/* * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_ #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_ #include #include #include #include "typedefs.h" #include "rtcp_utility.h" #include "rtp_utility.h" #include "rtp_rtcp_defines.h" #include "scoped_ptr.h" #include "tmmbr_help.h" #include "modules/remote_bitrate_estimator/include/bwe_defines.h" #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" namespace webrtc { class ModuleRtpRtcpImpl; class NACKStringBuilder { public: NACKStringBuilder(); void PushNACK(uint16_t nack); std::string GetResult(); private: std::ostringstream _stream; int _count; uint16_t _prevNack; bool _consecutive; }; class RTCPSender { public: RTCPSender(const int32_t id, const bool audio, Clock* clock, ModuleRtpRtcpImpl* owner); virtual ~RTCPSender(); void ChangeUniqueId(const int32_t id); int32_t Init(); int32_t RegisterSendTransport(Transport* outgoingTransport); RTCPMethod Status() const; int32_t SetRTCPStatus(const RTCPMethod method); bool Sending() const; int32_t SetSendingStatus(const bool enabled); // combine the functions int32_t SetNackStatus(const bool enable); void SetStartTimestamp(uint32_t start_timestamp); void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms); void SetSSRC( const uint32_t ssrc); int32_t SetRemoteSSRC( const uint32_t ssrc); int32_t SetCameraDelay(const int32_t delayMS); int32_t CNAME(char cName[RTCP_CNAME_SIZE]); int32_t SetCNAME(const char cName[RTCP_CNAME_SIZE]); int32_t AddMixedCNAME(const uint32_t SSRC, const char cName[RTCP_CNAME_SIZE]); int32_t RemoveMixedCNAME(const uint32_t SSRC); uint32_t SendTimeOfSendReport(const uint32_t sendReport); bool TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP = false) const; uint32_t LastSendReport(uint32_t& lastRTCPTime); int32_t SendRTCP(const uint32_t rtcpPacketTypeFlags, const int32_t nackSize = 0, const uint16_t* nackList = 0, const bool repeat = false, const uint64_t pictureID = 0); int32_t AddReportBlock(const uint32_t SSRC, const RTCPReportBlock* receiveBlock); int32_t RemoveReportBlock(const uint32_t SSRC); /* * REMB */ bool REMB() const; int32_t SetREMBStatus(const bool enable); int32_t SetREMBData(const uint32_t bitrate, const uint8_t numberOfSSRC, const uint32_t* SSRC); /* * TMMBR */ bool TMMBR() const; int32_t SetTMMBRStatus(const bool enable); int32_t SetTMMBN(const TMMBRSet* boundingSet, const uint32_t maxBitrateKbit); /* * Extended jitter report */ bool IJ() const; int32_t SetIJStatus(const bool enable); /* * */ int32_t SetApplicationSpecificData(const uint8_t subType, const uint32_t name, const uint8_t* data, const uint16_t length); int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric); int32_t SetCSRCs(const uint32_t arrOfCSRC[kRtpCsrcSize], const uint8_t arrLength); int32_t SetCSRCStatus(const bool include); void SetTargetBitrate(unsigned int target_bitrate); private: int32_t SendToNetwork(const uint8_t* dataBuffer, const uint16_t length); void UpdatePacketRate(); int32_t AddReportBlocks(uint8_t* rtcpbuffer, uint32_t& pos, uint8_t& numberOfReportBlocks, const RTCPReportBlock* received, const uint32_t NTPsec, const uint32_t NTPfrac); int32_t BuildSR(uint8_t* rtcpbuffer, uint32_t& pos, const uint32_t NTPsec, const uint32_t NTPfrac, const RTCPReportBlock* received = NULL); int32_t BuildRR(uint8_t* rtcpbuffer, uint32_t& pos, const uint32_t NTPsec, const uint32_t NTPfrac, const RTCPReportBlock* received = NULL); int32_t BuildExtendedJitterReport( uint8_t* rtcpbuffer, uint32_t& pos, const uint32_t jitterTransmissionTimeOffset); int32_t BuildSDEC(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildPLI(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildREMB(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildTMMBR(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildTMMBN(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildAPP(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildVoIPMetric(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildBYE(uint8_t* rtcpbuffer, uint32_t& pos); int32_t BuildFIR(uint8_t* rtcpbuffer, uint32_t& pos, bool repeat); int32_t BuildSLI(uint8_t* rtcpbuffer, uint32_t& pos, const uint8_t pictureID); int32_t BuildRPSI(uint8_t* rtcpbuffer, uint32_t& pos, const uint64_t pictureID, const uint8_t payloadType); int32_t BuildNACK(uint8_t* rtcpbuffer, uint32_t& pos, const int32_t nackSize, const uint16_t* nackList, std::string* nackString); private: int32_t _id; const bool _audio; Clock* _clock; RTCPMethod _method; ModuleRtpRtcpImpl& _rtpRtcp; CriticalSectionWrapper* _criticalSectionTransport; Transport* _cbTransport; CriticalSectionWrapper* _criticalSectionRTCPSender; bool _usingNack; bool _sending; bool _sendTMMBN; bool _REMB; bool _sendREMB; bool _TMMBR; bool _IJ; int64_t _nextTimeToSendRTCP; uint32_t start_timestamp_; uint32_t last_rtp_timestamp_; int64_t last_frame_capture_time_ms_; uint32_t _SSRC; uint32_t _remoteSSRC; // SSRC that we receive on our RTP channel char _CNAME[RTCP_CNAME_SIZE]; std::map _reportBlocks; std::map _csrcCNAMEs; int32_t _cameraDelayMS; // Sent uint32_t _lastSendReport[RTCP_NUMBER_OF_SR]; // allow packet loss and RTT above 1 sec uint32_t _lastRTCPTime[RTCP_NUMBER_OF_SR]; // send CSRCs uint8_t _CSRCs; uint32_t _CSRC[kRtpCsrcSize]; bool _includeCSRCs; // Full intra request uint8_t _sequenceNumberFIR; // REMB uint8_t _lengthRembSSRC; uint8_t _sizeRembSSRC; uint32_t* _rembSSRC; uint32_t _rembBitrate; TMMBRHelp _tmmbrHelp; uint32_t _tmmbr_Send; uint32_t _packetOH_Send; // APP bool _appSend; uint8_t _appSubType; uint32_t _appName; uint8_t* _appData; uint16_t _appLength; // XR VoIP metric bool _xrSendVoIPMetric; RTCPVoIPMetric _xrVoIPMetric; // Counters uint32_t _nackCount; uint32_t _pliCount; uint32_t _fullIntraRequestCount; }; } // namespace webrtc #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_