niklase@google.com | 470e71d | 2011-07-07 08:21:25 +0000 | [diff] [blame^] | 1 | /* |
| 2 | * Copyright (c) 2011 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 | #include "rtcp_utility.h" |
| 13 | |
| 14 | #include <string.h> // memcpy |
| 15 | #include <cassert> // assert |
| 16 | #include <cstdlib> // rand |
| 17 | |
| 18 | #include "trace.h" |
| 19 | #include "tick_util.h" |
| 20 | #include "common_types.h" |
| 21 | #include "critical_section_wrapper.h" |
| 22 | |
| 23 | namespace webrtc { |
| 24 | RTCPSender::RTCPSender(const WebRtc_Word32 id, |
| 25 | const bool audio, |
| 26 | ModuleRtpRtcpPrivate& callback) : |
| 27 | _id(id), |
| 28 | _audio(audio), |
| 29 | _method(kRtcpOff), |
| 30 | _cbRtcpPrivate(callback), |
| 31 | _criticalSectionTransport(*CriticalSectionWrapper::CreateCriticalSection()), |
| 32 | _cbTransport(NULL), |
| 33 | |
| 34 | _criticalSectionRTCPSender(*CriticalSectionWrapper::CreateCriticalSection()), |
| 35 | _usingNack(false), |
| 36 | _sending(false), |
| 37 | _sendTMMBN(false), |
| 38 | _TMMBR(false), |
| 39 | _nextTimeToSendRTCP(0), |
| 40 | _SSRC(0), |
| 41 | _remoteSSRC(0), |
| 42 | _CNAME(), |
| 43 | _reportBlocks(), |
| 44 | _csrcCNAMEs(), |
| 45 | |
| 46 | _cameraDelayMS(0), |
| 47 | |
| 48 | _lastSendReport(), |
| 49 | _lastRTCPTime(), |
| 50 | |
| 51 | _CSRCs(0), |
| 52 | _CSRC(), |
| 53 | _includeCSRCs(true), |
| 54 | |
| 55 | _sequenceNumberFIR(0), |
| 56 | _lastTimeFIR(0), |
| 57 | |
| 58 | _tmmbrHelp(audio), |
| 59 | _tmmbr_Send(0), |
| 60 | _packetOH_Send(0), |
| 61 | _remoteRateControl(), |
| 62 | |
| 63 | _appSend(false), |
| 64 | _appSubType(0), |
| 65 | _appName(), |
| 66 | _appData(NULL), |
| 67 | _appLength(0), |
| 68 | _xrSendVoIPMetric(false), |
| 69 | _xrVoIPMetric() |
| 70 | { |
| 71 | memset(_CNAME, 0, sizeof(_CNAME)); |
| 72 | memset(_lastSendReport, 0, sizeof(_lastSendReport)); |
| 73 | memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); |
| 74 | |
| 75 | WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); |
| 76 | } |
| 77 | |
| 78 | RTCPSender::~RTCPSender() |
| 79 | { |
| 80 | if(_appData) |
| 81 | { |
| 82 | delete [] _appData; |
| 83 | } |
| 84 | |
| 85 | MapItem* item = _reportBlocks.First(); |
| 86 | while(item) |
| 87 | { |
| 88 | RTCPReportBlock* ptr = (RTCPReportBlock*)(item->GetItem()); |
| 89 | if(ptr) |
| 90 | { |
| 91 | delete ptr; |
| 92 | } |
| 93 | _reportBlocks.Erase(item); |
| 94 | item = _reportBlocks.First(); |
| 95 | } |
| 96 | item = _csrcCNAMEs.First(); |
| 97 | while(item) |
| 98 | { |
| 99 | RTCPUtility::RTCPCnameInformation* ptr = (RTCPUtility::RTCPCnameInformation*)(item->GetItem()); |
| 100 | if(ptr) |
| 101 | { |
| 102 | delete ptr; |
| 103 | } |
| 104 | _csrcCNAMEs.Erase(item); |
| 105 | item = _csrcCNAMEs.First(); |
| 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 | _SSRC = 0; |
| 125 | _remoteSSRC = 0; |
| 126 | _cameraDelayMS = 0; |
| 127 | _sequenceNumberFIR = 0; |
| 128 | _tmmbr_Send = 0; |
| 129 | _packetOH_Send = 0; |
| 130 | _remoteRateControl.Reset(); |
| 131 | _nextTimeToSendRTCP = 0; |
| 132 | _CSRCs = 0; |
| 133 | _appSend = false; |
| 134 | _appSubType = 0; |
| 135 | |
| 136 | if(_appData) |
| 137 | { |
| 138 | delete [] _appData; |
| 139 | _appData = NULL; |
| 140 | } |
| 141 | _appLength = 0; |
| 142 | |
| 143 | _xrSendVoIPMetric = false; |
| 144 | |
| 145 | memset(&_xrVoIPMetric, 0, sizeof(_xrVoIPMetric)); |
| 146 | memset(_CNAME, 0, sizeof(_CNAME)); |
| 147 | memset(_lastSendReport, 0, sizeof(_lastSendReport)); |
| 148 | memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); |
| 149 | return 0; |
| 150 | } |
| 151 | |
| 152 | void |
| 153 | RTCPSender::ChangeUniqueId(const WebRtc_Word32 id) |
| 154 | { |
| 155 | _id = id; |
| 156 | } |
| 157 | |
| 158 | WebRtc_Word32 |
| 159 | RTCPSender::RegisterSendTransport(Transport* outgoingTransport) |
| 160 | { |
| 161 | CriticalSectionScoped lock(_criticalSectionTransport); |
| 162 | _cbTransport = outgoingTransport; |
| 163 | return 0; |
| 164 | } |
| 165 | |
| 166 | RTCPMethod |
| 167 | RTCPSender::Status() const |
| 168 | { |
| 169 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 170 | return _method; |
| 171 | } |
| 172 | |
| 173 | WebRtc_Word32 |
| 174 | RTCPSender::SetRTCPStatus(const RTCPMethod method) |
| 175 | { |
| 176 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 177 | if(method != kRtcpOff) |
| 178 | { |
| 179 | if(_audio) |
| 180 | { |
| 181 | _nextTimeToSendRTCP = ModuleRTPUtility::GetTimeInMS() + (RTCP_INTERVAL_AUDIO_MS/2); |
| 182 | } else |
| 183 | { |
| 184 | _nextTimeToSendRTCP = ModuleRTPUtility::GetTimeInMS() + (RTCP_INTERVAL_VIDEO_MS/2); |
| 185 | } |
| 186 | } |
| 187 | _method = method; |
| 188 | return 0; |
| 189 | } |
| 190 | |
| 191 | bool |
| 192 | RTCPSender::Sending() const |
| 193 | { |
| 194 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 195 | return _sending; |
| 196 | } |
| 197 | |
| 198 | WebRtc_Word32 |
| 199 | RTCPSender::SetSendingStatus(const bool sending) |
| 200 | { |
| 201 | bool sendRTCPBye = false; |
| 202 | { |
| 203 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 204 | |
| 205 | if(_method != kRtcpOff) |
| 206 | { |
| 207 | if(sending == false && _sending == true) |
| 208 | { |
| 209 | // Trigger RTCP bye |
| 210 | sendRTCPBye = true; |
| 211 | } |
| 212 | } |
| 213 | _sending = sending; |
| 214 | } |
| 215 | if(sendRTCPBye) |
| 216 | { |
| 217 | return SendRTCP(kRtcpBye); |
| 218 | } |
| 219 | return 0; |
| 220 | } |
| 221 | |
| 222 | bool |
| 223 | RTCPSender::TMMBR() const |
| 224 | { |
| 225 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 226 | return _TMMBR; |
| 227 | } |
| 228 | |
| 229 | WebRtc_Word32 |
| 230 | RTCPSender::SetTMMBRStatus(const bool enable) |
| 231 | { |
| 232 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 233 | _TMMBR = enable; |
| 234 | return 0; |
| 235 | } |
| 236 | |
| 237 | void |
| 238 | RTCPSender::SetSSRC( const WebRtc_UWord32 ssrc) |
| 239 | { |
| 240 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 241 | |
| 242 | if(_SSRC != 0) |
| 243 | { |
| 244 | // not first SetSSRC, probably due to a collision |
| 245 | // schedule a new RTCP report |
| 246 | _nextTimeToSendRTCP = ModuleRTPUtility::GetTimeInMS() + 100; // make sure that we send a RTP packet |
| 247 | } |
| 248 | _SSRC = ssrc; |
| 249 | } |
| 250 | |
| 251 | WebRtc_Word32 |
| 252 | RTCPSender::SetRemoteSSRC( const WebRtc_UWord32 ssrc) |
| 253 | { |
| 254 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 255 | _remoteSSRC = ssrc; |
| 256 | _remoteRateControl.Reset(); |
| 257 | return 0; |
| 258 | } |
| 259 | |
| 260 | WebRtc_Word32 |
| 261 | RTCPSender::SetCameraDelay(const WebRtc_Word32 delayMS) |
| 262 | { |
| 263 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 264 | if(delayMS > 1000 || delayMS < -1000) |
| 265 | { |
| 266 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, delay can't be larger than 1 sec", __FUNCTION__); |
| 267 | return -1; |
| 268 | } |
| 269 | _cameraDelayMS = delayMS; |
| 270 | return 0; |
| 271 | } |
| 272 | |
| 273 | WebRtc_Word32 |
| 274 | RTCPSender::CNAME(WebRtc_Word8 cName[RTCP_CNAME_SIZE]) |
| 275 | { |
| 276 | if(cName == NULL) |
| 277 | { |
| 278 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 279 | return -1; |
| 280 | } |
| 281 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 282 | memcpy(cName, _CNAME, RTCP_CNAME_SIZE); |
| 283 | return 0; |
| 284 | } |
| 285 | |
| 286 | WebRtc_Word32 |
| 287 | RTCPSender::SetCNAME(const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) |
| 288 | { |
| 289 | if(cName == NULL) |
| 290 | { |
| 291 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 292 | return -1; |
| 293 | } |
| 294 | WebRtc_Word32 length = (WebRtc_Word32)strlen(cName); |
| 295 | if(length > RTCP_CNAME_SIZE) |
| 296 | { |
| 297 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, too long cName", __FUNCTION__); |
| 298 | return -1; |
| 299 | } |
| 300 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 301 | |
| 302 | memcpy(_CNAME, cName, length+1); |
| 303 | return 0; |
| 304 | } |
| 305 | |
| 306 | WebRtc_Word32 |
| 307 | RTCPSender::AddMixedCNAME(const WebRtc_UWord32 SSRC, |
