libmpdemux/demux_rtp.cpp

   1 /*
   2  * routines (with C-linkage) that interface between MPlayer
   3  * and the "LIVE555 Streaming Media" libraries
   4  *
   5  * This file is part of MPlayer.
   6  *
   7  * MPlayer is free software; you can redistribute it and/or modify
   8  * it under the terms of the GNU General Public License as published by
   9  * the Free Software Foundation; either version 2 of the License, or
  10  * (at your option) any later version.
  11  *
  12  * MPlayer is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  15  * GNU General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU General Public License along
  18  * with MPlayer; if not, write to the Free Software Foundation, Inc.,
  19  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  20  */
  21
  22 extern "C" {
  23 // on MinGW, we must include windows.h before the things it conflicts
  24 #ifdef __MINGW32__    // with.  they are each protected from
  25 #include <windows.h>  // windows.h, but not the other way around.
  26 #endif
  27 #include "demux_rtp.h"
  28 #include "stheader.h"
  29 }
  30 #include "demux_rtp_internal.h"
  31
  32 #include "BasicUsageEnvironment.hh"
  33 #include "liveMedia.hh"
  34 #include "GroupsockHelper.hh"
  35 #include <unistd.h>
  36
  37 // A data structure representing input data for each stream:
  38 class ReadBufferQueue {
  39 public:
  40   ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
  41                   char const* tag);
  42   virtual ~ReadBufferQueue();
  43
  44   FramedSource* readSource() const { return fReadSource; }
  45   RTPSource* rtpSource() const { return fRTPSource; }
  46   demuxer_t* ourDemuxer() const { return fOurDemuxer; }
  47   char const* tag() const { return fTag; }
  48
  49   char blockingFlag; // used to implement synchronous reads
  50
  51   // For A/V synchronization:
  52   Boolean prevPacketWasSynchronized;
  53   float prevPacketPTS;
  54   ReadBufferQueue** otherQueue;
  55
  56   // The 'queue' actually consists of just a single "demux_packet_t"
  57   // (because the underlying OS does the actual queueing/buffering):
  58   demux_packet_t* dp;
  59
  60   // However, we sometimes inspect buffers before delivering them.
  61   // For this, we maintain a queue of pending buffers:
  62   void savePendingBuffer(demux_packet_t* dp);
  63   demux_packet_t* getPendingBuffer();
  64
  65   // For H264 over rtsp using AVParser, the next packet has to be saved
  66   demux_packet_t* nextpacket;
  67
  68 private:
  69   demux_packet_t* pendingDPHead;
  70   demux_packet_t* pendingDPTail;
  71
  72   FramedSource* fReadSource;
  73   RTPSource* fRTPSource;
  74   demuxer_t* fOurDemuxer;
  75   char const* fTag; // used for debugging
  76 };
  77
  78 // A structure of RTP-specific state, kept so that we can cleanly
  79 // reclaim it:
  80 typedef struct RTPState {
  81   char const* sdpDescription;
  82   RTSPClient* rtspClient;
  83   SIPClient* sipClient;
  84   MediaSession* mediaSession;
  85   ReadBufferQueue* audioBufferQueue;
  86   ReadBufferQueue* videoBufferQueue;
  87   unsigned flags;
  88   struct timeval firstSyncTime;
  89 };
  90
  91 extern "C" char* network_username;
  92 extern "C" char* network_password;
  93 static char* openURL_rtsp(RTSPClient* client, char const* url) {
  94   // If we were given a user name (and optional password), then use them:
  95   if (network_username != NULL) {
  96     char const* password = network_password == NULL ? "" : network_password;
  97     return client->describeWithPassword(url, network_username, password);
  98   } else {
  99     return client->describeURL(url);
 100   }
 101 }
 102
 103 static char* openURL_sip(SIPClient* client, char const* url) {
 104   // If we were given a user name (and optional password), then use them:
 105   if (network_username != NULL) {
 106     char const* password = network_password == NULL ? "" : network_password;
 107     return client->inviteWithPassword(url, network_username, password);
 108   } else {
 109     return client->invite(url);
 110   }
 111 }
 112
 113 #ifdef CONFIG_LIBNEMESI
 114 extern int rtsp_transport_tcp;
 115 #else
 116 int rtsp_transport_tcp = 0;
 117 #endif
 118
 119 extern int rtsp_port;
 120
 121 extern "C" int audio_id, video_id, dvdsub_id;
 122 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
 123   Boolean success = False;
 124   do {
