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 #include "options.h"
  30 }
  31 #include "demux_rtp_internal.h"
  32
  33 #include "BasicUsageEnvironment.hh"
  34 #include "liveMedia.hh"
  35 #include "GroupsockHelper.hh"
  36 #include <unistd.h>
  37
  38 // A data structure representing input data for each stream:
  39 class ReadBufferQueue {
  40 public:
  41   ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
  42                   char const* tag);
  43   virtual ~ReadBufferQueue();
  44
  45   FramedSource* readSource() const { return fReadSource; }
  46   RTPSource* rtpSource() const { return fRTPSource; }
  47   demuxer_t* ourDemuxer() const { return fOurDemuxer; }
  48   char const* tag() const { return fTag; }
  49
  50   char blockingFlag; // used to implement synchronous reads
  51
  52   // For A/V synchronization:
  53   Boolean prevPacketWasSynchronized;
  54   float prevPacketPTS;
  55   ReadBufferQueue** otherQueue;
  56
  57   // The 'queue' actually consists of just a single "demux_packet_t"
  58   // (because the underlying OS does the actual queueing/buffering):
  59   demux_packet_t* dp;
  60
  61   // However, we sometimes inspect buffers before delivering them.
  62   // For this, we maintain a queue of pending buffers:
  63   void savePendingBuffer(demux_packet_t* dp);
  64   demux_packet_t* getPendingBuffer();
  65
  66   // For H264 over rtsp using AVParser, the next packet has to be saved
  67   demux_packet_t* nextpacket;
  68
  69 private:
  70   demux_packet_t* pendingDPHead;
  71   demux_packet_t* pendingDPTail;
  72
  73   FramedSource* fReadSource;
  74   RTPSource* fRTPSource;
  75   demuxer_t* fOurDemuxer;
  76   char const* fTag; // used for debugging
  77 };
  78
  79 // A structure of RTP-specific state, kept so that we can cleanly
  80 // reclaim it:
  81 typedef struct RTPState {
  82   char const* sdpDescription;
  83   RTSPClient* rtspClient;
  84   SIPClient* sipClient;
  85   MediaSession* mediaSession;
  86   ReadBufferQueue* audioBufferQueue;
  87   ReadBufferQueue* videoBufferQueue;
  88   unsigned flags;
  89   struct timeval firstSyncTime;
  90 };
  91
  92 extern "C" char* network_username;
  93 extern "C" char* network_password;
  94 static char* openURL_rtsp(RTSPClient* client, char const* url) {
  95   // If we were given a user name (and optional password), then use them:
  96   if (network_username != NULL) {
  97     char const* password = network_password == NULL ? "" : network_password;
  98     return client->describeWithPassword(url, network_username, password);
  99   } else {
 100     return client->describeURL(url);
 101   }
 102 }
 103
 104 static char* openURL_sip(SIPClient* client, char const* url) {
 105   // If we were given a user name (and optional password), then use them:
 106   if (network_username != NULL) {
 107     char const* password = network_password == NULL ? "" : network_password;
 108     return client->inviteWithPassword(url, network_username, password);
 109   } else {
 110     return client->invite(url);
 111   }
 112 }
 113
 114 #ifdef CONFIG_LIBNEMESI
 115 extern int rtsp_transport_tcp;
 116 extern int rtsp_transport_http;
 117 #else
 118 int rtsp_transport_tcp = 0;
 119 int rtsp_transport_http = 0;
 120 #endif
 121
 122 extern int rtsp_port;
 123 #ifdef CONFIG_LIBAVCODEC
 124 extern AVCodecContext *avcctx;
 125 #endif
 126
 127 extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
 128   struct MPOpts *opts = demuxer->opts;
 129   Boolean success = False;
 130   do {
 131     TaskScheduler* scheduler = BasicTaskScheduler::createNew();
 132     if (scheduler == NULL) break;
 133     UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
 134     if (env == NULL) break;
 135
 136     RTSPClient* rtspClient = NULL;
 137     SIPClient* sipClient = NULL;
 138
 139     if (demuxer == NULL || demuxer->stream == NULL) break;  // shouldn't happen
 140     demuxer->stream->eof = 0; // just in case
 141
 142     // Look at the stream's 'priv' field to see if we were initiated
 143     // via a SDP description:
