/* * 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 "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/typedefs.h" namespace webrtc { class ModuleRtpRtcpImpl; class RTCPReceiver; class NACKStringBuilder { public: NACKStringBuilder(); ~NACKStringBuilder(); void PushNACK(uint16_t nack); std::string GetResult(); private: std::ostringstream _stream; int _count; uint16_t _prevNack; bool _consecutive; }; class RTCPSender { public: struct FeedbackState { explicit FeedbackState(ModuleRtpRtcpImpl* module); FeedbackState(); uint8_t send_payload_type; uint32_t frequency_hz; uint32_t packet_count_sent; uint32_t byte_count_sent; uint32_t send_bitrate; uint32_t last_rr_ntp_secs; uint32_t last_rr_ntp_frac; uint32_t remote_sr; bool has_last_xr_rr; RtcpReceiveTimeInfo last_xr_rr; // Used when generating TMMBR. ModuleRtpRtcpImpl* module; }; RTCPSender(const int32_t id, const bool audio, Clock* clock, ReceiveStatistics* receive_statistics); 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 FeedbackState& feedback_state, 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); void SetRemoteSSRC(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 SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const; bool TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP = false) const; uint32_t LastSendReport(uint32_t& lastRTCPTime); int32_t SendRTCP( const FeedbackState& feedback_state, uint32_t rtcpPacketTypeFlags, int32_t nackSize = 0, const uint16_t* nackList = 0, bool repeat = false, uint64_t pictureID = 0); int32_t AddExternalReportBlock( uint32_t SSRC, const RTCPReportBlock* receiveBlock); int32_t RemoveExternalReportBlock(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); void SendRtcpXrReceiverReferenceTime(bool enable); 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 WriteAllReportBlocksToBuffer(uint8_t* rtcpbuffer, int pos, uint8_t& numberOfReportBlocks, const uint32_t NTPsec, const uint32_t NTPfrac); int32_t WriteReportBlocksToBuffer( uint8_t* rtcpbuffer, int32_t position, const std::map& report_blocks); int32_t AddReportBlock( uint32_t SSRC, std::map* report_blocks, const RTCPReportBlock* receiveBlock); bool PrepareReport(const FeedbackState& feedback_state, StreamStatistician* statistician, RTCPReportBlock* report_block, uint32_t* ntp_secs, uint32_t* ntp_frac); int32_t BuildSR(const FeedbackState& feedback_state, uint8_t* rtcpbuffer, int& pos, uint32_t NTPsec, uint32_t NTPfrac); int32_t BuildRR(uint8_t* rtcpbuffer, int& pos, const uint32_t NTPsec, const uint32_t NTPfrac); int PrepareRTCP( const FeedbackState& feedback_state, uint32_t packetTypeFlags, int32_t nackSize, const uint16_t* nackList, bool repeat, uint64_t pictureID, uint8_t* rtcp_buffer, int buffer_size); bool ShouldSendReportBlocks(uint32_t rtcp_packet_type) const; int32_t BuildExtendedJitterReport( uint8_t* rtcpbuffer, int& pos, const uint32_t jitterTransmissionTimeOffset); int32_t BuildSDEC(uint8_t* rtcpbuffer, int& pos); int32_t BuildPLI(uint8_t* rtcpbuffer, int& pos); int32_t BuildREMB(uint8_t* rtcpbuffer, int& pos); int32_t BuildTMMBR(ModuleRtpRtcpImpl* module, uint8_t* rtcpbuffer, int& pos); int32_t BuildTMMBN(uint8_t* rtcpbuffer, int& pos); int32_t BuildAPP(uint8_t* rtcpbuffer, int& pos); int32_t BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos); int32_t BuildBYE(uint8_t* rtcpbuffer, int& pos); int32_t BuildFIR(uint8_t* rtcpbuffer, int& pos, bool repeat); int32_t BuildSLI(uint8_t* rtcpbuffer, int& pos, const uint8_t pictureID); int32_t BuildRPSI(uint8_t* rtcpbuffer, int& pos, const uint64_t pictureID, const uint8_t payloadType); int32_t BuildNACK(uint8_t* rtcpbuffer, int& pos, const int32_t nackSize, const uint16_t* nackList, std::string* nackString); int32_t BuildReceiverReferenceTime(uint8_t* buffer, int& pos, uint32_t ntp_sec, uint32_t ntp_frac); int32_t BuildDlrr(uint8_t* buffer, int& pos, const RtcpReceiveTimeInfo& info); private: int32_t _id; const bool _audio; Clock* _clock; RTCPMethod _method; 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]; ReceiveStatistics* receive_statistics_; std::map internal_report_blocks_; std::map external_report_blocks_; 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]; // Sent XR receiver reference time report. // . std::map last_xr_rr_; // 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; // True if sending of XR Receiver reference time report is enabled. bool xrSendReceiverReferenceTimeEnabled_; // 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_