| 308 | const WebRtc_Word8 cName[RTCP_CNAME_SIZE]) |
| 309 | { |
| 310 | if(cName == NULL) |
| 311 | { |
| 312 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 313 | return -1; |
| 314 | } |
| 315 | WebRtc_Word32 length = (WebRtc_Word32)strlen(cName); |
| 316 | if(length > RTCP_CNAME_SIZE) |
| 317 | { |
| 318 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, too long cName", __FUNCTION__); |
| 319 | return -1; |
| 320 | } |
| 321 | |
| 322 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 323 | if(_csrcCNAMEs.Size() == kRtpCsrcSize) |
| 324 | { |
| 325 | return -1; |
| 326 | } |
| 327 | RTCPUtility::RTCPCnameInformation* ptr= new RTCPUtility::RTCPCnameInformation(); |
| 328 | |
| 329 | memcpy(ptr->name, cName, length+1); |
| 330 | ptr->length = (WebRtc_UWord8)length; |
| 331 | _csrcCNAMEs.Insert(SSRC, ptr); |
| 332 | return 0; |
| 333 | } |
| 334 | |
| 335 | WebRtc_Word32 |
| 336 | RTCPSender::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) |
| 337 | { |
| 338 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 339 | MapItem* item= _csrcCNAMEs.Find(SSRC); |
| 340 | if(item) |
| 341 | { |
| 342 | RTCPUtility::RTCPCnameInformation* ptr= (RTCPUtility::RTCPCnameInformation*)(item->GetItem()); |
| 343 | if(ptr) |
| 344 | { |
| 345 | delete ptr; |
| 346 | } |
| 347 | _csrcCNAMEs.Erase(item); |
| 348 | return 0; |
| 349 | } |
| 350 | return -1; |
| 351 | } |
| 352 | |
| 353 | bool |
| 354 | RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const |
| 355 | { |
| 356 | /* |
| 357 | For audio we use a fix 5 sec interval |
| 358 | |
| 359 | For video we use 1 sec interval fo a BW smaller than 360 kbit/s, |
| 360 | technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare |
| 361 | |
| 362 | |
| 363 | From RFC 3550 |
| 364 | |
| 365 | MAX RTCP BW is 5% if the session BW |
| 366 | A send report is approximately 65 bytes inc CNAME |
| 367 | A report report is approximately 28 bytes |
| 368 | |
| 369 | The RECOMMENDED value for the reduced minimum in seconds is 360 |
| 370 | divided by the session bandwidth in kilobits/second. This minimum |
| 371 | is smaller than 5 seconds for bandwidths greater than 72 kb/s. |
| 372 | |
| 373 | If the participant has not yet sent an RTCP packet (the variable |
| 374 | initial is true), the constant Tmin is set to 2.5 seconds, else it |
| 375 | is set to 5 seconds. |
| 376 | |
| 377 | The interval between RTCP packets is varied randomly over the |
| 378 | range [0.5,1.5] times the calculated interval to avoid unintended |
| 379 | synchronization of all participants |
| 380 | |
| 381 | if we send |
| 382 | If the participant is a sender (we_sent true), the constant C is |
| 383 | set to the average RTCP packet size (avg_rtcp_size) divided by 25% |
| 384 | of the RTCP bandwidth (rtcp_bw), and the constant n is set to the |
| 385 | number of senders. |
| 386 | |
| 387 | if we receive only |
| 388 | If we_sent is not true, the constant C is set |
| 389 | to the average RTCP packet size divided by 75% of the RTCP |
| 390 | bandwidth. The constant n is set to the number of receivers |
| 391 | (members - senders). If the number of senders is greater than |
| 392 | 25%, senders and receivers are treated together. |
| 393 | |
| 394 | reconsideration NOT required for peer-to-peer |
| 395 | "timer reconsideration" is |
| 396 | employed. This algorithm implements a simple back-off mechanism |
| 397 | which causes users to hold back RTCP packet transmission if the |
| 398 | group sizes are increasing. |
| 399 | |
| 400 | n = number of members |
| 401 | C = avg_size/(rtcpBW/4) |
| 402 | |
| 403 | 3. The deterministic calculated interval Td is set to max(Tmin, n*C). |
| 404 | |
| 405 | 4. The calculated interval T is set to a number uniformly distributed |
| 406 | between 0.5 and 1.5 times the deterministic calculated interval. |
| 407 | |
| 408 | 5. The resulting value of T is divided by e-3/2=1.21828 to compensate |
| 409 | for the fact that the timer reconsideration algorithm converges to |
| 410 | a value of the RTCP bandwidth below the intended average |
| 411 | */ |
| 412 | |
| 413 | if(_method == kRtcpOff) |
| 414 | { |
| 415 | return false; |
| 416 | } |
| 417 | |
| 418 | WebRtc_UWord32 now = ModuleRTPUtility::GetTimeInMS(); |
| 419 | |
| 420 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 421 | |
| 422 | if(!_audio && sendKeyframeBeforeRTP) |
| 423 | { |
| 424 | // for video key-frames we want to send the RTCP before the large key-frame |
| 425 | // if we have a 100 ms margin |
| 426 | now += RTCP_SEND_BEFORE_KEY_FRAME_MS; |
| 427 | } |
| 428 | |
| 429 | if(now > _nextTimeToSendRTCP) |
| 430 | { |
| 431 | return true; |
| 432 | |
| 433 | } else if(now < 0x0000ffff && _nextTimeToSendRTCP > 0xffff0000) // 65 sec margin |
| 434 | { |
| 435 | // wrap |
| 436 | return true; |
| 437 | } |
| 438 | return false; |
| 439 | } |
| 440 | |
| 441 | WebRtc_UWord32 |
| 442 | RTCPSender::LastSendReport( WebRtc_UWord32& lastRTCPTime) |
| 443 | { |
| 444 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 445 | |
| 446 | lastRTCPTime = _lastRTCPTime[0]; |
| 447 | return _lastSendReport[0]; |
| 448 | } |
| 449 | |
| 450 | WebRtc_UWord32 |
| 451 | RTCPSender::SendTimeOfSendReport(const WebRtc_UWord32 sendReport) |
| 452 | { |
| 453 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 454 | |
| 455 | // This is only saved when we are the sender |
| 456 | if((_lastSendReport[0] == 0) || (sendReport == 0)) |
| 457 | { |
| 458 | return 0; // will be ignored |
| 459 | } else |
| 460 | { |
| 461 | for(int i = 0; i < RTCP_NUMBER_OF_SR; ++i) |
| 462 | { |
| 463 | if( _lastSendReport[i] == sendReport) |
| 464 | { |
| 465 | return _lastRTCPTime[i]; |
| 466 | } |
| 467 | } |
| 468 | } |
| 469 | return 0; |
| 470 | } |
| 471 | |
| 472 | WebRtc_Word32 |
| 473 | RTCPSender::AddReportBlock(const WebRtc_UWord32 SSRC, |
| 474 | const RTCPReportBlock* reportBlock) |
| 475 | { |
| 476 | if(reportBlock == NULL) |
| 477 | { |
| 478 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 479 | return -1; |
| 480 | } |
| 481 | |
| 482 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 483 | |
| 484 | if(_reportBlocks.Size() >= RTCP_MAX_REPORT_BLOCKS) |
| 485 | { |
| 486 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 487 | return -1; |
| 488 | } |
| 489 | RTCPReportBlock* copyReportBlock = new RTCPReportBlock(); |
| 490 | memcpy(copyReportBlock, reportBlock, sizeof(RTCPReportBlock)); |
| 491 | _reportBlocks.Insert(SSRC, copyReportBlock); |
| 492 | return 0; |
| 493 | } |
| 494 | |
| 495 | WebRtc_Word32 |
| 496 | RTCPSender::RemoveReportBlock(const WebRtc_UWord32 SSRC) |
| 497 | { |
| 498 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 499 | |
| 500 | MapItem* item= _reportBlocks.Find(SSRC); |
| 501 | if(item) |
| 502 | { |
| 503 | RTCPReportBlock* ptr= (RTCPReportBlock*)(item->GetItem()); |
| 504 | if(ptr) |
| 505 | { |
| 506 | delete ptr; |
| 507 | } |
| 508 | _reportBlocks.Erase(item); |
| 509 | return 0; |
| 510 | } |
| 511 | return -1; |
| 512 | } |
| 513 | |
| 514 | WebRtc_Word32 |
| 515 | RTCPSender::BuildSR(WebRtc_UWord8* rtcpbuffer, |
| 516 | WebRtc_UWord32& pos, |
| 517 | const WebRtc_UWord32 NTPsec, |
| 518 | const WebRtc_UWord32 NTPfrac, |
| 519 | const RTCPReportBlock* received) |
| 520 | { |
| 521 | // sanity |
| 522 | if(pos + 52 >= IP_PACKET_SIZE) |
| 523 | { |
| 524 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 525 | return -2; |
| 526 | } |
| 527 | WebRtc_UWord32 RTPtime; |
| 528 | WebRtc_UWord32 BackTimedNTPsec; |
| 529 | WebRtc_UWord32 BackTimedNTPfrac; |
| 530 | |
| 531 | WebRtc_UWord32 posNumberOfReportBlocks = pos; |
| 532 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| 533 | |
| 534 | // Sender report |
| 535 | rtcpbuffer[pos++]=(WebRtc_UWord8)200; |
| 536 | |
| 537 | for(int i = (RTCP_NUMBER_OF_SR-2); i >= 0; i--) |
| 538 | { |
| 539 | // shift old |
| 540 | _lastSendReport[i+1] = _lastSendReport[i]; |
| 541 | _lastRTCPTime[i+1] =_lastRTCPTime[i]; |
| 542 | } |
| 543 | |
| 544 | _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); // before video cam compensation |
| 545 | |