 144     char* sdpDescription = (char*)(demuxer->stream->priv);
 145     if (sdpDescription == NULL) {
 146       // We weren't given a SDP description directly, so assume that
 147       // we were given a RTSP or SIP URL:
 148       char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
 149       char const* url = demuxer->stream->streaming_ctrl->url->url;
 150       extern int verbose;
 151       if (strcmp(protocol, "rtsp") == 0) {
 152         if (rtsp_transport_http == 1) {
 153           rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port;
 154           rtsp_transport_tcp = 1;
 155         }
 156         rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http);
 157         if (rtspClient == NULL) {
 158           fprintf(stderr, "Failed to create RTSP client: %s\n",
 159                   env->getResultMsg());
 160           break;
 161         }
 162         sdpDescription = openURL_rtsp(rtspClient, url);
 163       } else { // SIP
 164         unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
 165         sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
 166                                          verbose, "MPlayer");
 167         if (sipClient == NULL) {
 168           fprintf(stderr, "Failed to create SIP client: %s\n",
 169                   env->getResultMsg());
 170           break;
 171         }
 172         sipClient->setClientStartPortNum(8000);
 173         sdpDescription = openURL_sip(sipClient, url);
 174       }
 175
 176       if (sdpDescription == NULL) {
 177         fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
 178                 url, env->getResultMsg());
 179         break;
 180       }
 181     }
 182
 183     // Now that we have a SDP description, create a MediaSession from it:
 184     MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
 185     if (mediaSession == NULL) break;
 186
 187
 188     // Create a 'RTPState' structure containing the state that we just created,
 189     // and store it in the demuxer's 'priv' field, for future reference:
 190     RTPState* rtpState = new RTPState;
 191     rtpState->sdpDescription = sdpDescription;
 192     rtpState->rtspClient = rtspClient;
 193     rtpState->sipClient = sipClient;
 194     rtpState->mediaSession = mediaSession;
 195     rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
 196     rtpState->flags = 0;
 197     rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
 198     demuxer->priv = rtpState;
 199
 200     int audiofound = 0, videofound = 0;
 201     // Create RTP receivers (sources) for each subsession:
 202     MediaSubsessionIterator iter(*mediaSession);
 203     MediaSubsession* subsession;
 204     unsigned desiredReceiveBufferSize;
 205     while ((subsession = iter.next()) != NULL) {
 206       // Ignore any subsession that's not audio or video:
 207       if (strcmp(subsession->mediumName(), "audio") == 0) {
 208         if (audiofound) {
 209           fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName());
 210           continue;
 211         }
 212         desiredReceiveBufferSize = 100000;
 213       } else if (strcmp(subsession->mediumName(), "video") == 0) {
 214         if (videofound) {
 215           fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName());
 216           continue;
 217         }
 218         desiredReceiveBufferSize = 2000000;
 219       } else {
 220         continue;
 221       }
 222
 223       if (rtsp_port)
 224           subsession->setClientPortNum (rtsp_port);
 225
 226       if (!subsession->initiate()) {
 227         fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
 228       } else {
 229         fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum());
 230
 231         // Set the OS's socket receive buffer sufficiently large to avoid
 232         // incoming packets getting dropped between successive reads from this
 233         // subsession's demuxer.  Depending on the bitrate(s) that you expect,
 234         // you may wish to tweak the "desiredReceiveBufferSize" values above.
