libmpdemux/demux_rtp.cpp

   1 ////////// Routines (with C-linkage) that interface between "MPlayer"
   2 ////////// and the "LIVE555 Streaming Media" libraries:
   3
   4 extern "C" {
   5 // on MinGW, we must include windows.h before the things it conflicts
   6 #ifdef __MINGW32__    // with.  they are each protected from
   7 #include <windows.h>  // windows.h, but not the other way around.
   8 #endif
   9 #include "demux_rtp.h"
  10 #include "stheader.h"
  11 }
  12 #include "demux_rtp_internal.h"
  13
  14 #include "BasicUsageEnvironment.hh"
  15 #include "liveMedia.hh"
  16 #include "GroupsockHelper.hh"
  17 #include <unistd.h>
  18
  19 // A data structure representing input data for each stream:
  20 class ReadBufferQueue {
  21 public:
  22   ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
  23                   char const* tag);
  24   virtual ~ReadBufferQueue();
  25
  26   FramedSource* readSource() const { return fReadSource; }
  27   RTPSource* rtpSource() const { return fRTPSource; }
  28   demuxer_t* ourDemuxer() const { return fOurDemuxer; }
  29   char const* tag() const { return fTag; }
  30
  31   char blockingFlag; // used to implement synchronous reads
  32
  33   // For A/V synchronization:
  34   Boolean prevPacketWasSynchronized;
  35   float prevPacketPTS;
  36   ReadBufferQueue** otherQueue;
  37
  38   // The 'queue' actually consists of just a single "demux_packet_t"
  39   // (because the underlying OS does the actual queueing/buffering):
  40   demux_packet_t* dp;
  41
  42   // However, we sometimes inspect buffers before delivering them.
  43   // For this, we maintain a queue of pending buffers:
  44   void savePendingBuffer(demux_packet_t* dp);
  45   demux_packet_t* getPendingBuffer();
  46
  47   // For H264 over rtsp using AVParser, the next packet has to be saved
  48   demux_packet_t* nextpacket;
  49
  50 private:
  51   demux_packet_t* pendingDPHead;
  52   demux_packet_t* pendingDPTail;
  53
  54   FramedSource* fReadSource;
  55   RTPSource* fRTPSource;
  56   demuxer_t* fOurDemuxer;
  57   char const* fTag; // used for debugging
  58 };
  59
  60 // A structure of RTP-specific state, kept so that we can cleanly
  61 // reclaim it:
  62 typedef struct RTPState {
  63   char const* sdpDescription;
  64   RTSPClient* rtspClient;
  65   SIPClient* sipClient;
  66   MediaSession* mediaSession;
  67   ReadBufferQueue* audioBufferQueue;
  68   ReadBufferQueue* videoBufferQueue;
  69   unsigned flags;
  70   struct timeval firstSyncTime;
  71 };
  72
  73 extern "C" char* network_username;
  74 extern "C" char* network_password;
  75 static char* openURL_rtsp(RTSPClient* client, char const* url) {
  76   // If we were given a user name (and optional password), then use them:
  77   if (network_username != NULL) {
  78     char const* password = network_password == NULL ? "" : network_password;
  79     return client->describeWithPassword(url, network_username, password);
  80   } else {
  81     return client->describeURL(url);
  82   }
  83 }
  84
  85 static char* openURL_sip(SIPClient* client, char const* url) {
  86   // If we were given a user name (and optional password), then use them:
  87   if (network_username != NULL) {
  88     char const* password = network_password == NULL ? "" : network_password;
  89     return client->inviteWithPassword(url, network_username, password);
  90   } else {
  91     return client->invite(url);
  92   }
  93 }
  94
  95 int rtspStreamOverTCP = 0;
  96 extern int rtsp_port;
  97
  98 extern "C" int audio_id, video_id, dvdsub_id;
  99 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
 100   Boolean success = False;
 101   do {
 102     TaskScheduler* scheduler = BasicTaskScheduler::createNew();
 103     if (scheduler == NULL) break;
 104     UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
 105     if (env == NULL) break;
 106
 107     RTSPClient* rtspClient = NULL;
 108     SIPClient* sipClient = NULL;
 109
 110     if (demuxer == NULL || demuxer->stream == NULL) break;  // shouldn't happen