| 546 | if(_cameraDelayMS >= 0) |
| 547 | { |
| 548 | // fraction of a second as an unsigned word32 4.294 967 296E9 |
| 549 | WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)_cameraDelayMS* 4294967; // note camera delay can't be larger than +/-1000ms |
| 550 | if(NTPfrac > cameraDelayFixFrac) |
| 551 | { |
| 552 | // no problem just reduce the fraction part |
| 553 | BackTimedNTPfrac = NTPfrac - cameraDelayFixFrac; |
| 554 | BackTimedNTPsec = NTPsec; |
| 555 | } else |
| 556 | { |
| 557 | // we need to reduce the sec and add that sec to the frac |
| 558 | BackTimedNTPsec = NTPsec - 1; |
| 559 | BackTimedNTPfrac = 0xffffffff - (cameraDelayFixFrac - NTPfrac); |
| 560 | } |
| 561 | } else |
| 562 | { |
| 563 | // fraction of a second as an unsigned word32 4.294 967 296E9 |
| 564 | WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)(-_cameraDelayMS)* 4294967; // note camera delay can't be larger than +/-1000ms |
| 565 | if(NTPfrac > 0xffffffff - cameraDelayFixFrac) |
| 566 | { |
| 567 | // we need to add the sec and add that sec to the frac |
| 568 | BackTimedNTPsec = NTPsec + 1; |
| 569 | BackTimedNTPfrac = cameraDelayFixFrac + NTPfrac; // this will wrap but that is ok |
| 570 | } else |
| 571 | { |
| 572 | // no problem just add the fraction part |
| 573 | BackTimedNTPsec = NTPsec; |
| 574 | BackTimedNTPfrac = NTPfrac + cameraDelayFixFrac; |
| 575 | } |
| 576 | } |
| 577 | _lastSendReport[0] = (BackTimedNTPsec <<16) + (BackTimedNTPfrac >> 16); |
| 578 | |
| 579 | // RTP timestamp |
| 580 | // This should have a ramdom start value added |
| 581 | // RTP is counted from NTP not the acctual RTP |
| 582 | // This reflects the perfect RTP time |
| 583 | // we solve this by initiating RTP to our NTP :) |
| 584 | |
| 585 | WebRtc_UWord32 freqHz = 90000; // For video |
| 586 | if(_audio) |
| 587 | { |
| 588 | freqHz = _cbRtcpPrivate.CurrentSendFrequencyHz(); |
| 589 | RTPtime = ModuleRTPUtility::CurrentRTP(freqHz); |
| 590 | } |
| 591 | else // video |
| 592 | { |
| 593 | // used to be (WebRtc_UWord32)(((float)BackTimedNTPfrac/(float)FRAC)* 90000) |
| 594 | WebRtc_UWord32 tmp = 9*(BackTimedNTPfrac/429496); |
| 595 | RTPtime = BackTimedNTPsec*freqHz + tmp; |
| 596 | } |
| 597 | |
| 598 | |
| 599 | |
| 600 | |
| 601 | // Add sender data |
| 602 | // Save for our length field |
| 603 | pos++; |
| 604 | pos++; |
| 605 | |
| 606 | // Add our own SSRC |
| 607 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 608 | pos += 4; |
| 609 | // NTP |
| 610 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPsec); |
| 611 | pos += 4; |
| 612 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPfrac); |
| 613 | pos += 4; |
| 614 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, RTPtime); |
| 615 | pos += 4; |
| 616 | |
| 617 | //sender's packet count |
| 618 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _cbRtcpPrivate.PacketCountSent()); |
| 619 | pos += 4; |
| 620 | |
| 621 | //sender's octet count |
| 622 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _cbRtcpPrivate.ByteCountSent()); |
| 623 | pos += 4; |
| 624 | |
| 625 | WebRtc_UWord8 numberOfReportBlocks = 0; |
| 626 | WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac); |
| 627 | if(retVal < 0) |
| 628 | { |
| 629 | // |
| 630 | return retVal ; |
| 631 | } |
| 632 | rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; |
| 633 | |
| 634 | WebRtc_UWord16 len = WebRtc_UWord16((pos/4) -1); |
| 635 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); |
| 636 | return 0; |
| 637 | } |
| 638 | |
| 639 | |
| 640 | WebRtc_Word32 |
| 641 | RTCPSender::BuildSDEC(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 642 | { |
| 643 | WebRtc_UWord32 lengthCname =(WebRtc_UWord32)strlen((char*)_CNAME); |
| 644 | |
| 645 | // sanity max is 255 |
| 646 | if(lengthCname > RTCP_CNAME_SIZE) |
| 647 | { |
| 648 | lengthCname = RTCP_CNAME_SIZE; |
| 649 | } |
| 650 | // sanity |
| 651 | if(pos + 12+ lengthCname >= IP_PACKET_SIZE) |
| 652 | { |
| 653 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 654 | return -2; |
| 655 | } |
| 656 | // SDEC Source Description |
| 657 | |
| 658 | // We always need to add SDES CNAME |
| 659 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1 + _csrcCNAMEs.Size(); // source counts |
| 660 | rtcpbuffer[pos++]=(WebRtc_UWord8)202; |
| 661 | |
| 662 | // handle SDES length later on |
| 663 | WebRtc_UWord32 SDESLengthPos = pos; |
| 664 | pos++; |
| 665 | pos++; |
| 666 | |
| 667 | // Add our own SSRC |
| 668 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 669 | pos += 4; |
| 670 | |
| 671 | // CNAME = 1 |
| 672 | rtcpbuffer[pos++]=(WebRtc_UWord8)1; |
| 673 | |
| 674 | // |
| 675 | rtcpbuffer[pos++]=(WebRtc_UWord8)lengthCname; |
| 676 | |
| 677 | WebRtc_UWord16 SDESLength = 10; |
| 678 | |
| 679 | memcpy(&rtcpbuffer[pos],_CNAME,lengthCname); |
| 680 | pos += lengthCname; |
| 681 | SDESLength += (WebRtc_UWord16)lengthCname; |
| 682 | |
| 683 | WebRtc_UWord16 padding =0; |
| 684 | |
| 685 | // We must have a zero field even if we have an even multiple of 4 bytes |
| 686 | if((pos % 4) ==0) |
| 687 | { |
| 688 | padding++; |
| 689 | rtcpbuffer[pos++]=0; |
| 690 | } |
| 691 | while((pos % 4) !=0) |
| 692 | { |
| 693 | padding++; |
| 694 | rtcpbuffer[pos++]=0; |
| 695 | } |
| 696 | SDESLength += padding; |
| 697 | |
| 698 | MapItem* item = _csrcCNAMEs.First(); |
| 699 | |
| 700 | for(int i = 0; item && i < _csrcCNAMEs.Size(); i++) |
| 701 | { |
| 702 | RTCPUtility::RTCPCnameInformation* cname = (RTCPUtility::RTCPCnameInformation*)(item->GetItem()); |
| 703 | WebRtc_UWord32 SSRC = item->GetUnsignedId(); |
| 704 | |
| 705 | // Add SSRC |
| 706 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, SSRC); |
| 707 | pos += 4; |
| 708 | |
| 709 | // CNAME = 1 |
| 710 | rtcpbuffer[pos++]=(WebRtc_UWord8)1; |
| 711 | |
| 712 | rtcpbuffer[pos++]= cname->length; |
| 713 | |
| 714 | SDESLength += 6; |
| 715 | |
| 716 | memcpy(&rtcpbuffer[pos],cname->name, cname->length); |
| 717 | pos += cname->length; |
| 718 | SDESLength += cname->length; |
| 719 | |
| 720 | WebRtc_UWord16 padding =0; |
| 721 | |
| 722 | // We must have a zero field even if we have an even multiple of 4 bytes |
| 723 | if((pos % 4) ==0) |
| 724 | { |
| 725 | padding++; |
| 726 | rtcpbuffer[pos++]=0; |
| 727 | } |
| 728 | while((pos % 4) !=0) |
| 729 | { |
| 730 | padding++; |
| 731 | rtcpbuffer[pos++]=0; |
| 732 | } |
| 733 | SDESLength += padding; |
| 734 | |
| 735 | item = _csrcCNAMEs.Next(item); |
| 736 | } |
| 737 | WebRtc_UWord16 length = SDESLength; |
| 738 | length= (length/4) - 1; // in 32-bit words minus one and we dont count the header |
| 739 | |
| 740 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+SDESLengthPos, length); |
| 741 | return 0; |
| 742 | } |
| 743 | |
| 744 | WebRtc_Word32 |
| 745 | RTCPSender::BuildRR(WebRtc_UWord8* rtcpbuffer, |
| 746 | WebRtc_UWord32& pos, |
| 747 | const WebRtc_UWord32 NTPsec, |
| 748 | const WebRtc_UWord32 NTPfrac, |
| 749 | const RTCPReportBlock* received) |
| 750 | { |
| 751 | // sanity one block |
| 752 | if(pos + 32 >= IP_PACKET_SIZE) |
| 753 | { |
| 754 | return -2; |
| 755 | } |
| 756 | WebRtc_UWord32 posNumberOfReportBlocks = pos; |
| 757 | |
| 758 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| 759 | rtcpbuffer[pos++]=(WebRtc_UWord8)201; |
| 760 | |
| 761 | // Save for our length field |
| 762 | pos++; |
| 763 | pos++; |
| 764 | |
| 765 | // Add our own SSRC |
| 766 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 767 | pos += 4; |
| 768 | |
| 769 | WebRtc_UWord8 numberOfReportBlocks = 0; |
| 770 | WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac); |
| 771 | if(retVal < 0) |
| 772 | { |
| 773 | return retVal; |
| 774 | } |
| 775 | rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; |
| 776 | |
| 777 | WebRtc_UWord16 len = WebRtc_UWord16((pos)/4 -1); |
| 778 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); |
| 779 | return 0; |
| 780 | } |
| 781 | |
| 782 | WebRtc_Word32 |
| 783 | RTCPSender::BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 784 | { |
| 785 | // sanity |
| 786 | if(pos + 12 >= IP_PACKET_SIZE) |
| 787 | { |
