andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 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 | #include "rtcp_sender.h" |
| 12 | |
| 13 | #include <cassert> // assert |
| 14 | #include <cstdlib> // rand |
| 15 | #include <string.h> // memcpy |
| 16 | |
| 17 | #include "common_types.h" |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 18 | #include "modules/rtp_rtcp/source/rtp_rtcp_impl.h" |
| 19 | #include "system_wrappers/interface/critical_section_wrapper.h" |
| 20 | #include "system_wrappers/interface/trace.h" |
| 21 | |
| 22 | namespace webrtc { |
| 23 | |
| 24 | using RTCPUtility::RTCPCnameInformation; |
| 25 | |
| 26 | RTCPSender::RTCPSender(const WebRtc_Word32 id, |
| 27 | const bool audio, |
| 28 | RtpRtcpClock* clock, |
| 29 | ModuleRtpRtcpImpl* owner) : |
| 30 | _id(id), |
| 31 | _audio(audio), |
| 32 | _clock(*clock), |
| 33 | _method(kRtcpOff), |
| 34 | _rtpRtcp(*owner), |
| 35 | _criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()), |
| 36 | _cbTransport(NULL), |
| 37 | |
| 38 | _criticalSectionRTCPSender(CriticalSectionWrapper::CreateCriticalSection()), |
| 39 | _usingNack(false), |
| 40 | _sending(false), |
| 41 | _sendTMMBN(false), |
| 42 | _REMB(false), |
| 43 | _sendREMB(false), |
| 44 | _TMMBR(false), |
| 45 | _IJ(false), |
| 46 | _nextTimeToSendRTCP(0), |
| 47 | start_timestamp_(0), |
| 48 | last_rtp_timestamp_(0), |
| 49 | last_frame_capture_time_ms_(-1), |
| 50 | _SSRC(0), |
| 51 | _remoteSSRC(0), |
| 52 | _CNAME(), |
| 53 | _reportBlocks(), |
| 54 | _csrcCNAMEs(), |
| 55 | |
| 56 | _cameraDelayMS(0), |
| 57 | |
| 58 | _lastSendReport(), |
| 59 | _lastRTCPTime(), |
| 60 | |
| 61 | _CSRCs(0), |
| 62 | _CSRC(), |
| 63 | _includeCSRCs(true), |
| 64 | |
| 65 | _sequenceNumberFIR(0), |
| 66 | |
| 67 | _lengthRembSSRC(0), |
| 68 | _sizeRembSSRC(0), |
| 69 | _rembSSRC(NULL), |
| 70 | _rembBitrate(0), |
| 71 | |
| 72 | _tmmbrHelp(), |
| 73 | _tmmbr_Send(0), |
| 74 | _packetOH_Send(0), |
| 75 | |
| 76 | _appSend(false), |
| 77 | _appSubType(0), |
| 78 | _appName(), |
| 79 | _appData(NULL), |
| 80 | _appLength(0), |
| 81 | _xrSendVoIPMetric(false), |
| 82 | _xrVoIPMetric() |
| 83 | { |
| 84 | memset(_CNAME, 0, sizeof(_CNAME)); |
| 85 | memset(_lastSendReport, 0, sizeof(_lastSendReport)); |
| 86 | memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); |
| 87 | |
| 88 | WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); |
| 89 | } |
| 90 | |
| 91 | RTCPSender::~RTCPSender() { |
| 92 | delete [] _rembSSRC; |
| 93 | delete [] _appData; |
| 94 | |
| 95 | while (!_reportBlocks.empty()) { |
| 96 | std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it = |
| 97 | _reportBlocks.begin(); |
| 98 | delete it->second; |
| 99 | _reportBlocks.erase(it); |
| 100 | } |
| 101 | while (!_csrcCNAMEs.empty()) { |
| 102 | std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it = |
| 103 | _csrcCNAMEs.begin(); |
| 104 | delete it->second; |
| 105 | _csrcCNAMEs.erase(it); |
| 106 | } |
| 107 | delete _criticalSectionTransport; |
| 108 | delete _criticalSectionRTCPSender; |
| 109 | |
| 110 | WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__); |
| 111 | } |
| 112 | |
| 113 | WebRtc_Word32 |
| 114 | RTCPSender::Init() |
| 115 | { |
| 116 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 117 | |
| 118 | _method = kRtcpOff; |
| 119 | _cbTransport = NULL; |
| 120 | _usingNack = false; |
| 121 | _sending = false; |
| 122 | _sendTMMBN = false; |
| 123 | _TMMBR = false; |
| 124 | _IJ = false; |
| 125 | _REMB = false; |
| 126 | _sendREMB = false; |
| 127 | last_rtp_timestamp_ = 0; |
| 128 | last_frame_capture_time_ms_ = -1; |
| 129 | start_timestamp_ = -1; |
| 130 | _SSRC = 0; |
| 131 | _remoteSSRC = 0; |
| 132 | _cameraDelayMS = 0; |
| 133 | _sequenceNumberFIR = 0; |
| 134 | _tmmbr_Send = 0; |
| 135 | _packetOH_Send = 0; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 136 | _nextTimeToSendRTCP = 0; |
| 137 | _CSRCs = 0; |
| 138 | _appSend = false; |
| 139 | _appSubType = 0; |
| 140 | |
| 141 | if(_appData) |
| 142 | { |
| 143 | delete [] _appData; |
| 144 | _appData = NULL; |
| 145 | } |
| 146 | _appLength = 0; |
| 147 | |
| 148 | _xrSendVoIPMetric = false; |
| 149 | |
| 150 | memset(&_xrVoIPMetric, 0, sizeof(_xrVoIPMetric)); |
| 151 | memset(_CNAME, 0, sizeof(_CNAME)); |
| 152 | memset(_lastSendReport, 0, sizeof(_lastSendReport)); |
| 153 | memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); |
| 154 | return 0; |
| 155 | } |
| 156 | |
| 157 | void |
| 158 | RTCPSender::ChangeUniqueId(const WebRtc_Word32 id) |
| 159 | { |
| 160 | _id = id; |
| 161 | } |
| 162 | |
| 163 | WebRtc_Word32 |
| 164 | RTCPSender::RegisterSendTransport(Transport* outgoingTransport) |
| 165 | { |
| 166 | CriticalSectionScoped lock(_criticalSectionTransport); |
| 167 | _cbTransport = outgoingTransport; |
| 168 | return 0; |
| 169 | } |
| 170 | |
| 171 | RTCPMethod |
| 172 | RTCPSender::Status() const |
| 173 | { |
| 174 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 175 | return _method; |
| 176 | } |
| 177 | |
| 178 | WebRtc_Word32 |
| 179 | RTCPSender::SetRTCPStatus(const RTCPMethod method) |
| 180 | { |
| 181 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 182 | if(method != kRtcpOff) |
| 183 | { |
| 184 | if(_audio) |
| 185 | { |
| 186 | _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_AUDIO_MS/2); |
| 187 | } else |
| 188 | { |
| 189 | _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_VIDEO_MS/2); |
| 190 | } |
| 191 | } |
| 192 | _method = method; |
| 193 | return 0; |
| 194 | } |
| 195 | |
| 196 | bool |
| 197 | RTCPSender::Sending() const |
| 198 | { |
| 199 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 200 | return _sending; |
| 201 | } |
| 202 | |
| 203 | WebRtc_Word32 |
| 204 | RTCPSender::SetSendingStatus(const bool sending) |
| 205 | { |
| 206 | bool sendRTCPBye = false; |
| 207 | { |
| 208 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 209 | |
| 210 | if(_method != kRtcpOff) |
| 211 | { |
| 212 | if(sending == false && _sending == true) |
| 213 | { |
| 214 | // Trigger RTCP bye |
| 215 | sendRTCPBye = true; |
| 216 | } |
| 217 | } |
| 218 | _sending = sending; |
| 219 | } |
| 220 | if(sendRTCPBye) |
| 221 | { |
| 222 | return SendRTCP(kRtcpBye); |
| 223 | } |
| 224 | return 0; |
| 225 | } |
| 226 | |
| 227 | bool |
| 228 | RTCPSender::REMB() const |
| 229 | { |
| 230 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 231 | return _REMB; |
| 232 | } |
| 233 | |
| 234 | WebRtc_Word32 |
| 235 | RTCPSender::SetREMBStatus(const bool enable) |
| 236 | { |
| 237 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 238 | _REMB = enable; |
| 239 | return 0; |
| 240 | } |
| 241 | |
| 242 | WebRtc_Word32 |
| 243 | RTCPSender::SetREMBData(const WebRtc_UWord32 bitrate, |
| 244 | const WebRtc_UWord8 numberOfSSRC, |
| 245 | const WebRtc_UWord32* SSRC) |
| 246 | { |
| 247 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 248 | _rembBitrate = bitrate; |
| 249 | |
| 250 | if(_sizeRembSSRC < numberOfSSRC) |
| 251 | { |
| 252 | delete [] _rembSSRC; |
| 253 | _rembSSRC = new WebRtc_UWord32[numberOfSSRC]; |
| 254 | _sizeRembSSRC = numberOfSSRC; |
| 255 | } |
| 256 | |
| 257 | _lengthRembSSRC = numberOfSSRC; |
| 258 | for (int i = 0; i < numberOfSSRC; i++) |
| 259 | { |
| 260 | _rembSSRC[i] = SSRC[i]; |
| 261 | } |
| 262 | _sendREMB = true; |
| 263 | return 0; |
| 264 | } |
| 265 | |
| 266 | bool |
| 267 | RTCPSender::TMMBR() const |
| 268 | { |
| 269 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 270 | return _TMMBR; |
| 271 | } |
| 272 | |
| 273 | WebRtc_Word32 |
| 274 | RTCPSender::SetTMMBRStatus(const bool enable) |
| 275 | { |
| 276 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 277 | _TMMBR = enable; |
| 278 | return 0; |
| 279 | } |
| 280 | |
| 281 | bool |
| 282 | RTCPSender::IJ() const |
| 283 | { |
| 284 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 285 | return _IJ; |
| 286 | } |
| 287 | |
| 288 | WebRtc_Word32 |
| 289 | RTCPSender::SetIJStatus(const bool enable) |
| 290 | { |
| 291 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 292 | _IJ = enable; |
| 293 | return 0; |
| 294 | } |
| 295 | |
| 296 | void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) { |
| 297 | start_timestamp_ = start_timestamp; |
| 298 | } |
| 299 | |
| 300 | void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp, |
| 301 | int64_t capture_time_ms) { |
| 302 | last_rtp_timestamp_ = rtp_timestamp; |
| 303 | if (capture_time_ms < 0) { |
| 304 | // We don't currently get a capture time from VoiceEngine. |
| 305 | last_frame_capture_time_ms_ = _clock.GetTimeInMS(); |
| 306 | } else { |
| 307 | last_frame_capture_time_ms_ = capture_time_ms; |
| 308 | } |
| 309 | } |
