Add support for unmixed audio from remote WebRTC remote tracks.

content/renderer/media/webrtc/webrtc_media_stream_adapter.cc

 // Copyright 2014 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/webrtc_media_stream_adapter.h"
 
 #include "base/logging.h"
 #include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/webrtc/media_stream_video_webrtc_sink.h"
@@ skipping 59 matching lines @@
       }
     }
   }
 }
 
 void WebRtcMediaStreamAdapter::CreateAudioTrack(
     const blink::WebMediaStreamTrack& track) {
   DCHECK_EQ(track.source().type(), blink::WebMediaStreamSource::TypeAudio);
   // A media stream is connected to a peer connection, enable the
   // peer connection mode for the sources.
-  MediaStreamTrack* native_track = MediaStreamTrack::GetTrack(track);
+  MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::GetTrack(track);
   if (!native_track || !native_track->is_local_track()) {
     // We don't support connecting remote audio tracks to PeerConnection yet.
     // See issue http://crbug/344303.
     // TODO(xians): Remove this after we support connecting remote audio track
     // to PeerConnection.
     NOTIMPLEMENTED() << "webrtc audio track can not be created from a remote "
                      << "audio track.";
     return;
   }
 
   // This is a local audio track.
   const blink::WebMediaStreamSource& source = track.source();
   MediaStreamAudioSource* audio_source =
       static_cast<MediaStreamAudioSource*>(source.extraData());
   if (audio_source && audio_source->GetAudioCapturer().get())
     audio_source->GetAudioCapturer()->EnablePeerConnectionMode();

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

Fyi, here is interesting code... This is were I would like to create adapter
PeerConnection needs and do whatever EnablePeerConnectionMode() is doing. 

Ie,  get rid of the native_track->GetAudioAdaptuer all together and create a
sink to the audio track that adapts to the webrtc interface. The other way
around from what you just do in the MediaStreamRemoteAudioTrack.

Anyway, not related to this cl.

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

Ack.  I need to wrap my head around what's going on inside
EnablePeerConnectionMode() and why it's needed.  At the moment there are big
differences for the default renderers in how local and remote audio is rendered
(e.g. because of WebRTC's mixer), so we have to continue to have some specific
handling.

 
   webrtc_media_stream_->AddTrack(native_track->GetAudioAdapter());
 }
 
 void WebRtcMediaStreamAdapter::CreateVideoTrack(
     const blink::WebMediaStreamTrack& track) {
   DCHECK_EQ(track.source().type(), blink::WebMediaStreamSource::TypeVideo);
   MediaStreamVideoWebRtcSink* adapter =
       new MediaStreamVideoWebRtcSink(track, factory_);
   video_adapters_.push_back(adapter);
   webrtc_media_stream_->AddTrack(adapter->webrtc_video_track());
 }
 
 }  // namespace content