 235         int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
 236         int receiveBufferSize
 237           = increaseReceiveBufferTo(*env, rtpSocketNum,
 238                                     desiredReceiveBufferSize);
 239         if (verbose > 0) {
 240           fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
 241                   subsession->mediumName(), receiveBufferSize);
 242         }
 243
 244         if (rtspClient != NULL) {
 245           // Issue a RTSP "SETUP" command on the chosen subsession:
 246           if (!rtspClient->setupMediaSubsession(*subsession, False,
 247                                                 rtsp_transport_tcp)) break;
 248           if (!strcmp(subsession->mediumName(), "audio"))
 249             audiofound = 1;
 250           if (!strcmp(subsession->mediumName(), "video"))
 251             videofound = 1;
 252         }
 253       }
 254     }
 255
 256     if (rtspClient != NULL) {
 257       // Issue a RTSP aggregate "PLAY" command on the whole session:
 258       if (!rtspClient->playMediaSession(*mediaSession)) break;
 259     } else if (sipClient != NULL) {
 260       sipClient->sendACK(); // to start the stream flowing
 261     }
 262
 263     // Now that the session is ready to be read, do additional
 264     // MPlayer codec-specific initialization on each subsession:
 265     iter.reset();
 266     while ((subsession = iter.next()) != NULL) {
 267       if (subsession->readSource() == NULL) continue; // not reading this
 268
 269       unsigned flags = 0;
 270       if (strcmp(subsession->mediumName(), "audio") == 0) {
 271         rtpState->audioBufferQueue
 272           = new ReadBufferQueue(subsession, demuxer, "audio");
 273         rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
 274         rtpCodecInitialize_audio(demuxer, subsession, flags);
 275       } else if (strcmp(subsession->mediumName(), "video") == 0) {
 276         rtpState->videoBufferQueue
 277           = new ReadBufferQueue(subsession, demuxer, "video");
 278         rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
 279         rtpCodecInitialize_video(demuxer, subsession, flags);
 280       }
 281       rtpState->flags |= flags;
 282     }
 283     success = True;
 284   } while (0);
 285   if (!success) return NULL; // an error occurred
 286
 287   // Hack: If audio and video are demuxed together on a single RTP stream,
 288   // then create a new "demuxer_t" structure to allow the higher-level
 289   // code to recognize this:
 290   if (demux_is_multiplexed_rtp_stream(demuxer)) {
 291     stream_t* s = new_ds_stream(demuxer->video);
 292     demuxer_t* od = demux_open(opts, s, DEMUXER_TYPE_UNKNOWN,
 293                                opts->audio_id, opts->video_id, opts->sub_id,
 294                                NULL);
 295     demuxer = new_demuxers_demuxer(od, od, od);
 296   }
 297
 298   return demuxer;
 299 }
 300
 301 extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
 302   // Get the RTP state that was stored in the demuxer's 'priv' field:
 303   RTPState* rtpState = (RTPState*)(demuxer->priv);
 304
 305   return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
 306 }
 307
 308 extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) {
 309   // Get the RTP state that was stored in the demuxer's 'priv' field:
 310   RTPState* rtpState = (RTPState*)(demuxer->priv);
 311
 312   return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0;
 313 }
 314
 315 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
 316                                  Boolean mustGetNewData,
 317                                  float& ptsBehind); // forward
 318
 319 extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
 320   // Get a filled-in "demux_packet" from the RTP source, and deliver it.
 321   // Note that this is called as a synchronous read operation, so it needs
 322   // to block in the (hopefully infrequent) case where no packet is
 323   // immediately available.
 324
 325   while (1) {
 326     float ptsBehind;
 327     demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
 328     if (dp == NULL) return 0;
 329
 330     if (demuxer->stream->eof) return 0; // source stream has closed down
 331
 332     // Before using this packet, check to make sure that its presentation
 333     // time is not far behind the other stream (if any).  If it is,
 334     // then we discard this packet, and get another instead.  (The rest of
 335     // MPlayer doesn't always do a good job of synchronizing when the
 336     // audio and video streams get this far apart.)
 337     // (We don't do this when streaming over TCP, because then the audio and
 338     // video streams are interleaved.)
 339     // (Also, if the stream is *excessively* far behind, then we allow
 340     // the packet, because in this case it probably means that there was
 341     // an error in the source's timestamp synchronization.)
 342     const float ptsBehindThreshold = 1.0; // seconds
 343     const float ptsBehindLimit = 60.0; // seconds
 344     if (ptsBehind < ptsBehindThreshold ||
 345         ptsBehind > ptsBehindLimit ||
 346         rtsp_transport_tcp) { // packet's OK
 347       ds_add_packet(ds, dp);
 348       break;
 349     }
 350
 351 #ifdef DEBUG_PRINT_DISCARDED_PACKETS
 352     RTPState* rtpState = (RTPState*)(demuxer->priv);
 353     ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue;
 354     fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind);
 355 #endif
 356     free_demux_packet(dp); // give back this packet, and get another one
 357   }
 358
 359   return 1;
 360 }
 361
 362 Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds,
 363                        unsigned char*& packetData, unsigned& packetDataLen,
 364                        float& pts) {
 365   // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
 366   // is not delivered to the "demux_stream".
