1.1 --- /dev/null Thu Jan 01 00:00:00 1970 +0000
1.2 +++ b/media/webrtc/signaling/src/media-conduit/VideoConduit.h Wed Dec 31 06:09:35 2014 +0100
1.3 @@ -0,0 +1,339 @@
1.4 +/* This Source Code Form is subject to the terms of the Mozilla Public
1.5 + * License, v. 2.0. If a copy of the MPL was not distributed with this file,
1.6 + * You can obtain one at http://mozilla.org/MPL/2.0/. */
1.7 +
1.8 +#ifndef VIDEO_SESSION_H_
1.9 +#define VIDEO_SESSION_H_
1.10 +
1.11 +#include "mozilla/Attributes.h"
1.12 +
1.13 +#include "MediaConduitInterface.h"
1.14 +#include "MediaEngineWrapper.h"
1.15 +
1.16 +// Video Engine Includes
1.17 +#include "webrtc/common_types.h"
1.18 +#ifdef FF
1.19 +#undef FF // Avoid name collision between scoped_ptr.h and nsCRTGlue.h.
1.20 +#endif
1.21 +#include "webrtc/modules/video_coding/codecs/interface/video_codec_interface.h"
1.22 +#include "webrtc/video_engine/include/vie_base.h"
1.23 +#include "webrtc/video_engine/include/vie_capture.h"
1.24 +#include "webrtc/video_engine/include/vie_codec.h"
1.25 +#include "webrtc/video_engine/include/vie_external_codec.h"
1.26 +#include "webrtc/video_engine/include/vie_render.h"
1.27 +#include "webrtc/video_engine/include/vie_network.h"
1.28 +#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
1.29 +
1.30 +/** This file hosts several structures identifying different aspects
1.31 + * of a RTP Session.
1.32 + */
1.33 +
1.34 + using webrtc::ViEBase;
1.35 + using webrtc::ViENetwork;
1.36 + using webrtc::ViECodec;
1.37 + using webrtc::ViECapture;
1.38 + using webrtc::ViERender;
1.39 + using webrtc::ViEExternalCapture;
1.40 + using webrtc::ViEExternalCodec;
1.41 +
1.42 +namespace mozilla {
1.43 +
1.44 +class WebrtcAudioConduit;
1.45 +
1.46 +// Interface of external video encoder for WebRTC.
1.47 +class WebrtcVideoEncoder:public VideoEncoder
1.48 + ,public webrtc::VideoEncoder
1.49 +{};
1.50 +
1.51 +// Interface of external video decoder for WebRTC.
1.52 +class WebrtcVideoDecoder:public VideoDecoder
1.53 + ,public webrtc::VideoDecoder
1.54 +{};
1.55 +
1.56 +/**
1.57 + * Concrete class for Video session. Hooks up
1.58 + * - media-source and target to external transport
1.59 + */
1.60 +class WebrtcVideoConduit:public VideoSessionConduit
1.61 + ,public webrtc::Transport
1.62 + ,public webrtc::ExternalRenderer
1.63 +{
1.64 +public:
1.65 + //VoiceEngine defined constant for Payload Name Size.
1.66 + static const unsigned int CODEC_PLNAME_SIZE;
1.67 +
1.68 + /**
1.69 + * Set up A/V sync between this (incoming) VideoConduit and an audio conduit.
1.70 + */
1.71 + void SyncTo(WebrtcAudioConduit *aConduit);
1.72 +
1.73 + /**
1.74 + * Function to attach Renderer end-point for the Media-Video conduit.
1.75 + * @param aRenderer : Reference to the concrete Video renderer implementation
1.76 + * Note: Multiple invocations of this API shall remove an existing renderer
1.77 + * and attaches the new to the Conduit.