| 310 | |
| 311 | void |
| 312 | RTCPSender::SetSSRC( const WebRtc_UWord32 ssrc) |
| 313 | { |
| 314 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 315 | |
| 316 | if(_SSRC != 0) |
| 317 | { |
| 318 | // not first SetSSRC, probably due to a collision |
| 319 | // schedule a new RTCP report |
| 320 | // make sure that we send a RTP packet |
| 321 | _nextTimeToSendRTCP = _clock.GetTimeInMS() + 100; |
| 322 | } |
| 323 | _SSRC = ssrc; |
| 324 | } |
| 325 | |
| 326 | WebRtc_Word32 |
| 327 | RTCPSender::SetRemoteSSRC( const WebRtc_UWord32 ssrc) |
| 328 | { |
| 329 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 330 | _remoteSSRC = ssrc; |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 331 | return 0; |
| 332 | } |
| 333 | |
| 334 | WebRtc_Word32 |
| 335 | RTCPSender::SetCameraDelay(const WebRtc_Word32 delayMS) |
| 336 | { |
| 337 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 338 | if(delayMS > 1000 || delayMS < -1000) |
| 339 | { |
| 340 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, delay can't be larger than 1 sec", __FUNCTION__); |
| 341 | return -1; |
| 342 | } |
| 343 | _cameraDelayMS = delayMS; |
| 344 | return 0; |
| 345 | } |
| 346 | |
| 347 | WebRtc_Word32 RTCPSender::CNAME(char cName[RTCP_CNAME_SIZE]) { |
| 348 | assert(cName); |
| 349 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 350 | cName[RTCP_CNAME_SIZE - 1] = 0; |
| 351 | strncpy(cName, _CNAME, RTCP_CNAME_SIZE - 1); |
| 352 | return 0; |
| 353 | } |
| 354 | |
| 355 | WebRtc_Word32 RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) { |
| 356 | if (!cName) |
| 357 | return -1; |
| 358 | |
| 359 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 360 | _CNAME[RTCP_CNAME_SIZE - 1] = 0; |
| 361 | strncpy(_CNAME, cName, RTCP_CNAME_SIZE - 1); |
| 362 | return 0; |
| 363 | } |
| 364 | |
| 365 | WebRtc_Word32 RTCPSender::AddMixedCNAME(const WebRtc_UWord32 SSRC, |
| 366 | const char cName[RTCP_CNAME_SIZE]) { |
| 367 | assert(cName); |
| 368 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 369 | if (_csrcCNAMEs.size() >= kRtpCsrcSize) { |
| 370 | return -1; |
| 371 | } |
| 372 | RTCPCnameInformation* ptr = new RTCPCnameInformation(); |
| 373 | ptr->name[RTCP_CNAME_SIZE - 1] = 0; |
| 374 | strncpy(ptr->name, cName, RTCP_CNAME_SIZE - 1); |
| 375 | _csrcCNAMEs[SSRC] = ptr; |
| 376 | return 0; |
| 377 | } |
| 378 | |
| 379 | WebRtc_Word32 RTCPSender::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) { |
| 380 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 381 | std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it = |
| 382 | _csrcCNAMEs.find(SSRC); |
| 383 | |
| 384 | if (it == _csrcCNAMEs.end()) { |
| 385 | return -1; |
| 386 | } |
| 387 | delete it->second; |
| 388 | _csrcCNAMEs.erase(it); |
| 389 | return 0; |
| 390 | } |
| 391 | |
| 392 | bool |
| 393 | RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const |
| 394 | { |
| 395 | /* |
| 396 | For audio we use a fix 5 sec interval |
| 397 | |
| 398 | For video we use 1 sec interval fo a BW smaller than 360 kbit/s, |
| 399 | technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare |
| 400 | |
| 401 | |
| 402 | From RFC 3550 |
| 403 | |
| 404 | MAX RTCP BW is 5% if the session BW |
| 405 | A send report is approximately 65 bytes inc CNAME |
| 406 | A report report is approximately 28 bytes |
| 407 | |
| 408 | The RECOMMENDED value for the reduced minimum in seconds is 360 |
| 409 | divided by the session bandwidth in kilobits/second. This minimum |
| 410 | is smaller than 5 seconds for bandwidths greater than 72 kb/s. |
| 411 | |
| 412 | If the participant has not yet sent an RTCP packet (the variable |
| 413 | initial is true), the constant Tmin is set to 2.5 seconds, else it |
| 414 | is set to 5 seconds. |
| 415 | |
| 416 | The interval between RTCP packets is varied randomly over the |
| 417 | range [0.5,1.5] times the calculated interval to avoid unintended |
| 418 | synchronization of all participants |
| 419 | |
| 420 | if we send |
| 421 | If the participant is a sender (we_sent true), the constant C is |
| 422 | set to the average RTCP packet size (avg_rtcp_size) divided by 25% |
| 423 | of the RTCP bandwidth (rtcp_bw), and the constant n is set to the |
| 424 | number of senders. |
| 425 | |
| 426 | if we receive only |
| 427 | If we_sent is not true, the constant C is set |
| 428 | to the average RTCP packet size divided by 75% of the RTCP |
| 429 | bandwidth. The constant n is set to the number of receivers |
| 430 | (members - senders). If the number of senders is greater than |
| 431 | 25%, senders and receivers are treated together. |
| 432 | |
| 433 | reconsideration NOT required for peer-to-peer |
| 434 | "timer reconsideration" is |
| 435 | employed. This algorithm implements a simple back-off mechanism |
| 436 | which causes users to hold back RTCP packet transmission if the |
| 437 | group sizes are increasing. |
| 438 | |
| 439 | n = number of members |
| 440 | C = avg_size/(rtcpBW/4) |
| 441 | |
| 442 | 3. The deterministic calculated interval Td is set to max(Tmin, n*C). |
| 443 | |
| 444 | 4. The calculated interval T is set to a number uniformly distributed |
| 445 | between 0.5 and 1.5 times the deterministic calculated interval. |
| 446 | |
| 447 | 5. The resulting value of T is divided by e-3/2=1.21828 to compensate |
| 448 | for the fact that the timer reconsideration algorithm converges to |
| 449 | a value of the RTCP bandwidth below the intended average |
| 450 | */ |
| 451 | |
| 452 | WebRtc_Word64 now = _clock.GetTimeInMS(); |
| 453 | |
| 454 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 455 | |
| 456 | if(_method == kRtcpOff) |
| 457 | { |
| 458 | return false; |
| 459 | } |
| 460 | |
| 461 | if(!_audio && sendKeyframeBeforeRTP) |
| 462 | { |
| 463 | // for video key-frames we want to send the RTCP before the large key-frame |
| 464 | // if we have a 100 ms margin |
| 465 | now += RTCP_SEND_BEFORE_KEY_FRAME_MS; |
| 466 | } |
| 467 | |
| 468 | if(now > _nextTimeToSendRTCP) |
| 469 | { |
| 470 | return true; |
| 471 | |
| 472 | } else if(now < 0x0000ffff && _nextTimeToSendRTCP > 0xffff0000) // 65 sec margin |
| 473 | { |
| 474 | // wrap |
| 475 | return true; |
| 476 | } |
| 477 | return false; |
| 478 | } |
| 479 | |
| 480 | WebRtc_UWord32 |
| 481 | RTCPSender::LastSendReport( WebRtc_UWord32& lastRTCPTime) |
| 482 | { |
| 483 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 484 | |
| 485 | lastRTCPTime = _lastRTCPTime[0]; |
| 486 | return _lastSendReport[0]; |
| 487 | } |
| 488 | |
| 489 | WebRtc_UWord32 |
| 490 | RTCPSender::SendTimeOfSendReport(const WebRtc_UWord32 sendReport) |
| 491 | { |
| 492 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 493 | |
| 494 | // This is only saved when we are the sender |
| 495 | if((_lastSendReport[0] == 0) || (sendReport == 0)) |
| 496 | { |
| 497 | return 0; // will be ignored |
| 498 | } else |
| 499 | { |
| 500 | for(int i = 0; i < RTCP_NUMBER_OF_SR; ++i) |
| 501 | { |
| 502 | if( _lastSendReport[i] == sendReport) |
| 503 | { |
| 504 | return _lastRTCPTime[i]; |
| 505 | } |
| 506 | } |
| 507 | } |
| 508 | return 0; |
| 509 | } |
| 510 | |
| 511 | WebRtc_Word32 RTCPSender::AddReportBlock(const WebRtc_UWord32 SSRC, |
| 512 | const RTCPReportBlock* reportBlock) { |
| 513 | if (reportBlock == NULL) { |
| 514 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| 515 | "%s invalid argument", __FUNCTION__); |
| 516 | return -1; |
| 517 | } |
| 518 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 519 | |
| 520 | if (_reportBlocks.size() >= RTCP_MAX_REPORT_BLOCKS) { |
| 521 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| 522 | "%s invalid argument", __FUNCTION__); |
| 523 | return -1; |
| 524 | } |
stefan@webrtc.org | cf4441c | 2012-12-03 14:01:46 +0000 | [diff] [blame^] | 525 | std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it = |
| 526 | _reportBlocks.find(SSRC); |
| 527 | if (it != _reportBlocks.end()) { |
| 528 | delete it->second; |
| 529 | _reportBlocks.erase(it); |
| 530 | } |
andrew@webrtc.org | b015cbe | 2012-10-22 18:19:23 +0000 | [diff] [blame] | 531 | RTCPReportBlock* copyReportBlock = new RTCPReportBlock(); |
| 532 | memcpy(copyReportBlock, reportBlock, sizeof(RTCPReportBlock)); |
| 533 | _reportBlocks[SSRC] = copyReportBlock; |
| 534 | return 0; |
| 535 | } |
| 536 | |
| 537 | WebRtc_Word32 RTCPSender::RemoveReportBlock(const WebRtc_UWord32 SSRC) { |
| 538 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 539 | |
| 540 | std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it = |
| 541 | _reportBlocks.find(SSRC); |
| 542 | |
| 543 | if (it == _reportBlocks.end()) { |
| 544 | return -1; |
| 545 | } |
| 546 | delete it->second; |
