Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/rtc_peer_connection_handler.cc

Index: content/renderer/media/rtc_peer_connection_handler.cc
===================================================================
--- content/renderer/media/rtc_peer_connection_handler.cc	(revision 166546)
+++ content/renderer/media/rtc_peer_connection_handler.cc	(working copy)
@@ -12,7 +12,10 @@
 #include "base/utf_string_conversions.h"
 #include "content/renderer/media/media_stream_dependency_factory.h"
 #include "content/renderer/media/rtc_media_constraints.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebMediaConstraints.h"
+#include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamComponent.h"
+#include "third_party/WebKit/Source/Platform/chromium/public/WebMediaStreamSource.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCConfiguration.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCICECandidate.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCPeerConnectionHandlerClient.h"
@@ -306,6 +309,24 @@
 bool RTCPeerConnectionHandler::addStream(
     const WebKit::WebMediaStreamDescriptor& stream,
     const WebKit::WebMediaConstraints& options) {
+  // See if we're adding a WebAudio MediaStream.
+  WebKit::WebVector<WebKit::WebMediaStreamComponent> audioComponents;
+  stream.audioSources(audioComponents);
+  if (audioComponents.size() > 0) {

[perkj_chrome, 2012/11/12 10:35:11]

This seems to be in the wrong place and should be done when the underlaying
source is constructed in the media_stream_dependency_factory one way or the
other. Maybe via the MediaStreamCenter.

How does this work conceptually?
1. GetUserMedia creates a Ms with tracks where the source is a camera and a mic.


2. What is this doing? Are you replacing the mic source in the the ms from 1
with WebAudio or is this a new ms or is it a new audio track with a new source
you have added to 1? 

If you are replacing the the source in an existing audio track you should be
able to do that through the MediaStreamCenter and the
MediaStreamDependencyFactory.

If the output from WebAudio is a new MediaStream with a new audio source (the
output from the webaudio filter) - the MediaStream should be constructed through
MediaStreamCenter. Please see the existing constructor in MediaStreamCenter. 

+    if (audioComponents[0].source().isWebAudioSource()) {
+      DVLOG(1) << "RTCPeerConnectionHandler: adding WebAudio MediaStream.";
+      // 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.
+      base::AutoLock auto_lock(audio_lock_);
+      webaudio_capturer_source_ = new WebAudioCapturerSource();

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

Should we first DCHECK that webaudioÖcapturer_source_ is NULL?

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

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

no chance that GetWebRtcAudioDevice() returns NULL or would that be a serious
error?

+      capturer->SetCapturerSource(webaudio_capturer_source_);
+      capturer->Start();
+    }
+  }
+
   RTCMediaConstraints constraints(options);
   return AddStream(stream, &constraints);
 }
@@ -320,6 +341,14 @@
   native_peer_connection_ = NULL;
 }
 
+void RTCPeerConnectionHandler::consumeAudio(
+    const WebKit::WebVector<const float*>& audio_data,
+    size_t number_of_frames) {
+  base::AutoLock auto_lock(audio_lock_);
+  if (webaudio_capturer_source_.get())

[tommi (sloooow) - chröme, 2012/11/12 09:46:18]

nit: scoped_refptr supports checking validity without .get() and that seems to
be the norm now.

+    webaudio_capturer_source_->HandleCapture(audio_data, number_of_frames);
+}
+
 void RTCPeerConnectionHandler::OnError() {
   // TODO(perkj): Implement.
   NOTIMPLEMENTED();