 125     TaskScheduler* scheduler = BasicTaskScheduler::createNew();
 126     if (scheduler == NULL) break;
 127     UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
 128     if (env == NULL) break;
 129
 130     RTSPClient* rtspClient = NULL;
 131     SIPClient* sipClient = NULL;
 132
 133     if (demuxer == NULL || demuxer->stream == NULL) break;  // shouldn't happen
 134     demuxer->stream->eof = 0; // just in case
 135
 136     // Look at the stream's 'priv' field to see if we were initiated
 137     // via a SDP description:
 138     char* sdpDescription = (char*)(demuxer->stream->priv);
 139     if (sdpDescription == NULL) {
 140       // We weren't given a SDP description directly, so assume that
 141       // we were given a RTSP or SIP URL:
 142       char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
 143       char const* url = demuxer->stream->streaming_ctrl->url->url;
 144       extern int verbose;
 145       if (strcmp(protocol, "rtsp") == 0) {
 146         rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
 147         if (rtspClient == NULL) {
 148           fprintf(stderr, "Failed to create RTSP client: %s\n",
 149                   env->getResultMsg());
 150           break;
 151         }
 152         sdpDescription = openURL_rtsp(rtspClient, url);
 153       } else { // SIP
 154         unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
 155         sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
 156                                          verbose, "MPlayer");
 157         if (sipClient == NULL) {
 158           fprintf(stderr, "Failed to create SIP client: %s\n",
 159                   env->getResultMsg());
 160           break;
 161         }
 162         sipClient->setClientStartPortNum(8000);
 163         sdpDescription = openURL_sip(sipClient, url);
 164       }
 165
 166       if (sdpDescription == NULL) {
 167         fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
 168                 url, env->getResultMsg());
 169         break;
 170       }
 171     }
 172
 173     // Now that we have a SDP description, create a MediaSession from it:
 174     MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
 175     if (mediaSession == NULL) break;
 176
 177
 178     // Create a 'RTPState' structure containing the state that we just created,
 179     // and store it in the demuxer's 'priv' field, for future reference:
 180     RTPState* rtpState = new RTPState;
 181     rtpState->sdpDescription = sdpDescription;
 182     rtpState->rtspClient = rtspClient;
 183     rtpState->sipClient = sipClient;
 184     rtpState->mediaSession = mediaSession;
 185     rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
 186     rtpState->flags = 0;
 187     rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
 188     demuxer->priv = rtpState;
 189
 190     int audiofound = 0, videofound = 0;
 191     // Create RTP receivers (sources) for each subsession:
 192     MediaSubsessionIterator iter(*mediaSession);
 193     MediaSubsession* subsession;
 194     unsigned desiredReceiveBufferSize;
 195     while ((subsession = iter.next()) != NULL) {
 196       // Ignore any subsession that's not audio or video:
 197       if (strcmp(subsession->mediumName(), "audio") == 0) {
 198         if (audiofound) {
 199           fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName());
 200           continue;
 201         }
 202         desiredReceiveBufferSize = 100000;
 203       } else if (strcmp(subsession->mediumName(), "video") == 0) {
 204         if (videofound) {
 205           fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName());
 206           continue;
 207         }
 208         desiredReceiveBufferSize = 2000000;
 209       } else {
 210         continue;
 211       }
 212
 213       if (rtsp_port)
 214           subsession->setClientPortNum (rtsp_port);
 215
 216       if (!subsession->initiate()) {
 217         fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
 218       } else {
 219         fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum());
 220
 221         // Set the OS's socket receive buffer sufficiently large to avoid
 222         // incoming packets getting dropped between successive reads from this
 223         // subsession's demuxer.  Depending on the bitrate(s) that you expect,
 224         // you may wish to tweak the "desiredReceiveBufferSize" values above.