| 547 | _reportBlocks.erase(it); |
| 548 | return 0; |
| 549 | } |
| 550 | |
| 551 | WebRtc_Word32 |
| 552 | RTCPSender::BuildSR(WebRtc_UWord8* rtcpbuffer, |
| 553 | WebRtc_UWord32& pos, |
| 554 | const WebRtc_UWord32 NTPsec, |
| 555 | const WebRtc_UWord32 NTPfrac, |
| 556 | const RTCPReportBlock* received) |
| 557 | { |
| 558 | // sanity |
| 559 | if(pos + 52 >= IP_PACKET_SIZE) |
| 560 | { |
| 561 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 562 | return -2; |
| 563 | } |
| 564 | WebRtc_UWord32 RTPtime; |
| 565 | |
| 566 | WebRtc_UWord32 posNumberOfReportBlocks = pos; |
| 567 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| 568 | |
| 569 | // Sender report |
| 570 | rtcpbuffer[pos++]=(WebRtc_UWord8)200; |
| 571 | |
| 572 | for(int i = (RTCP_NUMBER_OF_SR-2); i >= 0; i--) |
| 573 | { |
| 574 | // shift old |
| 575 | _lastSendReport[i+1] = _lastSendReport[i]; |
| 576 | _lastRTCPTime[i+1] =_lastRTCPTime[i]; |
| 577 | } |
| 578 | |
| 579 | _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); |
| 580 | _lastSendReport[0] = (NTPsec << 16) + (NTPfrac >> 16); |
| 581 | |
| 582 | WebRtc_UWord32 freqHz = 90000; // For video |
| 583 | if(_audio) { |
| 584 | freqHz = _rtpRtcp.CurrentSendFrequencyHz(); |
| 585 | } |
| 586 | // The timestamp of this RTCP packet should be estimated as the timestamp of |
| 587 | // the frame being captured at this moment. We are calculating that |
| 588 | // timestamp as the last frame's timestamp + the time since the last frame |
| 589 | // was captured. |
| 590 | RTPtime = start_timestamp_ + last_rtp_timestamp_ + (_clock.GetTimeInMS() - |
| 591 | last_frame_capture_time_ms_) * (freqHz / 1000); |
| 592 | |
| 593 | // Add sender data |
| 594 | // Save for our length field |
| 595 | pos++; |
| 596 | pos++; |
| 597 | |
| 598 | // Add our own SSRC |
| 599 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 600 | pos += 4; |
| 601 | // NTP |
| 602 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, NTPsec); |
| 603 | pos += 4; |
| 604 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, NTPfrac); |
| 605 | pos += 4; |
| 606 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, RTPtime); |
| 607 | pos += 4; |
| 608 | |
| 609 | //sender's packet count |
| 610 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.PacketCountSent()); |
| 611 | pos += 4; |
| 612 | |
| 613 | //sender's octet count |
| 614 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.ByteCountSent()); |
| 615 | pos += 4; |
| 616 | |
| 617 | WebRtc_UWord8 numberOfReportBlocks = 0; |
| 618 | WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac); |
| 619 | if(retVal < 0) |
| 620 | { |
| 621 | // |
| 622 | return retVal ; |
| 623 | } |
| 624 | rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; |
| 625 | |
| 626 | WebRtc_UWord16 len = WebRtc_UWord16((pos/4) -1); |
| 627 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); |
| 628 | return 0; |
| 629 | } |
| 630 | |
| 631 | |
| 632 | WebRtc_Word32 RTCPSender::BuildSDEC(WebRtc_UWord8* rtcpbuffer, |
| 633 | WebRtc_UWord32& pos) { |
| 634 | size_t lengthCname = strlen(_CNAME); |
| 635 | assert(lengthCname < RTCP_CNAME_SIZE); |
| 636 | |
| 637 | // sanity |
| 638 | if(pos + 12 + lengthCname >= IP_PACKET_SIZE) { |
| 639 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| 640 | "%s invalid argument", __FUNCTION__); |
| 641 | return -2; |
| 642 | } |
| 643 | // SDEC Source Description |
| 644 | |
| 645 | // We always need to add SDES CNAME |
| 646 | rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(0x80 + 1 + _csrcCNAMEs.size()); |
| 647 | rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(202); |
| 648 | |
| 649 | // handle SDES length later on |
| 650 | WebRtc_UWord32 SDESLengthPos = pos; |
| 651 | pos++; |
| 652 | pos++; |
| 653 | |
| 654 | // Add our own SSRC |
| 655 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 656 | pos += 4; |
| 657 | |
| 658 | // CNAME = 1 |
| 659 | rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1); |
| 660 | |
| 661 | // |
| 662 | rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(lengthCname); |
| 663 | |
| 664 | WebRtc_UWord16 SDESLength = 10; |
| 665 | |
| 666 | memcpy(&rtcpbuffer[pos], _CNAME, lengthCname); |
| 667 | pos += lengthCname; |
| 668 | SDESLength += (WebRtc_UWord16)lengthCname; |
| 669 | |
| 670 | WebRtc_UWord16 padding = 0; |
| 671 | // We must have a zero field even if we have an even multiple of 4 bytes |
| 672 | if ((pos % 4) == 0) { |
| 673 | padding++; |
| 674 | rtcpbuffer[pos++]=0; |
| 675 | } |
| 676 | while ((pos % 4) != 0) { |
| 677 | padding++; |
| 678 | rtcpbuffer[pos++]=0; |
| 679 | } |
| 680 | SDESLength += padding; |
| 681 | |
| 682 | std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*>::iterator it = |
| 683 | _csrcCNAMEs.begin(); |
| 684 | |
| 685 | for(; it != _csrcCNAMEs.end(); it++) { |
| 686 | RTCPCnameInformation* cname = it->second; |
| 687 | WebRtc_UWord32 SSRC = it->first; |
| 688 | |
| 689 | // Add SSRC |
| 690 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, SSRC); |
| 691 | pos += 4; |
| 692 | |
| 693 | // CNAME = 1 |
| 694 | rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1); |
| 695 | |
| 696 | size_t length = strlen(cname->name); |
| 697 | assert(length < RTCP_CNAME_SIZE); |
| 698 | |
| 699 | rtcpbuffer[pos++]= static_cast<WebRtc_UWord8>(length); |
| 700 | SDESLength += 6; |
| 701 | |
| 702 | memcpy(&rtcpbuffer[pos],cname->name, length); |
| 703 | |
| 704 | pos += length; |
| 705 | SDESLength += length; |
| 706 | WebRtc_UWord16 padding = 0; |
| 707 | |
| 708 | // We must have a zero field even if we have an even multiple of 4 bytes |
| 709 | if((pos % 4) == 0){ |
| 710 | padding++; |
| 711 | rtcpbuffer[pos++]=0; |
| 712 | } |
| 713 | while((pos % 4) != 0){ |
| 714 | padding++; |
| 715 | rtcpbuffer[pos++] = 0; |
| 716 | } |
| 717 | SDESLength += padding; |
| 718 | } |
| 719 | // in 32-bit words minus one and we don't count the header |
| 720 | WebRtc_UWord16 buffer_length = (SDESLength / 4) - 1; |
| 721 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, |
| 722 | buffer_length); |
| 723 | return 0; |
| 724 | } |
| 725 | |
| 726 | WebRtc_Word32 |
| 727 | RTCPSender::BuildRR(WebRtc_UWord8* rtcpbuffer, |
| 728 | WebRtc_UWord32& pos, |
| 729 | const WebRtc_UWord32 NTPsec, |
| 730 | const WebRtc_UWord32 NTPfrac, |
| 731 | const RTCPReportBlock* received) |
| 732 | { |
| 733 | // sanity one block |
| 734 | if(pos + 32 >= IP_PACKET_SIZE) |
| 735 | { |
| 736 | return -2; |
| 737 | } |
| 738 | WebRtc_UWord32 posNumberOfReportBlocks = pos; |
| 739 | |
| 740 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| 741 | rtcpbuffer[pos++]=(WebRtc_UWord8)201; |
| 742 | |
| 743 | // Save for our length field |
| 744 | pos++; |
| 745 | pos++; |
| 746 | |
| 747 | // Add our own SSRC |
| 748 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 749 | pos += 4; |
| 750 | |
| 751 | WebRtc_UWord8 numberOfReportBlocks = 0; |
| 752 | WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac); |
| 753 | if(retVal < 0) |
| 754 | { |
| 755 | return retVal; |
| 756 | } |
| 757 | rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; |
| 758 | |
| 759 | WebRtc_UWord16 len = WebRtc_UWord16((pos)/4 -1); |
| 760 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); |
| 761 | return 0; |
| 762 | } |
| 763 | |
| 764 | // From RFC 5450: Transmission Time Offsets in RTP Streams. |
| 765 | // 0 1 2 3 |
| 766 | // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
| 767 | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 768 | // hdr |V=2|P| RC | PT=IJ=195 | length | |
| 769 | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 770 | // | inter-arrival jitter | |
| 771 | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 772 | // . . |
| 773 | // . . |
| 774 | // . . |
| 775 | // | inter-arrival jitter | |
| 776 | // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 777 | // |
| 778 | // If present, this RTCP packet must be placed after a receiver report |
| 779 | // (inside a compound RTCP packet), and MUST have the same value for RC |
| 780 | // (reception report count) as the receiver report. |
| 781 | |
| 782 | WebRtc_Word32 |
| 783 | RTCPSender::BuildExtendedJitterReport( |
| 784 | WebRtc_UWord8* rtcpbuffer, |
| 785 | WebRtc_UWord32& pos, |
| 786 | const WebRtc_UWord32 jitterTransmissionTimeOffset) |
| 787 | { |
| 788 | if (_reportBlocks.size() > 0) |
| 789 | { |
| 790 | WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "Not implemented."); |
| 791 | return 0; |
| 792 | } |
| 793 | |
| 794 | // sanity |
| 795 | if(pos + 8 >= IP_PACKET_SIZE) |
| 796 | { |
| 797 | return -2; |
| 798 | } |
| 799 | // add picture loss indicator |
| 800 | WebRtc_UWord8 RC = 1; |
| 801 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + RC; |
