Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/media_stream_dependency_factory.cc

Index: content/renderer/media/media_stream_dependency_factory.cc
===================================================================
--- content/renderer/media/media_stream_dependency_factory.cc	(revision 172673)
+++ content/renderer/media/media_stream_dependency_factory.cc	(working copy)
@@ -237,26 +237,59 @@
   // Add audio tracks.
   WebKit::WebVector<WebKit::WebMediaStreamComponent> audio_components;
   description->audioSources(audio_components);
+
   for (size_t i = 0; i < audio_components.size(); ++i) {
-    const WebKit::WebMediaStreamSource& source = audio_components[i].source();
-    MediaStreamSourceExtraData* source_data =
-        static_cast<MediaStreamSourceExtraData*>(source.extraData());
-    if (!source_data) {
-      // TODO(perkj): Implement support for sources from remote MediaStreams.
-      NOTIMPLEMENTED();
-      continue;
+    WebKit::WebMediaStreamSource source = audio_components[i].source();
+
+    // See if we're adding a WebAudio MediaStream.
+    if (source.requiresAudioConsumer()) {
+      if (!webaudio_capturer_source_.get()) {
+        DVLOG(1) << "CreateNativeLocalMediaStream: WebAudio MediaStream.";
+        DCHECK(GetWebRtcAudioDevice());
+
+        // TODO(crogers, xians): In reality we should be able to send a unique
+        // audio stream to each PeerConnection separately.  But currently WebRTC
+        // is only able to handle a global audio stream sent to ALL peers.
+        webaudio_capturer_source_ = new WebAudioCapturerSource();

[perkj_chrome, 2012/12/17 10:59:20]

Why is webauido_capturer_source_ stored at all in MediaStreamDependencyFactory?
If it is never used again just loos this reference. If you need it again I
suggest you create  a MediaStreamSourceExtraData and store it in there. That way
Webkit controls the lifetime and it can be fetched everywhere a MediaStream is
used.

+        WebRtcAudioCapturer* capturer = GetWebRtcAudioDevice()->capturer();

[henrika (OOO until Aug 14), 2012/12/18 12:23:02]

Note that this guy can return NULL for non-supported sample rates, e.g. 88200,
22050 etc. We must ensure that these cases does not crash.

+        capturer->SetCapturerSource(webaudio_capturer_source_);
+        capturer->Start();
+
+        // For lifetime, we're relying on the fact that
+        // |webaudio_capturer_source_| will live longer than any
+        // MediaStreamSource, since we're never calling removeAudioConsumer().
+        source.addAudioConsumer(webaudio_capturer_source_.get());

[perkj_chrome, 2012/12/17 10:59:20]

Does addAudioConsumer ref count webaudio_capturer_source_ so it is guaranteed to
live even if a user creates another WebAudio MediaStream.

+      } else {
+        // TODO(crogers): this is very likely to be less important, but
+        // in theory we should be able to "connect" multiple WebAudio
+        // MediaStreams to a single peer, mixing their results.

[perkj_chrome, 2012/12/17 10:59:20]

Multiple MediaStreams will result in multiple calls to
MediaStreamDependencyFactory::CreateNativeLocalMediaStream. You can end up here
if a JS creates a new MediaStream based on another MediaStream. Ex creating a
copy of a MediaStream.

+        // Instead we just ignore additional ones after the first.
+        DVLOG(1)
+            << "Multiple MediaStreamAudioDestinationNodes not yet supported!";
+      }
+    } else {
+        MediaStreamSourceExtraData* source_data =
+            static_cast<MediaStreamSourceExtraData*>(source.extraData());
+
+        if (!source_data) {
+          // TODO(perkj): Implement support for sources from
+          // remote MediaStreams.
+          NOTIMPLEMENTED();
+          continue;
+        }
+
+        // TODO(perkj): Refactor the creation of audio tracks to use a proper
+        // interface for receiving audio input data. Currently NULL is passed
+        // since the |audio_device| is the wrong class and is unused.
+        scoped_refptr<webrtc::LocalAudioTrackInterface> audio_track(

[perkj_chrome, 2012/12/17 15:38:24]

line 284 to 287 must be done even for webaudio sources. The native
repressentation of the track must be created and added to the native stream.

[henrika (OOO until Aug 14), 2012/12/18 13:35:15]

I tried this but it did not work. Per adviced me to try out:

scoped_refptr<webrtc::LocalAudioTrackInterface> audio_track(
            CreateLocalAudioTrack(label + "a0", NULL));

and that worked.

[perkj_chrome, 2012/12/18 14:09:04]

Yes- the tracks labels must follow the msid rfc (see dreft alvestrand 02) a0
stand for the first audio track in the stream with label |label|. It does not
make sence that chrome must know this, so feel free to add a todo with my name
to remove the label from track creation. 

+            CreateLocalAudioTrack(UTF16ToUTF8(source.id()), NULL));
+        native_stream->AddTrack(audio_track);
+        audio_track->set_enabled(audio_components[i].isEnabled());
+        // TODO(xians): This set the source of all audio tracks to the same
+        // microphone. Implement support for setting the source per audio track
+        // instead.
+        SetAudioDeviceSessionId(source_data->device_info().session_id);
     }
-    // TODO(perkj): Refactor the creation of audio tracks to use a proper
-    // interface for receiving audio input data. Currently NULL is passed since
-    // the |audio_device| is the wrong class and is unused.
-    scoped_refptr<webrtc::LocalAudioTrackInterface> audio_track(
-        CreateLocalAudioTrack(UTF16ToUTF8(source.id()), NULL));
-    native_stream->AddTrack(audio_track);
-    audio_track->set_enabled(audio_components[i].isEnabled());
-    // TODO(xians): This set the source of all audio tracks to the same
-    // microphone. Implement support for setting the source per audio track
-    // instead.
-    SetAudioDeviceSessionId(source_data->device_info().session_id);
   }
 
   // Add video tracks.