| 788 | return -2; |
| 789 | } |
| 790 | // add picture loss indicator |
| 791 | WebRtc_UWord8 FMT = 1; |
| 792 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 793 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 794 | |
| 795 | //Used fixed length of 2 |
| 796 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 797 | rtcpbuffer[pos++]=(WebRtc_UWord8)(2); |
| 798 | |
| 799 | // Add our own SSRC |
| 800 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 801 | pos += 4; |
| 802 | |
| 803 | // Add the remote SSRC |
| 804 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 805 | pos += 4; |
| 806 | return 0; |
| 807 | } |
| 808 | |
| 809 | WebRtc_Word32 |
| 810 | RTCPSender::BuildFIR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, const WebRtc_UWord32 RTT) |
| 811 | { |
| 812 | bool firRepeat = false; |
| 813 | WebRtc_UWord32 diff = ModuleRTPUtility::GetTimeInMS() - _lastTimeFIR; |
| 814 | if(diff < RTT + 3) // 3 is processing jitter |
| 815 | { |
| 816 | // we have recently sent a FIR |
| 817 | // don't send another |
| 818 | return 0; |
| 819 | |
| 820 | } else |
| 821 | { |
| 822 | if(diff < (RTT*2 + RTCP_MIN_FRAME_LENGTH_MS)) |
| 823 | { |
| 824 | // send a FIR_REPEAT instead of a FIR |
| 825 | firRepeat = true; |
| 826 | } |
| 827 | } |
| 828 | _lastTimeFIR = ModuleRTPUtility::GetTimeInMS(); |
| 829 | if(!firRepeat) |
| 830 | { |
| 831 | _sequenceNumberFIR++; // do not increase if repetition |
| 832 | } |
| 833 | |
| 834 | // sanity |
| 835 | if(pos + 20 >= IP_PACKET_SIZE) |
| 836 | { |
| 837 | return -2; |
| 838 | } |
| 839 | |
| 840 | // add full intra request indicator |
| 841 | WebRtc_UWord8 FMT = 4; |
| 842 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 843 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 844 | |
| 845 | //Length of 4 |
| 846 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 847 | rtcpbuffer[pos++]=(WebRtc_UWord8)(4); |
| 848 | |
| 849 | // Add our own SSRC |
| 850 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 851 | pos += 4; |
| 852 | |
| 853 | // RFC 5104 4.3.1.2. Semantics |
| 854 | |
| 855 | // SSRC of media source |
| 856 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 857 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 858 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 859 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 860 | |
| 861 | // Additional Feedback Control Information (FCI) |
| 862 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 863 | pos += 4; |
| 864 | |
| 865 | rtcpbuffer[pos++]=(WebRtc_UWord8)(_sequenceNumberFIR); |
| 866 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 867 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 868 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 869 | return 0; |
| 870 | } |
| 871 | |
| 872 | /* |
| 873 | 0 1 2 3 |
| 874 | 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 |
| 875 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 876 | | First | Number | PictureID | |
| 877 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 878 | */ |
| 879 | WebRtc_Word32 |
| 880 | RTCPSender::BuildSLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, const WebRtc_UWord8 pictureID) |
| 881 | { |
| 882 | // sanity |
| 883 | if(pos + 16 >= IP_PACKET_SIZE) |
| 884 | { |
| 885 | return -2; |
| 886 | } |
| 887 | // add slice loss indicator |
| 888 | WebRtc_UWord8 FMT = 2; |
| 889 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 890 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 891 | |
| 892 | //Used fixed length of 3 |
| 893 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 894 | rtcpbuffer[pos++]=(WebRtc_UWord8)(3); |
| 895 | |
| 896 | // Add our own SSRC |
| 897 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 898 | pos += 4; |
| 899 | |
| 900 | // Add the remote SSRC |
| 901 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 902 | pos += 4; |
| 903 | |
| 904 | // Add first, number & picture ID 6 bits |
| 905 | // first = 0, 13 - bits |
| 906 | // number = 0x1fff, 13 - bits only ones for now |
| 907 | WebRtc_UWord32 sliField = (0x1fff << 6)+ (0x3f & pictureID); |
| 908 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, sliField); |
| 909 | pos += 4; |
| 910 | return 0; |
| 911 | } |
| 912 | |
| 913 | /* |
| 914 | 0 1 2 3 |
| 915 | 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 |
| 916 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 917 | | PB |0| Payload Type| Native RPSI bit string | |
| 918 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 919 | | defined per codec ... | Padding (0) | |
| 920 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| 921 | */ |
| 922 | /* |
| 923 | * Note: not generic made for VP8 |
| 924 | */ |
| 925 | WebRtc_Word32 |
| 926 | RTCPSender::BuildRPSI(WebRtc_UWord8* rtcpbuffer, |
| 927 | WebRtc_UWord32& pos, |
| 928 | const WebRtc_UWord64 pictureID, |
| 929 | const WebRtc_UWord8 payloadType) |
| 930 | { |
| 931 | // sanity |
| 932 | if(pos + 24 >= IP_PACKET_SIZE) |
| 933 | { |
| 934 | return -2; |
| 935 | } |
| 936 | // add Reference Picture Selection Indication |
| 937 | WebRtc_UWord8 FMT = 3; |
| 938 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 939 | rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| 940 | |
| 941 | // calc length |
| 942 | WebRtc_UWord32 bitsRequired = 7; |
| 943 | WebRtc_UWord8 bytesRequired = 1; |
| 944 | while((pictureID>>bitsRequired) > 0) |
| 945 | { |
| 946 | bitsRequired += 7; |
| 947 | bytesRequired++; |
| 948 | } |
| 949 | |
| 950 | WebRtc_UWord8 size = 3; |
| 951 | if(bytesRequired > 6) |
| 952 | { |
| 953 | size = 5; |
| 954 | } else if(bytesRequired > 2) |
| 955 | { |
| 956 | size = 4; |
| 957 | } |
| 958 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 959 | rtcpbuffer[pos++]=size; |
| 960 | |
| 961 | // Add our own SSRC |
| 962 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 963 | pos += 4; |
| 964 | |
| 965 | // Add the remote SSRC |
| 966 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 967 | pos += 4; |
| 968 | |
| 969 | // calc padding length |
| 970 | WebRtc_UWord8 paddingBytes = 4-((2+bytesRequired)%4); |
| 971 | if(paddingBytes == 4) |
| 972 | { |
| 973 | paddingBytes = 0; |
| 974 | } |
| 975 | // add padding length in bits |
| 976 | rtcpbuffer[pos] = paddingBytes*8; // padding can be 0, 8, 16 or 24 |
| 977 | pos++; |
| 978 | |
| 979 | // add payload type |
| 980 | rtcpbuffer[pos] = payloadType; |
| 981 | pos++; |
| 982 | |
| 983 | // add picture ID |
| 984 | for(int i = bytesRequired-1; i > 0; i--) |
| 985 | { |
| 986 | rtcpbuffer[pos] = 0x80 | WebRtc_UWord8(pictureID >> (i*7)); |
| 987 | pos++; |
| 988 | } |
| 989 | // add last byte of picture ID |
| 990 | rtcpbuffer[pos] = WebRtc_UWord8(pictureID & 0x7f); |
| 991 | pos++; |
| 992 | |
| 993 | // add padding |
| 994 | for(int j = 0; j <paddingBytes; j++) |
| 995 | { |
| 996 | rtcpbuffer[pos] = 0; |
| 997 | pos++; |
| 998 | } |
| 999 | return 0; |
| 1000 | } |
| 1001 | |
| 1002 | WebRtc_Word32 |
| 1003 | RTCPSender::BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, WebRtc_UWord32 RTT) |
| 1004 | { |
| 1005 | // Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate |
| 1006 | // If the sender is an owner of the TMMBN -> send TMMBR |
| 1007 | // If not an owner but the TMMBR would enter the TMMBN -> send TMMBR |
| 1008 | |
| 1009 | // About to send TMMBR, first run remote rate control |
| 1010 | // to get a target bit rate. |
| 1011 | _tmmbr_Send = _remoteRateControl.TargetBitRate(RTT) / 1000; |
| 1012 | |
| 1013 | // get current bounding set from RTCP receiver |
| 1014 | bool tmmbrOwner = false; |
| 1015 | TMMBRSet* candidateSet = _tmmbrHelp.CandidateSet(); // store in candidateSet, allocates one extra slot |
| 1016 | |
| 1017 | // holding _criticalSectionRTCPSender while calling RTCPreceiver which will accuire _criticalSectionRTCPReceiver |
| 1018 | // is a potental deadlock but since RTCPreceiver is not doing the revese we should be fine |
| 1019 | WebRtc_Word32 lengthOfBoundingSet = _cbRtcpPrivate.BoundingSet(tmmbrOwner, candidateSet); |
| 1020 | |
| 1021 | if(lengthOfBoundingSet > 0) |
| 1022 | { |