 225         int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
 226         int receiveBufferSize
 227           = increaseReceiveBufferTo(*env, rtpSocketNum,
 228                                     desiredReceiveBufferSize);
 229         if (verbose > 0) {
 230           fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
 231                   subsession->mediumName(), receiveBufferSize);
 232         }
 233
 234         if (rtspClient != NULL) {
 235           // Issue a RTSP "SETUP" command on the chosen subsession:
 236           if (!rtspClient->setupMediaSubsession(*subsession, False,
 237                                                 rtsp_transport_tcp)) break;
 238           if (!strcmp(subsession->mediumName(), "audio"))
 239             audiofound = 1;
 240           if (!strcmp(subsession->mediumName(), "video"))
 241             videofound = 1;
 242         }
 243       }
 244     }
 245
 246     if (rtspClient != NULL) {
 247       // Issue a RTSP aggregate "PLAY" command on the whole session:
 248       if (!rtspClient->playMediaSession(*mediaSession)) break;
 249     } else if (sipClient != NULL) {
 250       sipClient->sendACK(); // to start the stream flowing
 251     }
 252
 253     // Now that the session is ready to be read, do additional
 254     // MPlayer codec-specific initialization on each subsession:
 255     iter.reset();
 256     while ((subsession = iter.next()) != NULL) {
 257       if (subsession->readSource() == NULL) continue; // not reading this
 258
 259       unsigned flags = 0;
 260       if (strcmp(subsession->mediumName(), "audio") == 0) {
 261         rtpState->audioBufferQueue
 262           = new ReadBufferQueue(subsession, demuxer, "audio");
 263         rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
 264         rtpCodecInitialize_audio(demuxer, subsession, flags);
 265       } else if (strcmp(subsession->mediumName(), "video") == 0) {
 266         rtpState->videoBufferQueue
 267           = new ReadBufferQueue(subsession, demuxer, "video");
 268         rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
 269         rtpCodecInitialize_video(demuxer, subsession, flags);
 270       }
 271       rtpState->flags |= flags;
 272     }
 273     success = True;
 274   } while (0);
 275   if (!success) return NULL; // an error occurred
 276
 277   // Hack: If audio and video are demuxed together on a single RTP stream,
 278   // then create a new "demuxer_t" structure to allow the higher-level
 279   // code to recognize this:
 280   if (demux_is_multiplexed_rtp_stream(demuxer)) {
 281     stream_t* s = new_ds_stream(demuxer->video);
 282     demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN,
 283                                audio_id, video_id, dvdsub_id, NULL);
 284     demuxer = new_demuxers_demuxer(od, od, od);
 285   }
 286
 287   return demuxer;
 288 }
 289
 290 extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
 291   // Get the RTP state that was stored in the demuxer's 'priv' field:
 292   RTPState* rtpState = (RTPState*)(demuxer->priv);
 293
 294   return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
 295 }
 296
 297 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) {
 298   // Get the RTP state that was stored in the demuxer's 'priv' field:
 299   RTPState* rtpState = (RTPState*)(demuxer->priv);
 300
 301   return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0;
 302 }
 303
 304 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
 305                                  Boolean mustGetNewData,
 306                                  float& ptsBehind); // forward
 307
 308 extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
 309   // Get a filled-in "demux_packet" from the RTP source, and deliver it.
 310   // Note that this is called as a synchronous read operation, so it needs
 311   // to block in the (hopefully infrequent) case where no packet is
 312   // immediately available.
 313
 314   while (1) {
 315     float ptsBehind;
 316     demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
 317     if (dp == NULL) return 0;
 318
 319     if (demuxer->stream->eof) return 0; // source stream has closed down
 320
 321     // Before using this packet, check to make sure that its presentation
 322     // time is not far behind the other stream (if any).  If it is,
 323     // then we discard this packet, and get another instead.  (The rest of
 324     // MPlayer doesn't always do a good job of synchronizing when the
 325     // audio and video streams get this far apart.)
 326     // (We don't do this when streaming over TCP, because then the audio and
 327     // video streams are interleaved.)
 328     // (Also, if the stream is *excessively* far behind, then we allow
 329     // the packet, because in this case it probably means that there was
 330     // an error in the source's timestamp synchronization.)
 331     const float ptsBehindThreshold = 1.0; // seconds
 332     const float ptsBehindLimit = 60.0; // seconds
 333     if (ptsBehind < ptsBehindThreshold ||
 334         ptsBehind > ptsBehindLimit ||
 335         rtsp_transport_tcp) { // packet's OK
 336       ds_add_packet(ds, dp);
 337       break;
 338     }
 339
 340 #ifdef DEBUG_PRINT_DISCARDED_PACKETS
 341     RTPState* rtpState = (RTPState*)(demuxer->priv);
 342     ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue;
 343     fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind);
 344 #endif
 345     free_demux_packet(dp); // give back this packet, and get another one
 346   }
 347
 348   return 1;
 349 }
 350
 351 Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds,
 352                        unsigned char*& packetData, unsigned& packetDataLen,
 353                        float& pts) {
 354   // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
 355   // is not delivered to the "demux_stream".
