Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/rtc_peer_connection_handler.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
 #define CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
 
 #include "base/basictypes.h"
 #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;
 }
 
 namespace content {
 
 // 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
     : public PeerConnectionHandlerBase,
       NON_EXPORTED_BASE(public WebKit::WebRTCPeerConnectionHandler) {
  public:
   RTCPeerConnectionHandler(
       WebKit::WebRTCPeerConnectionHandlerClient* client,
       MediaStreamDependencyFactory* dependency_factory);
   virtual ~RTCPeerConnectionHandler();
@@ skipping 35 matching lines @@
       const WebKit::WebRTCICECandidate& candidate) OVERRIDE;
 
   virtual bool addStream(
       const WebKit::WebMediaStreamDescriptor& stream,
       const WebKit::WebMediaConstraints& options) OVERRIDE;
   virtual void removeStream(
       const WebKit::WebMediaStreamDescriptor& stream) OVERRIDE;
   // 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;
   virtual void OnAddStream(webrtc::MediaStreamInterface* stream) OVERRIDE;
   virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream) OVERRIDE;
   virtual void OnIceCandidate(
       const webrtc::IceCandidateInterface* candidate) OVERRIDE;
   virtual void OnIceComplete() OVERRIDE;
   virtual void OnRenegotiationNeeded() OVERRIDE;
 
  private:
   webrtc::SessionDescriptionInterface* CreateNativeSessionDescription(
       const WebKit::WebRTCSessionDescription& description);
 
   // |client_| is a weak pointer, and is valid until stop() has returned.
   WebKit::WebRTCPeerConnectionHandlerClient* client_;
 
   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);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_