| 1023 | for (WebRtc_Word32 i = 0; i < lengthOfBoundingSet; i++) |
| 1024 | { |
| 1025 | if( candidateSet->ptrTmmbrSet[i] == _tmmbr_Send && |
| 1026 | candidateSet->ptrPacketOHSet[i] == _packetOH_Send) |
| 1027 | { |
| 1028 | // do not send the same tuple |
| 1029 | return 0; |
| 1030 | } |
| 1031 | } |
| 1032 | if(!tmmbrOwner) |
| 1033 | { |
| 1034 | // use received bounding set as candidate set |
| 1035 | // add current tuple |
| 1036 | candidateSet->ptrTmmbrSet[lengthOfBoundingSet] = _tmmbr_Send; |
| 1037 | candidateSet->ptrPacketOHSet[lengthOfBoundingSet] = _packetOH_Send; |
| 1038 | candidateSet->ptrSsrcSet[lengthOfBoundingSet] = _SSRC; |
| 1039 | int numCandidates = lengthOfBoundingSet+ 1; |
| 1040 | |
| 1041 | // find bounding set |
| 1042 | TMMBRSet* boundingSet = NULL; |
| 1043 | int numBoundingSet = _tmmbrHelp.FindTMMBRBoundingSet(boundingSet); |
| 1044 | if(numBoundingSet > 0 || numBoundingSet <= numCandidates) |
| 1045 | { |
| 1046 | tmmbrOwner = _tmmbrHelp.IsOwner(_SSRC, numBoundingSet); |
| 1047 | } |
| 1048 | if(!tmmbrOwner) |
| 1049 | { |
| 1050 | // did not enter bounding set, no meaning to send this request |
| 1051 | return 0; |
| 1052 | } |
| 1053 | } |
| 1054 | } |
| 1055 | |
| 1056 | if(_tmmbr_Send) |
| 1057 | { |
| 1058 | // sanity |
| 1059 | if(pos + 20 >= IP_PACKET_SIZE) |
| 1060 | { |
| 1061 | return -2; |
| 1062 | } |
| 1063 | // add TMMBR indicator |
| 1064 | WebRtc_UWord8 FMT = 3; |
| 1065 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 1066 | rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| 1067 | |
| 1068 | //Length of 4 |
| 1069 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1070 | rtcpbuffer[pos++]=(WebRtc_UWord8)(4); |
| 1071 | |
| 1072 | // Add our own SSRC |
| 1073 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1074 | pos += 4; |
| 1075 | |
| 1076 | // RFC 5104 4.2.1.2. Semantics |
| 1077 | |
| 1078 | // SSRC of media source |
| 1079 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1080 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1081 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1082 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1083 | |
| 1084 | // Additional Feedback Control Information (FCI) |
| 1085 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 1086 | pos += 4; |
| 1087 | |
| 1088 | WebRtc_UWord32 bitRate = _tmmbr_Send*1000; |
| 1089 | WebRtc_UWord32 mmbrExp = 0; |
| 1090 | for(WebRtc_UWord32 i=0;i<64;i++) |
| 1091 | { |
| 1092 | if(bitRate <= ((WebRtc_UWord32)131071 << i)) |
| 1093 | { |
| 1094 | mmbrExp = i; |
| 1095 | break; |
| 1096 | } |
| 1097 | } |
| 1098 | WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp); |
| 1099 | |
| 1100 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03)); |
| 1101 | rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7); |
| 1102 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((_packetOH_Send >> 8)& 0x01)); |
| 1103 | rtcpbuffer[pos++]=(WebRtc_UWord8)(_packetOH_Send); |
| 1104 | } |
| 1105 | return 0; |
| 1106 | } |
| 1107 | |
| 1108 | WebRtc_Word32 |
| 1109 | RTCPSender::BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1110 | { |
| 1111 | TMMBRSet* boundingSet = _tmmbrHelp.BoundingSetToSend(); |
| 1112 | if(boundingSet == NULL) |
| 1113 | { |
| 1114 | return -1; |
| 1115 | } |
| 1116 | // sanity |
| 1117 | if(pos + 12 + boundingSet->lengthOfSet*8 >= IP_PACKET_SIZE) |
| 1118 | { |
| 1119 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1120 | return -2; |
| 1121 | } |
| 1122 | WebRtc_UWord8 FMT = 4; |
| 1123 | // add TMMBN indicator |
| 1124 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 1125 | rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| 1126 | |
| 1127 | //Add length later |
| 1128 | int posLength = pos; |
| 1129 | pos++; |
| 1130 | pos++; |
| 1131 | |
| 1132 | // Add our own SSRC |
| 1133 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1134 | pos += 4; |
| 1135 | |
| 1136 | // RFC 5104 4.2.2.2. Semantics |
| 1137 | |
| 1138 | // SSRC of media source |
| 1139 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1140 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1141 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1142 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1143 | |
| 1144 | // Additional Feedback Control Information (FCI) |
| 1145 | int numBoundingSet = 0; |
| 1146 | for(WebRtc_UWord32 n=0; n< boundingSet->lengthOfSet; n++) |
| 1147 | { |
| 1148 | if (boundingSet->ptrTmmbrSet[n] > 0) |
| 1149 | { |
| 1150 | WebRtc_UWord32 tmmbrSSRC = boundingSet->ptrSsrcSet[n]; |
| 1151 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, tmmbrSSRC); |
| 1152 | pos += 4; |
| 1153 | |
| 1154 | WebRtc_UWord32 bitRate = boundingSet->ptrTmmbrSet[n] * 1000; |
| 1155 | WebRtc_UWord32 mmbrExp = 0; |
| 1156 | for(int i=0; i<64; i++) |
| 1157 | { |
| 1158 | if(bitRate <= ((WebRtc_UWord32)131071 << i)) |
| 1159 | { |
| 1160 | mmbrExp = i; |
| 1161 | break; |
| 1162 | } |
| 1163 | } |
| 1164 | WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp); |
| 1165 | WebRtc_UWord32 measuredOH = boundingSet->ptrPacketOHSet[n]; |
| 1166 | |
| 1167 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03)); |
| 1168 | rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7); |
| 1169 | rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((measuredOH >> 8)& 0x01)); |
| 1170 | rtcpbuffer[pos++]=(WebRtc_UWord8)(measuredOH); |
| 1171 | numBoundingSet++; |
| 1172 | } |
| 1173 | } |
| 1174 | WebRtc_UWord16 length= (WebRtc_UWord16)(2+2*numBoundingSet); |
| 1175 | rtcpbuffer[posLength++]=(WebRtc_UWord8)(length>>8); |
| 1176 | rtcpbuffer[posLength]=(WebRtc_UWord8)(length); |
| 1177 | return 0; |
| 1178 | } |
| 1179 | |
| 1180 | WebRtc_Word32 |
| 1181 | RTCPSender::BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1182 | { |
| 1183 | // sanity |
| 1184 | if(_appData == NULL) |
| 1185 | { |
| 1186 | WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__); |
| 1187 | return -1; |
| 1188 | } |
| 1189 | if(pos + 12 + _appLength >= IP_PACKET_SIZE) |
| 1190 | { |
| 1191 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1192 | return -2; |
| 1193 | } |
| 1194 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + _appSubType; |
| 1195 | |
| 1196 | // Add APP ID |
| 1197 | rtcpbuffer[pos++]=(WebRtc_UWord8)204; |
| 1198 | |
| 1199 | WebRtc_UWord16 length = (_appLength>>2) + 2; // include SSRC and name |
| 1200 | rtcpbuffer[pos++]=(WebRtc_UWord8)(length>>8); |
| 1201 | rtcpbuffer[pos++]=(WebRtc_UWord8)(length); |
| 1202 | |
| 1203 | // Add our own SSRC |
| 1204 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1205 | pos += 4; |
| 1206 | |
| 1207 | // Add our application name |
| 1208 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _appName); |
| 1209 | pos += 4; |
| 1210 | |
| 1211 | // Add the data |
| 1212 | memcpy(rtcpbuffer +pos, _appData,_appLength); |
| 1213 | pos += _appLength; |
| 1214 | return 0; |
| 1215 | } |
| 1216 | |
| 1217 | WebRtc_Word32 |
| 1218 | RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer, |
| 1219 | WebRtc_UWord32& pos, |
| 1220 | const WebRtc_Word32 nackSize, |
| 1221 | const WebRtc_UWord16* nackList) |
| 1222 | { |
| 1223 | // sanity |
| 1224 | if(pos + 16 >= IP_PACKET_SIZE) |
| 1225 | { |
| 1226 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1227 | return -2; |
| 1228 | } |
| 1229 | |
| 1230 | // int size, WebRtc_UWord16* nackList |
| 1231 | // add nack list |
| 1232 | WebRtc_UWord8 FMT = 1; |
| 1233 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| 1234 | rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| 1235 | |
| 1236 | rtcpbuffer[pos++]=(WebRtc_UWord8) 0; |
| 1237 | int nackSizePos = pos; |
| 1238 | rtcpbuffer[pos++]=(WebRtc_UWord8)(3); //setting it to one kNACK signal as default |
| 1239 | |
| 1240 | // Add our own SSRC |
| 1241 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1242 | pos += 4; |
| 1243 | |
| 1244 | // Add the remote SSRC |
| 1245 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 1246 | pos += 4; |
| 1247 | |
| 1248 | // add the list |
| 1249 | int i = 0; |
| 1250 | int numOfNackFields = 0; |
