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