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

content/renderer/media/webrtc/webrtc_media_stream_adapter.cc

Index: content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
diff --git a/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc b/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
index 4f5d8347f98068b07e5e0a6aae9514752f4b3592..b55575942ae5180b1b780e7e2a44b0d7f9cae52d 100644
--- a/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
+++ b/content/renderer/media/webrtc/webrtc_media_stream_adapter.cc
@@ -4,12 +4,15 @@
 
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 
+#include "base/bind.h"
 #include "base/logging.h"
-#include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_audio_track.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/webrtc/media_stream_video_webrtc_sink.h"
 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
+#include "content/renderer/media/webrtc/peer_connection_remote_audio_source.h"
+#include "content/renderer/media/webrtc/processed_local_audio_track.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebString.h"
@@ -78,27 +81,34 @@ void WebRtcMediaStreamAdapter::CreateAudioTrack(
   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.
-  MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::GetTrack(track);
+  MediaStreamAudioTrack* native_track = MediaStreamAudioTrack::Get(track);
   if (!native_track) {
     DLOG(ERROR) << "No native track for blink audio track.";
     return;
   }
 
-  webrtc::AudioTrackInterface* audio_track = native_track->GetAudioAdapter();
-  if (!audio_track) {
-    DLOG(ERROR) << "Audio track doesn't support webrtc.";
-    return;
-  }
-
-  if (native_track->is_local_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();
+  // If we have an instance of ProcessedLocalAudioTrack or
+  // PeerConnectionRemoteAudioTrack, use its webrtc::AudioTrackInterface
+  // implementation.  Otherwise, create a place-holder instance for tracks
+  // providing audio data from other sources.
+  scoped_refptr<webrtc::AudioTrackInterface> track_interface;
+  if (ProcessedLocalAudioTrack* local_rtc_track =
+          ProcessedLocalAudioTrack::From(native_track)) {
+    track_interface = local_rtc_track->adapter();
+  } else if (PeerConnectionRemoteAudioTrack* remote_pc_track =
+                 PeerConnectionRemoteAudioTrack::From(native_track)) {
+    track_interface = remote_pc_track->track_interface();
+  } else {
+    const scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter =
+        WebRtcLocalAudioTrackAdapter::Create(
+            track.id().utf8(), nullptr, factory_->GetWebRtcSignalingThread());
+    adapter->SetMediaStreamAudioTrack(native_track);
+    native_track->AddStopObserver(
+        base::Bind(&WebRtcLocalAudioTrackAdapter::SetMediaStreamAudioTrack,
+                   adapter, nullptr));
+    track_interface = adapter;
   }
-
-  webrtc_media_stream_->AddTrack(audio_track);
+  webrtc_media_stream_->AddTrack(track_interface.get());
 }
 
 void WebRtcMediaStreamAdapter::CreateVideoTrack(