| 1251 | while(nackSize > i && numOfNackFields < 253) |
| 1252 | { |
| 1253 | WebRtc_UWord16 nack = nackList[i]; |
| 1254 | // put dow our sequence number |
| 1255 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, nack); |
| 1256 | pos += 2; |
| 1257 | |
| 1258 | i++; |
| 1259 | numOfNackFields++; |
| 1260 | if(nackSize > i) |
| 1261 | { |
| 1262 | bool moreThan16Away = (WebRtc_UWord16(nack+16) < nackList[i])?true: false; |
| 1263 | if(!moreThan16Away) |
| 1264 | { |
| 1265 | // check for a wrap |
| 1266 | if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| 1267 | { |
| 1268 | // wrap |
| 1269 | moreThan16Away = true; |
| 1270 | } |
| 1271 | } |
| 1272 | if(moreThan16Away) |
| 1273 | { |
| 1274 | // next is more than 16 away |
| 1275 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1276 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1277 | } else |
| 1278 | { |
| 1279 | // build our bitmask |
| 1280 | WebRtc_UWord16 bitmask = 0; |
| 1281 | |
| 1282 | bool within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false; |
| 1283 | if(within16Away) |
| 1284 | { |
| 1285 | // check for a wrap |
| 1286 | if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| 1287 | { |
| 1288 | // wrap |
| 1289 | within16Away = false; |
| 1290 | } |
| 1291 | } |
| 1292 | |
| 1293 | while( nackSize > i && within16Away) |
| 1294 | { |
| 1295 | WebRtc_Word16 shift = (nackList[i]-nack)-1; |
| 1296 | assert(!(shift > 15) && !(shift < 0)); |
| 1297 | |
| 1298 | bitmask += (1<< shift); |
| 1299 | i++; |
| 1300 | if(nackSize > i) |
| 1301 | { |
| 1302 | within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false; |
| 1303 | if(within16Away) |
| 1304 | { |
| 1305 | // check for a wrap |
| 1306 | if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| 1307 | { |
| 1308 | // wrap |
| 1309 | within16Away = false; |
| 1310 | } |
| 1311 | } |
| 1312 | } |
| 1313 | } |
| 1314 | ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, bitmask); |
| 1315 | pos += 2; |
| 1316 | } |
| 1317 | // sanity do we have room from one more 4 byte block? |
| 1318 | if(pos + 4 >= IP_PACKET_SIZE) |
| 1319 | { |
| 1320 | return -2; |
| 1321 | } |
| 1322 | } else |
| 1323 | { |
| 1324 | // no more in the list |
| 1325 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1326 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1327 | } |
| 1328 | } |
| 1329 | rtcpbuffer[nackSizePos]=(WebRtc_UWord8)(2+numOfNackFields); |
| 1330 | return 0; |
| 1331 | } |
| 1332 | |
| 1333 | WebRtc_Word32 |
| 1334 | RTCPSender::BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1335 | { |
| 1336 | // sanity |
| 1337 | if(pos + 8 >= IP_PACKET_SIZE) |
| 1338 | { |
| 1339 | return -2; |
| 1340 | } |
| 1341 | if(_includeCSRCs) |
| 1342 | { |
| 1343 | // Add a bye packet |
| 1344 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1 + _CSRCs; // number of SSRC+CSRCs |
| 1345 | rtcpbuffer[pos++]=(WebRtc_UWord8)203; |
| 1346 | |
| 1347 | // length |
| 1348 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1349 | rtcpbuffer[pos++]=(WebRtc_UWord8)(1 + _CSRCs); |
| 1350 | |
| 1351 | // Add our own SSRC |
| 1352 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1353 | pos += 4; |
| 1354 | |
| 1355 | // add CSRCs |
| 1356 | for(int i = 0; i < _CSRCs; i++) |
| 1357 | { |
| 1358 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _CSRC[i]); |
| 1359 | pos += 4; |
| 1360 | } |
| 1361 | } else |
| 1362 | { |
| 1363 | // Add a bye packet |
| 1364 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1; // number of SSRC+CSRCs |
| 1365 | rtcpbuffer[pos++]=(WebRtc_UWord8)203; |
| 1366 | |
| 1367 | // length |
| 1368 | rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| 1369 | rtcpbuffer[pos++]=(WebRtc_UWord8)1; |
| 1370 | |
| 1371 | // Add our own SSRC |
| 1372 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1373 | pos += 4; |
| 1374 | } |
| 1375 | return 0; |
| 1376 | } |
| 1377 | |
| 1378 | WebRtc_Word32 |
| 1379 | RTCPSender::BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| 1380 | { |
| 1381 | // sanity |
| 1382 | if(pos + 44 >= IP_PACKET_SIZE) |
| 1383 | { |
| 1384 | return -2; |
| 1385 | } |
| 1386 | |
| 1387 | // Add XR header |
| 1388 | rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| 1389 | rtcpbuffer[pos++]=(WebRtc_UWord8)207; |
| 1390 | |
| 1391 | WebRtc_UWord32 XRLengthPos = pos; |
| 1392 | |
| 1393 | // handle length later on |
| 1394 | pos++; |
| 1395 | pos++; |
| 1396 | |
| 1397 | // Add our own SSRC |
| 1398 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| 1399 | pos += 4; |
| 1400 | |
| 1401 | // Add a VoIP metrics block |
| 1402 | rtcpbuffer[pos++]=7; |
| 1403 | rtcpbuffer[pos++]=0; |
| 1404 | rtcpbuffer[pos++]=0; |
| 1405 | rtcpbuffer[pos++]=8; |
| 1406 | |
| 1407 | // Add the remote SSRC |
| 1408 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 1409 | pos += 4; |
| 1410 | |
| 1411 | rtcpbuffer[pos++] = _xrVoIPMetric.lossRate; |
| 1412 | rtcpbuffer[pos++] = _xrVoIPMetric.discardRate; |
| 1413 | rtcpbuffer[pos++] = _xrVoIPMetric.burstDensity; |
| 1414 | rtcpbuffer[pos++] = _xrVoIPMetric.gapDensity; |
| 1415 | |
| 1416 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration >> 8); |
| 1417 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration); |
| 1418 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration >> 8); |
| 1419 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration); |
| 1420 | |
| 1421 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay >> 8); |
| 1422 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay); |
| 1423 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay >> 8); |
| 1424 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay); |
| 1425 | |
| 1426 | rtcpbuffer[pos++] = _xrVoIPMetric.signalLevel; |
| 1427 | rtcpbuffer[pos++] = _xrVoIPMetric.noiseLevel; |
| 1428 | rtcpbuffer[pos++] = _xrVoIPMetric.RERL; |
| 1429 | rtcpbuffer[pos++] = _xrVoIPMetric.Gmin; |
| 1430 | |
| 1431 | rtcpbuffer[pos++] = _xrVoIPMetric.Rfactor; |
| 1432 | rtcpbuffer[pos++] = _xrVoIPMetric.extRfactor; |
| 1433 | rtcpbuffer[pos++] = _xrVoIPMetric.MOSLQ; |
| 1434 | rtcpbuffer[pos++] = _xrVoIPMetric.MOSCQ; |
| 1435 | |
| 1436 | rtcpbuffer[pos++] = _xrVoIPMetric.RXconfig; |
| 1437 | rtcpbuffer[pos++] = 0; // reserved |
| 1438 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal >> 8); |
| 1439 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal); |
| 1440 | |
| 1441 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax >> 8); |
| 1442 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax); |
| 1443 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax >> 8); |
| 1444 | rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax); |
| 1445 | |
| 1446 | rtcpbuffer[XRLengthPos]=(WebRtc_UWord8)(0); |
| 1447 | rtcpbuffer[XRLengthPos+1]=(WebRtc_UWord8)(10); |
| 1448 | return 0; |
| 1449 | } |
| 1450 | |
| 1451 | WebRtc_Word32 |
| 1452 | RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags, |
| 1453 | const WebRtc_Word32 nackSize, // NACK |
| 1454 | const WebRtc_UWord16* nackList, // NACK |
| 1455 | const WebRtc_UWord32 RTT, // FIR |
| 1456 | const WebRtc_UWord64 pictureID) // SLI & RPSI |
| 1457 | { |
| 1458 | WebRtc_UWord32 rtcpPacketTypeFlags = packetTypeFlags; |
| 1459 | WebRtc_UWord32 pos = 0; |
| 1460 | WebRtc_UWord8 rtcpbuffer[IP_PACKET_SIZE]; |
| 1461 | |
| 1462 | if(_method == kRtcpOff) |
| 1463 | { |
| 1464 | WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__); |
| 1465 | return -1; |
| 1466 | } |
| 1467 | |
| 1468 | do // only to be able to use break :) (and the critsect must be inside its own scope) |
| 1469 | { |
| 1470 | // collect the received information |
| 1471 | RTCPReportBlock received; |
| 1472 | bool hasReceived = false; |
| 1473 | WebRtc_UWord32 NTPsec = 0; |
| 1474 | WebRtc_UWord32 NTPfrac = 0; |
| 1475 | |
| 1476 | if( _method == kRtcpCompound || |
| 1477 | rtcpPacketTypeFlags & kRtcpReport || |
| 1478 | rtcpPacketTypeFlags & kRtcpSr || |
| 1479 | rtcpPacketTypeFlags & kRtcpRr) |
| 1480 | { |
| 1481 | // get statistics from our RTPreceiver outside critsect |
| 1482 | if(_cbRtcpPrivate.ReportBlockStatistics(&received.fractionLost, |