| 802 | rtcpbuffer[pos++]=(WebRtc_UWord8)195; |
| 803 | |
| 804 | // Used fixed length of 2 |
| 805 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 806 | rtcpbuffer[pos++]=(WebRtc_UWord8)(1); |
| 807 | |
| 808 | // Add inter-arrival jitter |
| 809 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, |
| 810 | jitterTransmissionTimeOffset); |
| 811 | pos += 4; |
| 812 | return 0; |
| 813 | } |
| 814 | |
| 815 | WebRtc_Word32 |
| 816 | RTCPSender::BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 817 | { |
| 818 | // sanity |
| 819 | if(pos + 12 >= IP_PACKET_SIZE) |
| 820 | { |
| 821 | return -2; |
| 822 | } |
| 823 | // add picture loss indicator |
| 824 | WebRtc_UWord8 FMT = 1; |
| 825 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 826 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 827 | |
| 828 | //Used fixed length of 2 |
| 829 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 830 | rtcpbuffer[pos++]=(WebRtc_UWord8)(2); |
| 831 | |
| 832 | // Add our own SSRC |
| 833 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 834 | pos += 4; |
| 835 | |
| 836 | // Add the remote SSRC |
| 837 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 838 | pos += 4; |
| 839 | return 0; |
| 840 | } |
| 841 | |
| 842 | WebRtc_Word32 RTCPSender::BuildFIR(WebRtc_UWord8* rtcpbuffer, |
| 843 | WebRtc_UWord32& pos, |
| 844 | bool repeat) { |
| 845 | // sanity |
| 846 | if(pos + 20 >= IP_PACKET_SIZE) { |
| 847 | return -2; |
| 848 | } |
| 849 | if (!repeat) { |
| 850 | _sequenceNumberFIR++; // do not increase if repetition |
| 851 | } |
| 852 | |
| 853 | // add full intra request indicator |
| 854 | WebRtc_UWord8 FMT = 4; |
| 855 | rtcpbuffer[pos++] = (WebRtc_UWord8)0x80 + FMT; |
| 856 | rtcpbuffer[pos++] = (WebRtc_UWord8)206; |
| 857 | |
| 858 | //Length of 4 |
| 859 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 860 | rtcpbuffer[pos++] = (WebRtc_UWord8)(4); |
| 861 | |
| 862 | // Add our own SSRC |
| 863 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC); |
| 864 | pos += 4; |
| 865 | |
| 866 | // RFC 5104 4.3.1.2. Semantics |
| 867 | // SSRC of media source |
| 868 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 869 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 870 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 871 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 872 | |
| 873 | // Additional Feedback Control Information (FCI) |
| 874 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC); |
| 875 | pos += 4; |
| 876 | |
| 877 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_sequenceNumberFIR); |
| 878 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 879 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 880 | rtcpbuffer[pos++] = (WebRtc_UWord8)0; |
| 881 | return 0; |
| 882 | } |
| 883 | |
| 884 | /* |
| 885 | 0 1 2 3 |
| 886 | 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
| 887 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 888 | | First | Number | PictureID | |
| 889 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 890 | */ |
| 891 | WebRtc_Word32 |
| 892 | RTCPSender::BuildSLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, const WebRtc_UWord8 pictureID) |
| 893 | { |
| 894 | // sanity |
| 895 | if(pos + 16 >= IP_PACKET_SIZE) |
| 896 | { |
| 897 | return -2; |
| 898 | } |
| 899 | // add slice loss indicator |
| 900 | WebRtc_UWord8 FMT = 2; |
| 901 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 902 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 903 | |
| 904 | //Used fixed length of 3 |
| 905 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 906 | rtcpbuffer[pos++]=(WebRtc_UWord8)(3); |
| 907 | |
| 908 | // Add our own SSRC |
| 909 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 910 | pos += 4; |
| 911 | |
| 912 | // Add the remote SSRC |
| 913 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 914 | pos += 4; |
| 915 | |
| 916 | // Add first, number & picture ID 6 bits |
| 917 | // first = 0, 13 - bits |
| 918 | // number = 0x1fff, 13 - bits only ones for now |
| 919 | WebRtc_UWord32 sliField = (0x1fff << 6)+ (0x3f & pictureID); |
| 920 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, sliField); |
| 921 | pos += 4; |
| 922 | return 0; |
| 923 | } |
| 924 | |
| 925 | /* |
| 926 | 0 1 2 3 |
| 927 | 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
| 928 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 929 | | PB |0| Payload Type| Native RPSI bit string | |
| 930 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 931 | | defined per codec ... | Padding (0) | |
| 932 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 933 | */ |
| 934 | /* |
| 935 | * Note: not generic made for VP8 |
| 936 | */ |
| 937 | WebRtc_Word32 |
| 938 | RTCPSender::BuildRPSI(WebRtc_UWord8* rtcpbuffer, |
| 939 | WebRtc_UWord32& pos, |
| 940 | const WebRtc_UWord64 pictureID, |
| 941 | const WebRtc_UWord8 payloadType) |
| 942 | { |
| 943 | // sanity |
| 944 | if(pos + 24 >= IP_PACKET_SIZE) |
| 945 | { |
| 946 | return -2; |
| 947 | } |
| 948 | // add Reference Picture Selection Indication |
| 949 | WebRtc_UWord8 FMT = 3; |
| 950 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 951 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 952 | |
| 953 | // calc length |
| 954 | WebRtc_UWord32 bitsRequired = 7; |
| 955 | WebRtc_UWord8 bytesRequired = 1; |
| 956 | while((pictureID>>bitsRequired) > 0) |
| 957 | { |
| 958 | bitsRequired += 7; |
| 959 | bytesRequired++; |
| 960 | } |
| 961 | |
| 962 | WebRtc_UWord8 size = 3; |
| 963 | if(bytesRequired > 6) |
| 964 | { |
| 965 | size = 5; |
| 966 | } else if(bytesRequired > 2) |
| 967 | { |
| 968 | size = 4; |
| 969 | } |
| 970 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 971 | rtcpbuffer[pos++]=size; |
| 972 | |
| 973 | // Add our own SSRC |
| 974 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 975 | pos += 4; |
| 976 | |
| 977 | // Add the remote SSRC |
| 978 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 979 | pos += 4; |
| 980 | |
| 981 | // calc padding length |
| 982 | WebRtc_UWord8 paddingBytes = 4-((2+bytesRequired)%4); |
| 983 | if(paddingBytes == 4) |
| 984 | { |
| 985 | paddingBytes = 0; |
| 986 | } |
| 987 | // add padding length in bits |
| 988 | rtcpbuffer[pos] = paddingBytes*8; // padding can be 0, 8, 16 or 24 |
| 989 | pos++; |
| 990 | |
| 991 | // add payload type |
| 992 | rtcpbuffer[pos] = payloadType; |
| 993 | pos++; |
| 994 | |
| 995 | // add picture ID |
| 996 | for(int i = bytesRequired-1; i > 0; i--) |
| 997 | { |
| 998 | rtcpbuffer[pos] = 0x80 | WebRtc_UWord8(pictureID >> (i*7)); |
| 999 | pos++; |
| 1000 | } |
| 1001 | // add last byte of picture ID |
| 1002 | rtcpbuffer[pos] = WebRtc_UWord8(pictureID & 0x7f); |
| 1003 | pos++; |
| 1004 | |
| 1005 | // add padding |
| 1006 | for(int j = 0; j <paddingBytes; j++) |
| 1007 | { |
| 1008 | rtcpbuffer[pos] = 0; |
| 1009 | pos++; |
| 1010 | } |
| 1011 | return 0; |
| 1012 | } |
| 1013 | |
| 1014 | WebRtc_Word32 |
| 1015 | RTCPSender::BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1016 | { |
| 1017 | // sanity |
| 1018 | if(pos + 20 + 4 * _lengthRembSSRC >= IP_PACKET_SIZE) |
| 1019 | { |
| 1020 | return -2; |
| 1021 | } |
| 1022 | // add application layer feedback |
| 1023 | WebRtc_UWord8 FMT = 15; |
| 1024 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 1025 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 1026 | |
| 1027 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1028 | rtcpbuffer[pos++]=_lengthRembSSRC + 4; |
| 1029 | |
| 1030 | // Add our own SSRC |
| 1031 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1032 | pos += 4; |
| 1033 | |
| 1034 | // Remote SSRC must be 0 |
| 1035 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, 0); |
| 1036 | pos += 4; |
| 1037 | |
| 1038 | rtcpbuffer[pos++]='R'; |
| 1039 | rtcpbuffer[pos++]='E'; |
| 1040 | rtcpbuffer[pos++]='M'; |
| 1041 | rtcpbuffer[pos++]='B'; |
| 1042 | |
| 1043 | rtcpbuffer[pos++] = _lengthRembSSRC; |
| 1044 | // 6 bit Exp |
| 1045 | // 18 bit mantissa |
| 1046 | WebRtc_UWord8 brExp = 0; |
| 1047 | for(WebRtc_UWord32 i=0; i<64; i++) |
| 1048 | { |
| 1049 | if(_rembBitrate <= ((WebRtc_UWord32)262143 << i)) |
| 1050 | { |
| 1051 | brExp = i; |
| 1052 | break; |
| 1053 | } |
| 1054 | } |
| 1055 | const WebRtc_UWord32 brMantissa = (_rembBitrate >> brExp); |
| 1056 | rtcpbuffer[pos++]=(WebRtc_UWord8)((brExp << 2) + ((brMantissa >> 16) & 0x03)); |
| 1057 | rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa >> 8); |
| 1058 | rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa); |
| 1059 | |
| 1060 | for (int i = 0; i < _lengthRembSSRC; i++) |
| 1061 | { |