1.78 + */
1.79 + virtual MediaConduitErrorCode AttachRenderer(mozilla::RefPtr<VideoRenderer> aVideoRenderer);
1.80 + virtual void DetachRenderer();
1.81 +
1.82 + /**
1.83 + * APIs used by the registered external transport to this Conduit to
1.84 + * feed in received RTP Frames to the VideoEngine for decoding
1.85 + */
1.86 + virtual MediaConduitErrorCode ReceivedRTPPacket(const void *data, int len);
1.87 +
1.88 + /**
1.89 + * APIs used by the registered external transport to this Conduit to
1.90 + * feed in received RTP Frames to the VideoEngine for decoding
1.91 + */
1.92 + virtual MediaConduitErrorCode ReceivedRTCPPacket(const void *data, int len);
1.93 +
1.94 + /**
1.95 + * Function to configure send codec for the video session
1.96 + * @param sendSessionConfig: CodecConfiguration
1.97 + * @result: On Success, the video engine is configured with passed in codec for send
1.98 + * On failure, video engine transmit functionality is disabled.
1.99 + * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
1.100 + * transmission sub-system on the engine.
1.101 + */
1.102 + virtual MediaConduitErrorCode ConfigureSendMediaCodec(const VideoCodecConfig* codecInfo);
1.103 +
1.104 + /**
1.105 + * Function to configure list of receive codecs for the video session
1.106 + * @param sendSessionConfig: CodecConfiguration
1.107 + * @result: On Success, the video engine is configured with passed in codec for send
1.108 + * Also the playout is enabled.
1.109 + * On failure, video engine transmit functionality is disabled.
1.110 + * NOTE: This API can be invoked multiple time. Invoking this API may involve restarting
1.111 + * transmission sub-system on the engine.
1.112 + */
1.113 + virtual MediaConduitErrorCode ConfigureRecvMediaCodecs(
1.114 + const std::vector<VideoCodecConfig* >& codecConfigList);
1.115 +
1.116 + /**
1.117 + * Register Transport for this Conduit. RTP and RTCP frames from the VideoEngine
1.118 + * shall be passed to the registered transport for transporting externally.
1.119 + */
1.120 + virtual MediaConduitErrorCode AttachTransport(mozilla::RefPtr<TransportInterface> aTransport);
1.121 +
1.122 + /**
1.123 + * Function to select and change the encoding resolution based on incoming frame size
1.124 + * and current available bandwidth.
1.125 + * @param width, height: dimensions of the frame
1.126 + */
1.127 + virtual bool SelectSendResolution(unsigned short width,
1.128 + unsigned short height);
1.129 +
1.130 + /**
1.131 + * Function to deliver a capture video frame for encoding and transport
1.132 + * @param video_frame: pointer to captured video-frame.
1.133 + * @param video_frame_length: size of the frame
1.134 + * @param width, height: dimensions of the frame
1.135 + * @param video_type: Type of the video frame - I420, RAW
1.136 + * @param captured_time: timestamp when the frame was captured.
1.137 + * if 0 timestamp is automatcally generated by the engine.
1.138 + *NOTE: ConfigureSendMediaCodec() SHOULD be called before this function can be invoked
1.139 + * This ensures the inserted video-frames can be transmitted by the conduit
1.140 + */
1.141 + virtual MediaConduitErrorCode SendVideoFrame(unsigned char* video_frame,
1.142 + unsigned int video_frame_length,
1.143 + unsigned short width,
1.144 + unsigned short height,
1.145 + VideoType video_type,
1.146 + uint64_t capture_time);
1.147 +
1.148 + /**
1.149 + * Set an external encoder object |encoder| to the payload type |pltype|
1.150 + * for sender side codec.
1.151 + */
1.152 + virtual MediaConduitErrorCode SetExternalSendCodec(int pltype,
1.153 + VideoEncoder* encoder);
1.154 +
1.155 + /**
1.156 + * Set an external decoder object |decoder| to the payload type |pltype|
1.157 + * for receiver side codec.
1.158 + */
1.159 + virtual MediaConduitErrorCode SetExternalRecvCodec(int pltype,
1.160 + VideoDecoder* decoder);
1.161 +
1.162 +
1.163 + /**
1.164 + * Webrtc transport implementation to send and receive RTP packet.
1.165 + * VideoConduit registers itself as ExternalTransport to the VideoEngine
1.166 + */
1.167 + virtual int SendPacket(int channel, const void *data, int len) ;
1.168 +
1.169 + /**
1.170 + * Webrtc transport implementation to send and receive RTCP packet.
