 5f93d0a140
			
		
	
	5f93d0a140
	
	
	
		
			
			BUG=2133 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
		
			
				
	
	
		
			336 lines
		
	
	
		
			12 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			336 lines
		
	
	
		
			12 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
| /*
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|  * libjingle
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|  * Copyright 2004 Google Inc.
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|  *
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|  * Redistribution and use in source and binary forms, with or without
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|  * modification, are permitted provided that the following conditions are met:
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|  *
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|  *  1. Redistributions of source code must retain the above copyright notice,
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|  *     this list of conditions and the following disclaimer.
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|  *  2. Redistributions in binary form must reproduce the above copyright notice,
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|  *     this list of conditions and the following disclaimer in the documentation
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|  *     and/or other materials provided with the distribution.
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|  *  3. The name of the author may not be used to endorse or promote products
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|  *     derived from this software without specific prior written permission.
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|  *
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|  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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|  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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|  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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|  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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|  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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|  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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|  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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|  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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|  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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|  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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|  */
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| 
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| #ifndef TALK_MEDIA_BASE_FILEMEDIAENGINE_H_
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| #define TALK_MEDIA_BASE_FILEMEDIAENGINE_H_
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| 
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| #include <string>
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| #include <vector>
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| 
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| #include "talk/media/base/codec.h"
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| #include "talk/media/base/mediachannel.h"
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| #include "talk/media/base/mediaengine.h"
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| #include "webrtc/base/scoped_ptr.h"
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| #include "webrtc/base/stream.h"
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| 
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| namespace rtc {
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| class StreamInterface;
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| }
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| 
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| namespace cricket {
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| 
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| // A media engine contains a capturer, an encoder, and a sender in the sender
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| // side and a receiver, a decoder, and a renderer in the receiver side.
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| // FileMediaEngine simulates the capturer and the encoder via an input RTP dump
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| // stream and simulates the decoder and the renderer via an output RTP dump
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| // stream. Depending on the parameters of the constructor, FileMediaEngine can
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| // act as file voice engine, file video engine, or both. Currently, we use
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| // only the RTP dump packets. TODO(whyuan): Enable RTCP packets.
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| class FileMediaEngine : public MediaEngineInterface {
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|  public:
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|   FileMediaEngine() : rtp_sender_thread_(NULL) {}
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|   virtual ~FileMediaEngine() {}
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| 
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|   // Set the file name of the input or output RTP dump for voice or video.
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|   // Should be called before the channel is created.
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|   void set_voice_input_filename(const std::string& filename) {
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|     voice_input_filename_ = filename;
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|   }
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|   void set_voice_output_filename(const std::string& filename) {
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|     voice_output_filename_ = filename;
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|   }
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|   void set_video_input_filename(const std::string& filename) {
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|     video_input_filename_ = filename;
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|   }
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|   void set_video_output_filename(const std::string& filename) {
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|     video_output_filename_ = filename;
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|   }
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| 
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|   // Should be called before codecs() and video_codecs() are called. We need to
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|   // set the voice and video codecs; otherwise, Jingle initiation will fail.
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|   void set_voice_codecs(const std::vector<AudioCodec>& codecs) {
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|     voice_codecs_ = codecs;
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|   }
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|   void set_video_codecs(const std::vector<VideoCodec>& codecs) {
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|     video_codecs_ = codecs;
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|   }
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| 
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|   // Implement pure virtual methods of MediaEngine.
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|   virtual bool Init(rtc::Thread* worker_thread) {
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|     return true;
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|   }
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|   virtual void Terminate() {}
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|   virtual int GetCapabilities();
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|   virtual VoiceMediaChannel* CreateChannel();
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|   virtual VideoMediaChannel* CreateVideoChannel(const VideoOptions& options,
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|                                                 VoiceMediaChannel* voice_ch);
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|   virtual SoundclipMedia* CreateSoundclip() { return NULL; }
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|   virtual AudioOptions GetAudioOptions() const { return AudioOptions(); }
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|   virtual bool SetAudioOptions(const AudioOptions& options) { return true; }
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|   virtual bool SetAudioDelayOffset(int offset) { return true; }
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|   virtual bool SetDefaultVideoEncoderConfig(const VideoEncoderConfig& config) {
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|     return true;
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|   }
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|   virtual bool SetSoundDevices(const Device* in_dev, const Device* out_dev) {
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|     return true;
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|   }
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|   virtual bool SetVideoCaptureDevice(const Device* cam_device) { return true; }
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|   virtual bool SetVideoCapturer(VideoCapturer* /*capturer*/) {
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|     return true;
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|   }
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|   virtual VideoCapturer* GetVideoCapturer() const {