 367   float ptsBehind;
 368   demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking
 369   if (dp == NULL) return False;
 370
 371   packetData = dp->buffer;
 372   packetDataLen = dp->len;
 373   pts = dp->pts;
 374
 375   return True;
 376 }
 377
 378 static void teardownRTSPorSIPSession(RTPState* rtpState); // forward
 379
 380 extern "C" void demux_close_rtp(demuxer_t* demuxer) {
 381   // Reclaim all RTP-related state:
 382
 383   // Get the RTP state that was stored in the demuxer's 'priv' field:
 384   RTPState* rtpState = (RTPState*)(demuxer->priv);
 385   if (rtpState == NULL) return;
 386
 387   teardownRTSPorSIPSession(rtpState);
 388
 389   UsageEnvironment* env = NULL;
 390   TaskScheduler* scheduler = NULL;
 391   if (rtpState->mediaSession != NULL) {
 392     env = &(rtpState->mediaSession->envir());
 393     scheduler = &(env->taskScheduler());
 394   }
 395   Medium::close(rtpState->mediaSession);
 396   Medium::close(rtpState->rtspClient);
 397   Medium::close(rtpState->sipClient);
 398   delete rtpState->audioBufferQueue;
 399   delete rtpState->videoBufferQueue;
 400   delete[] rtpState->sdpDescription;
 401   delete rtpState;
 402 #ifdef CONFIG_LIBAVCODEC
 403   av_freep(&avcctx);
 404 #endif
 405
 406   env->reclaim(); delete scheduler;
 407 }
 408
 409 ////////// Extra routines that help implement the above interface functions:
 410
 411 #define MAX_RTP_FRAME_SIZE 5000000
 412     // >= the largest conceivable frame composed from one or more RTP packets
 413
 414 static void afterReading(void* clientData, unsigned frameSize,
 415                          unsigned /*numTruncatedBytes*/,
 416                          struct timeval presentationTime,
 417                          unsigned /*durationInMicroseconds*/) {
 418   int headersize = 0;
 419   if (frameSize >= MAX_RTP_FRAME_SIZE) {
 420     fprintf(stderr, "Saw an input frame too large (>=%d).  Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
 421             MAX_RTP_FRAME_SIZE);
 422   }
 423   ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
 424   demuxer_t* demuxer = bufferQueue->ourDemuxer();
 425   RTPState* rtpState = (RTPState*)(demuxer->priv);
 426
 427   if (frameSize > 0) demuxer->stream->eof = 0;
 428
 429   demux_packet_t* dp = bufferQueue->dp;
 430
 431   if (bufferQueue->readSource()->isAMRAudioSource())
 432     headersize = 1;
 433   else if (bufferQueue == rtpState->videoBufferQueue &&
 434       ((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) {
 435     dp->buffer[0]=0x00;
 436     dp->buffer[1]=0x00;
 437     dp->buffer[2]=0x01;
 438     headersize = 3;
 439   }
 440
 441   resize_demux_packet(dp, frameSize + headersize);
 442
 443   // Set the packet's presentation time stamp, depending on whether or
 444   // not our RTP source's timestamps have been synchronized yet:
 445   Boolean hasBeenSynchronized
 446     = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
 447   if (hasBeenSynchronized) {
 448     if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
 449       fprintf(stderr, "%s stream has been synchronized using RTCP \n",
 450               bufferQueue->tag());
 451     }
 452
 453     struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
 454     if (fst->tv_sec == 0 && fst->tv_usec == 0) {
 455       *fst = presentationTime;
 456     }
 457
 458     // For the "pts" field, use the time differential from the first
 459     // synchronized time, rather than absolute time, in order to avoid
 460     // round-off errors when converting to a float:
 461     dp->pts = presentationTime.tv_sec - fst->tv_sec
 462       + (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
 463     bufferQueue->prevPacketPTS = dp->pts;
 464   } else {
 465     if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) {
 466       fprintf(stderr, "%s stream is no longer RTCP-synchronized \n",
 467               bufferQueue->tag());
 468     }
 469
 470     // use the previous packet's "pts" once again:
 471     dp->pts = bufferQueue->prevPacketPTS;
 472   }
 473   bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized;
 474
