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);
 
   // Add an ECDSA certificate to |config| in case it contains no certificate.
   static void SetDefaultCertificate(
       webrtc::PeerConnectionInterface::RTCConfiguration* config);
 
   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 24 matching lines @@
 
   // Starts recording an RTC event log.
   virtual void StopRtcEventLog();
 
   WebRtcAudioDeviceImpl* GetWebRtcAudioDevice();
 
   void EnsureInitialized();
   scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcWorkerThread() const;
   scoped_refptr<base::SingleThreadTaskRunner> GetWebRtcSignalingThread() const;
 
- protected:
-  // Asks the PeerConnection factory to create a Local Audio Source.
+  // Called by ProcessedLocalAudioSource to have the PeerConnection factory
+  // create the corresponding WebRtc-internal instance.
   virtual scoped_refptr<webrtc::AudioSourceInterface> CreateLocalAudioSource(
       const cricket::AudioOptions& options);
 
-  // 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);
-
+ protected:
   // 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();
 
-  // 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);
@@ skipping 34 matching lines @@
   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_