20ed36dada
The goal with this new clock interface is to have something which is used all over WebRTC to make it easier to switch clock implementation depending on where the components are used. This is a first step in that direction. Next steps will be to, step by step, move all modules, video engine and voice engine over to the new interface, effectively deprecating the old clock interfaces. Long-term my vision is that we should be able to deprecate the clock of WebRTC and rely on the user providing the implementation. TEST=vie_auto_test, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1041004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 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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Clock* 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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Clock& _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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