Rename WebKit namespace to blink (part 3)

content/renderer/media/media_stream_dependency_factory.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_MEDIA_STREAM_DEPENDENCY_FACTORY_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_DEPENDENCY_FACTORY_H_
 
 #include <string>
 
 #include "base/basictypes.h"
@@ skipping 12 matching lines @@
 namespace talk_base {
 class NetworkManager;
 class PacketSocketFactory;
 class Thread;
 }
 
 namespace webrtc {
 class PeerConnection;
 }
 
-namespace WebKit {
+namespace blink {
 class WebFrame;
 class WebMediaConstraints;
 class WebMediaStream;
 class WebRTCPeerConnectionHandler;
 class WebRTCPeerConnectionHandlerClient;
 }
 
 namespace content {
 
 class IpcNetworkManager;
 class IpcPacketSocketFactory;
 class RTCMediaConstraints;
 class VideoCaptureImplManager;
 class WebAudioCapturerSource;
 class WebRtcAudioCapturer;
 class WebRtcAudioDeviceImpl;
 class WebRtcLoggingHandlerImpl;
 class WebRtcLoggingMessageFilter;
 struct StreamDeviceInfo;
 
 #if defined(GOOGLE_TV)
 class RTCVideoDecoderFactoryTv;
 #endif
 
 // Object factory for RTC MediaStreams and RTC PeerConnections.
 class CONTENT_EXPORT MediaStreamDependencyFactory
     : NON_EXPORTED_BASE(public base::NonThreadSafe) {
  public:
   // MediaSourcesCreatedCallback is used in CreateNativeMediaSources.
-  typedef base::Callback<void(WebKit::WebMediaStream* web_stream,
+  typedef base::Callback<void(blink::WebMediaStream* web_stream,
                               bool live)> MediaSourcesCreatedCallback;
   MediaStreamDependencyFactory(
       VideoCaptureImplManager* vc_manager,
       P2PSocketDispatcher* p2p_socket_dispatcher);
   virtual ~MediaStreamDependencyFactory();
 
   // Create a RTCPeerConnectionHandler object that implements the
   // WebKit WebRTCPeerConnectionHandler interface.
-  WebKit::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler(
-      WebKit::WebRTCPeerConnectionHandlerClient* client);
+  blink::WebRTCPeerConnectionHandler* CreateRTCPeerConnectionHandler(
+      blink::WebRTCPeerConnectionHandlerClient* client);
 
   // CreateNativeMediaSources creates libjingle representations of
   // the underlying sources to the tracks in |web_stream|.
   // |sources_created| is invoked when the sources have either been created and
   // transitioned to a live state or failed.
   // The libjingle sources is stored in the extra data field of
   // WebMediaStreamSource.
   // |audio_constraints| and |video_constraints| set parameters for the sources.
   void CreateNativeMediaSources(
       int render_view_id,
-      const WebKit::WebMediaConstraints& audio_constraints,
-      const WebKit::WebMediaConstraints& video_constraints,
-      WebKit::WebMediaStream* web_stream,
+      const blink::WebMediaConstraints& audio_constraints,
+      const blink::WebMediaConstraints& video_constraints,
+      blink::WebMediaStream* web_stream,
       const MediaSourcesCreatedCallback& sources_created);
 
   // Creates a libjingle representation of a MediaStream and stores
   // it in the extra data field of |web_stream|.
   void CreateNativeLocalMediaStream(
-      WebKit::WebMediaStream* web_stream);
+      blink::WebMediaStream* web_stream);
 
   // Creates a libjingle representation of a MediaStream and stores
   // it in the extra data field of |web_stream|.
   // |stream_stopped| is a callback that is run when a MediaStream have been
   // stopped.
   void CreateNativeLocalMediaStream(
-      WebKit::WebMediaStream* web_stream,
+      blink::WebMediaStream* web_stream,
       const MediaStreamExtraData::StreamStopCallback& stream_stop);
 
   // Adds a libjingle representation of a MediaStreamTrack to |stream| based
   // on the source of |track|.
-  bool AddNativeMediaStreamTrack(const WebKit::WebMediaStream& stream,
-                                 const WebKit::WebMediaStreamTrack& track);
+  bool AddNativeMediaStreamTrack(const blink::WebMediaStream& stream,
+                                 const blink::WebMediaStreamTrack& track);
 