| 1062 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rembSSRC[i]); |
| 1063 | pos += 4; |
| 1064 | } |
| 1065 | return 0; |
| 1066 | } |
| 1067 | |
| 1068 | void |
| 1069 | RTCPSender::SetTargetBitrate(unsigned int target_bitrate) |
| 1070 | { |
| 1071 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1072 | _tmmbr_Send = target_bitrate / 1000; |
| 1073 | } |
| 1074 | |
| 1075 | WebRtc_Word32 |
| 1076 | RTCPSender::BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1077 | { |
| 1078 | // Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate |
| 1079 | // If the sender is an owner of the TMMBN -> send TMMBR |
| 1080 | // If not an owner but the TMMBR would enter the TMMBN -> send TMMBR |
| 1081 | |
| 1082 | // get current bounding set from RTCP receiver |
| 1083 | bool tmmbrOwner = false; |
| 1084 | // store in candidateSet, allocates one extra slot |
| 1085 | TMMBRSet* candidateSet = _tmmbrHelp.CandidateSet(); |
| 1086 | |
| 1087 | // holding _criticalSectionRTCPSender while calling RTCPreceiver which |
| 1088 | // will accuire _criticalSectionRTCPReceiver is a potental deadlock but |
| 1089 | // since RTCPreceiver is not doing the reverse we should be fine |
| 1090 | WebRtc_Word32 lengthOfBoundingSet |
| 1091 | = _rtpRtcp.BoundingSet(tmmbrOwner, candidateSet); |
| 1092 | |
| 1093 | if(lengthOfBoundingSet > 0) |
| 1094 | { |
| 1095 | for (WebRtc_Word32 i = 0; i < lengthOfBoundingSet; i++) |
| 1096 | { |
| 1097 | if( candidateSet->Tmmbr(i) == _tmmbr_Send && |
| 1098 | candidateSet->PacketOH(i) == _packetOH_Send) |
| 1099 | { |
| 1100 | // do not send the same tuple |
| 1101 | return 0; |
| 1102 | } |
| 1103 | } |
| 1104 | if(!tmmbrOwner) |
| 1105 | { |
| 1106 | // use received bounding set as candidate set |
| 1107 | // add current tuple |
| 1108 | candidateSet->SetEntry(lengthOfBoundingSet, |
| 1109 | _tmmbr_Send, |
| 1110 | _packetOH_Send, |
| 1111 | _SSRC); |
| 1112 | int numCandidates = lengthOfBoundingSet+ 1; |
| 1113 | |
| 1114 | // find bounding set |
| 1115 | TMMBRSet* boundingSet = NULL; |
| 1116 | int numBoundingSet = _tmmbrHelp.FindTMMBRBoundingSet(boundingSet); |
| 1117 | if(numBoundingSet > 0 || numBoundingSet <= numCandidates) |
| 1118 | { |
| 1119 | tmmbrOwner = _tmmbrHelp.IsOwner(_SSRC, numBoundingSet); |
| 1120 | } |
| 1121 | if(!tmmbrOwner) |
| 1122 | { |
| 1123 | // did not enter bounding set, no meaning to send this request |
| 1124 | return 0; |
| 1125 | } |
| 1126 | } |
| 1127 | } |
| 1128 | |
| 1129 | if(_tmmbr_Send) |
| 1130 | { |
| 1131 | // sanity |
| 1132 | if(pos + 20 >= IP_PACKET_SIZE) |
| 1133 | { |
| 1134 | return -2; |
| 1135 | } |
| 1136 | // add TMMBR indicator |
| 1137 | WebRtc_UWord8 FMT = 3; |
| 1138 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 1139 | rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| 1140 | |
| 1141 | //Length of 4 |
| 1142 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1143 | rtcpbuffer[pos++]=(WebRtc_UWord8)(4); |
| 1144 | |
| 1145 | // Add our own SSRC |
| 1146 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1147 | pos += 4; |
| 1148 | |
| 1149 | // RFC 5104 4.2.1.2. Semantics |
| 1150 | |
| 1151 | // SSRC of media source |
| 1152 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1153 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1154 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1155 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1156 | |
| 1157 | // Additional Feedback Control Information (FCI) |
| 1158 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 1159 | pos += 4; |
| 1160 | |
| 1161 | WebRtc_UWord32 bitRate = _tmmbr_Send*1000; |
| 1162 | WebRtc_UWord32 mmbrExp = 0; |
| 1163 | for(WebRtc_UWord32 i=0;i<64;i++) |
| 1164 | { |
| 1165 | if(bitRate <= ((WebRtc_UWord32)131071 << i)) |
| 1166 | { |
| 1167 | mmbrExp = i; |
| 1168 | break; |
| 1169 | } |
| 1170 | } |
| 1171 | WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp); |
| 1172 | |
| 1173 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03)); |
| 1174 | rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7); |
| 1175 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((_packetOH_Send >> 8)& 0x01)); |
| 1176 | rtcpbuffer[pos++]=(WebRtc_UWord8)(_packetOH_Send); |
| 1177 | } |
| 1178 | return 0; |
| 1179 | } |
| 1180 | |
| 1181 | WebRtc_Word32 |
| 1182 | RTCPSender::BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1183 | { |
| 1184 | TMMBRSet* boundingSet = _tmmbrHelp.BoundingSetToSend(); |
| 1185 | if(boundingSet == NULL) |
| 1186 | { |
| 1187 | return -1; |
| 1188 | } |
| 1189 | // sanity |
| 1190 | if(pos + 12 + boundingSet->lengthOfSet()*8 >= IP_PACKET_SIZE) |
| 1191 | { |
| 1192 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1193 | return -2; |
| 1194 | } |
| 1195 | WebRtc_UWord8 FMT = 4; |
| 1196 | // add TMMBN indicator |
| 1197 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 1198 | rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| 1199 | |
| 1200 | //Add length later |
| 1201 | int posLength = pos; |
| 1202 | pos++; |
| 1203 | pos++; |
| 1204 | |
| 1205 | // Add our own SSRC |
| 1206 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1207 | pos += 4; |
| 1208 | |
| 1209 | // RFC 5104 4.2.2.2. Semantics |
| 1210 | |
| 1211 | // SSRC of media source |
| 1212 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1213 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1214 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1215 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1216 | |
| 1217 | // Additional Feedback Control Information (FCI) |
| 1218 | int numBoundingSet = 0; |
| 1219 | for(WebRtc_UWord32 n=0; n< boundingSet->lengthOfSet(); n++) |
| 1220 | { |
| 1221 | if (boundingSet->Tmmbr(n) > 0) |
| 1222 | { |
| 1223 | WebRtc_UWord32 tmmbrSSRC = boundingSet->Ssrc(n); |
| 1224 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, tmmbrSSRC); |
| 1225 | pos += 4; |
| 1226 | |
| 1227 | WebRtc_UWord32 bitRate = boundingSet->Tmmbr(n) * 1000; |
| 1228 | WebRtc_UWord32 mmbrExp = 0; |
| 1229 | for(int i=0; i<64; i++) |
| 1230 | { |
| 1231 | if(bitRate <= ((WebRtc_UWord32)131071 << i)) |
| 1232 | { |
| 1233 | mmbrExp = i; |
| 1234 | break; |
| 1235 | } |
| 1236 | } |
| 1237 | WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp); |
| 1238 | WebRtc_UWord32 measuredOH = boundingSet->PacketOH(n); |
| 1239 | |
| 1240 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03)); |
| 1241 | rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7); |
| 1242 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((measuredOH >> 8)& 0x01)); |
| 1243 | rtcpbuffer[pos++]=(WebRtc_UWord8)(measuredOH); |
| 1244 | numBoundingSet++; |
| 1245 | } |
| 1246 | } |
| 1247 | WebRtc_UWord16 length= (WebRtc_UWord16)(2+2*numBoundingSet); |
| 1248 | rtcpbuffer[posLength++]=(WebRtc_UWord8)(length>>8); |
| 1249 | rtcpbuffer[posLength]=(WebRtc_UWord8)(length); |
| 1250 | return 0; |
| 1251 | } |
| 1252 | |
| 1253 | WebRtc_Word32 |
| 1254 | RTCPSender::BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1255 | { |
| 1256 | // sanity |
| 1257 | if(_appData == NULL) |
| 1258 | { |
| 1259 | WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__); |
| 1260 | return -1; |
| 1261 | } |
| 1262 | if(pos + 12 + _appLength >= IP_PACKET_SIZE) |
| 1263 | { |
| 1264 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1265 | return -2; |
| 1266 | } |
| 1267 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + _appSubType; |
| 1268 | |
| 1269 | // Add APP ID |
| 1270 | rtcpbuffer[pos++]=(WebRtc_UWord8)204; |
| 1271 | |
| 1272 | WebRtc_UWord16 length = (_appLength>>2) + 2; // include SSRC and name |
| 1273 | rtcpbuffer[pos++]=(WebRtc_UWord8)(length>>8); |
| 1274 | rtcpbuffer[pos++]=(WebRtc_UWord8)(length); |
| 1275 | |
| 1276 | // Add our own SSRC |
| 1277 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1278 | pos += 4; |
| 1279 | |
| 1280 | // Add our application name |
| 1281 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _appName); |
| 1282 | pos += 4; |
| 1283 | |
| 1284 | // Add the data |
| 1285 | memcpy(rtcpbuffer +pos, _appData,_appLength); |
| 1286 | pos += _appLength; |
| 1287 | return 0; |
| 1288 | } |
| 1289 | |
| 1290 | WebRtc_Word32 |
| 1291 | RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer, |
| 1292 | WebRtc_UWord32& pos, |
| 1293 | const WebRtc_Word32 nackSize, |
| 1294 | const WebRtc_UWord16* nackList) |
| 1295 | { |
| 1296 | // sanity |
| 1297 | if(pos + 16 >= IP_PACKET_SIZE) |
| 1298 | { |
| 1299 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1300 | return -2; |
| 1301 | } |
| 1302 | |
| 1303 | // int size, WebRtc_UWord16* nackList |
| 1304 | // add nack list |
| 1305 | WebRtc_UWord8 FMT = 1; |