 356   float ptsBehind;
 357   demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking
 358   if (dp == NULL) return False;
 359
 360   packetData = dp->buffer;
 361   packetDataLen = dp->len;
 362   pts = dp->pts;
 363
 364   return True;
 365 }
 366
 367 static void teardownRTSPorSIPSession(RTPState* rtpState); // forward
 368
 369 extern "C" void demux_close_rtp(demuxer_t* demuxer) {
 370   // Reclaim all RTP-related state:
 371
 372   // Get the RTP state that was stored in the demuxer's 'priv' field:
 373   RTPState* rtpState = (RTPState*)(demuxer->priv);
 374   if (rtpState == NULL) return;
 375
 376   teardownRTSPorSIPSession(rtpState);
 377
 378   UsageEnvironment* env = NULL;
 379   TaskScheduler* scheduler = NULL;
 380   if (rtpState->mediaSession != NULL) {
 381     env = &(rtpState->mediaSession->envir());
 382     scheduler = &(env->taskScheduler());
 383   }
 384   Medium::close(rtpState->mediaSession);
 385   Medium::close(rtpState->rtspClient);
 386   Medium::close(rtpState->sipClient);
 387   delete rtpState->audioBufferQueue;
 388   delete rtpState->videoBufferQueue;
 389   delete rtpState->sdpDescription;
 390   delete rtpState;
 391
 392   env->reclaim(); delete scheduler;
 393 }
 394
 395 ////////// Extra routines that help implement the above interface functions:
 396
 397 #define MAX_RTP_FRAME_SIZE 5000000
 398     // >= the largest conceivable frame composed from one or more RTP packets
 399
 400 static void afterReading(void* clientData, unsigned frameSize,
 401                          unsigned /*numTruncatedBytes*/,
 402                          struct timeval presentationTime,
 403                          unsigned /*durationInMicroseconds*/) {
 404   int headersize = 0;
 405   if (frameSize >= MAX_RTP_FRAME_SIZE) {
 406     fprintf(stderr, "Saw an input frame too large (>=%d).  Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
 407             MAX_RTP_FRAME_SIZE);
 408   }
 409   ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
 410   demuxer_t* demuxer = bufferQueue->ourDemuxer();
 411   RTPState* rtpState = (RTPState*)(demuxer->priv);
 412
 413   if (frameSize > 0) demuxer->stream->eof = 0;
 414
 415   demux_packet_t* dp = bufferQueue->dp;
 416
 417   if (bufferQueue->readSource()->isAMRAudioSource())
 418     headersize = 1;
 419   else if (bufferQueue == rtpState->videoBufferQueue &&
 420       ((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) {
 421     dp->buffer[0]=0x00;
 422     dp->buffer[1]=0x00;
 423     dp->buffer[2]=0x01;
 424     headersize = 3;
 425   }
 426
 427   resize_demux_packet(dp, frameSize + headersize);
 428
 429   // Set the packet's presentation time stamp, depending on whether or
 430   // not our RTP source's timestamps have been synchronized yet:
 431   Boolean hasBeenSynchronized
 432     = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
 433   if (hasBeenSynchronized) {
 434     if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
 435       fprintf(stderr, "%s stream has been synchronized using RTCP \n",
 436               bufferQueue->tag());
 437     }
 438
 439     struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
 440     if (fst->tv_sec == 0 && fst->tv_usec == 0) {
 441       *fst = presentationTime;
 442     }
 443
 444     // For the "pts" field, use the time differential from the first
 445     // synchronized time, rather than absolute time, in order to avoid
 446     // round-off errors when converting to a float:
 447     dp->pts = presentationTime.tv_sec - fst->tv_sec
 448       + (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
 449     bufferQueue->prevPacketPTS = dp->pts;
 450   } else {
 451     if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) {
 452       fprintf(stderr, "%s stream is no longer RTCP-synchronized \n",
 453               bufferQueue->tag());
 454     }
 455
 456     // use the previous packet's "pts" once again:
 457     dp->pts = bufferQueue->prevPacketPTS;
 458   }
 459   bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized;
 460
 461   dp->pos = demuxer->filepos;
