Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one.

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/files/file.h"
@@ skipping 29 matching lines @@
 class WebMediaStreamSource;
 class WebMediaStreamTrack;
 class WebRTCPeerConnectionHandler;
 class WebRTCPeerConnectionHandlerClient;
 }
 
 namespace content {
 
 class IpcNetworkManager;
 class IpcPacketSocketFactory;
+class MediaStreamAudioSource;
+class WebAudioCapturerSource;
+class WebRtcAudioCapturer;
 class WebRtcAudioDeviceImpl;
+class WebRtcLocalAudioTrack;
 class WebRtcLoggingHandlerImpl;
 class WebRtcLoggingMessageFilter;
 class WebRtcVideoCapturerAdapter;
 struct StreamDeviceInfo;
 
 // Object factory for RTC PeerConnections.
 class CONTENT_EXPORT PeerConnectionDependencyFactory
     : NON_EXPORTED_BASE(base::MessageLoop::DestructionObserver),
       NON_EXPORTED_BASE(public base::NonThreadSafe) {
  public:
   PeerConnectionDependencyFactory(
       P2PSocketDispatcher* p2p_socket_dispatcher);
   ~PeerConnectionDependencyFactory() override;
 
   // Create a RTCPeerConnectionHandler object that implements the
   // WebKit WebRTCPeerConnectionHandler interface.
   blink::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler(
       blink::WebRTCPeerConnectionHandlerClient* client);
 
   // Generate an ECDSA certificate.
   static rtc::scoped_refptr<rtc::RTCCertificate> GenerateDefaultCertificate();
 
   // Asks the PeerConnection factory to create a Local MediaStream object.
   virtual scoped_refptr<webrtc::MediaStreamInterface>
       CreateLocalMediaStream(const std::string& label);
 
+  // InitializeMediaStreamAudioSource initialize a MediaStream source object
+  // for audio input.
+  bool InitializeMediaStreamAudioSource(
+      int render_frame_id,
+      const blink::WebMediaConstraints& audio_constraints,
+      MediaStreamAudioSource* source_data);
+
   // Creates an implementation of a cricket::VideoCapturer object that can be
   // used when creating a libjingle webrtc::VideoTrackSourceInterface object.
   virtual WebRtcVideoCapturerAdapter* CreateVideoCapturer(
       bool is_screen_capture);
 
+  // Creates an instance of WebRtcLocalAudioTrack and stores it
+  // in the extraData field of |track|.
+  void CreateLocalAudioTrack(const blink::WebMediaStreamTrack& track);
+
+  // Creates an instance of MediaStreamRemoteAudioTrack and associates with the
+  // |track| object.
+  void CreateRemoteAudioTrack(const blink::WebMediaStreamTrack& track);
+
   // Asks the PeerConnection factory to create a Local VideoTrack object.
   virtual scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
       const std::string& id,
       webrtc::VideoTrackSourceInterface* source);
 
   // Asks the PeerConnection factory to create a Video Source.
   // The video source takes ownership of |capturer|.
   virtual scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(
       cricket::VideoCapturer* capturer);
 
@@ skipping 22 matching lines @@
   // Starts recording an RTC event log.
   virtual bool StartRtcEventLog(base::PlatformFile file);
 
   // Starts recording an RTC event log.
   virtual void StopRtcEventLog();
 
   WebRtcAudioDeviceImpl* GetWebRtcAudioDevice();
 
   void EnsureInitialized();
   scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const;
-  virtual scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread()
-      const;
+  scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() const;
 
-  // Called by ProcessedLocalAudioSource to have the PeerConnection factory
-  // create the corresponding WebRtc-internal instance.
+ protected:
+  // Asks the PeerConnection factory to create a Local Audio Source.
   virtual scoped_refptr<webrtc::AudioSourceInterface> CreateLocalAudioSource(
       const cricket::AudioOptions& options);
 
- protected:
+  // Creates a media::AudioCapturerSource with an implementation that is
+  // specific for a WebAudio source. The created WebAudioCapturerSource
+  // instance will function as audio source instead of the default
+  // WebRtcAudioCapturer. Ownership of the new WebAudioCapturerSource is
+  // transferred to |source|.
+  virtual void CreateWebAudioSource(blink::WebMediaStreamSource* source);
+
   // Asks the PeerConnection factory to create a Local VideoTrack object with
   // the video source using |capturer|.
   virtual scoped_refptr<webrtc::VideoTrackInterface>
       CreateLocalVideoTrack(const std::string& id,
                             cricket::VideoCapturer* capturer);
 
   virtual const scoped_refptr<webrtc::PeerConnectionFactoryInterface>&
       GetPcFactory();
   virtual bool PeerConnectionFactoryCreated();
 
-  // Helper method to create a WebRtcAudioDeviceImpl.
-  void EnsureWebRtcAudioDeviceImpl();
+  // Returns a new capturer or existing capturer based on the |render_frame_id|
+  // and |device_info|; if both are valid, it reuses existing capture if any --
+  // otherwise it creates a new capturer.
+  virtual std::unique_ptr<WebRtcAudioCapturer> CreateAudioCapturer(
+      int render_frame_id,
+      const StreamDeviceInfo& device_info,
+      const blink::WebMediaConstraints& constraints,
+      MediaStreamAudioSource* audio_source);
 
  private:
   // Implement base::MessageLoop::DestructionObserver.
   // This makes sure the libjingle PeerConnectionFactory is released before
   // the renderer message loop is destroyed.
   void WillDestroyCurrentMessageLoop() override;
 
   // Functions related to Stun probing trial to determine how fast we could send
   // Stun request without being dropped by NAT.
   void TryScheduleStunProbeTrial();
   void StartStunProbeTrialOnWorkerThread(const std::string& params);
 
   // Creates |pc_factory_|, which in turn is used for
   // creating PeerConnection objects.
   void CreatePeerConnectionFactory();
 
   void InitializeSignalingThread(
       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_;
   std::unique_ptr<IpcPacketSocketFactory> socket_factory_;
 
   scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
 
   scoped_refptr<P2PSocketDispatcher> p2p_socket_dispatcher_;
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device_;
 
   std::unique_ptr<StunProberTrial> stun_trial_;
 
   // 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_