   // Creates and adds libjingle representation of a MediaStreamTrack to |stream|
   // based on the desired |track_id| and |capturer|.
   bool AddNativeVideoMediaTrack(const std::string& track_id,
-                                WebKit::WebMediaStream* stream,
+                                blink::WebMediaStream* stream,
                                 cricket::VideoCapturer* capturer);
 
-  bool RemoveNativeMediaStreamTrack(const WebKit::WebMediaStream& stream,
-                                    const WebKit::WebMediaStreamTrack& track);
+  bool RemoveNativeMediaStreamTrack(const blink::WebMediaStream& stream,
+                                    const blink::WebMediaStreamTrack& track);
 
   // Asks the libjingle PeerConnection factory to create a libjingle
   // PeerConnection object.
   // The PeerConnection object is owned by PeerConnectionHandler.
   virtual scoped_refptr<webrtc::PeerConnectionInterface>
       CreatePeerConnection(
           const webrtc::PeerConnectionInterface::IceServers& ice_servers,
           const webrtc::MediaConstraintsInterface* constraints,
-          WebKit::WebFrame* web_frame,
+          blink::WebFrame* web_frame,
           webrtc::PeerConnectionObserver* observer);
 
   // Creates a libjingle representation of a Session description. Used by a
   // RTCPeerConnectionHandler instance.
   virtual webrtc::SessionDescriptionInterface* CreateSessionDescription(
       const std::string& type,
       const std::string& sdp,
       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();
 
 #if defined(GOOGLE_TV)
   RTCVideoDecoderFactoryTv* decoder_factory_tv() { return decoder_factory_tv_; }
 #endif
 
   static void AddNativeTrackToBlinkTrack(
       webrtc::MediaStreamTrackInterface* native_track,
-      const WebKit::WebMediaStreamTrack& webkit_track);
+      const blink::WebMediaStreamTrack& webkit_track);
 
   static webrtc::MediaStreamInterface* GetNativeMediaStream(
-      const WebKit::WebMediaStream& stream);
+      const blink::WebMediaStream& stream);
 
   static webrtc::MediaStreamTrackInterface* GetNativeMediaStreamTrack(
-      const WebKit::WebMediaStreamTrack& track);
+      const blink::WebMediaStreamTrack& track);
 
  protected:
   // Asks the PeerConnection factory to create a Local MediaStream object.
   virtual scoped_refptr<webrtc::MediaStreamInterface>
       CreateLocalMediaStream(const std::string& label);
 
   // Asks the PeerConnection factory to create a Local Audio Source.
   virtual scoped_refptr<webrtc::AudioSourceInterface>
       CreateLocalAudioSource(
           const webrtc::MediaConstraintsInterface* constraints);
 
   // Asks the PeerConnection factory to create a Local Video Source.
   virtual scoped_refptr<webrtc::VideoSourceInterface>
       CreateLocalVideoSource(
           int video_session_id,
           bool is_screen_cast,
           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.
   // The |constraints| will be modified to include the default, mandatory
   // WebAudio constraints.
   virtual scoped_refptr<WebAudioCapturerSource> CreateWebAudioSource(
-      WebKit::WebMediaStreamSource* source, RTCMediaConstraints* constraints);
+      blink::WebMediaStreamSource* source, RTCMediaConstraints* constraints);
 
   // Asks the PeerConnection factory to create a Local AudioTrack object.
   virtual scoped_refptr<webrtc::AudioTrackInterface>
       CreateLocalAudioTrack(
           const std::string& id,
           const scoped_refptr<WebRtcAudioCapturer>& capturer,
           WebAudioCapturerSource* webaudio_source,
           webrtc::AudioSourceInterface* source,
           const webrtc::MediaConstraintsInterface* constraints);
 
@@ skipping 51 matching lines @@
   talk_base::Thread* signaling_thread_;
   talk_base::Thread* worker_thread_;
   base::Thread chrome_worker_thread_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamDependencyFactory);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_DEPENDENCY_FACTORY_H_