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|     return NULL;
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|   }
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|   virtual bool GetOutputVolume(int* level) {
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|     *level = 0;
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|     return true;
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|   }
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|   virtual bool SetOutputVolume(int level) { return true; }
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|   virtual int GetInputLevel() { return 0; }
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|   virtual bool SetLocalMonitor(bool enable) { return true; }
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|   // TODO(whyuan): control channel send?
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|   virtual bool SetVideoCapture(bool capture) { return true; }
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|   virtual const std::vector<AudioCodec>& audio_codecs() {
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|     return voice_codecs_;
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|   }
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|   virtual const std::vector<VideoCodec>& video_codecs() {
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|     return video_codecs_;
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|   }
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|   virtual const std::vector<RtpHeaderExtension>& audio_rtp_header_extensions() {
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|     return audio_rtp_header_extensions_;
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|   }
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|   virtual const std::vector<RtpHeaderExtension>& video_rtp_header_extensions() {
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|     return video_rtp_header_extensions_;
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|   }
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| 
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|   virtual bool FindAudioCodec(const AudioCodec& codec) { return true; }
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|   virtual bool FindVideoCodec(const VideoCodec& codec) { return true; }
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|   virtual void SetVoiceLogging(int min_sev, const char* filter) {}
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|   virtual void SetVideoLogging(int min_sev, const char* filter) {}
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|   virtual bool StartAecDump(rtc::PlatformFile) { return false; }
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| 
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|   virtual bool RegisterVideoProcessor(VideoProcessor* processor) {
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|     return true;
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|   }
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|   virtual bool UnregisterVideoProcessor(VideoProcessor* processor) {
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|     return true;
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|   }
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|   virtual bool RegisterVoiceProcessor(uint32 ssrc,
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|                                       VoiceProcessor* processor,
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|                                       MediaProcessorDirection direction) {
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|     return true;
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|   }
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|   virtual bool UnregisterVoiceProcessor(uint32 ssrc,
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|                                         VoiceProcessor* processor,
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|                                         MediaProcessorDirection direction) {
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|     return true;
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|   }
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|   VideoFormat GetStartCaptureFormat() const {
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|     return VideoFormat();
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|   }
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| 
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|   virtual sigslot::repeater2<VideoCapturer*, CaptureState>&
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|       SignalVideoCaptureStateChange() {
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|     return signal_state_change_;
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|   }
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| 
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|   void set_rtp_sender_thread(rtc::Thread* thread) {
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|     rtp_sender_thread_ = thread;
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|   }
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| 
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|  private:
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|   std::string voice_input_filename_;
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|   std::string voice_output_filename_;
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|   std::string video_input_filename_;
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|   std::string video_output_filename_;
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|   std::vector<AudioCodec> voice_codecs_;
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|   std::vector<VideoCodec> video_codecs_;
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|   std::vector<RtpHeaderExtension> audio_rtp_header_extensions_;
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|   std::vector<RtpHeaderExtension> video_rtp_header_extensions_;
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|   sigslot::repeater2<VideoCapturer*, CaptureState>
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|      signal_state_change_;
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|   rtc::Thread* rtp_sender_thread_;
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| 
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|   DISALLOW_COPY_AND_ASSIGN(FileMediaEngine);
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| };
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| 
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| class RtpSenderReceiver;  // Forward declaration. Defined in the .cc file.
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| 
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| class FileVoiceChannel : public VoiceMediaChannel {
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|  public:
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|   FileVoiceChannel(rtc::StreamInterface* input_file_stream,
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|       rtc::StreamInterface* output_file_stream,
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|       rtc::Thread* rtp_sender_thread);
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|   virtual ~FileVoiceChannel();
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| 
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|   // Implement pure virtual methods of VoiceMediaChannel.
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|   virtual bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
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|     return true;
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|   }
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|   virtual bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
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|   virtual bool SetRecvRtpHeaderExtensions(
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|       const std::vector<RtpHeaderExtension>& extensions) {
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|     return true;
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|   }
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|   virtual bool SetSendRtpHeaderExtensions(
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|       const std::vector<RtpHeaderExtension>& extensions) {
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|     return true;
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|   }
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|   virtual bool SetPlayout(bool playout) { return true; }
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|   virtual bool SetSend(SendFlags flag);
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|   virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
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|     return false;
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|   }
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|   virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
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|     return false;
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|   }
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|   virtual bool GetActiveStreams(AudioInfo::StreamList* actives) { return true; }
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|   virtual int GetOutputLevel() { return 0; }
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|   virtual int GetTimeSinceLastTyping() { return -1; }
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|   virtual void SetTypingDetectionParameters(int time_window,
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|     int cost_per_typing, int reporting_threshold, int penalty_decay,
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|     int type_event_delay) {}
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| 
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|   virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
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|     return false;
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|   }