1.171 + * VideoConduit registers itself as ExternalTransport to the VideoEngine
1.172 + */
1.173 + virtual int SendRTCPPacket(int channel, const void *data, int len) ;
1.174 +
1.175 +
1.176 + /**
1.177 + * Webrtc External Renderer Implementation APIs.
1.178 + * Raw I420 Frames are delivred to the VideoConduit by the VideoEngine
1.179 + */
1.180 + virtual int FrameSizeChange(unsigned int, unsigned int, unsigned int);
1.181 +
1.182 + virtual int DeliverFrame(unsigned char*,int, uint32_t , int64_t,
1.183 + void *handle);
1.184 +
1.185 + /**
1.186 + * Does DeliverFrame() support a null buffer and non-null handle
1.187 + * (video texture)?
1.188 + * B2G support it (when using HW video decoder with graphic buffer output).
1.189 + * XXX Investigate! Especially for Android
1.190 + */
1.191 + virtual bool IsTextureSupported() {
1.192 +#ifdef WEBRTC_GONK
1.193 + return true;
1.194 +#else
1.195 + return false;
1.196 +#endif
1.197 + }
1.198 +
1.199 + unsigned short SendingWidth() {
1.200 + return mSendingWidth;
1.201 + }
1.202 +
1.203 + unsigned short SendingHeight() {
1.204 + return mSendingHeight;
1.205 + }
1.206 +
1.207 + unsigned int SendingMaxFs() {
1.208 + if(mCurSendCodecConfig) {
1.209 + return mCurSendCodecConfig->mMaxFrameSize;
1.210 + }
1.211 + return 0;
1.212 + }
1.213 +
1.214 + unsigned int SendingMaxFr() {
1.215 + if(mCurSendCodecConfig) {
1.216 + return mCurSendCodecConfig->mMaxFrameRate;
1.217 + }
1.218 + return 0;
1.219 + }
1.220 +
1.221 + WebrtcVideoConduit():
1.222 + mOtherDirection(nullptr),
1.223 + mShutDown(false),
1.224 + mVideoEngine(nullptr),
1.225 + mTransport(nullptr),
1.226 + mRenderer(nullptr),
1.227 + mPtrExtCapture(nullptr),
1.228 + mEngineTransmitting(false),
1.229 + mEngineReceiving(false),
1.230 + mChannel(-1),
1.231 + mCapId(-1),
1.232 + mCurSendCodecConfig(nullptr),
1.233 + mSendingWidth(0),
1.234 + mSendingHeight(0),
1.235 + mReceivingWidth(640),
1.236 + mReceivingHeight(480),
1.237 + mVideoLatencyTestEnable(false),
1.238 + mVideoLatencyAvg(0)
1.239 + {
1.240 + }
1.241 +
1.242 + virtual ~WebrtcVideoConduit() ;
1.243 +
1.244 + MediaConduitErrorCode Init(WebrtcVideoConduit *other);
1.245 +
1.246 + int GetChannel() { return mChannel; }
1.247 + webrtc::VideoEngine* GetVideoEngine() { return mVideoEngine; }
1.248 + bool GetLocalSSRC(unsigned int* ssrc);
1.249 + bool GetRemoteSSRC(unsigned int* ssrc);
1.250 + bool GetAVStats(int32_t* jitterBufferDelayMs,
1.251 + int32_t* playoutBufferDelayMs,
1.252 + int32_t* avSyncOffsetMs);
1.253 + bool GetRTPStats(unsigned int* jitterMs, unsigned int* cumulativeLost);
1.254 + bool GetRTCPReceiverReport(DOMHighResTimeStamp* timestamp,
1.255 + uint32_t* jitterMs,
1.256 + uint32_t* packetsReceived,
1.257 + uint64_t* bytesReceived,
1.258 + uint32_t* cumulativeLost,
1.259 + int32_t* rttMs);
1.260 + bool GetRTCPSenderReport(DOMHighResTimeStamp* timestamp,
1.261 + unsigned int* packetsSent,
1.262 + uint64_t* bytesSent);
1.263 + uint64_t MozVideoLatencyAvg();
1.264 +
1.265 +private:
1.266 +
1.267 + WebrtcVideoConduit(const WebrtcVideoConduit& other) MOZ_DELETE;
1.268 + void operator=(const WebrtcVideoConduit& other) MOZ_DELETE;
1.269 +
1.270 + //Local database of currently applied receive codecs
