media/webrtc/trunk/src/modules/rtp_rtcp/source/rtcp_sender.h

   1 /*
   2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
   3  *
   4  *  Use of this source code is governed by a BSD-style license
   5  *  that can be found in the LICENSE file in the root of the source
   6  *  tree. An additional intellectual property rights grant can be found
   7  *  in the file PATENTS.  All contributing project authors may
   8  *  be found in the AUTHORS file in the root of the source tree.
   9  */
  10
  11 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
  12 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
  13
  14 #include <map>
  15
  16 #include "typedefs.h"
  17 #include "rtcp_utility.h"
  18 #include "rtp_utility.h"
  19 #include "rtp_rtcp_defines.h"
  20 #include "scoped_ptr.h"
  21 #include "tmmbr_help.h"
  22 #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
  23 #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
  24
  25 namespace webrtc {
  26
  27 class ModuleRtpRtcpImpl;
  28
  29 class RTCPSender
  30 {
  31 public:
  32     RTCPSender(const WebRtc_Word32 id, const bool audio,
  33                RtpRtcpClock* clock, ModuleRtpRtcpImpl* owner);
  34     virtual ~RTCPSender();
  35
  36     void ChangeUniqueId(const WebRtc_Word32 id);
  37
  38     WebRtc_Word32 Init();
  39
  40     WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
  41
  42     RTCPMethod Status() const;
  43     WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
  44
  45     bool Sending() const;
  46     WebRtc_Word32 SetSendingStatus(const bool enabled); // combine the functions
  47
  48     WebRtc_Word32 SetNackStatus(const bool enable);
  49
  50     void SetSSRC( const WebRtc_UWord32 ssrc);
  51
  52     WebRtc_Word32 SetRemoteSSRC( const WebRtc_UWord32 ssrc);
  53
  54     WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
  55
  56     WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
  57     WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
  58
  59     WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
  60                                 const char cName[RTCP_CNAME_SIZE]);
  61
  62     WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
  63
  64     WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
  65
  66     bool TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP = false) const;
  67
  68     WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
  69
  70     WebRtc_Word32 SendRTCP(const WebRtc_UWord32 rtcpPacketTypeFlags,
  71                            const WebRtc_Word32 nackSize = 0,
  72                            const WebRtc_UWord16* nackList = 0,
  73                            const bool repeat = false,
  74                            const WebRtc_UWord64 pictureID = 0);
  75
  76     WebRtc_Word32 AddReportBlock(const WebRtc_UWord32 SSRC,
  77                                  const RTCPReportBlock* receiveBlock);
  78
  79     WebRtc_Word32 RemoveReportBlock(const WebRtc_UWord32 SSRC);
  80
  81     /*
  82     *  REMB
  83     */
  84     bool REMB() const;
  85
  86     WebRtc_Word32 SetREMBStatus(const bool enable);
  87
  88     WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
  89                               const WebRtc_UWord8 numberOfSSRC,
  90                               const WebRtc_UWord32* SSRC);
  91
  92     /*
  93     *   TMMBR
  94     */
  95     bool TMMBR() const;
  96
  97     WebRtc_Word32 SetTMMBRStatus(const bool enable);
  98
  99     WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
 100                            const WebRtc_UWord32 maxBitrateKbit);
 101
 102     /*
 103     *   Extended jitter report
 104     */
 105     bool IJ() const;
 106
 107     WebRtc_Word32 SetIJStatus(const bool enable);
 108
 109     /*
 110     *
 111     */
 112
 113     WebRtc_Word32 SetApplicationSpecificData(const WebRtc_UWord8 subType,
 114                                              const WebRtc_UWord32 name,
 115                                              const WebRtc_UWord8* data,
 116                                              const WebRtc_UWord16 length);
 117
 118     WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
 119
 120     WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
 121                            const WebRtc_UWord8 arrLength);
 122
 123     WebRtc_Word32 SetCSRCStatus(const bool include);
 124
 125     void SetTargetBitrate(unsigned int target_bitrate);
 126
 127 private:
 128     WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
 129                                 const WebRtc_UWord16 length);
 130
 131     void UpdatePacketRate();
 132
 133     WebRtc_Word32 AddReportBlocks(WebRtc_UWord8* rtcpbuffer,
 134                                 WebRtc_UWord32& pos,