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|   virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
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|     return false;
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|   }
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|   virtual bool SetRingbackTone(const char* buf, int len) { return true; }
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|   virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
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|     return true;
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|   }
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|   virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags) {
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|     return false;
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|   }
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|   virtual bool GetStats(VoiceMediaInfo* info) { return true; }
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| 
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|   // Implement pure virtual methods of MediaChannel.
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|   virtual void OnPacketReceived(rtc::Buffer* packet,
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|                                 const rtc::PacketTime& packet_time);
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|   virtual void OnRtcpReceived(rtc::Buffer* packet,
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|                               const rtc::PacketTime& packet_time) {}
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|   virtual void OnReadyToSend(bool ready) {}
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|   virtual bool AddSendStream(const StreamParams& sp);
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|   virtual bool RemoveSendStream(uint32 ssrc);
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|   virtual bool AddRecvStream(const StreamParams& sp) { return true; }
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|   virtual bool RemoveRecvStream(uint32 ssrc) { return true; }
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|   virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
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|   virtual bool SetMaxSendBandwidth(int bps) { return true; }
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|   virtual bool SetOptions(const AudioOptions& options) {
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|     options_ = options;
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|     return true;
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|   }
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|   virtual bool GetOptions(AudioOptions* options) const {
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|     *options = options_;
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|     return true;
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|   }
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| 
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|  private:
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|   uint32 send_ssrc_;
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|   rtc::scoped_ptr<RtpSenderReceiver> rtp_sender_receiver_;
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|   AudioOptions options_;
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| 
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|   DISALLOW_COPY_AND_ASSIGN(FileVoiceChannel);
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| };
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| 
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| class FileVideoChannel : public VideoMediaChannel {
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|  public:
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|   FileVideoChannel(rtc::StreamInterface* input_file_stream,
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|       rtc::StreamInterface* output_file_stream,
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|       rtc::Thread* rtp_sender_thread);
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|   virtual ~FileVideoChannel();
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| 
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|   // Implement pure virtual methods of VideoMediaChannel.
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|   virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
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|     return true;
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|   }
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|   virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
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|   virtual bool GetSendCodec(VideoCodec* send_codec) {
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|     *send_codec = VideoCodec();
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|     return true;
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|   }
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|   virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) {
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|     return true;
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|   }
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|   virtual bool SetRecvRtpHeaderExtensions(
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|       const std::vector<RtpHeaderExtension>& extensions) {
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|     return true;
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|   }
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|   virtual bool SetSendRtpHeaderExtensions(
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|       const std::vector<RtpHeaderExtension>& extensions) {
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|     return true;
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|   }
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|   virtual bool SetRender(bool render) { return true; }
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|   virtual bool SetSend(bool send);
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|   virtual bool SetRenderer(uint32 ssrc, VideoRenderer* renderer) {
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|     return true;
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|   }
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|   virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
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|     return false;
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|   }
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|   virtual bool GetStats(const StatsOptions& options, VideoMediaInfo* info) {
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|     return true;
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|   }
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|   virtual bool SendIntraFrame() { return false; }
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|   virtual bool RequestIntraFrame() { return false; }
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| 
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|   // Implement pure virtual methods of MediaChannel.
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|   virtual void OnPacketReceived(rtc::Buffer* packet,
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|                                 const rtc::PacketTime& packet_time);
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|   virtual void OnRtcpReceived(rtc::Buffer* packet,
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|                               const rtc::PacketTime& packet_time) {}
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|   virtual void OnReadyToSend(bool ready) {}
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|   virtual bool AddSendStream(const StreamParams& sp);
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|   virtual bool RemoveSendStream(uint32 ssrc);
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|   virtual bool AddRecvStream(const StreamParams& sp) { return true; }
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|   virtual bool RemoveRecvStream(uint32 ssrc) { return true; }
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|   virtual bool MuteStream(uint32 ssrc, bool on) { return false; }
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|   virtual bool SetMaxSendBandwidth(int bps) { return true; }
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|   virtual bool SetOptions(const VideoOptions& options) {
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|     options_ = options;
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|     return true;
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|   }
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|   virtual bool GetOptions(VideoOptions* options) const {
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|     *options = options_;
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|     return true;
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|   }
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|   virtual void UpdateAspectRatio(int ratio_w, int ratio_h) {}
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| 
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|  private:
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|   uint32 send_ssrc_;
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|   rtc::scoped_ptr<RtpSenderReceiver> rtp_sender_receiver_;
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|   VideoOptions options_;
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| 
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|   DISALLOW_COPY_AND_ASSIGN(FileVideoChannel);
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| };
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| 
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| }  // namespace cricket
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| 
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| #endif  // TALK_MEDIA_BASE_FILEMEDIAENGINE_H_
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