| 1483 | &received.cumulativeLost, |
| 1484 | &received.extendedHighSeqNum, |
| 1485 | &received.jitter) == 0) |
| 1486 | { |
| 1487 | hasReceived = true; |
| 1488 | |
| 1489 | WebRtc_UWord32 lastReceivedRRNTPsecs = 0; |
| 1490 | WebRtc_UWord32 lastReceivedRRNTPfrac = 0; |
| 1491 | WebRtc_UWord32 remoteSR = 0; |
| 1492 | |
| 1493 | // ok even if we have not received a SR, we will send 0 in that case |
| 1494 | _cbRtcpPrivate.LastReceivedNTP(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac, remoteSR); |
| 1495 | |
| 1496 | // get our NTP as late as possible to avoid a race |
| 1497 | ModuleRTPUtility::CurrentNTP(NTPsec, NTPfrac); |
| 1498 | |
| 1499 | // Delay since last received report |
| 1500 | WebRtc_UWord32 delaySinceLastReceivedSR = 0; |
| 1501 | if((lastReceivedRRNTPsecs !=0) || (lastReceivedRRNTPfrac !=0)) |
| 1502 | { |
| 1503 | // get the 16 lowest bits of seconds and the 16 higest bits of fractions |
| 1504 | WebRtc_UWord32 now=NTPsec&0x0000FFFF; |
| 1505 | now <<=16; |
| 1506 | now += (NTPfrac&0xffff0000)>>16; |
| 1507 | |
| 1508 | WebRtc_UWord32 receiveTime = lastReceivedRRNTPsecs&0x0000FFFF; |
| 1509 | receiveTime <<=16; |
| 1510 | receiveTime += (lastReceivedRRNTPfrac&0xffff0000)>>16; |
| 1511 | |
| 1512 | delaySinceLastReceivedSR = now-receiveTime; |
| 1513 | } |
| 1514 | received.delaySinceLastSR = delaySinceLastReceivedSR; |
| 1515 | received.lastSR = remoteSR; |
| 1516 | } else |
| 1517 | { |
| 1518 | // we need to send our NTP even if we dont have received any reports |
| 1519 | ModuleRTPUtility::CurrentNTP(NTPsec, NTPfrac); |
| 1520 | } |
| 1521 | } |
| 1522 | |
| 1523 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1524 | |
| 1525 | if(_TMMBR ) // attach TMMBR to send and receive reports |
| 1526 | { |
| 1527 | rtcpPacketTypeFlags |= kRtcpTmmbr; |
| 1528 | } |
| 1529 | if(_appSend) |
| 1530 | { |
| 1531 | rtcpPacketTypeFlags |= kRtcpApp; |
| 1532 | _appSend = false; |
| 1533 | } |
| 1534 | if(_xrSendVoIPMetric) |
| 1535 | { |
| 1536 | rtcpPacketTypeFlags |= kRtcpXrVoipMetric; |
| 1537 | _xrSendVoIPMetric = false; |
| 1538 | } |
| 1539 | if(_sendTMMBN) // set when having received a TMMBR |
| 1540 | { |
| 1541 | rtcpPacketTypeFlags |= kRtcpTmmbn; |
| 1542 | _sendTMMBN = false; |
| 1543 | } |
| 1544 | |
| 1545 | if(_method == kRtcpCompound) |
| 1546 | { |
| 1547 | if(_sending) |
| 1548 | { |
| 1549 | rtcpPacketTypeFlags |= kRtcpSr; |
| 1550 | } else |
| 1551 | { |
| 1552 | rtcpPacketTypeFlags |= kRtcpRr; |
| 1553 | } |
| 1554 | } else if(_method == kRtcpNonCompound) |
| 1555 | { |
| 1556 | if(rtcpPacketTypeFlags & kRtcpReport) |
| 1557 | { |
| 1558 | if(_sending) |
| 1559 | { |
| 1560 | rtcpPacketTypeFlags |= kRtcpSr; |
| 1561 | } else |
| 1562 | { |
| 1563 | rtcpPacketTypeFlags |= kRtcpRr; |
| 1564 | } |
| 1565 | } |
| 1566 | } |
| 1567 | if( rtcpPacketTypeFlags & kRtcpRr || |
| 1568 | rtcpPacketTypeFlags & kRtcpSr) |
| 1569 | { |
| 1570 | // generate next time to send a RTCP report |
| 1571 | // seeded from RTP constructor |
| 1572 | WebRtc_Word32 random = rand() % 1000; |
| 1573 | WebRtc_Word32 timeToNext = RTCP_INTERVAL_AUDIO_MS; |
| 1574 | |
| 1575 | if(_audio) |
| 1576 | { |
| 1577 | timeToNext = (RTCP_INTERVAL_AUDIO_MS/2) + (RTCP_INTERVAL_AUDIO_MS*random/1000); |
| 1578 | }else |
| 1579 | { |
| 1580 | WebRtc_UWord32 minIntervalMs = RTCP_INTERVAL_AUDIO_MS; |
| 1581 | if(_sending) |
| 1582 | { |
| 1583 | // calc bw for video 360/sendBW in kbit/s |
| 1584 | WebRtc_Word32 sendBitrateKbit = _cbRtcpPrivate.BitrateSent()/1000; |
| 1585 | if(sendBitrateKbit != 0) |
| 1586 | { |
| 1587 | minIntervalMs = 360000/sendBitrateKbit; |
| 1588 | } |
| 1589 | } |
| 1590 | if(minIntervalMs > RTCP_INTERVAL_VIDEO_MS) |
| 1591 | { |
| 1592 | minIntervalMs = RTCP_INTERVAL_VIDEO_MS; |
| 1593 | } |
| 1594 | timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000); |
| 1595 | } |
| 1596 | _nextTimeToSendRTCP = ModuleRTPUtility::GetTimeInMS() + timeToNext; |
| 1597 | } |
| 1598 | |
| 1599 | // if the data does not fitt in the packet we fill it as much as possible |
| 1600 | WebRtc_Word32 buildVal = 0; |
| 1601 | |
| 1602 | if(rtcpPacketTypeFlags & kRtcpSr) |
| 1603 | { |
| 1604 | if(hasReceived) |
| 1605 | { |
| 1606 | buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac, &received); |
| 1607 | } else |
| 1608 | { |
| 1609 | buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac); |
| 1610 | } |
| 1611 | if(buildVal == -1) |
| 1612 | { |
| 1613 | return -1; // error |
| 1614 | |
| 1615 | }else if(buildVal == -2) |
| 1616 | { |
| 1617 | break; // out of buffer |
| 1618 | } |
| 1619 | buildVal = BuildSDEC(rtcpbuffer, pos); |
| 1620 | if(buildVal == -1) |
| 1621 | { |
| 1622 | return -1; // error |
| 1623 | |
| 1624 | }else if(buildVal == -2) |
| 1625 | { |
| 1626 | break; // out of buffer |
| 1627 | } |
| 1628 | |
| 1629 | }else if(rtcpPacketTypeFlags & kRtcpRr) |
| 1630 | { |
| 1631 | if(hasReceived) |
| 1632 | { |
| 1633 | buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac,&received); |
| 1634 | }else |
| 1635 | { |
| 1636 | buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac); |
| 1637 | } |
| 1638 | if(buildVal == -1) |
| 1639 | { |
| 1640 | return -1; // error |
| 1641 | |
| 1642 | }else if(buildVal == -2) |
| 1643 | { |
| 1644 | break; // out of buffer |
| 1645 | } |
| 1646 | // only of set |
| 1647 | if(_CNAME[0] != 0) |
| 1648 | { |
| 1649 | buildVal = BuildSDEC(rtcpbuffer, pos); |
| 1650 | if(buildVal == -1) |
| 1651 | { |
| 1652 | return -1; // error |
| 1653 | } |
| 1654 | } |
| 1655 | } |
| 1656 | if(rtcpPacketTypeFlags & kRtcpPli) |
| 1657 | { |
| 1658 | buildVal = BuildPLI(rtcpbuffer, pos); |
| 1659 | if(buildVal == -1) |
| 1660 | { |
| 1661 | return -1; // error |
| 1662 | |
| 1663 | }else if(buildVal == -2) |
| 1664 | { |
| 1665 | break; // out of buffer |
| 1666 | } |
| 1667 | } |
| 1668 | if(rtcpPacketTypeFlags & kRtcpFir) |
| 1669 | { |
| 1670 | buildVal = BuildFIR(rtcpbuffer, pos, RTT); |
| 1671 | if(buildVal == -1) |
| 1672 | { |
| 1673 | return -1; // error |
| 1674 | |
| 1675 | }else if(buildVal == -2) |
| 1676 | { |
| 1677 | break; // out of buffer |
| 1678 | } |
| 1679 | } |
| 1680 | if(rtcpPacketTypeFlags & kRtcpSli) |
| 1681 | { |
| 1682 | buildVal = BuildSLI(rtcpbuffer, pos, (WebRtc_UWord8)pictureID); |
| 1683 | if(buildVal == -1) |
| 1684 | { |
| 1685 | return -1; // error |
| 1686 | |
| 1687 | }else if(buildVal == -2) |
| 1688 | { |
| 1689 | break; // out of buffer |
| 1690 | } |
| 1691 | } |
| 1692 | if(rtcpPacketTypeFlags & kRtcpRpsi) |
| 1693 | { |
| 1694 | const WebRtc_Word8 payloadType = _cbRtcpPrivate.SendPayloadType(); |
| 1695 | if(payloadType == -1) |
| 1696 | { |
| 1697 | return -1; |
| 1698 | } |
| 1699 | buildVal = BuildRPSI(rtcpbuffer, pos, pictureID, (WebRtc_UWord8)payloadType); |
| 1700 | if(buildVal == -1) |
| 1701 | { |
| 1702 | return -1; // error |
| 1703 | |
| 1704 | }else if(buildVal == -2) |
| 1705 | { |
| 1706 | break; // out of buffer |
| 1707 | } |
| 1708 | } |
| 1709 | if(rtcpPacketTypeFlags & kRtcpBye) |
| 1710 | { |
| 1711 | buildVal = BuildBYE(rtcpbuffer, pos); |
| 1712 | if(buildVal == -1) |
| 1713 | { |
| 1714 | return -1; // error |
| 1715 | |
| 1716 | }else if(buildVal == -2) |
| 1717 | { |
| 1718 | break; // out of buffer |
| 1719 | } |
| 1720 | } |
| 1721 | if(rtcpPacketTypeFlags & kRtcpApp) |
| 1722 | { |
| 1723 | buildVal = BuildAPP(rtcpbuffer, pos); |
| 1724 | if(buildVal == -1) |
| 1725 | { |
| 1726 | return -1; // error |
| 1727 | |
| 1728 | }else if(buildVal == -2) |
| 1729 | { |
| 1730 | break; // out of buffer |
| 1731 | } |
| 1732 | } |
| 1733 | if(rtcpPacketTypeFlags & kRtcpTmmbr) |
| 1734 | { |
| 1735 | buildVal = BuildTMMBR(rtcpbuffer, pos, RTT); |
| 1736 | if(buildVal == -1) |
| 1737 | { |
| 1738 | return -1; // error |
| 1739 | |
| 1740 | }else if(buildVal == -2) |
| 1741 | { |
| 1742 | break; // out of buffer |
| 1743 | } |
| 1744 | } |
| 1745 | if(rtcpPacketTypeFlags & kRtcpTmmbn) |
| 1746 | { |
| 1747 | buildVal = BuildTMMBN(rtcpbuffer, pos); |
| 1748 | if(buildVal == -1) |
| 1749 | { |
| 1750 | return -1; // error |
| 1751 | |
| 1752 | }else if(buildVal == -2) |
