The Tor Browser: media/webrtc/signaling/src/media-conduit/VideoConduit.h@6474c204b198 (annotated)

media/webrtc/signaling/src/media-conduit/VideoConduit.h@6474c204b198 (annotated)

media/webrtc/signaling/src/media-conduit/VideoConduit.h

Wed, 31 Dec 2014 06:09:35 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Wed, 31 Dec 2014 06:09:35 +0100
changeset 0: 6474c204b198
permissions: -rwxr-xr-x

Cloned upstream origin tor-browser at tor-browser-31.3.0esr-4.5-1-build1
revision ID fc1c9ff7c1b2defdbc039f12214767608f46423f for hacking purpose.

 /* This Source Code Form is subject to the terms of the Mozilla Public
  * License, v. 2.0. If a copy of the MPL was not distributed with this file,
  * You can obtain one at http://mozilla.org/MPL/2.0/. */
 #ifndef VIDEO_SESSION_H_
 #define VIDEO_SESSION_H_
 #include "mozilla/Attributes.h"
 #include "MediaConduitInterface.h"
 #include "MediaEngineWrapper.h"
 // Video Engine Includes
 #include "webrtc/common_types.h"
 #ifdef FF
 #undef FF // Avoid name collision between scoped_ptr.h and nsCRTGlue.h.
 #endif
 #include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
 #include "webrtc/video_engine/include/vie_base.h"
 #include "webrtc/video_engine/include/vie_capture.h"
 #include "webrtc/video_engine/include/vie_codec.h"
 #include "webrtc/video_engine/include/vie_external_codec.h"
 #include "webrtc/video_engine/include/vie_render.h"
 #include "webrtc/video_engine/include/vie_network.h"
 #include "webrtc/video_engine/include/vie_rtp_rtcp.h"
 /** This file hosts several structures identifying different aspects
  * of a RTP Session.
  */
  using  webrtc::ViEBase;
  using  webrtc::ViENetwork;
  using  webrtc::ViECodec;
  using  webrtc::ViECapture;
  using  webrtc::ViERender;
  using  webrtc::ViEExternalCapture;
  using  webrtc::ViEExternalCodec;
 namespace mozilla {
 class WebrtcAudioConduit;
 // Interface of external video encoder for WebRTC.
 class WebrtcVideoEncoder:public VideoEncoder
                          ,public webrtc::VideoEncoder
 {};
 // Interface of external video decoder for WebRTC.
 class WebrtcVideoDecoder:public VideoDecoder
                          ,public webrtc::VideoDecoder
 {};
 /**
  * Concrete class for Video session. Hooks up
  *  - media-source and target to external transport
  */
 class WebrtcVideoConduit:public VideoSessionConduit
                          ,public webrtc::Transport
                          ,public webrtc::ExternalRenderer
 {
 public:
   //VoiceEngine defined constant for Payload Name Size.
   static const unsigned int CODEC_PLNAME_SIZE;
   /**
    * Set up A/V sync between this (incoming) VideoConduit and an audio conduit.
    */
   void SyncTo(WebrtcAudioConduit *aConduit);
   /**
    * Function to attach Renderer end-point for the Media-Video conduit.
    * @param aRenderer : Reference to the concrete Video renderer implementation
    * Note: Multiple invocations of this API shall remove an existing renderer
    * and attaches the new to the Conduit.
    */
   virtual MediaConduitErrorCode AttachRenderer(mozilla::RefPtr<VideoRenderer> aVideoRenderer);
   virtual void DetachRenderer();
   /**
    * APIs used by the registered external transport to this Conduit to
    * feed in received RTP Frames to the VideoEngine for decoding
    */
   virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len);
   /**
    * APIs used by the registered external transport to this Conduit to
    * feed in received RTP Frames to the VideoEngine for decoding
    */
   virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len);
    /**
    * Function to configure send codec for the video session
    * @param sendSessionConfig: CodecConfiguration
    * @result: On Success, the video engine is configured with passed in codec for send
    *          On failure, video engine transmit functionality is disabled.
    * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
    *        transmission sub-system on the engine.
    */
   virtual MediaConduitErrorCode ConfigureSendMediaCodec(const VideoCodecConfig* codecInfo);
   /**
    * Function to configure list of receive codecs for the video session
    * @param sendSessionConfig: CodecConfiguration
    * @result: On Success, the video engine is configured with passed in codec for send
    *          Also the playout is enabled.
    *          On failure, video engine transmit functionality is disabled.
    * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