 462   demuxer->filepos += frameSize + headersize;
 463
 464   // Signal any pending 'doEventLoop()' call on this queue:
 465   bufferQueue->blockingFlag = ~0;
 466 }
 467
 468 static void onSourceClosure(void* clientData) {
 469   ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
 470   demuxer_t* demuxer = bufferQueue->ourDemuxer();
 471
 472   demuxer->stream->eof = 1;
 473
 474   // Signal any pending 'doEventLoop()' call on this queue:
 475   bufferQueue->blockingFlag = ~0;
 476 }
 477
 478 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
 479                                  Boolean mustGetNewData,
 480                                  float& ptsBehind) {
 481   // Begin by finding the buffer queue that we want to read from:
 482   // (Get this from the RTP state, which we stored in
 483   //  the demuxer's 'priv' field)
 484   RTPState* rtpState = (RTPState*)(demuxer->priv);
 485   ReadBufferQueue* bufferQueue = NULL;
 486   int headersize = 0;
 487   TaskToken task;
 488
 489   if (demuxer->stream->eof) return NULL;
 490
 491   if (ds == demuxer->video) {
 492     bufferQueue = rtpState->videoBufferQueue;
 493     if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4'))
 494       headersize = 3;
 495   } else if (ds == demuxer->audio) {
 496     bufferQueue = rtpState->audioBufferQueue;
 497     if (bufferQueue->readSource()->isAMRAudioSource())
 498       headersize = 1;
 499   } else {
 500     fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n");
 501     return NULL;
 502   }
 503
 504   if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
 505     fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
 506     return NULL;
 507   }
 508
 509   demux_packet_t* dp = NULL;
 510   if (!mustGetNewData) {
 511     // Check whether we have a previously-saved buffer that we can use:
 512     dp = bufferQueue->getPendingBuffer();
 513     if (dp != NULL) {
 514       ptsBehind = 0.0; // so that we always accept this data
 515       return dp;
 516     }
 517   }
 518
 519   // Allocate a new packet buffer, and arrange to read into it:
 520     if (!bufferQueue->nextpacket) {
 521   dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
 522   bufferQueue->dp = dp;
 523   if (dp == NULL) return NULL;
 524     }
 525
 526 #ifdef CONFIG_LIBAVCODEC
 527   extern AVCodecParserContext * h264parserctx;
 528   int consumed, poutbuf_size = 1;
 529   const uint8_t *poutbuf = NULL;
 530   float lastpts = 0.0;
 531
 532   do {
 533     if (!bufferQueue->nextpacket) {
 534 #endif
 535   // Schedule the read operation:
 536   bufferQueue->blockingFlag = 0;
 537   bufferQueue->readSource()->getNextFrame(&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize,
 538                                           afterReading, bufferQueue,
 539                                           onSourceClosure, bufferQueue);
 540   // Block ourselves until data becomes available:
 541   TaskScheduler& scheduler
 542     = bufferQueue->readSource()->envir().taskScheduler();
 543   int delay = 10000000;
 544   if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime())
 545     delay /= 10;
 546   task = scheduler.scheduleDelayedTask(delay, onSourceClosure, bufferQueue);
 547   scheduler.doEventLoop(&bufferQueue->blockingFlag);
 548   scheduler.unscheduleDelayedTask(task);
 549   if (demuxer->stream->eof) {
 550     free_demux_packet(dp);
 551     return NULL;
 552   }
 553
 554   if (headersize == 1) // amr
 555     dp->buffer[0] =
 556         ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader();
 557 #ifdef CONFIG_LIBAVCODEC
 558     } else {
 559       bufferQueue->dp = dp = bufferQueue->nextpacket;
 560       bufferQueue->nextpacket = NULL;
 561     }
 562     if (headersize == 3 && h264parserctx) { // h264
 563       consumed = h264parserctx->parser->parser_parse(h264parserctx,
 564                                NULL,
 565                                &poutbuf, &poutbuf_size,
 566                                dp->buffer, dp->len);
 567
 568       if (!consumed && !poutbuf_size)
 569         return NULL;
 570