| 1753 | { |
| 1754 | break; // out of buffer |
| 1755 | } |
| 1756 | } |
| 1757 | if(rtcpPacketTypeFlags & kRtcpNack) |
| 1758 | { |
| 1759 | buildVal = BuildNACK(rtcpbuffer, pos, nackSize, nackList); |
| 1760 | if(buildVal == -1) |
| 1761 | { |
| 1762 | return -1; // error |
| 1763 | |
| 1764 | }else if(buildVal == -2) |
| 1765 | { |
| 1766 | break; // out of buffer |
| 1767 | } |
| 1768 | } |
| 1769 | if(rtcpPacketTypeFlags & kRtcpXrVoipMetric) |
| 1770 | { |
| 1771 | buildVal = BuildVoIPMetric(rtcpbuffer, pos); |
| 1772 | if(buildVal == -1) |
| 1773 | { |
| 1774 | return -1; // error |
| 1775 | |
| 1776 | }else if(buildVal == -2) |
| 1777 | { |
| 1778 | break; // out of buffer |
| 1779 | } |
| 1780 | } |
| 1781 | }while (false); |
| 1782 | |
| 1783 | return SendToNetwork(rtcpbuffer, (WebRtc_UWord16)pos); |
| 1784 | } |
| 1785 | |
| 1786 | WebRtc_Word32 |
| 1787 | RTCPSender::SendToNetwork(const WebRtc_UWord8* dataBuffer, |
| 1788 | const WebRtc_UWord16 length) |
| 1789 | { |
| 1790 | CriticalSectionScoped lock(_criticalSectionTransport); |
| 1791 | if(_cbTransport) |
| 1792 | { |
| 1793 | if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0) |
| 1794 | { |
| 1795 | return 0; |
| 1796 | } |
| 1797 | } |
| 1798 | return -1; |
| 1799 | } |
| 1800 | |
| 1801 | WebRtc_Word32 |
| 1802 | RTCPSender::SetCSRCStatus(const bool include) |
| 1803 | { |
| 1804 | _includeCSRCs = include; |
| 1805 | return 0; |
| 1806 | } |
| 1807 | |
| 1808 | WebRtc_Word32 |
| 1809 | RTCPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], |
| 1810 | const WebRtc_UWord8 arrLength) |
| 1811 | { |
| 1812 | if(arrLength > kRtpCsrcSize) |
| 1813 | { |
| 1814 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1815 | assert(false); |
| 1816 | return -1; |
| 1817 | } |
| 1818 | |
| 1819 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1820 | |
| 1821 | for(int i = 0; i < arrLength;i++) |
| 1822 | { |
| 1823 | _CSRC[i] = arrOfCSRC[i]; |
| 1824 | } |
| 1825 | _CSRCs = arrLength; |
| 1826 | return 0; |
| 1827 | } |
| 1828 | |
| 1829 | WebRtc_Word32 |
| 1830 | RTCPSender::SetApplicationSpecificData(const WebRtc_UWord8 subType, |
| 1831 | const WebRtc_UWord32 name, |
| 1832 | const WebRtc_UWord8* data, |
| 1833 | const WebRtc_UWord16 length) |
| 1834 | { |
| 1835 | if(length %4 != 0) |
| 1836 | { |
| 1837 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1838 | return -1; |
| 1839 | } |
| 1840 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1841 | |
| 1842 | if(_appData) |
| 1843 | { |
| 1844 | delete [] _appData; |
| 1845 | } |
| 1846 | |
| 1847 | _appSend = true; |
| 1848 | _appSubType = subType; |
| 1849 | _appName = name; |
| 1850 | _appData = new WebRtc_UWord8[length]; |
| 1851 | _appLength = length; |
| 1852 | memcpy(_appData, data, length); |
| 1853 | return 0; |
| 1854 | } |
| 1855 | |
| 1856 | WebRtc_Word32 |
| 1857 | RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) |
| 1858 | { |
| 1859 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1860 | memcpy(&_xrVoIPMetric, VoIPMetric, sizeof(RTCPVoIPMetric)); |
| 1861 | |
| 1862 | _xrSendVoIPMetric = true; |
| 1863 | return 0; |
| 1864 | } |
| 1865 | |
| 1866 | // called under critsect _criticalSectionRTCPSender |
| 1867 | WebRtc_Word32 |
| 1868 | RTCPSender::AddReportBlocks(WebRtc_UWord8* rtcpbuffer, |
| 1869 | WebRtc_UWord32& pos, |
| 1870 | WebRtc_UWord8& numberOfReportBlocks, |
| 1871 | const RTCPReportBlock* received, |
| 1872 | const WebRtc_UWord32 NTPsec, |
| 1873 | const WebRtc_UWord32 NTPfrac) |
| 1874 | { |
| 1875 | // sanity one block |
| 1876 | if(pos + 24 >= IP_PACKET_SIZE) |
| 1877 | { |
| 1878 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1879 | return -1; |
| 1880 | } |
| 1881 | |
| 1882 | numberOfReportBlocks = _reportBlocks.Size(); |
| 1883 | if(received) |
| 1884 | { |
| 1885 | // add our multiple RR to numberOfReportBlocks |
| 1886 | numberOfReportBlocks++; |
| 1887 | } |
| 1888 | |
| 1889 | if(received) |
| 1890 | { |
| 1891 | // answer to the one that sends to me |
| 1892 | _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); |
| 1893 | |
| 1894 | // Remote SSRC |
| 1895 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| 1896 | pos += 4; |
| 1897 | |
| 1898 | // fraction lost |
| 1899 | rtcpbuffer[pos++]=received->fractionLost; |
| 1900 | |
| 1901 | // cumulative loss |
| 1902 | ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos, received->cumulativeLost); |
| 1903 | pos += 3; |
| 1904 | |
| 1905 | // extended highest seq_no, contain the highest sequence number received |
| 1906 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->extendedHighSeqNum); |
| 1907 | pos += 4; |
| 1908 | |
| 1909 | //Jitter |
| 1910 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->jitter); |
| 1911 | pos += 4; |
| 1912 | |
| 1913 | // Last SR timestamp, our NTP time when we received the last report |
| 1914 | // This is the value that we read from the send report packet not when we received it... |
| 1915 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->lastSR); |
| 1916 | pos += 4; |
| 1917 | |
| 1918 | // Delay since last received report,time since we received the report |
| 1919 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->delaySinceLastSR); |
| 1920 | pos += 4; |
| 1921 | } |
| 1922 | |
| 1923 | if(pos + _reportBlocks.Size()*24 >= IP_PACKET_SIZE) |
| 1924 | { |
| 1925 | WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| 1926 | return -1; |
| 1927 | } |
| 1928 | |
| 1929 | MapItem* item = _reportBlocks.First(); |
| 1930 | for(int i = 0; i < _reportBlocks.Size() && item; i++) |
| 1931 | { |
| 1932 | // we can have multiple report block in a conference |
| 1933 | WebRtc_UWord32 remoteSSRC = item->GetId(); |
| 1934 | RTCPReportBlock* reportBlock = (RTCPReportBlock*)item->GetItem(); |
| 1935 | if(reportBlock) |
| 1936 | { |
| 1937 | // Remote SSRC |
| 1938 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, remoteSSRC); |
| 1939 | pos += 4; |
| 1940 | |
| 1941 | // fraction lost |
| 1942 | rtcpbuffer[pos++]=(WebRtc_UWord8)(reportBlock->fractionLost); |
| 1943 | |
| 1944 | // cumulative loss |
| 1945 | ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos, reportBlock->cumulativeLost); |
| 1946 | pos += 3; |
| 1947 | |
| 1948 | // extended highest seq_no, contain the highest sequence number received |
| 1949 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, reportBlock->extendedHighSeqNum); |
| 1950 | pos += 4; |
| 1951 | |
| 1952 | //Jitter |
| 1953 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, reportBlock->jitter); |
| 1954 | pos += 4; |
| 1955 | |
| 1956 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, reportBlock->lastSR); |
| 1957 | pos += 4; |
| 1958 | |
| 1959 | ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, reportBlock->delaySinceLastSR); |
| 1960 | pos += 4; |
| 1961 | } |
| 1962 | item = _reportBlocks.Next(item); |
| 1963 | } |
| 1964 | return pos; |
| 1965 | } |
| 1966 | |
| 1967 | // no callbacks allowed inside this function |
| 1968 | WebRtc_Word32 |
| 1969 | RTCPSender::SetTMMBN(const TMMBRSet* boundingSet, |
| 1970 | const WebRtc_UWord32 maxBitrateKbit) |
| 1971 | { |
| 1972 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1973 | |
| 1974 | if (0 == _tmmbrHelp.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit)) |
| 1975 | { |
| 1976 | _sendTMMBN = true; |
| 1977 | return 0; |
| 1978 | } |
| 1979 | return -1; |
| 1980 | } |
| 1981 | |
| 1982 | WebRtc_Word32 |
| 1983 | RTCPSender::RequestTMMBR(WebRtc_UWord32 estimatedBW, WebRtc_UWord32 packetOH) |
| 1984 | { |
| 1985 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 1986 | if(_TMMBR) |
| 1987 | { |
| 1988 | _tmmbr_Send = estimatedBW; |
| 1989 | _packetOH_Send = packetOH; |
| 1990 | |
| 1991 | return 0; |
| 1992 | } |
| 1993 | return -1; |
| 1994 | } |
| 1995 | |
| 1996 | RateControlRegion |
| 1997 | RTCPSender::UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse) |
| 1998 | { |
| 1999 | CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| 2000 | return _remoteRateControl.Update(rateControlInput, firstOverUse); |
| 2001 | } |
| 2002 | } // namespace webrtc |