 475   dp->pos = demuxer->filepos;
 476   demuxer->filepos += frameSize + headersize;
 477
 478   // Signal any pending 'doEventLoop()' call on this queue:
 479   bufferQueue->blockingFlag = ~0;
 480 }
 481
 482 static void onSourceClosure(void* clientData) {
 483   ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
 484   demuxer_t* demuxer = bufferQueue->ourDemuxer();
 485
 486   demuxer->stream->eof = 1;
 487
 488   // Signal any pending 'doEventLoop()' call on this queue:
 489   bufferQueue->blockingFlag = ~0;
 490 }
 491
 492 static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
 493                                  Boolean mustGetNewData,
 494                                  float& ptsBehind) {
 495   // Begin by finding the buffer queue that we want to read from:
 496   // (Get this from the RTP state, which we stored in
 497   //  the demuxer's 'priv' field)
 498   RTPState* rtpState = (RTPState*)(demuxer->priv);
 499   ReadBufferQueue* bufferQueue = NULL;
 500   int headersize = 0;
 501   TaskToken task;
 502
 503   if (demuxer->stream->eof) return NULL;
 504
 505   if (ds == demuxer->video) {
 506     bufferQueue = rtpState->videoBufferQueue;
 507     if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4'))
 508       headersize = 3;
 509   } else if (ds == demuxer->audio) {
 510     bufferQueue = rtpState->audioBufferQueue;
 511     if (bufferQueue->readSource()->isAMRAudioSource())
 512       headersize = 1;
 513   } else {
 514     fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n");
 515     return NULL;
 516   }
 517
 518   if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
 519     fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
 520     return NULL;
 521   }
 522
 523   demux_packet_t* dp = NULL;
 524   if (!mustGetNewData) {
 525     // Check whether we have a previously-saved buffer that we can use:
 526     dp = bufferQueue->getPendingBuffer();
 527     if (dp != NULL) {
 528       ptsBehind = 0.0; // so that we always accept this data
 529       return dp;
 530     }
 531   }
 532
 533   // Allocate a new packet buffer, and arrange to read into it:
 534     if (!bufferQueue->nextpacket) {
 535   dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
 536   bufferQueue->dp = dp;
 537   if (dp == NULL) return NULL;
 538     }
 539
 540 #ifdef CONFIG_LIBAVCODEC
 541   extern AVCodecParserContext * h264parserctx;
 542   int consumed, poutbuf_size = 1;
 543   const uint8_t *poutbuf = NULL;
 544   float lastpts = 0.0;
 545
 546   do {
 547     if (!bufferQueue->nextpacket) {
 548 #endif
 549   // Schedule the read operation:
 550   bufferQueue->blockingFlag = 0;
 551   bufferQueue->readSource()->getNextFrame(&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize,
 552                                           afterReading, bufferQueue,
 553                                           onSourceClosure, bufferQueue);
 554   // Block ourselves until data becomes available:
 555   TaskScheduler& scheduler
 556     = bufferQueue->readSource()->envir().taskScheduler();
 557   int delay = 10000000;
 558   if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime())
 559     delay /= 10;
 560   task = scheduler.scheduleDelayedTask(delay, onSourceClosure, bufferQueue);
 561   scheduler.doEventLoop(&bufferQueue->blockingFlag);
 562   scheduler.unscheduleDelayedTask(task);
 563   if (demuxer->stream->eof) {
 564     free_demux_packet(dp);
 565     return NULL;
 566   }
 567
 568   if (headersize == 1) // amr
 569     dp->buffer[0] =
 570         ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader();
 571 #ifdef CONFIG_LIBAVCODEC
 572     } else {
 573       bufferQueue->dp = dp = bufferQueue->nextpacket;
 574       bufferQueue->nextpacket = NULL;
 575     }
 576     if (headersize == 3 && h264parserctx) { // h264
 577       consumed = h264parserctx->parser->parser_parse(h264parserctx,
 578                                avcctx,
 579                                &poutbuf, &poutbuf_size,
 580                                dp->buffer, dp->len);
 581
 582       if (!consumed && !poutbuf_size)
 583         return NULL;
