Add generic interfaces for the sinks of the media stream audio track

content/renderer/media/media_stream_impl.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/media_stream_impl.h"
 
 #include <utility>
 
 #include "base/logging.h"
 #include "base/strings/string_util.h"
@@ skipping 253 matching lines @@
     return NULL;  // This is not a valid stream.
 
   DVLOG(1) << "MediaStreamImpl::GetAudioRenderer stream:"
            << UTF16ToUTF8(web_stream.id());
 
   MediaStreamExtraData* extra_data =
       static_cast<MediaStreamExtraData*>(web_stream.extraData());
 
   if (extra_data->is_local()) {
     // Create the local audio renderer if the stream contains audio tracks.
-    return CreateLocalAudioRenderer(extra_data->stream().get());
+    blink::WebVector<blink::WebMediaStreamTrack> audio_tracks;
+    web_stream.audioTracks(audio_tracks);
+    if (audio_tracks.isEmpty())
+      return NULL;
+
+    // TODO(xians): Add support for the case that the media stream contains
+    // multiple audio tracks.
+    return CreateLocalAudioRenderer(audio_tracks[0]);
   }
 
   webrtc::MediaStreamInterface* stream = extra_data->stream().get();
   if (!stream || stream->GetAudioTracks().empty())
     return NULL;
 
   // This is a remote media stream.
   WebRtcAudioDeviceImpl* audio_device =
       dependency_factory_->GetWebRtcAudioDevice();
 
@@ skipping 507 matching lines @@
                                                &buffer_size)) {
     GetDefaultOutputDeviceParams(&sample_rate, &buffer_size);
   }
 
   return new WebRtcAudioRenderer(RenderViewObserver::routing_id(),
       session_id, sample_rate, buffer_size);
 }
 
 scoped_refptr<WebRtcLocalAudioRenderer>
 MediaStreamImpl::CreateLocalAudioRenderer(
-    webrtc::MediaStreamInterface* stream) {
-  if (stream->GetAudioTracks().empty())
-    return NULL;
-
-  DVLOG(1) << "MediaStreamImpl::CreateLocalAudioRenderer label:"
-           << stream->label();
-
-  webrtc::AudioTrackVector audio_tracks = stream->GetAudioTracks();
-  DCHECK_EQ(audio_tracks.size(), 1u);
-  webrtc::AudioTrackInterface* audio_track = audio_tracks[0];
-  DVLOG(1) << "audio_track.kind   : " << audio_track->kind()
-           << "audio_track.id     : " << audio_track->id()
-           << "audio_track.enabled: " << audio_track->enabled();
+    const blink::WebMediaStreamTrack& audio_track) {
+  DVLOG(1) << "MediaStreamImpl::CreateLocalAudioRenderer";
 
   int session_id = 0, sample_rate = 0, buffer_size = 0;
   if (!GetAuthorizedDeviceInfoForAudioRenderer(&session_id,
                                                &sample_rate,
                                                &buffer_size)) {
     GetDefaultOutputDeviceParams(&sample_rate, &buffer_size);
   }
 
   // Create a new WebRtcLocalAudioRenderer instance and connect it to the
   // existing WebRtcAudioCapturer so that the renderer can use it as source.
   return new WebRtcLocalAudioRenderer(
-      static_cast<WebRtcLocalAudioTrack*>(audio_track),
+      audio_track,
       RenderViewObserver::routing_id(),
       session_id,
       buffer_size);
 }
 
 bool MediaStreamImpl::GetAuthorizedDeviceInfoForAudioRenderer(
     int* session_id,
     int* output_sample_rate,
     int* output_frames_per_buffer) {
   DCHECK(CalledOnValidThread());
@@ skipping 58 matching lines @@
       enable_automatic_output_device_selection(
           enable_automatic_output_device_selection),
       frame(frame),
       request(request) {
 }
 
 MediaStreamImpl::UserMediaRequestInfo::~UserMediaRequestInfo() {
 }
 
 }  // namespace content