 111     demuxer->stream->eof = 0; // just in case
 112
 113     // Look at the stream's 'priv' field to see if we were initiated
 114     // via a SDP description:
 115     char* sdpDescription = (char*)(demuxer->stream->priv);
 116     if (sdpDescription == NULL) {
 117       // We weren't given a SDP description directly, so assume that
 118       // we were given a RTSP or SIP URL:
 119       char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
 120       char const* url = demuxer->stream->streaming_ctrl->url->url;
 121       extern int verbose;
 122       if (strcmp(protocol, "rtsp") == 0) {
 123         rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
 124         if (rtspClient == NULL) {
 125           fprintf(stderr, "Failed to create RTSP client: %s\n",
 126                   env->getResultMsg());
 127           break;
 128         }
 129         sdpDescription = openURL_rtsp(rtspClient, url);
 130       } else { // SIP
 131         unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
 132         sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
 133                                          verbose, "MPlayer");
 134         if (sipClient == NULL) {
 135           fprintf(stderr, "Failed to create SIP client: %s\n",
 136                   env->getResultMsg());
 137           break;
 138         }
 139         sipClient->setClientStartPortNum(8000);
 140         sdpDescription = openURL_sip(sipClient, url);
 141       }
 142
 143       if (sdpDescription == NULL) {
 144         fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
 145                 url, env->getResultMsg());
 146         break;
 147       }
 148     }
 149
 150     // Now that we have a SDP description, create a MediaSession from it:
 151     MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
 152     if (mediaSession == NULL) break;
 153
 154
 155     // Create a 'RTPState' structure containing the state that we just created,
 156     // and store it in the demuxer's 'priv' field, for future reference:
 157     RTPState* rtpState = new RTPState;
 158     rtpState->sdpDescription = sdpDescription;
 159     rtpState->rtspClient = rtspClient;
 160     rtpState->sipClient = sipClient;
 161     rtpState->mediaSession = mediaSession;
 162     rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
 163     rtpState->flags = 0;
 164     rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
 165     demuxer->priv = rtpState;
 166
 167     int audiofound = 0, videofound = 0;
 168     // Create RTP receivers (sources) for each subsession:
 169     MediaSubsessionIterator iter(*mediaSession);
 170     MediaSubsession* subsession;
 171     unsigned desiredReceiveBufferSize;
 172     while ((subsession = iter.next()) != NULL) {
 173       // Ignore any subsession that's not audio or video:
 174       if (strcmp(subsession->mediumName(), "audio") == 0) {
 175         if (audiofound) {
 176           fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName());
 177           continue;
 178         }
 179         desiredReceiveBufferSize = 100000;
 180       } else if (strcmp(subsession->mediumName(), "video") == 0) {
 181         if (videofound) {
 182           fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName());
 183           continue;
 184         }
 185         desiredReceiveBufferSize = 2000000;
 186       } else {
 187         continue;
 188       }
 189
 190       if (rtsp_port)
 191           subsession->setClientPortNum (rtsp_port);
 192
 193       if (!subsession->initiate()) {
 194         fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
 195       } else {
 196         fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum());
 197
 198         // Set the OS's socket receive buffer sufficiently large to avoid
 199         // incoming packets getting dropped between successive reads from this
 200         // subsession's demuxer.  Depending on the bitrate(s) that you expect,
 201         // you may wish to tweak the "desiredReceiveBufferSize" values above.
