14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
256 lines
8.6 KiB
C++
256 lines
8.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
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#include <map>
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#include "typedefs.h"
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#include "rtcp_utility.h"
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#include "rtp_utility.h"
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#include "rtp_rtcp_defines.h"
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#include "scoped_ptr.h"
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#include "tmmbr_help.h"
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#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
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#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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namespace webrtc {
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class ModuleRtpRtcpImpl;
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class RTCPSender
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{
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public:
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RTCPSender(const WebRtc_Word32 id, const bool audio,
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RtpRtcpClock* clock, ModuleRtpRtcpImpl* owner);
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virtual ~RTCPSender();
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void ChangeUniqueId(const WebRtc_Word32 id);
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WebRtc_Word32 Init();
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WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
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RTCPMethod Status() const;
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WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
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bool Sending() const;
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WebRtc_Word32 SetSendingStatus(const bool enabled); // combine the functions
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WebRtc_Word32 SetNackStatus(const bool enable);
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void SetStartTimestamp(uint32_t start_timestamp);
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void SetLastRtpTime(uint32_t rtp_timestamp,
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int64_t capture_time_ms);
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void SetSSRC( const WebRtc_UWord32 ssrc);
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WebRtc_Word32 SetRemoteSSRC( const WebRtc_UWord32 ssrc);
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WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
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WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
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WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
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WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
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const char cName[RTCP_CNAME_SIZE]);
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WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
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WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
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bool TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP = false) const;
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WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
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WebRtc_Word32 SendRTCP(const WebRtc_UWord32 rtcpPacketTypeFlags,
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const WebRtc_Word32 nackSize = 0,
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const WebRtc_UWord16* nackList = 0,
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const bool repeat = false,
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const WebRtc_UWord64 pictureID = 0);
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WebRtc_Word32 AddReportBlock(const WebRtc_UWord32 SSRC,
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const RTCPReportBlock* receiveBlock);
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WebRtc_Word32 RemoveReportBlock(const WebRtc_UWord32 SSRC);
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/*
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* REMB
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*/
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bool REMB() const;
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WebRtc_Word32 SetREMBStatus(const bool enable);
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WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
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const WebRtc_UWord8 numberOfSSRC,
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const WebRtc_UWord32* SSRC);
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/*
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* TMMBR
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*/
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bool TMMBR() const;
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WebRtc_Word32 SetTMMBRStatus(const bool enable);
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WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
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const WebRtc_UWord32 maxBitrateKbit);
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/*
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* Extended jitter report
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*/
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bool IJ() const;
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WebRtc_Word32 SetIJStatus(const bool enable);
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/*
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*
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*/
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WebRtc_Word32 SetApplicationSpecificData(const WebRtc_UWord8 subType,
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const WebRtc_UWord32 name,
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const WebRtc_UWord8* data,
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const WebRtc_UWord16 length);
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WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
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WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
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const WebRtc_UWord8 arrLength);
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WebRtc_Word32 SetCSRCStatus(const bool include);
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void SetTargetBitrate(unsigned int target_bitrate);
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private:
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WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
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const WebRtc_UWord16 length);
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void UpdatePacketRate();
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WebRtc_Word32 AddReportBlocks(WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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WebRtc_UWord8& numberOfReportBlocks,
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const RTCPReportBlock* received,
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const WebRtc_UWord32 NTPsec,
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const WebRtc_UWord32 NTPfrac);
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WebRtc_Word32 BuildSR(WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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const WebRtc_UWord32 NTPsec,
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const WebRtc_UWord32 NTPfrac,
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const RTCPReportBlock* received = NULL);
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WebRtc_Word32 BuildRR(WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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const WebRtc_UWord32 NTPsec,
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const WebRtc_UWord32 NTPfrac,
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const RTCPReportBlock* received = NULL);
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WebRtc_Word32 BuildExtendedJitterReport(
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WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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const WebRtc_UWord32 jitterTransmissionTimeOffset);
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WebRtc_Word32 BuildSDEC(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
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WebRtc_Word32 BuildFIR(WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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bool repeat);
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WebRtc_Word32 BuildSLI(WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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const WebRtc_UWord8 pictureID);
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WebRtc_Word32 BuildRPSI(WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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const WebRtc_UWord64 pictureID,
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const WebRtc_UWord8 payloadType);
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WebRtc_Word32 BuildNACK(WebRtc_UWord8* rtcpbuffer,
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WebRtc_UWord32& pos,
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const WebRtc_Word32 nackSize,
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const WebRtc_UWord16* nackList);
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private:
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WebRtc_Word32 _id;
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const bool _audio;
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RtpRtcpClock& _clock;
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RTCPMethod _method;
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ModuleRtpRtcpImpl& _rtpRtcp;
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CriticalSectionWrapper* _criticalSectionTransport;
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Transport* _cbTransport;
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CriticalSectionWrapper* _criticalSectionRTCPSender;
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bool _usingNack;
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bool _sending;
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bool _sendTMMBN;
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bool _REMB;
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bool _sendREMB;
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bool _TMMBR;
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bool _IJ;
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WebRtc_Word64 _nextTimeToSendRTCP;
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uint32_t start_timestamp_;
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uint32_t last_rtp_timestamp_;
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int64_t last_frame_capture_time_ms_;
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WebRtc_UWord32 _SSRC;
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WebRtc_UWord32 _remoteSSRC; // SSRC that we receive on our RTP channel
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char _CNAME[RTCP_CNAME_SIZE];
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std::map<WebRtc_UWord32, RTCPReportBlock*> _reportBlocks;
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std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs;
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WebRtc_Word32 _cameraDelayMS;
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// Sent
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WebRtc_UWord32 _lastSendReport[RTCP_NUMBER_OF_SR]; // allow packet loss and RTT above 1 sec
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WebRtc_UWord32 _lastRTCPTime[RTCP_NUMBER_OF_SR];
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// send CSRCs
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WebRtc_UWord8 _CSRCs;
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WebRtc_UWord32 _CSRC[kRtpCsrcSize];
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bool _includeCSRCs;
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// Full intra request
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WebRtc_UWord8 _sequenceNumberFIR;
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// REMB
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WebRtc_UWord8 _lengthRembSSRC;
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WebRtc_UWord8 _sizeRembSSRC;
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WebRtc_UWord32* _rembSSRC;
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WebRtc_UWord32 _rembBitrate;
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TMMBRHelp _tmmbrHelp;
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WebRtc_UWord32 _tmmbr_Send;
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WebRtc_UWord32 _packetOH_Send;
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// APP
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bool _appSend;
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WebRtc_UWord8 _appSubType;
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WebRtc_UWord32 _appName;
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WebRtc_UWord8* _appData;
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WebRtc_UWord16 _appLength;
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// XR VoIP metric
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bool _xrSendVoIPMetric;
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RTCPVoIPMetric _xrVoIPMetric;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
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