 571       if (!poutbuf_size) {
 572         lastpts=dp->pts;
 573         free_demux_packet(dp);
 574         bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
 575       } else {
 576         bufferQueue->nextpacket = dp;
 577         bufferQueue->dp = dp = new_demux_packet(poutbuf_size);
 578         memcpy(dp->buffer, poutbuf, poutbuf_size);
 579         dp->pts=lastpts;
 580       }
 581     }
 582   } while (!poutbuf_size);
 583 #endif
 584
 585   // Set the "ptsBehind" result parameter:
 586   if (bufferQueue->prevPacketPTS != 0.0
 587       && bufferQueue->prevPacketWasSynchronized
 588       && *(bufferQueue->otherQueue) != NULL
 589       && (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0
 590       && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) {
 591     ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS
 592                  - bufferQueue->prevPacketPTS;
 593   } else {
 594     ptsBehind = 0.0;
 595   }
 596
 597   if (mustGetNewData) {
 598     // Save this buffer for future reads:
 599     bufferQueue->savePendingBuffer(dp);
 600   }
 601
 602   return dp;
 603 }
 604
 605 static void teardownRTSPorSIPSession(RTPState* rtpState) {
 606   MediaSession* mediaSession = rtpState->mediaSession;
 607   if (mediaSession == NULL) return;
 608   if (rtpState->rtspClient != NULL) {
 609     rtpState->rtspClient->teardownMediaSession(*mediaSession);
 610   } else if (rtpState->sipClient != NULL) {
 611     rtpState->sipClient->sendBYE();
 612   }
 613 }
 614
 615 ////////// "ReadBuffer" and "ReadBufferQueue" implementation:
 616
 617 ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession,
 618                                  demuxer_t* demuxer, char const* tag)
 619   : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL),
 620     dp(NULL), nextpacket(NULL),
 621     pendingDPHead(NULL), pendingDPTail(NULL),
 622     fReadSource(subsession == NULL ? NULL : subsession->readSource()),
 623     fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
 624     fOurDemuxer(demuxer), fTag(strdup(tag)) {
 625 }
 626
 627 ReadBufferQueue::~ReadBufferQueue() {
 628   delete fTag;
 629
 630   // Free any pending buffers (that never got delivered):
 631   demux_packet_t* dp = pendingDPHead;
 632   while (dp != NULL) {
 633     demux_packet_t* dpNext = dp->next;
 634     dp->next = NULL;
 635     free_demux_packet(dp);
 636     dp = dpNext;
 637   }
 638 }
 639
 640 void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) {
 641   // Keep this buffer around, until MPlayer asks for it later:
 642   if (pendingDPTail == NULL) {
 643     pendingDPHead = pendingDPTail = dp;
 644   } else {
 645     pendingDPTail->next = dp;
 646     pendingDPTail = dp;
 647   }
 648   dp->next = NULL;
 649 }
 650
 651 demux_packet_t* ReadBufferQueue::getPendingBuffer() {
 652   demux_packet_t* dp = pendingDPHead;
 653   if (dp != NULL) {
 654     pendingDPHead = dp->next;
 655     if (pendingDPHead == NULL) pendingDPTail = NULL;
 656
 657     dp->next = NULL;
 658   }
 659
 660   return dp;
 661 }
 662
 663 static int demux_rtp_control(struct demuxer_st *demuxer, int cmd, void *arg) {
 664   double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime();
 665
 666   switch(cmd) {
 667     case DEMUXER_CTRL_GET_TIME_LENGTH:
 668       if (endpts <= 0)
 669         return DEMUXER_CTRL_DONTKNOW;
 670       *((double *)arg) = endpts;
 671       return DEMUXER_CTRL_OK;
 672
 673     case DEMUXER_CTRL_GET_PERCENT_POS:
 674       if (endpts <= 0)
 675         return DEMUXER_CTRL_DONTKNOW;
 676       *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts);
 677       return DEMUXER_CTRL_OK;
 678
 679     default:
 680       return DEMUXER_CTRL_NOTIMPL;
 681     }
 682 }
 683
 684 demuxer_desc_t demuxer_desc_rtp = {
 685   "LIVE555 RTP demuxer",
 686   "live555",
 687   "",
 688   "Ross Finlayson",
 689   "requires LIVE555 Streaming Media library",
 690   DEMUXER_TYPE_RTP,
 691   0, // no autodetect
 692   NULL,
 693   demux_rtp_fill_buffer,
 694   demux_open_rtp,
 695   demux_close_rtp,
 696   NULL,
 697   demux_rtp_control
 698 };