 202         int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
 203         int receiveBufferSize
 204           = increaseReceiveBufferTo(*env, rtpSocketNum,
 205                                     desiredReceiveBufferSize);
 206         if (verbose > 0) {
 207           fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
 208                   subsession->mediumName(), receiveBufferSize);
 209         }
 210
 211         if (rtspClient != NULL) {
 212           // Issue a RTSP "SETUP" command on the chosen subsession:
 213           if (!rtspClient->setupMediaSubsession(*subsession, False,
 214                                                 rtspStreamOverTCP)) break;
 215           if (!strcmp(subsession->mediumName(), "audio"))
 216             audiofound = 1;
 217           if (!strcmp(subsession->mediumName(), "video"))
 218             videofound = 1;
 219         }
 220       }
 221     }
 222
 223     if (rtspClient != NULL) {
 224       // Issue a RTSP aggregate "PLAY" command on the whole session:
 225       if (!rtspClient->playMediaSession(*mediaSession)) break;
 226     } else if (sipClient != NULL) {
 227       sipClient->sendACK(); // to start the stream flowing
 228     }
 229
 230     // Now that the session is ready to be read, do additional
 231     // MPlayer codec-specific initialization on each subsession:
 232     iter.reset();
 233     while ((subsession = iter.next()) != NULL) {
 234       if (subsession->readSource() == NULL) continue; // not reading this
 235
 236       unsigned flags = 0;
 237       if (strcmp(subsession->mediumName(), "audio") == 0) {
 238         rtpState->audioBufferQueue
 239           = new ReadBufferQueue(subsession, demuxer, "audio");
 240         rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
 241         rtpCodecInitialize_audio(demuxer, subsession, flags);
 242       } else if (strcmp(subsession->mediumName(), "video") == 0) {
 243         rtpState->videoBufferQueue
 244           = new ReadBufferQueue(subsession, demuxer, "video");
 245         rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
 246         rtpCodecInitialize_video(demuxer, subsession, flags);
 247       }
 248       rtpState->flags |= flags;
 249     }
 250     success = True;
 251   } while (0);
 252   if (!success) return NULL; // an error occurred
 253
 254   // Hack: If audio and video are demuxed together on a single RTP stream,
 255   // then create a new "demuxer_t" structure to allow the higher-level
 256   // code to recognize this:
 257   if (demux_is_multiplexed_rtp_stream(demuxer)) {
 258     stream_t* s = new_ds_stream(demuxer->video);
 259     demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN,
 260                                audio_id, video_id, dvdsub_id, NULL);
 261     demuxer = new_demuxers_demuxer(od, od, od);
 262   }
 263
 264   return demuxer;
 265 }
 266
 267 extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
 268   // Get the RTP state that was stored in the demuxer's 'priv' field:
 269   RTPState* rtpState = (RTPState*)(demuxer->priv);
 270
 271   return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
 272 }
 273
 274 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) {
 275   // Get the RTP state that was stored in the demuxer's 'priv' field:
 276   RTPState* rtpState = (RTPState*)(demuxer->priv);
 277
 278   return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0;
 279 }
 280
 281 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
 282                                  Boolean mustGetNewData,
 283                                  float& ptsBehind); // forward
 284
 285 extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
 286   // Get a filled-in "demux_packet" from the RTP source, and deliver it.
 287   // Note that this is called as a synchronous read operation, so it needs
 288   // to block in the (hopefully infrequent) case where no packet is
 289   // immediately available.
 290
 291   while (1) {
 292     float ptsBehind;
 293     demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
 294     if (dp == NULL) return 0;
 295
 296     if (demuxer->stream->eof) return 0; // source stream has closed down
 297
 298     // Before using this packet, check to make sure that its presentation
 299     // time is not far behind the other stream (if any).  If it is,
 300     // then we discard this packet, and get another instead.  (The rest of
 301     // MPlayer doesn't always do a good job of synchronizing when the
 302     // audio and video streams get this far apart.)
 303     // (We don't do this when streaming over TCP, because then the audio and
 304     // video streams are interleaved.)
 305     // (Also, if the stream is *excessively* far behind, then we allow
 306     // the packet, because in this case it probably means that there was
 307     // an error in the source's timestamp synchronization.)
 308     const float ptsBehindThreshold = 1.0; // seconds
 309     const float ptsBehindLimit = 60.0; // seconds
 310     if (ptsBehind < ptsBehindThreshold ||
 311         ptsBehind > ptsBehindLimit ||
 312         rtspStreamOverTCP) { // packet's OK
 313       ds_add_packet(ds, dp);
 314       break;
 315     }
 316
 317 #ifdef DEBUG_PRINT_DISCARDED_PACKETS
 318     RTPState* rtpState = (RTPState*)(demuxer->priv);
 319     ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue;
 320     fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind);
 321 #endif
 322     free_demux_packet(dp); // give back this packet, and get another one
 323   }
 324
 325   return 1;
 326 }
 327
 328 Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds,
 329                        unsigned char*& packetData, unsigned& packetDataLen,
 330                        float& pts) {
 331   // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
 332   // is not delivered to the "demux_stream".
