Clean up the media stream audio track code

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 37 matching lines @@
 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(public base::NonThreadSafe),
-      NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
+    : NON_EXPORTED_BASE(public base::NonThreadSafe) {
  public:
   PeerConnectionDependencyFactory(
       P2PSocketDispatcher* p2p_socket_dispatcher);
   virtual ~PeerConnectionDependencyFactory();
 
   // Create a RTCPeerConnectionHandler object that implements the
   // WebKit WebRTCPeerConnectionHandler interface.
   blink::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler(
       blink::WebRTCPeerConnectionHandlerClient* client);
 
@@ skipping 49 matching lines @@
   virtual webrtc::IceCandidateInterface* CreateIceCandidate(
       const std::string& sdp_mid,
       int sdp_mline_index,
       const std::string& sdp);
 
   WebRtcAudioDeviceImpl* GetWebRtcAudioDevice();
 
   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;
-
  protected:
   // Asks the PeerConnection factory to create a Local Audio Source.
   virtual scoped_refptr<webrtc::AudioSourceInterface>
       CreateLocalAudioSource(
           const webrtc::MediaConstraintsInterface* constraints);
 
   // 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.
@@ skipping 45 matching lines @@
   // 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_