    *        transmission sub-system on the engine.
    */
    virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
                                const std::vector<VideoCodecConfig* >& codecConfigList);
   /**
    * Register Transport for this Conduit. RTP and RTCP frames from the VideoEngine
    * shall be passed to the registered transport for transporting externally.
    */
   virtual MediaConduitErrorCode AttachTransport(mozilla::RefPtr<TransportInterface> aTransport);
   /**
    * Function to select and change the encoding resolution based on incoming frame size
    * and current available bandwidth.
    * @param width, height: dimensions of the frame
    */
   virtual bool SelectSendResolution(unsigned short width,
                                     unsigned short height);
   /**
    * Function to deliver a capture video frame for encoding and transport
    * @param video_frame: pointer to captured video-frame.
    * @param video_frame_length: size of the frame
    * @param width, height: dimensions of the frame
    * @param video_type: Type of the video frame - I420, RAW
    * @param captured_time: timestamp when the frame was captured.
    *                       if 0 timestamp is automatcally generated by the engine.
    *NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can be invoked
    *       This ensures the inserted video-frames can be transmitted by the conduit
    */
   virtual MediaConduitErrorCode SendVideoFrame(unsigned char* video_frame,
                                                 unsigned int video_frame_length,
                                                 unsigned short width,
                                                 unsigned short height,
                                                 VideoType video_type,
                                                 uint64_t capture_time);
   /**
    * Set an external encoder object |encoder| to the payload type |pltype|
    * for sender side codec.
    */
   virtual MediaConduitErrorCode SetExternalSendCodec(int pltype,
                                                      VideoEncoder* encoder);
   /**
    * Set an external decoder object |decoder| to the payload type |pltype|
    * for receiver side codec.
    */
   virtual MediaConduitErrorCode SetExternalRecvCodec(int pltype,
                                                      VideoDecoder* decoder);
   /**
    * Webrtc transport implementation to send and receive RTP packet.
    * VideoConduit registers itself as ExternalTransport to the VideoEngine
    */
   virtual int SendPacket(int channel, const void *data, int len) ;
   /**
    * Webrtc transport implementation to send and receive RTCP packet.
    * VideoConduit registers itself as ExternalTransport to the VideoEngine
    */
   virtual int SendRTCPPacket(int channel, const void *data, int len) ;
   /**
    * Webrtc External Renderer Implementation APIs.
    * Raw I420 Frames are delivred to the VideoConduit by the VideoEngine
    */
   virtual int FrameSizeChange(unsigned int, unsigned int, unsigned int);
   virtual int DeliverFrame(unsigned char*,int, uint32_t , int64_t,
                            void *handle);
   /**
    * Does DeliverFrame() support a null buffer and non-null handle
    * (video texture)?
    * B2G support it (when using HW video decoder with graphic buffer output).
    * XXX Investigate!  Especially for Android
    */
   virtual bool IsTextureSupported() {
 #ifdef WEBRTC_GONK
     return true;
 #else
     return false;
 #endif
   }
   unsigned short SendingWidth() {
     return mSendingWidth;
   }
   unsigned short SendingHeight() {
     return mSendingHeight;
   }
   unsigned int SendingMaxFs() {
     if(mCurSendCodecConfig) {
       return mCurSendCodecConfig->mMaxFrameSize;
     }
     return 0;
   }
   unsigned int SendingMaxFr() {
     if(mCurSendCodecConfig) {
       return mCurSendCodecConfig->mMaxFrameRate;
     }
     return 0;
   }
   WebrtcVideoConduit():
                       mOtherDirection(nullptr),
                       mShutDown(false),
                       mVideoEngine(nullptr),
                       mTransport(nullptr),
                       mRenderer(nullptr),
                       mPtrExtCapture(nullptr),
                       mEngineTransmitting(false),
                       mEngineReceiving(false),
                       mChannel(-1),
                       mCapId(-1),
                       mCurSendCodecConfig(nullptr),
                       mSendingWidth(0),