 333   float ptsBehind;
 334   demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking
 335   if (dp == NULL) return False;
 336
 337   packetData = dp->buffer;
 338   packetDataLen = dp->len;
 339   pts = dp->pts;
 340
 341   return True;
 342 }
 343
 344 static void teardownRTSPorSIPSession(RTPState* rtpState); // forward
 345
 346 extern "C" void demux_close_rtp(demuxer_t* demuxer) {
 347   // Reclaim all RTP-related state:
 348
 349   // Get the RTP state that was stored in the demuxer's 'priv' field:
 350   RTPState* rtpState = (RTPState*)(demuxer->priv);
 351   if (rtpState == NULL) return;
 352
 353   teardownRTSPorSIPSession(rtpState);
 354
 355   UsageEnvironment* env = NULL;
 356   TaskScheduler* scheduler = NULL;
 357   if (rtpState->mediaSession != NULL) {
 358     env = &(rtpState->mediaSession->envir());
 359     scheduler = &(env->taskScheduler());
 360   }
 361   Medium::close(rtpState->mediaSession);
 362   Medium::close(rtpState->rtspClient);
 363   Medium::close(rtpState->sipClient);
 364   delete rtpState->audioBufferQueue;
 365   delete rtpState->videoBufferQueue;
 366   delete rtpState->sdpDescription;
 367   delete rtpState;
 368
 369   env->reclaim(); delete scheduler;
 370 }
 371
 372 ////////// Extra routines that help implement the above interface functions:
 373
 374 #define MAX_RTP_FRAME_SIZE 5000000
 375     // >= the largest conceivable frame composed from one or more RTP packets
 376
 377 static void afterReading(void* clientData, unsigned frameSize,
 378                          unsigned /*numTruncatedBytes*/,
 379                          struct timeval presentationTime,
 380                          unsigned /*durationInMicroseconds*/) {
 381   int headersize = 0;
 382   if (frameSize >= MAX_RTP_FRAME_SIZE) {
 383     fprintf(stderr, "Saw an input frame too large (>=%d).  Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
 384             MAX_RTP_FRAME_SIZE);
 385   }
 386   ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
 387   demuxer_t* demuxer = bufferQueue->ourDemuxer();
 388   RTPState* rtpState = (RTPState*)(demuxer->priv);
 389
 390   if (frameSize > 0) demuxer->stream->eof = 0;
 391
 392   demux_packet_t* dp = bufferQueue->dp;
 393
 394   if (bufferQueue->readSource()->isAMRAudioSource())
 395     headersize = 1;
 396   else if (bufferQueue == rtpState->videoBufferQueue &&
 397       ((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) {
 398     dp->buffer[0]=0x00;
 399     dp->buffer[1]=0x00;
 400     dp->buffer[2]=0x01;
 401     headersize = 3;
 402   }
 403
 404   resize_demux_packet(dp, frameSize + headersize);
 405
 406   // Set the packet's presentation time stamp, depending on whether or
 407   // not our RTP source's timestamps have been synchronized yet:
 408   Boolean hasBeenSynchronized
 409     = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
 410   if (hasBeenSynchronized) {
 411     if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
 412       fprintf(stderr, "%s stream has been synchronized using RTCP \n",
 413               bufferQueue->tag());
 414     }
 415
 416     struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
 417     if (fst->tv_sec == 0 && fst->tv_usec == 0) {
 418       *fst = presentationTime;
 419     }
 420
 421     // For the "pts" field, use the time differential from the first
 422     // synchronized time, rather than absolute time, in order to avoid
 423     // round-off errors when converting to a float:
 424     dp->pts = presentationTime.tv_sec - fst->tv_sec
 425       + (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
 426     bufferQueue->prevPacketPTS = dp->pts;
 427   } else {
 428     if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) {
 429       fprintf(stderr, "%s stream is no longer RTCP-synchronized \n",
 430               bufferQueue->tag());
 431     }
 432
 433     // use the previous packet's "pts" once again:
 434     dp->pts = bufferQueue->prevPacketPTS;
 435   }
 436   bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized;
 437
 438   dp->pos = demuxer->filepos;
