Split libjingle's signaling thread from the UI thread

content/renderer/media/webrtc/peer_connection_dependency_factory.h

 // Copyright 2014 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_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_
 
 #include <string>
 
 #include "base/basictypes.h"
@@ skipping 105 matching lines @@
       webrtc::SdpParseError* error);
 
   // Creates a libjingle representation of an ice candidate.
   virtual webrtc::IceCandidateInterface* CreateIceCandidate(
       const std::string& sdp_mid,
       int sdp_mline_index,
       const std::string& sdp);
 
   WebRtcAudioDeviceImpl* GetWebRtcAudioDevice();
 
-  static void AddNativeAudioTrackToBlinkTrack(
-      webrtc::MediaStreamTrackInterface* native_track,
-      const blink::WebMediaStreamTrack& webkit_track,
-      bool is_local_track);
-
   scoped_refptr<base::MessageLoopProxy> GetWebRtcWorkerThread() const;
   scoped_refptr<base::MessageLoopProxy> GetWebRtcSignalingThread() const;
 
   // AecDumpMessageFilter::AecDumpDelegate implementation.
   // TODO(xians): Remove when option to disable audio track processing is
   // removed.
   void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
   void OnDisableAecDump() override;
   void OnIpcClosing() override;
 
@@ skipping 31 matching lines @@
   // Adds the audio device as a sink to the audio track and starts the local
   // audio track. This is virtual for test purposes since no real audio device
   // exist in unit tests.
   virtual void StartLocalAudioTrack(WebRtcLocalAudioTrack* audio_track);
 
  private:
   // Creates |pc_factory_|, which in turn is used for
   // creating PeerConnection objects.
   void CreatePeerConnectionFactory();
 
+  void InitializeSignalingThread(
+      const scoped_refptr<media::GpuVideoAcceleratorFactories>& gpu_factories,
+      base::WaitableEvent* event);
+
   void InitializeWorkerThread(rtc::Thread** thread,
                               base::WaitableEvent* event);
 
   void CreateIpcNetworkManagerOnWorkerThread(base::WaitableEvent* event);
   void DeleteIpcNetworkManager();
   void CleanupPeerConnectionFactory();
 
   // Helper method to create a WebRtcAudioDeviceImpl.
   void EnsureWebRtcAudioDeviceImpl();
 
   // We own network_manager_, must be deleted on the worker thread.
   // The network manager uses |p2p_socket_dispatcher_|.
   IpcNetworkManager* network_manager_;
   scoped_ptr<IpcPacketSocketFactory> socket_factory_;
 
   scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
 
   scoped_refptr<P2PSocketDispatcher> p2p_socket_dispatcher_;
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device_;
 
   // This is only used if audio track processing is disabled.
   // TODO(xians): Remove when option to disable audio track processing is
   // removed.
   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
 
   // PeerConnection threads. signaling_thread_ is created from the
   // "current" chrome thread.
   rtc::Thread* signaling_thread_;
   rtc::Thread* worker_thread_;
+  base::Thread chrome_signaling_thread_;
   base::Thread chrome_worker_thread_;
 
   DISALLOW_COPY_AND_ASSIGN(PeerConnectionDependencyFactory);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_PEER_CONNECTION_DEPENDENCY_FACTORY_H_