| 1306 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 1307 | rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| 1308 | |
| 1309 | rtcpbuffer[pos++]=(WebRtc_UWord8) 0; |
| 1310 | int nackSizePos = pos; |
| 1311 | rtcpbuffer[pos++]=(WebRtc_UWord8)(3); //setting it to one kNACK signal as default |
| 1312 | |
| 1313 | // Add our own SSRC |
| 1314 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1315 | pos += 4; |
| 1316 | |
| 1317 | // Add the remote SSRC |
| 1318 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 1319 | pos += 4; |
| 1320 | |
| 1321 | // add the list |
| 1322 | int i = 0; |
| 1323 | int numOfNackFields = 0; |
| 1324 | while(nackSize > i && numOfNackFields < 253) |
| 1325 | { |
| 1326 | WebRtc_UWord16 nack = nackList[i]; |
| 1327 | // put dow our sequence number |
| 1328 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, nack); |
| 1329 | pos += 2; |
| 1330 | |
| 1331 | i++; |
| 1332 | numOfNackFields++; |
| 1333 | if(nackSize > i) |
| 1334 | { |
| 1335 | bool moreThan16Away = (WebRtc_UWord16(nack+16) < nackList[i])?true: false; |
| 1336 | if(!moreThan16Away) |
| 1337 | { |
| 1338 | // check for a wrap |
| 1339 | if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| 1340 | { |
| 1341 | // wrap |
| 1342 | moreThan16Away = true; |
| 1343 | } |
| 1344 | } |
| 1345 | if(moreThan16Away) |
| 1346 | { |
| 1347 | // next is more than 16 away |
| 1348 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1349 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1350 | } else |
| 1351 | { |
| 1352 | // build our bitmask |
| 1353 | WebRtc_UWord16 bitmask = 0; |
| 1354 | |
| 1355 | bool within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false; |
| 1356 | if(within16Away) |
| 1357 | { |
| 1358 | // check for a wrap |
| 1359 | if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| 1360 | { |
| 1361 | // wrap |
| 1362 | within16Away = false; |
| 1363 | } |
| 1364 | } |
| 1365 | |
| 1366 | while( nackSize > i && within16Away) |
| 1367 | { |
| 1368 | WebRtc_Word16 shift = (nackList[i]-nack)-1; |
| 1369 | assert(!(shift > 15) && !(shift < 0)); |
| 1370 | |
| 1371 | bitmask += (1<< shift); |
| 1372 | i++; |
| 1373 | if(nackSize > i) |
| 1374 | { |
| 1375 | within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false; |
| 1376 | if(within16Away) |
| 1377 | { |
| 1378 | // check for a wrap |
| 1379 | if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| 1380 | { |
| 1381 | // wrap |
| 1382 | within16Away = false; |
| 1383 | } |
| 1384 | } |
| 1385 | } |
| 1386 | } |
| 1387 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, bitmask); |
| 1388 | pos += 2; |
| 1389 | } |
| 1390 | // sanity do we have room from one more 4 byte block? |
| 1391 | if(pos + 4 >= IP_PACKET_SIZE) |
| 1392 | { |
| 1393 | return -2; |
| 1394 | } |
| 1395 | } else |
| 1396 | { |
| 1397 | // no more in the list |
| 1398 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1399 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1400 | } |
| 1401 | } |
| 1402 | rtcpbuffer[nackSizePos]=(WebRtc_UWord8)(2+numOfNackFields); |
| 1403 | return 0; |
| 1404 | } |
| 1405 | |
| 1406 | WebRtc_Word32 |
| 1407 | RTCPSender::BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1408 | { |
| 1409 | // sanity |
| 1410 | if(pos + 8 >= IP_PACKET_SIZE) |
| 1411 | { |
| 1412 | return -2; |
| 1413 | } |
| 1414 | if(_includeCSRCs) |
| 1415 | { |
| 1416 | // Add a bye packet |
| 1417 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1 + _CSRCs; // number of SSRC+CSRCs |
| 1418 | rtcpbuffer[pos++]=(WebRtc_UWord8)203; |
| 1419 | |
| 1420 | // length |
| 1421 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1422 | rtcpbuffer[pos++]=(WebRtc_UWord8)(1 + _CSRCs); |
| 1423 | |
| 1424 | // Add our own SSRC |
| 1425 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1426 | pos += 4; |
| 1427 | |
| 1428 | // add CSRCs |
| 1429 | for(int i = 0; i < _CSRCs; i++) |
| 1430 | { |
| 1431 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _CSRC[i]); |
| 1432 | pos += 4; |
| 1433 | } |
| 1434 | } else |
| 1435 | { |
| 1436 | // Add a bye packet |
| 1437 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1; // number of SSRC+CSRCs |
| 1438 | rtcpbuffer[pos++]=(WebRtc_UWord8)203; |
| 1439 | |
| 1440 | // length |
| 1441 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1442 | rtcpbuffer[pos++]=(WebRtc_UWord8)1; |
| 1443 | |
| 1444 | // Add our own SSRC |
| 1445 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1446 | pos += 4; |
| 1447 | } |
| 1448 | return 0; |
| 1449 | } |
| 1450 | |
| 1451 | WebRtc_Word32 |
| 1452 | RTCPSender::BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1453 | { |
| 1454 | // sanity |
| 1455 | if(pos + 44 >= IP_PACKET_SIZE) |
| 1456 | { |
| 1457 | return -2; |
| 1458 | } |
| 1459 | |
| 1460 | // Add XR header |
| 1461 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| 1462 | rtcpbuffer[pos++]=(WebRtc_UWord8)207; |
| 1463 | |
| 1464 | WebRtc_UWord32 XRLengthPos = pos; |
| 1465 | |
| 1466 | // handle length later on |
| 1467 | pos++; |
| 1468 | pos++; |
| 1469 | |
| 1470 | // Add our own SSRC |
| 1471 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1472 | pos += 4; |
| 1473 | |
| 1474 | // Add a VoIP metrics block |
| 1475 | rtcpbuffer[pos++]=7; |
| 1476 | rtcpbuffer[pos++]=0; |
| 1477 | rtcpbuffer[pos++]=0; |
| 1478 | rtcpbuffer[pos++]=8; |
| 1479 | |
| 1480 | // Add the remote SSRC |
| 1481 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 1482 | pos += 4; |
| 1483 | |
| 1484 | rtcpbuffer[pos++] = _xrVoIPMetric.lossRate; |
| 1485 | rtcpbuffer[pos++] = _xrVoIPMetric.discardRate; |
| 1486 | rtcpbuffer[pos++] = _xrVoIPMetric.burstDensity; |
| 1487 | rtcpbuffer[pos++] = _xrVoIPMetric.gapDensity; |
| 1488 | |
| 1489 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration >> 8); |
| 1490 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration); |
| 1491 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration >> 8); |
| 1492 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration); |
| 1493 | |
| 1494 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay >> 8); |
| 1495 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay); |
| 1496 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay >> 8); |
| 1497 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay); |
| 1498 | |
| 1499 | rtcpbuffer[pos++] = _xrVoIPMetric.signalLevel; |
| 1500 | rtcpbuffer[pos++] = _xrVoIPMetric.noiseLevel; |
| 1501 | rtcpbuffer[pos++] = _xrVoIPMetric.RERL; |
| 1502 | rtcpbuffer[pos++] = _xrVoIPMetric.Gmin; |
| 1503 | |
| 1504 | rtcpbuffer[pos++] = _xrVoIPMetric.Rfactor; |
| 1505 | rtcpbuffer[pos++] = _xrVoIPMetric.extRfactor; |
| 1506 | rtcpbuffer[pos++] = _xrVoIPMetric.MOSLQ; |
| 1507 | rtcpbuffer[pos++] = _xrVoIPMetric.MOSCQ; |
| 1508 | |
| 1509 | rtcpbuffer[pos++] = _xrVoIPMetric.RXconfig; |
| 1510 | rtcpbuffer[pos++] = 0; // reserved |
| 1511 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal >> 8); |
| 1512 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal); |
| 1513 | |
| 1514 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax >> 8); |
| 1515 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax); |
| 1516 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax >> 8); |
| 1517 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax); |
| 1518 | |
| 1519 | rtcpbuffer[XRLengthPos]=(WebRtc_UWord8)(0); |
| 1520 | rtcpbuffer[XRLengthPos+1]=(WebRtc_UWord8)(10); |
| 1521 | return 0; |
| 1522 | } |
| 1523 | |
| 1524 | WebRtc_Word32 |
| 1525 | RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags, |
| 1526 | const WebRtc_Word32 nackSize, // NACK |
| 1527 | const WebRtc_UWord16* nackList, // NACK |
| 1528 | const bool repeat, // FIR |
| 1529 | const WebRtc_UWord64 pictureID) // SLI & RPSI |
| 1530 | { |
| 1531 | WebRtc_UWord32 rtcpPacketTypeFlags = packetTypeFlags; |
| 1532 | WebRtc_UWord32 pos = 0; |
| 1533 | WebRtc_UWord8 rtcpbuffer[IP_PACKET_SIZE]; |
| 1534 | |
| 1535 | do // only to be able to use break :) (and the critsect must be inside its own scope) |
| 1536 | { |
| 1537 | // collect the received information |
| 1538 | RTCPReportBlock received; |
| 1539 | bool hasReceived = false; |
| 1540 | WebRtc_UWord32 NTPsec = 0; |
| 1541 | WebRtc_UWord32 NTPfrac = 0; |
| 1542 | bool rtcpCompound = false; |
| 1543 | WebRtc_UWord32 jitterTransmissionOffset = 0; |
| 1544 | |
| 1545 | { |
| 1546 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1547 | if(_method == kRtcpOff) |
| 1548 | { |
| 1549 | WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, |
| 1550 | "%s invalid state", __FUNCTION__); |
| 1551 | return -1; |
| 1552 | } |
| 1553 | rtcpCompound = (_method == kRtcpCompound) ? true : false; |
| 1554 | } |
| 1555 | |
| 1556 | if (rtcpCompound || |