 439   demuxer->filepos += frameSize + headersize;
 440
 441   // Signal any pending 'doEventLoop()' call on this queue:
 442   bufferQueue->blockingFlag = ~0;
 443 }
 444
 445 static void onSourceClosure(void* clientData) {
 446   ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
 447   demuxer_t* demuxer = bufferQueue->ourDemuxer();
 448
 449   demuxer->stream->eof = 1;
 450
 451   // Signal any pending 'doEventLoop()' call on this queue:
 452   bufferQueue->blockingFlag = ~0;
 453 }
 454
 455 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
 456                                  Boolean mustGetNewData,
 457                                  float& ptsBehind) {
 458   // Begin by finding the buffer queue that we want to read from:
 459   // (Get this from the RTP state, which we stored in
 460   //  the demuxer's 'priv' field)
 461   RTPState* rtpState = (RTPState*)(demuxer->priv);
 462   ReadBufferQueue* bufferQueue = NULL;
 463   int headersize = 0;
 464   TaskToken task;
 465
 466   if (demuxer->stream->eof) return NULL;
 467
 468   if (ds == demuxer->video) {
 469     bufferQueue = rtpState->videoBufferQueue;
 470     if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4'))
 471       headersize = 3;
 472   } else if (ds == demuxer->audio) {
 473     bufferQueue = rtpState->audioBufferQueue;
 474     if (bufferQueue->readSource()->isAMRAudioSource())
 475       headersize = 1;
 476   } else {
 477     fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n");
 478     return NULL;
 479   }
 480
 481   if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
 482     fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
 483     return NULL;
 484   }
 485
 486   demux_packet_t* dp = NULL;
 487   if (!mustGetNewData) {
 488     // Check whether we have a previously-saved buffer that we can use:
 489     dp = bufferQueue->getPendingBuffer();
 490     if (dp != NULL) {
 491       ptsBehind = 0.0; // so that we always accept this data
 492       return dp;
 493     }
 494   }
 495
 496   // Allocate a new packet buffer, and arrange to read into it:
 497     if (!bufferQueue->nextpacket) {
 498   dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
 499   bufferQueue->dp = dp;
 500   if (dp == NULL) return NULL;
 501     }
 502
 503 #ifdef CONFIG_LIBAVCODEC
 504   extern AVCodecParserContext * h264parserctx;
 505   int consumed, poutbuf_size = 1;
 506   const uint8_t *poutbuf = NULL;
 507   float lastpts = 0.0;
 508
 509   do {
 510     if (!bufferQueue->nextpacket) {
 511 #endif
 512   // Schedule the read operation:
 513   bufferQueue->blockingFlag = 0;
 514   bufferQueue->readSource()->getNextFrame(&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize,
 515                                           afterReading, bufferQueue,
 516                                           onSourceClosure, bufferQueue);
 517   // Block ourselves until data becomes available:
 518   TaskScheduler& scheduler
 519     = bufferQueue->readSource()->envir().taskScheduler();
 520   int delay = 10000000;
 521   if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime())
 522     delay /= 10;
 523   task = scheduler.scheduleDelayedTask(delay, onSourceClosure, bufferQueue);
 524   scheduler.doEventLoop(&bufferQueue->blockingFlag);
 525   scheduler.unscheduleDelayedTask(task);
 526   if (demuxer->stream->eof) {
 527     free_demux_packet(dp);
 528     return NULL;
 529   }
 530
 531   if (headersize == 1) // amr
 532     dp->buffer[0] =
 533         ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader();
 534 #ifdef CONFIG_LIBAVCODEC
 535     } else {
 536       bufferQueue->dp = dp = bufferQueue->nextpacket;
 537       bufferQueue->nextpacket = NULL;
 538     }
 539     if (headersize == 3 && h264parserctx) { // h264
 540       consumed = h264parserctx->parser->parser_parse(h264parserctx,
 541                                NULL,
 542                                &poutbuf, &poutbuf_size,
 543                                dp->buffer, dp->len);
 544
 545       if (!consumed && !poutbuf_size)
 546         return NULL;
 547
