       continue;
     webrtc::AudioSourceInterface* source =
         track->GetAudioAdapter()->GetSource();
     DCHECK(source);
     if (!state->playing()) {
       if (RemovePlayingState(source, state))
@@ skipping 80 matching lines @@
     sink_params_ = new_sink_params;
     fifo_delay_milliseconds_ = new_fifo_delay_milliseconds;
     if (new_audio_fifo.get())
       audio_fifo_ = new_audio_fifo.Pass();
   }
 
   sink_->Initialize(new_sink_params, this);
 }
 
 }  // namespace content