| 1557 | rtcpPacketTypeFlags & kRtcpReport || |
| 1558 | rtcpPacketTypeFlags & kRtcpSr || |
| 1559 | rtcpPacketTypeFlags & kRtcpRr) |
| 1560 | { |
| 1561 | // get statistics from our RTPreceiver outside critsect |
| 1562 | if(_rtpRtcp.ReportBlockStatistics(&received.fractionLost, |
| 1563 | &received.cumulativeLost, |
| 1564 | &received.extendedHighSeqNum, |
| 1565 | &received.jitter, |
| 1566 | &jitterTransmissionOffset) == 0) |
| 1567 | { |
| 1568 | hasReceived = true; |
| 1569 | |
| 1570 | WebRtc_UWord32 lastReceivedRRNTPsecs = 0; |
| 1571 | WebRtc_UWord32 lastReceivedRRNTPfrac = 0; |
| 1572 | WebRtc_UWord32 remoteSR = 0; |
| 1573 | |
| 1574 | // ok even if we have not received a SR, we will send 0 in that case |
| 1575 | _rtpRtcp.LastReceivedNTP(lastReceivedRRNTPsecs, |
| 1576 | lastReceivedRRNTPfrac, |
| 1577 | remoteSR); |
| 1578 | |
| 1579 | // get our NTP as late as possible to avoid a race |
| 1580 | _clock.CurrentNTP(NTPsec, NTPfrac); |
| 1581 | |
| 1582 | // Delay since last received report |
| 1583 | WebRtc_UWord32 delaySinceLastReceivedSR = 0; |
| 1584 | if((lastReceivedRRNTPsecs !=0) || (lastReceivedRRNTPfrac !=0)) |
| 1585 | { |
| 1586 | // get the 16 lowest bits of seconds and the 16 higest bits of fractions |
| 1587 | WebRtc_UWord32 now=NTPsec&0x0000FFFF; |
| 1588 | now <<=16; |
| 1589 | now += (NTPfrac&0xffff0000)>>16; |
| 1590 | |
| 1591 | WebRtc_UWord32 receiveTime = lastReceivedRRNTPsecs&0x0000FFFF; |
| 1592 | receiveTime <<=16; |
| 1593 | receiveTime += (lastReceivedRRNTPfrac&0xffff0000)>>16; |
| 1594 | |
| 1595 | delaySinceLastReceivedSR = now-receiveTime; |
| 1596 | } |
| 1597 | received.delaySinceLastSR = delaySinceLastReceivedSR; |
| 1598 | received.lastSR = remoteSR; |
| 1599 | } else |
| 1600 | { |
| 1601 | // we need to send our NTP even if we dont have received any reports |
| 1602 | _clock.CurrentNTP(NTPsec, NTPfrac); |
| 1603 | } |
| 1604 | } |
| 1605 | |
| 1606 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1607 | |
| 1608 | if(_TMMBR ) // attach TMMBR to send and receive reports |
| 1609 | { |
| 1610 | rtcpPacketTypeFlags |= kRtcpTmmbr; |
| 1611 | } |
| 1612 | if(_appSend) |
| 1613 | { |
| 1614 | rtcpPacketTypeFlags |= kRtcpApp; |
| 1615 | _appSend = false; |
| 1616 | } |
| 1617 | if(_REMB && _sendREMB) |
| 1618 | { |
| 1619 | // Always attach REMB to SR if that is configured. Note that REMB is |
| 1620 | // only sent on one of the RTP modules in the REMB group. |
| 1621 | rtcpPacketTypeFlags |= kRtcpRemb; |
| 1622 | } |
| 1623 | if(_xrSendVoIPMetric) |
| 1624 | { |
| 1625 | rtcpPacketTypeFlags |= kRtcpXrVoipMetric; |
| 1626 | _xrSendVoIPMetric = false; |
| 1627 | } |
| 1628 | if(_sendTMMBN) // set when having received a TMMBR |
| 1629 | { |
| 1630 | rtcpPacketTypeFlags |= kRtcpTmmbn; |
| 1631 | _sendTMMBN = false; |
| 1632 | } |
| 1633 | |
| 1634 | if(_method == kRtcpCompound) |
| 1635 | { |
| 1636 | if(_sending) |
| 1637 | { |
| 1638 | rtcpPacketTypeFlags |= kRtcpSr; |
| 1639 | } else |
| 1640 | { |
| 1641 | rtcpPacketTypeFlags |= kRtcpRr; |
| 1642 | } |
| 1643 | if (_IJ && hasReceived) |
| 1644 | { |
| 1645 | rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset; |
| 1646 | } |
| 1647 | } else if(_method == kRtcpNonCompound) |
| 1648 | { |
| 1649 | if(rtcpPacketTypeFlags & kRtcpReport) |
| 1650 | { |
| 1651 | if(_sending) |
| 1652 | { |
| 1653 | rtcpPacketTypeFlags |= kRtcpSr; |
| 1654 | } else |
| 1655 | { |
| 1656 | rtcpPacketTypeFlags |= kRtcpRr; |
| 1657 | } |
| 1658 | } |
| 1659 | } |
| 1660 | if( rtcpPacketTypeFlags & kRtcpRr || |
| 1661 | rtcpPacketTypeFlags & kRtcpSr) |
| 1662 | { |
| 1663 | // generate next time to send a RTCP report |
| 1664 | // seeded from RTP constructor |
| 1665 | WebRtc_Word32 random = rand() % 1000; |
| 1666 | WebRtc_Word32 timeToNext = RTCP_INTERVAL_AUDIO_MS; |
| 1667 | |
| 1668 | if(_audio) |
| 1669 | { |
| 1670 | timeToNext = (RTCP_INTERVAL_AUDIO_MS/2) + (RTCP_INTERVAL_AUDIO_MS*random/1000); |
| 1671 | }else |
| 1672 | { |
| 1673 | WebRtc_UWord32 minIntervalMs = RTCP_INTERVAL_AUDIO_MS; |
| 1674 | if(_sending) |
| 1675 | { |
| 1676 | // calc bw for video 360/sendBW in kbit/s |
| 1677 | WebRtc_UWord32 sendBitrateKbit = 0; |
| 1678 | WebRtc_UWord32 videoRate = 0; |
| 1679 | WebRtc_UWord32 fecRate = 0; |
| 1680 | WebRtc_UWord32 nackRate = 0; |
| 1681 | _rtpRtcp.BitrateSent(&sendBitrateKbit, |
| 1682 | &videoRate, |
| 1683 | &fecRate, |
| 1684 | &nackRate); |
| 1685 | sendBitrateKbit /= 1000; |
| 1686 | if(sendBitrateKbit != 0) |
| 1687 | { |
| 1688 | minIntervalMs = 360000/sendBitrateKbit; |
| 1689 | } |
| 1690 | } |
| 1691 | if(minIntervalMs > RTCP_INTERVAL_VIDEO_MS) |
| 1692 | { |
| 1693 | minIntervalMs = RTCP_INTERVAL_VIDEO_MS; |
| 1694 | } |
| 1695 | timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000); |
| 1696 | } |
| 1697 | _nextTimeToSendRTCP = _clock.GetTimeInMS() + timeToNext; |
| 1698 | } |
| 1699 | |
| 1700 | // if the data does not fitt in the packet we fill it as much as possible |
| 1701 | WebRtc_Word32 buildVal = 0; |
| 1702 | |
| 1703 | if(rtcpPacketTypeFlags & kRtcpSr) |
| 1704 | { |
| 1705 | if(hasReceived) |
| 1706 | { |
| 1707 | buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac, &received); |
| 1708 | } else |
| 1709 | { |
| 1710 | buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac); |
| 1711 | } |
| 1712 | if(buildVal == -1) |
| 1713 | { |
| 1714 | return -1; // error |
| 1715 | |
| 1716 | }else if(buildVal == -2) |
| 1717 | { |
| 1718 | break; // out of buffer |
| 1719 | } |
| 1720 | buildVal = BuildSDEC(rtcpbuffer, pos); |
| 1721 | if(buildVal == -1) |
| 1722 | { |
| 1723 | return -1; // error |
| 1724 | |
| 1725 | }else if(buildVal == -2) |
| 1726 | { |
| 1727 | break; // out of buffer |
| 1728 | } |
| 1729 | |
| 1730 | }else if(rtcpPacketTypeFlags & kRtcpRr) |
| 1731 | { |
| 1732 | if(hasReceived) |
| 1733 | { |
| 1734 | buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac,&received); |
| 1735 | }else |
| 1736 | { |
| 1737 | buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac); |
| 1738 | } |
| 1739 | if(buildVal == -1) |
| 1740 | { |
| 1741 | return -1; // error |
| 1742 | |
| 1743 | }else if(buildVal == -2) |
| 1744 | { |
| 1745 | break; // out of buffer |
| 1746 | } |
| 1747 | // only of set |
| 1748 | if(_CNAME[0] != 0) |
| 1749 | { |
| 1750 | buildVal = BuildSDEC(rtcpbuffer, pos); |
| 1751 | if(buildVal == -1) |
| 1752 | { |
| 1753 | return -1; // error |
| 1754 | } |
| 1755 | } |
| 1756 | } |
| 1757 | if(rtcpPacketTypeFlags & kRtcpTransmissionTimeOffset) |
| 1758 | { |
| 1759 | // If present, this RTCP packet must be placed after a |
| 1760 | // receiver report. |
| 1761 | buildVal = BuildExtendedJitterReport(rtcpbuffer, |
| 1762 | pos, |
| 1763 | jitterTransmissionOffset); |
| 1764 | if(buildVal == -1) |
| 1765 | { |
| 1766 | return -1; // error |
| 1767 | } |
| 1768 | else if(buildVal == -2) |
| 1769 | { |
| 1770 | break; // out of buffer |
| 1771 | } |
| 1772 | } |
| 1773 | if(rtcpPacketTypeFlags & kRtcpPli) |
| 1774 | { |
| 1775 | buildVal = BuildPLI(rtcpbuffer, pos); |
| 1776 | if(buildVal == -1) |
| 1777 | { |
| 1778 | return -1; // error |
| 1779 | |
| 1780 | }else if(buildVal == -2) |
| 1781 | { |
| 1782 | break; // out of buffer |
| 1783 | } |
| 1784 | } |
| 1785 | if(rtcpPacketTypeFlags & kRtcpFir) |
| 1786 | { |
| 1787 | buildVal = BuildFIR(rtcpbuffer, pos, repeat); |
| 1788 | if(buildVal == -1) |
| 1789 | { |
| 1790 | return -1; // error |
| 1791 | |
| 1792 | }else if(buildVal == -2) |
| 1793 | { |
| 1794 | break; // out of buffer |
| 1795 | } |
| 1796 | } |
| 1797 | if(rtcpPacketTypeFlags & kRtcpSli) |
| 1798 | { |
| 1799 | buildVal = BuildSLI(rtcpbuffer, pos, (WebRtc_UWord8)pictureID); |
| 1800 | if(buildVal == -1) |
| 1801 | { |
| 1802 | return -1; // error |
| 1803 | |
| 1804 | }else if(buildVal == -2) |
| 1805 | { |
| 1806 | break; // out of buffer |
| 1807 | } |
| 1808 | } |
| 1809 | if(rtcpPacketTypeFlags & kRtcpRpsi) |
| 1810 | { |
| 1811 | const WebRtc_Word8 payloadType = _rtpRtcp.SendPayloadType(); |
| 1812 | if(payloadType == -1) |
| 1813 | { |
| 1814 | return -1; |
| 1815 | } |
| 1816 | buildVal = BuildRPSI(rtcpbuffer, pos, pictureID, (WebRtc_UWord8)payloadType); |
| 1817 | if(buildVal == -1) |
| 1818 | { |
| 1819 | return -1; // error |
| 1820 | |
| 1821 | }else if(buildVal == -2) |
| 1822 | { |
| 1823 | break; // out of buffer |
| 1824 | } |
| 1825 | } |
| 1826 | if(rtcpPacketTypeFlags & kRtcpRemb) |
| 1827 | { |
| 1828 | buildVal = BuildREMB(rtcpbuffer, pos); |
| 1829 | if(buildVal == -1) |
| 1830 | { |
| 1831 | return -1; // error |
| 1832 | |
| 1833 | }else if(buildVal == -2) |
| 1834 | { |
| 1835 | break; // out of buffer |
| 1836 | } |
| 1837 | } |