 135                                 WebRtc_UWord8& numberOfReportBlocks,
 136                                 const RTCPReportBlock* received,
 137                                 const WebRtc_UWord32 NTPsec,
 138                                 const WebRtc_UWord32 NTPfrac);
 139
 140     WebRtc_Word32 BuildSR(WebRtc_UWord8* rtcpbuffer,
 141                         WebRtc_UWord32& pos,
 142                         const WebRtc_UWord32 NTPsec,
 143                         const WebRtc_UWord32 NTPfrac,
 144                         const RTCPReportBlock* received = NULL);
 145
 146     WebRtc_Word32 BuildRR(WebRtc_UWord8* rtcpbuffer,
 147                         WebRtc_UWord32& pos,
 148                         const WebRtc_UWord32 NTPsec,
 149                         const WebRtc_UWord32 NTPfrac,
 150                         const RTCPReportBlock* received = NULL);
 151
 152     WebRtc_Word32 BuildExtendedJitterReport(
 153         WebRtc_UWord8* rtcpbuffer,
 154         WebRtc_UWord32& pos,
 155         const WebRtc_UWord32 jitterTransmissionTimeOffset);
 156
 157     WebRtc_Word32 BuildSDEC(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 158     WebRtc_Word32 BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 159     WebRtc_Word32 BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 160     WebRtc_Word32 BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 161     WebRtc_Word32 BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 162     WebRtc_Word32 BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 163     WebRtc_Word32 BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 164     WebRtc_Word32 BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
 165     WebRtc_Word32 BuildFIR(WebRtc_UWord8* rtcpbuffer,
 166                            WebRtc_UWord32& pos,
 167                            bool repeat);
 168     WebRtc_Word32 BuildSLI(WebRtc_UWord8* rtcpbuffer,
 169                          WebRtc_UWord32& pos,
 170                          const WebRtc_UWord8 pictureID);
 171     WebRtc_Word32 BuildRPSI(WebRtc_UWord8* rtcpbuffer,
 172                          WebRtc_UWord32& pos,
 173                          const WebRtc_UWord64 pictureID,
 174                          const WebRtc_UWord8 payloadType);
 175
 176     WebRtc_Word32 BuildNACK(WebRtc_UWord8* rtcpbuffer,
 177                           WebRtc_UWord32& pos,
 178                           const WebRtc_Word32 nackSize,
 179                           const WebRtc_UWord16* nackList);
 180
 181 private:
 182     WebRtc_Word32            _id;
 183     const bool               _audio;
 184     RtpRtcpClock&            _clock;
 185     RTCPMethod               _method;
 186
 187     ModuleRtpRtcpImpl&      _rtpRtcp;
 188
 189     CriticalSectionWrapper* _criticalSectionTransport;
 190     Transport*              _cbTransport;
 191
 192     CriticalSectionWrapper* _criticalSectionRTCPSender;
 193     bool                    _usingNack;
 194     bool                    _sending;
 195     bool                    _sendTMMBN;
 196     bool                    _REMB;
 197     bool                    _sendREMB;
 198     bool                    _TMMBR;
 199     bool                    _IJ;
 200
 201     WebRtc_Word64        _nextTimeToSendRTCP;
 202
 203     WebRtc_UWord32 _SSRC;
 204     WebRtc_UWord32 _remoteSSRC;  // SSRC that we receive on our RTP channel
 205     char _CNAME[RTCP_CNAME_SIZE];
 206
 207     std::map<WebRtc_UWord32, RTCPReportBlock*> _reportBlocks;
 208     std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs;
 209
 210     WebRtc_Word32         _cameraDelayMS;
 211
 212     // Sent
 213     WebRtc_UWord32        _lastSendReport[RTCP_NUMBER_OF_SR];  // allow packet loss and RTT above 1 sec
 214     WebRtc_UWord32        _lastRTCPTime[RTCP_NUMBER_OF_SR];
 215
 216     // send CSRCs
 217     WebRtc_UWord8         _CSRCs;
 218     WebRtc_UWord32        _CSRC[kRtpCsrcSize];
 219     bool                _includeCSRCs;
 220
 221     // Full intra request
 222     WebRtc_UWord8         _sequenceNumberFIR;
 223
 224     // REMB
 225     WebRtc_UWord8       _lengthRembSSRC;
 226     WebRtc_UWord8       _sizeRembSSRC;
 227     WebRtc_UWord32*     _rembSSRC;
 228     WebRtc_UWord32      _rembBitrate;
 229
 230     TMMBRHelp           _tmmbrHelp;
 231     WebRtc_UWord32      _tmmbr_Send;
 232     WebRtc_UWord32      _packetOH_Send;
 233
 234     // APP
 235     bool                 _appSend;
 236     WebRtc_UWord8        _appSubType;
 237     WebRtc_UWord32       _appName;
 238     WebRtc_UWord8*       _appData;
 239     WebRtc_UWord16       _appLength;
 240
 241     // XR VoIP metric
 242     bool                _xrSendVoIPMetric;
 243     RTCPVoIPMetric      _xrVoIPMetric;
 244 };
 245 } // namespace webrtc
 246
 247 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_