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

content/renderer/media/webrtc_audio_renderer.cc

Index: content/renderer/media/webrtc_audio_renderer.cc
diff --git a/content/renderer/media/webrtc_audio_renderer.cc b/content/renderer/media/webrtc_audio_renderer.cc
index 49fc1857f83b3771212249d1dc3dec6efa067201..e993dc26f74fb923a5a3fb1211e366e2783fb65d 100644
--- a/content/renderer/media/webrtc_audio_renderer.cc
+++ b/content/renderer/media/webrtc_audio_renderer.cc
@@ -15,6 +15,7 @@
 #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"
@@ -586,14 +587,17 @@ void WebRtcAudioRenderer::OnPlayStateChanged(
   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))