MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

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 <utility>
 
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/strings/string_util.h"
 #include "base/strings/stringprintf.h"
 #include "build/build_config.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/peer_connection_remote_audio_source.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/webrtc/api/mediastreaminterface.h"
 #include "third_party/webrtc/media/base/audiorenderer.h"
 
@@ skipping 551 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) {
-    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())
+    PeerConnectionRemoteAudioTrack* const remote_track =
+        PeerConnectionRemoteAudioTrack::From(
+            MediaStreamAudioTrack::Get(web_track));
+    if (!remote_track)
       continue;
     webrtc::AudioSourceInterface* source =
-        track->GetAudioAdapter()->GetSource();
+        static_cast<webrtc::AudioTrackInterface*>(remote_track->track_interface())
+            ->GetSource();
     DCHECK(source);
     if (!state->playing()) {
       if (RemovePlayingState(source, state))
         EnterPauseState();
     } else if (AddPlayingState(source, state)) {
       EnterPlayState();
     }
     UpdateSourceVolume(source);
   }
 }
@@ skipping 56 matching lines @@
           base::Bind(&WebRtcAudioRenderer::SourceCallback,
                      base::Unretained(this))));
     }
     sink_params_ = new_sink_params;
   }
 
   sink_->Initialize(new_sink_params, this);
 }
 
 }  // namespace content