Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/rtc_peer_connection_handler.h

Index: content/renderer/media/rtc_peer_connection_handler.h
===================================================================
--- content/renderer/media/rtc_peer_connection_handler.h	(revision 166546)
+++ content/renderer/media/rtc_peer_connection_handler.h	(working copy)
@@ -9,7 +9,9 @@
 #include "base/compiler_specific.h"
 #include "content/common/content_export.h"
 #include "content/renderer/media/peer_connection_handler_base.h"
+#include "content/renderer/media/webaudio_capturer_source.h"
 #include "third_party/WebKit/Source/Platform/chromium/public/WebRTCPeerConnectionHandler.h"
+#include "third_party/WebKit/Source/Platform/chromium/public/WebVector.h"
 
 namespace WebKit {
 class WebFrame;
@@ -20,7 +22,8 @@
 // RTCPeerConnectionHandler is a delegate for the RTC PeerConnection API
 // messages going between WebKit and native PeerConnection in libjingle. It's
 // owned by WebKit.
-// WebKit call all of these methods on the main render thread.
+// WebKit call all of these methods on the main render thread
+// except for consumeAudio().
 // Callbacks to the webrtc::PeerConnectionObserver implementation also occur on
 // the main render thread.
 class CONTENT_EXPORT RTCPeerConnectionHandler
@@ -76,6 +79,9 @@
   // We will be deleted by WebKit after stop has been returned.
   virtual void stop() OVERRIDE;
 
+  virtual void consumeAudio(const WebKit::WebVector<const float*>&,

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

missing param name (I know webkit is different)

+      size_t number_of_frames) OVERRIDE;
+
   // webrtc::PeerConnectionObserver implementation
   virtual void OnError() OVERRIDE;
   virtual void OnStateChange(StateType state_changed) OVERRIDE;
@@ -95,6 +101,12 @@
 
   WebKit::WebFrame* frame_;
 
+  // Handles audio from MediaStreamAudioDestinationNode.
+  scoped_refptr<WebAudioCapturerSource> webaudio_capturer_source_;
+
+  // Synchronizes uses of |webaudio_capturer_source_|.
+  base::Lock audio_lock_;
+
   DISALLOW_COPY_AND_ASSIGN(RTCPeerConnectionHandler);
 };