 548       if (!poutbuf_size) {
 549         lastpts=dp->pts;
 550         free_demux_packet(dp);
 551         bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
 552       } else {
 553         bufferQueue->nextpacket = dp;
 554         bufferQueue->dp = dp = new_demux_packet(poutbuf_size);
 555         memcpy(dp->buffer, poutbuf, poutbuf_size);
 556         dp->pts=lastpts;
 557       }
 558     }
 559   } while (!poutbuf_size);
 560 #endif
 561
 562   // Set the "ptsBehind" result parameter:
 563   if (bufferQueue->prevPacketPTS != 0.0
 564       && bufferQueue->prevPacketWasSynchronized
 565       && *(bufferQueue->otherQueue) != NULL
 566       && (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0
 567       && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) {
 568     ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS
 569                  - bufferQueue->prevPacketPTS;
 570   } else {
 571     ptsBehind = 0.0;
 572   }
 573
 574   if (mustGetNewData) {
 575     // Save this buffer for future reads:
 576     bufferQueue->savePendingBuffer(dp);
 577   }
 578
 579   return dp;
 580 }
 581
 582 static void teardownRTSPorSIPSession(RTPState* rtpState) {
 583   MediaSession* mediaSession = rtpState->mediaSession;
 584   if (mediaSession == NULL) return;
 585   if (rtpState->rtspClient != NULL) {
 586     rtpState->rtspClient->teardownMediaSession(*mediaSession);
 587   } else if (rtpState->sipClient != NULL) {
 588     rtpState->sipClient->sendBYE();
 589   }
 590 }
 591
 592 ////////// "ReadBuffer" and "ReadBufferQueue" implementation:
 593
 594 ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession,
 595                                  demuxer_t* demuxer, char const* tag)
 596   : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL),
 597     dp(NULL), nextpacket(NULL),
 598     pendingDPHead(NULL), pendingDPTail(NULL),
 599     fReadSource(subsession == NULL ? NULL : subsession->readSource()),
 600     fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
 601     fOurDemuxer(demuxer), fTag(strdup(tag)) {
 602 }
 603
 604 ReadBufferQueue::~ReadBufferQueue() {
 605   delete fTag;
 606
 607   // Free any pending buffers (that never got delivered):
 608   demux_packet_t* dp = pendingDPHead;
 609   while (dp != NULL) {
 610     demux_packet_t* dpNext = dp->next;
 611     dp->next = NULL;
 612     free_demux_packet(dp);
 613     dp = dpNext;
 614   }
 615 }
 616
 617 void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) {
 618   // Keep this buffer around, until MPlayer asks for it later:
 619   if (pendingDPTail == NULL) {
 620     pendingDPHead = pendingDPTail = dp;
 621   } else {
 622     pendingDPTail->next = dp;
 623     pendingDPTail = dp;
 624   }
 625   dp->next = NULL;
 626 }
 627
 628 demux_packet_t* ReadBufferQueue::getPendingBuffer() {
 629   demux_packet_t* dp = pendingDPHead;
 630   if (dp != NULL) {
 631     pendingDPHead = dp->next;
 632     if (pendingDPHead == NULL) pendingDPTail = NULL;
 633
 634     dp->next = NULL;
 635   }
 636
 637   return dp;
 638 }
 639
 640 static int demux_rtp_control(struct demuxer_st *demuxer, int cmd, void *arg) {
 641   double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime();
 642
 643   switch(cmd) {
 644     case DEMUXER_CTRL_GET_TIME_LENGTH:
 645       if (endpts <= 0)
 646         return DEMUXER_CTRL_DONTKNOW;
 647       *((double *)arg) = endpts;
 648       return DEMUXER_CTRL_OK;
 649
 650     case DEMUXER_CTRL_GET_PERCENT_POS:
 651       if (endpts <= 0)
 652         return DEMUXER_CTRL_DONTKNOW;
 653       *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts);
 654       return DEMUXER_CTRL_OK;
 655
 656     default:
 657       return DEMUXER_CTRL_NOTIMPL;
 658     }
 659 }
 660
 661 demuxer_desc_t demuxer_desc_rtp = {
 662   "LIVE555 RTP demuxer",
 663   "live555",
 664   "",
 665   "Ross Finlayson",
 666   "requires LIVE555 Streaming Media library",
 667   DEMUXER_TYPE_RTP,
 668   0, // no autodetect
 669   NULL,
 670   demux_rtp_fill_buffer,
 671   demux_open_rtp,
 672   demux_close_rtp,
 673   NULL,
 674   demux_rtp_control
 675 };