Add support for unmixed audio from remote WebRTC remote tracks.

content/renderer/media/rtc_peer_connection_handler.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/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
@@ skipping 1343 matching lines @@
   return new RtcDataChannelHandler(base::ThreadTaskRunnerHandle::Get(),
                                    webrtc_channel);
 }
 
 blink::WebRTCDTMFSenderHandler* RTCPeerConnectionHandler::createDTMFSender(
     const blink::WebMediaStreamTrack& track) {
   DCHECK(thread_checker_.CalledOnValidThread());
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::createDTMFSender");
   DVLOG(1) << "createDTMFSender.";
 
-  MediaStreamTrack* native_track = MediaStreamTrack::GetTrack(track);
+  MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::GetTrack(track);
   if (!native_track || !native_track->is_local_track() ||
       track.source().type() != blink::WebMediaStreamSource::TypeAudio) {
     DLOG(ERROR) << "The DTMF sender requires a local audio track.";
     return nullptr;
   }
 
   scoped_refptr<webrtc::AudioTrackInterface> audio_track =
       native_track->GetAudioAdapter();
   rtc::scoped_refptr<webrtc::DtmfSenderInterface> sender(
       native_peer_connection_->CreateDtmfSender(audio_track.get()));
@@ skipping 293 matching lines @@
 }
 
 void RTCPeerConnectionHandler::ResetUMAStats() {
   DCHECK(thread_checker_.CalledOnValidThread());
   num_local_candidates_ipv6_ = 0;
   num_local_candidates_ipv4_ = 0;
   ice_connection_checking_start_ = base::TimeTicks();
   memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
 }
 }  // namespace content