1.271 + typedef std::vector<VideoCodecConfig* > RecvCodecList;
1.272 +
1.273 + //Function to convert between WebRTC and Conduit codec structures
1.274 + void CodecConfigToWebRTCCodec(const VideoCodecConfig* codecInfo,
1.275 + webrtc::VideoCodec& cinst);
1.276 +
1.277 + // Function to copy a codec structure to Conduit's database
1.278 + bool CopyCodecToDB(const VideoCodecConfig* codecInfo);
1.279 +
1.280 + // Functions to verify if the codec passed is already in
1.281 + // conduits database
1.282 + bool CheckCodecForMatch(const VideoCodecConfig* codecInfo) const;
1.283 + bool CheckCodecsForMatch(const VideoCodecConfig* curCodecConfig,
1.284 + const VideoCodecConfig* codecInfo) const;
1.285 +
1.286 + //Checks the codec to be applied
1.287 + MediaConduitErrorCode ValidateCodecConfig(const VideoCodecConfig* codecInfo, bool send) const;
1.288 +
1.289 + //Utility function to dump recv codec database
1.290 + void DumpCodecDB() const;
1.291 +
1.292 + // Video Latency Test averaging filter
1.293 + void VideoLatencyUpdate(uint64_t new_sample);
1.294 +
1.295 + // The two sides of a send/receive pair of conduits each keep a pointer to the other.
1.296 + // They also share a single VideoEngine and mChannel. Shutdown must be coordinated
1.297 + // carefully to avoid double-freeing or accessing after one frees.
1.298 + WebrtcVideoConduit* mOtherDirection;
1.299 + // The other side has shut down our mChannel and related items already
1.300 + bool mShutDown;
1.301 +
1.302 + // A few of these are shared by both directions. They're released by the last
1.303 + // conduit to die.
1.304 + webrtc::VideoEngine* mVideoEngine; // shared
1.305 + mozilla::RefPtr<TransportInterface> mTransport;
1.306 + mozilla::RefPtr<VideoRenderer> mRenderer;
1.307 +
1.308 + ScopedCustomReleasePtr<webrtc::ViEBase> mPtrViEBase;
1.309 + ScopedCustomReleasePtr<webrtc::ViECapture> mPtrViECapture;
1.310 + ScopedCustomReleasePtr<webrtc::ViECodec> mPtrViECodec;
1.311 + ScopedCustomReleasePtr<webrtc::ViENetwork> mPtrViENetwork;
1.312 + ScopedCustomReleasePtr<webrtc::ViERender> mPtrViERender;
1.313 + ScopedCustomReleasePtr<webrtc::ViERTP_RTCP> mPtrRTP;
1.314 + ScopedCustomReleasePtr<webrtc::ViEExternalCodec> mPtrExtCodec;
1.315 +
1.316 + webrtc::ViEExternalCapture* mPtrExtCapture; // shared
1.317 +
1.318 + // Engine state we are concerned with.
1.319 + bool mEngineTransmitting; //If true ==> Transmit Sub-system is up and running
1.320 + bool mEngineReceiving; // if true ==> Receive Sus-sysmtem up and running
1.321 +
1.322 + int mChannel; // Video Channel for this conduit
1.323 + int mCapId; // Capturer for this conduit
1.324 + RecvCodecList mRecvCodecList;
1.325 + VideoCodecConfig* mCurSendCodecConfig;
1.326 + unsigned short mSendingWidth;
1.327 + unsigned short mSendingHeight;
1.328 + unsigned short mReceivingWidth;
1.329 + unsigned short mReceivingHeight;
1.330 + bool mVideoLatencyTestEnable;
1.331 + uint64_t mVideoLatencyAvg;
1.332 +
1.333 + static const unsigned int sAlphaNum = 7;
1.334 + static const unsigned int sAlphaDen = 8;
1.335 + static const unsigned int sRoundingPadding = 1024;
1.336 +
1.337 + mozilla::RefPtr<WebrtcAudioConduit> mSyncedTo;
1.338 +};
1.339 +
1.340 +} // end namespace
1.341 +
1.342 +#endif