| 1838 | if(rtcpPacketTypeFlags & kRtcpBye) |
| 1839 | { |
| 1840 | buildVal = BuildBYE(rtcpbuffer, pos); |
| 1841 | if(buildVal == -1) |
| 1842 | { |
| 1843 | return -1; // error |
| 1844 | |
| 1845 | }else if(buildVal == -2) |
| 1846 | { |
| 1847 | break; // out of buffer |
| 1848 | } |
| 1849 | } |
| 1850 | if(rtcpPacketTypeFlags & kRtcpApp) |
| 1851 | { |
| 1852 | buildVal = BuildAPP(rtcpbuffer, pos); |
| 1853 | if(buildVal == -1) |
| 1854 | { |
| 1855 | return -1; // error |
| 1856 | |
| 1857 | }else if(buildVal == -2) |
| 1858 | { |
| 1859 | break; // out of buffer |
| 1860 | } |
| 1861 | } |
| 1862 | if(rtcpPacketTypeFlags & kRtcpTmmbr) |
| 1863 | { |
| 1864 | buildVal = BuildTMMBR(rtcpbuffer, pos); |
| 1865 | if(buildVal == -1) |
| 1866 | { |
| 1867 | return -1; // error |
| 1868 | |
| 1869 | }else if(buildVal == -2) |
| 1870 | { |
| 1871 | break; // out of buffer |
| 1872 | } |
| 1873 | } |
| 1874 | if(rtcpPacketTypeFlags & kRtcpTmmbn) |
| 1875 | { |
| 1876 | buildVal = BuildTMMBN(rtcpbuffer, pos); |
| 1877 | if(buildVal == -1) |
| 1878 | { |
| 1879 | return -1; // error |
| 1880 | |
| 1881 | }else if(buildVal == -2) |
| 1882 | { |
| 1883 | break; // out of buffer |
| 1884 | } |
| 1885 | } |
| 1886 | if(rtcpPacketTypeFlags & kRtcpNack) |
| 1887 | { |
| 1888 | buildVal = BuildNACK(rtcpbuffer, pos, nackSize, nackList); |
| 1889 | if(buildVal == -1) |
| 1890 | { |
| 1891 | return -1; // error |
| 1892 | |
| 1893 | }else if(buildVal == -2) |
| 1894 | { |
| 1895 | break; // out of buffer |
| 1896 | } |
| 1897 | } |
| 1898 | if(rtcpPacketTypeFlags & kRtcpXrVoipMetric) |
| 1899 | { |
| 1900 | buildVal = BuildVoIPMetric(rtcpbuffer, pos); |
| 1901 | if(buildVal == -1) |
| 1902 | { |
| 1903 | return -1; // error |
| 1904 | |
| 1905 | }else if(buildVal == -2) |
| 1906 | { |
| 1907 | break; // out of buffer |
| 1908 | } |
| 1909 | } |
| 1910 | }while (false); |
| 1911 | // Sanity don't send empty packets. |
| 1912 | if (pos == 0) |
| 1913 | { |
| 1914 | return -1; |
| 1915 | } |
| 1916 | return SendToNetwork(rtcpbuffer, (WebRtc_UWord16)pos); |
| 1917 | } |
| 1918 | |
| 1919 | WebRtc_Word32 |
| 1920 | RTCPSender::SendToNetwork(const WebRtc_UWord8* dataBuffer, |
| 1921 | const WebRtc_UWord16 length) |
| 1922 | { |
| 1923 | CriticalSectionScoped lock(_criticalSectionTransport); |
| 1924 | if(_cbTransport) |
| 1925 | { |
| 1926 | if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0) |
| 1927 | { |
| 1928 | return 0; |
| 1929 | } |
| 1930 | } |
| 1931 | return -1; |
| 1932 | } |
| 1933 | |
| 1934 | WebRtc_Word32 |
| 1935 | RTCPSender::SetCSRCStatus(const bool include) |
| 1936 | { |
| 1937 | _includeCSRCs = include; |
| 1938 | return 0; |
| 1939 | } |
| 1940 | |
| 1941 | WebRtc_Word32 |
| 1942 | RTCPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], |
| 1943 | const WebRtc_UWord8 arrLength) |
| 1944 | { |
| 1945 | if(arrLength > kRtpCsrcSize) |
| 1946 | { |
| 1947 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1948 | assert(false); |
| 1949 | return -1; |
| 1950 | } |
| 1951 | |
| 1952 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1953 | |
| 1954 | for(int i = 0; i < arrLength;i++) |
| 1955 | { |
| 1956 | _CSRC[i] = arrOfCSRC[i]; |
| 1957 | } |
| 1958 | _CSRCs = arrLength; |
| 1959 | return 0; |
| 1960 | } |
| 1961 | |
| 1962 | WebRtc_Word32 |
| 1963 | RTCPSender::SetApplicationSpecificData(const WebRtc_UWord8 subType, |
| 1964 | const WebRtc_UWord32 name, |
| 1965 | const WebRtc_UWord8* data, |
| 1966 | const WebRtc_UWord16 length) |
| 1967 | { |
| 1968 | if(length %4 != 0) |
| 1969 | { |
| 1970 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1971 | return -1; |
| 1972 | } |
| 1973 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1974 | |
| 1975 | if(_appData) |
| 1976 | { |
| 1977 | delete [] _appData; |
| 1978 | } |
| 1979 | |
| 1980 | _appSend = true; |
| 1981 | _appSubType = subType; |
| 1982 | _appName = name; |
| 1983 | _appData = new WebRtc_UWord8[length]; |
| 1984 | _appLength = length; |
| 1985 | memcpy(_appData, data, length); |
| 1986 | return 0; |
| 1987 | } |
| 1988 | |
| 1989 | WebRtc_Word32 |
| 1990 | RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) |
| 1991 | { |
| 1992 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1993 | memcpy(&_xrVoIPMetric, VoIPMetric, sizeof(RTCPVoIPMetric)); |
| 1994 | |
| 1995 | _xrSendVoIPMetric = true; |
| 1996 | return 0; |
| 1997 | } |
| 1998 | |
| 1999 | // called under critsect _criticalSectionRTCPSender |
| 2000 | WebRtc_Word32 RTCPSender::AddReportBlocks(WebRtc_UWord8* rtcpbuffer, |
| 2001 | WebRtc_UWord32& pos, |
| 2002 | WebRtc_UWord8& numberOfReportBlocks, |
| 2003 | const RTCPReportBlock* received, |
| 2004 | const WebRtc_UWord32 NTPsec, |
| 2005 | const WebRtc_UWord32 NTPfrac) { |
| 2006 | // sanity one block |
| 2007 | if(pos + 24 >= IP_PACKET_SIZE) { |
| 2008 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| 2009 | "%s invalid argument", __FUNCTION__); |
| 2010 | return -1; |
| 2011 | } |
| 2012 | numberOfReportBlocks = _reportBlocks.size(); |
| 2013 | if (received) { |
| 2014 | // add our multiple RR to numberOfReportBlocks |
| 2015 | numberOfReportBlocks++; |
| 2016 | } |
| 2017 | if (received) { |
| 2018 | // answer to the one that sends to me |
| 2019 | _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); |
| 2020 | |
| 2021 | // Remote SSRC |
| 2022 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 2023 | pos += 4; |
| 2024 | |
| 2025 | // fraction lost |
| 2026 | rtcpbuffer[pos++]=received->fractionLost; |
| 2027 | |
| 2028 | // cumulative loss |
| 2029 | ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos, |
| 2030 | received->cumulativeLost); |
| 2031 | pos += 3; |
| 2032 | // extended highest seq_no, contain the highest sequence number received |
| 2033 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| 2034 | received->extendedHighSeqNum); |
| 2035 | pos += 4; |
| 2036 | |
| 2037 | //Jitter |
| 2038 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->jitter); |
| 2039 | pos += 4; |
| 2040 | |
| 2041 | // Last SR timestamp, our NTP time when we received the last report |
| 2042 | // This is the value that we read from the send report packet not when we |
| 2043 | // received it... |
| 2044 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->lastSR); |
| 2045 | pos += 4; |
| 2046 | |
| 2047 | // Delay since last received report,time since we received the report |
| 2048 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| 2049 | received->delaySinceLastSR); |
| 2050 | pos += 4; |
| 2051 | } |
| 2052 | if ((pos + _reportBlocks.size() * 24) >= IP_PACKET_SIZE) { |
| 2053 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| 2054 | "%s invalid argument", __FUNCTION__); |
| 2055 | return -1; |
| 2056 | } |
| 2057 | std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it = |
| 2058 | _reportBlocks.begin(); |
| 2059 | |
| 2060 | for (; it != _reportBlocks.end(); it++) { |
| 2061 | // we can have multiple report block in a conference |
| 2062 | WebRtc_UWord32 remoteSSRC = it->first; |
| 2063 | RTCPReportBlock* reportBlock = it->second; |
| 2064 | if (reportBlock) { |
| 2065 | // Remote SSRC |
| 2066 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, remoteSSRC); |
| 2067 | pos += 4; |
| 2068 | |
| 2069 | // fraction lost |
| 2070 | rtcpbuffer[pos++] = reportBlock->fractionLost; |
| 2071 | |
| 2072 | // cumulative loss |
| 2073 | ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos, |
| 2074 | reportBlock->cumulativeLost); |
| 2075 | pos += 3; |
| 2076 | |
| 2077 | // extended highest seq_no, contain the highest sequence number received |
| 2078 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| 2079 | reportBlock->extendedHighSeqNum); |
| 2080 | pos += 4; |
| 2081 | |
| 2082 | //Jitter |
| 2083 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| 2084 | reportBlock->jitter); |
| 2085 | pos += 4; |
| 2086 | |
| 2087 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| 2088 | reportBlock->lastSR); |
| 2089 | pos += 4; |
| 2090 | |
| 2091 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| 2092 | reportBlock->delaySinceLastSR); |
| 2093 | pos += 4; |
| 2094 | } |
| 2095 | } |
| 2096 | return pos; |
| 2097 | } |
| 2098 | |
| 2099 | // no callbacks allowed inside this function |
| 2100 | WebRtc_Word32 |
| 2101 | RTCPSender::SetTMMBN(const TMMBRSet* boundingSet, |
| 2102 | const WebRtc_UWord32 maxBitrateKbit) |
| 2103 | { |
| 2104 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 2105 | |
| 2106 | if (0 == _tmmbrHelp.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit)) |
| 2107 | { |
| 2108 | _sendTMMBN = true; |
| 2109 | return 0; |
| 2110 | } |
| 2111 | return -1; |
| 2112 | } |
| 2113 | } // namespace webrtc |