 584
 585       if (!poutbuf_size) {
 586         lastpts=dp->pts;
 587         free_demux_packet(dp);
 588         bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
 589       } else {
 590         bufferQueue->nextpacket = dp;
 591         bufferQueue->dp = dp = new_demux_packet(poutbuf_size);
 592         memcpy(dp->buffer, poutbuf, poutbuf_size);
 593         dp->pts=lastpts;
 594       }
 595     }
 596   } while (!poutbuf_size);
 597 #endif
 598
 599   // Set the "ptsBehind" result parameter:
 600   if (bufferQueue->prevPacketPTS != 0.0
 601       && bufferQueue->prevPacketWasSynchronized
 602       && *(bufferQueue->otherQueue) != NULL
 603       && (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0
 604       && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) {
 605     ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS
 606                  - bufferQueue->prevPacketPTS;
 607   } else {
 608     ptsBehind = 0.0;
 609   }
 610
 611   if (mustGetNewData) {
 612     // Save this buffer for future reads:
 613     bufferQueue->savePendingBuffer(dp);
 614   }
 615
 616   return dp;
 617 }
 618
 619 static void teardownRTSPorSIPSession(RTPState* rtpState) {
 620   MediaSession* mediaSession = rtpState->mediaSession;
 621   if (mediaSession == NULL) return;
 622   if (rtpState->rtspClient != NULL) {
 623     rtpState->rtspClient->teardownMediaSession(*mediaSession);
 624   } else if (rtpState->sipClient != NULL) {
 625     rtpState->sipClient->sendBYE();
 626   }
 627 }
 628
 629 ////////// "ReadBuffer" and "ReadBufferQueue" implementation:
 630
 631 ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession,
 632                                  demuxer_t* demuxer, char const* tag)
 633   : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL),
 634     dp(NULL), nextpacket(NULL),
 635     pendingDPHead(NULL), pendingDPTail(NULL),
 636     fReadSource(subsession == NULL ? NULL : subsession->readSource()),
 637     fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
 638     fOurDemuxer(demuxer), fTag(strdup(tag)) {
 639 }
 640
 641 ReadBufferQueue::~ReadBufferQueue() {
 642   free((void *)fTag);
 643
 644   // Free any pending buffers (that never got delivered):
 645   demux_packet_t* dp = pendingDPHead;
 646   while (dp != NULL) {
 647     demux_packet_t* dpNext = dp->next;
 648     dp->next = NULL;
 649     free_demux_packet(dp);
 650     dp = dpNext;
 651   }
 652 }
 653
 654 void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) {
 655   // Keep this buffer around, until MPlayer asks for it later:
 656   if (pendingDPTail == NULL) {
 657     pendingDPHead = pendingDPTail = dp;
 658   } else {
 659     pendingDPTail->next = dp;
 660     pendingDPTail = dp;
 661   }
 662   dp->next = NULL;
 663 }
 664
 665 demux_packet_t* ReadBufferQueue::getPendingBuffer() {
 666   demux_packet_t* dp = pendingDPHead;
 667   if (dp != NULL) {
 668     pendingDPHead = dp->next;
 669     if (pendingDPHead == NULL) pendingDPTail = NULL;
 670
 671     dp->next = NULL;
 672   }
 673
 674   return dp;
 675 }
 676
 677 static int demux_rtp_control(struct demuxer *demuxer, int cmd, void *arg) {
 678   double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime();
 679
 680   switch(cmd) {
 681     case DEMUXER_CTRL_GET_TIME_LENGTH:
 682       if (endpts <= 0)
 683         return DEMUXER_CTRL_DONTKNOW;
 684       *((double *)arg) = endpts;
 685       return DEMUXER_CTRL_OK;
 686
 687     case DEMUXER_CTRL_GET_PERCENT_POS:
 688       if (endpts <= 0)
 689         return DEMUXER_CTRL_DONTKNOW;
 690       *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts);
 691       return DEMUXER_CTRL_OK;
 692
 693     default:
 694       return DEMUXER_CTRL_NOTIMPL;
 695     }
 696 }
 697
 698 demuxer_desc_t demuxer_desc_rtp = {
 699   "LIVE555 RTP demuxer",
 700   "live555",
 701   "",
 702   "Ross Finlayson",
 703   "requires LIVE555 Streaming Media library",
 704   DEMUXER_TYPE_RTP,
 705   0, // no autodetect
 706   NULL,
 707   demux_rtp_fill_buffer,
 708   demux_open_rtp,
 709   demux_close_rtp,
 710   NULL,
 711   demux_rtp_control
 712 };