 		      mSendingHeight(0),
 		      mReceivingWidth(640),
 		      mReceivingHeight(480),
 		      mVideoLatencyTestEnable(false),
 		      mVideoLatencyAvg(0)
   {
   }
   virtual ~WebrtcVideoConduit() ;
   MediaConduitErrorCode Init(WebrtcVideoConduit *other);
   int GetChannel() { return mChannel; }
   webrtc::VideoEngine* GetVideoEngine() { return mVideoEngine; }
   bool GetLocalSSRC(unsigned int* ssrc);
   bool GetRemoteSSRC(unsigned int* ssrc);
   bool GetAVStats(int32_t* jitterBufferDelayMs,
                   int32_t* playoutBufferDelayMs,
                   int32_t* avSyncOffsetMs);
   bool GetRTPStats(unsigned int* jitterMs, unsigned int* cumulativeLost);
   bool GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
                              uint32_t* jitterMs,
                              uint32_t* packetsReceived,
                              uint64_t* bytesReceived,
                              uint32_t* cumulativeLost,
                              int32_t* rttMs);
   bool GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
                            unsigned int* packetsSent,
                            uint64_t* bytesSent);
   uint64_t MozVideoLatencyAvg();
 private:
   WebrtcVideoConduit(const WebrtcVideoConduit& other) MOZ_DELETE;
   void operator=(const WebrtcVideoConduit& other) MOZ_DELETE;
   //Local database of currently applied receive codecs
   typedef std::vector<VideoCodecConfig* > RecvCodecList;
   //Function to convert between WebRTC and Conduit codec structures
   void CodecConfigToWebRTCCodec(const VideoCodecConfig* codecInfo,
                                 webrtc::VideoCodec& cinst);
   // Function to copy a codec structure to Conduit's database
   bool CopyCodecToDB(const VideoCodecConfig* codecInfo);
   // Functions to verify if the codec passed is already in
   // conduits database
   bool CheckCodecForMatch(const VideoCodecConfig* codecInfo) const;
   bool CheckCodecsForMatch(const VideoCodecConfig* curCodecConfig,
                            const VideoCodecConfig* codecInfo) const;
   //Checks the codec to be applied
   MediaConduitErrorCode ValidateCodecConfig(const VideoCodecConfig* codecInfo, bool send) const;
   //Utility function to dump recv codec database
   void DumpCodecDB() const;
   // Video Latency Test averaging filter
   void VideoLatencyUpdate(uint64_t new_sample);
   // The two sides of a send/receive pair of conduits each keep a pointer to the other.
   // They also share a single VideoEngine and mChannel.  Shutdown must be coordinated
   // carefully to avoid double-freeing or accessing after one frees.
   WebrtcVideoConduit*  mOtherDirection;
   // The other side has shut down our mChannel and related items already
   bool mShutDown;
   // A few of these are shared by both directions.  They're released by the last
   // conduit to die.
   webrtc::VideoEngine* mVideoEngine;          // shared
   mozilla::RefPtr<TransportInterface> mTransport;
   mozilla::RefPtr<VideoRenderer> mRenderer;
   ScopedCustomReleasePtr<webrtc::ViEBase> mPtrViEBase;
   ScopedCustomReleasePtr<webrtc::ViECapture> mPtrViECapture;
   ScopedCustomReleasePtr<webrtc::ViECodec> mPtrViECodec;
   ScopedCustomReleasePtr<webrtc::ViENetwork> mPtrViENetwork;
   ScopedCustomReleasePtr<webrtc::ViERender> mPtrViERender;
   ScopedCustomReleasePtr<webrtc::ViERTP_RTCP> mPtrRTP;
   ScopedCustomReleasePtr<webrtc::ViEExternalCodec> mPtrExtCodec;
   webrtc::ViEExternalCapture* mPtrExtCapture; // shared
   // Engine state we are concerned with.
   bool mEngineTransmitting; //If true ==> Transmit Sub-system is up and running
   bool mEngineReceiving;    // if true ==> Receive Sus-sysmtem up and running
   int mChannel; // Video Channel for this conduit
   int mCapId;   // Capturer for this conduit
   RecvCodecList    mRecvCodecList;
   VideoCodecConfig* mCurSendCodecConfig;
   unsigned short mSendingWidth;
   unsigned short mSendingHeight;
   unsigned short mReceivingWidth;
   unsigned short mReceivingHeight;
   bool mVideoLatencyTestEnable;
   uint64_t mVideoLatencyAvg;
   static const unsigned int sAlphaNum = 7;
   static const unsigned int sAlphaDen = 8;
   static const unsigned int sRoundingPadding = 1024;
   mozilla::RefPtr<WebrtcAudioConduit> mSyncedTo;
 };
 } // end namespace
 #endif

The Tor Browser / annotate

media/webrtc/signaling/src/media-conduit/VideoConduit.h@6474c204b198 (annotated)

media/webrtc/signaling/src/media-conduit/VideoConduit.h