annotate libmpdemux/demux_rtp.cpp @ 33644:c15dabfa2380

Do a proper calculation of free RAM to be used as V4L buffers. The code uses sysinfo to query the available RAM, however it used ancient form available is some early development 2.3.x kernels. Newer form reports the size in memory units (usually same as page size), as result the code would fallback to 2 buffers even on multi GB system. The commit does: Improve the check in configure to ensure that we do use sysinfo struct with present mem_unit. Use free ram instead of total ram (to avoid swapping). Tweak memory constants and simplify code.

/*
 * routines (with C-linkage) that interface between MPlayer
 * and the "LIVE555 Streaming Media" libraries
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

extern "C" {
// on MinGW, we must include windows.h before the things it conflicts
#ifdef __MINGW32__    // with.  they are each protected from
#include <windows.h>  // windows.h, but not the other way around.
#endif
#include "demuxer.h"
#include "demux_rtp.h"
#include "stheader.h"
}
#include "demux_rtp_internal.h"

#include "BasicUsageEnvironment.hh"
#include "liveMedia.hh"
#include "GroupsockHelper.hh"
#include <unistd.h>

// A data structure representing input data for each stream:
class ReadBufferQueue {
public:
  ReadBufferQueue(MediaSubsession* subsession, demuxer_t* demuxer,
		  char const* tag);
  virtual ~ReadBufferQueue();

  FramedSource* readSource() const { return fReadSource; }
  RTPSource* rtpSource() const { return fRTPSource; }
  demuxer_t* ourDemuxer() const { return fOurDemuxer; }
  char const* tag() const { return fTag; }

  char blockingFlag; // used to implement synchronous reads

  // For A/V synchronization:
  Boolean prevPacketWasSynchronized;
  float prevPacketPTS;
  ReadBufferQueue** otherQueue;

  // The 'queue' actually consists of just a single "demux_packet_t"
  // (because the underlying OS does the actual queueing/buffering):
  demux_packet_t* dp;

  // However, we sometimes inspect buffers before delivering them.
  // For this, we maintain a queue of pending buffers:
  void savePendingBuffer(demux_packet_t* dp);
  demux_packet_t* getPendingBuffer();

  // For H264 over rtsp using AVParser, the next packet has to be saved
  demux_packet_t* nextpacket;

private:
  demux_packet_t* pendingDPHead;
  demux_packet_t* pendingDPTail;

  FramedSource* fReadSource;
  RTPSource* fRTPSource;
  demuxer_t* fOurDemuxer;
  char const* fTag; // used for debugging
};

// A structure of RTP-specific state, kept so that we can cleanly
// reclaim it:
struct RTPState {
  char const* sdpDescription;
  RTSPClient* rtspClient;
  SIPClient* sipClient;
  MediaSession* mediaSession;
  ReadBufferQueue* audioBufferQueue;
  ReadBufferQueue* videoBufferQueue;
  unsigned flags;
  struct timeval firstSyncTime;
};

extern "C" char* network_username;
extern "C" char* network_password;
static char* openURL_rtsp(RTSPClient* client, char const* url) {
  // If we were given a user name (and optional password), then use them:
  if (network_username != NULL) {
    char const* password = network_password == NULL ? "" : network_password;
    return client->describeWithPassword(url, network_username, password);
  } else {
    return client->describeURL(url);
  }
}

static char* openURL_sip(SIPClient* client, char const* url) {
  // If we were given a user name (and optional password), then use them:
  if (network_username != NULL) {
    char const* password = network_password == NULL ? "" : network_password;
    return client->inviteWithPassword(url, network_username, password);
  } else {
    return client->invite(url);
  }
}

#ifdef CONFIG_LIBNEMESI
extern int rtsp_transport_tcp;
extern int rtsp_transport_http;
#else
int rtsp_transport_tcp = 0;
int rtsp_transport_http = 0;
#endif

#ifdef CONFIG_FFMPEG
extern AVCodecContext *avcctx;
#endif

extern "C" int audio_id, video_id, dvdsub_id;
extern "C" demuxer_t* demux_open_rtp(demuxer_t* demuxer) {
  Boolean success = False;
  do {
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    if (scheduler == NULL) break;
    UsageEnvironment* env = BasicUsageEnvironment::createNew(*scheduler);
    if (env == NULL) break;

    RTSPClient* rtspClient = NULL;
    SIPClient* sipClient = NULL;

    if (demuxer == NULL || demuxer->stream == NULL) break;  // shouldn't happen
    demuxer->stream->eof = 0; // just in case

    // Look at the stream's 'priv' field to see if we were initiated
    // via a SDP description:
    char* sdpDescription = (char*)(demuxer->stream->priv);
    if (sdpDescription == NULL) {
      // We weren't given a SDP description directly, so assume that
      // we were given a RTSP or SIP URL:
      char const* protocol = demuxer->stream->streaming_ctrl->url->protocol;
      char const* url = demuxer->stream->streaming_ctrl->url->url;
      extern int verbose;
      if (strcmp(protocol, "rtsp") == 0) {
	if (rtsp_transport_http == 1) {
	  rtsp_transport_http = demuxer->stream->streaming_ctrl->url->port;
	  rtsp_transport_tcp = 1;
	}
	rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer", rtsp_transport_http);
	if (rtspClient == NULL) {
	  fprintf(stderr, "Failed to create RTSP client: %s\n",
		  env->getResultMsg());
	  break;
	}
	sdpDescription = openURL_rtsp(rtspClient, url);
      } else { // SIP
	unsigned char desiredAudioType = 0; // PCMU (use 3 for GSM)
	sipClient = SIPClient::createNew(*env, desiredAudioType, NULL,
					 verbose, "MPlayer");
	if (sipClient == NULL) {
	  fprintf(stderr, "Failed to create SIP client: %s\n",
		  env->getResultMsg());
	  break;
	}
	sipClient->setClientStartPortNum(8000);
	sdpDescription = openURL_sip(sipClient, url);
      }

      if (sdpDescription == NULL) {
	fprintf(stderr, "Failed to get a SDP description from URL \"%s\": %s\n",
		url, env->getResultMsg());
	break;
      }
    }

    // Now that we have a SDP description, create a MediaSession from it:
    MediaSession* mediaSession = MediaSession::createNew(*env, sdpDescription);
    if (mediaSession == NULL) break;


    // Create a 'RTPState' structure containing the state that we just created,
    // and store it in the demuxer's 'priv' field, for future reference:
    RTPState* rtpState = new RTPState;
    rtpState->sdpDescription = sdpDescription;
    rtpState->rtspClient = rtspClient;
    rtpState->sipClient = sipClient;
    rtpState->mediaSession = mediaSession;
    rtpState->audioBufferQueue = rtpState->videoBufferQueue = NULL;
    rtpState->flags = 0;
    rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;
    demuxer->priv = rtpState;

    int audiofound = 0, videofound = 0;
    // Create RTP receivers (sources) for each subsession:
    MediaSubsessionIterator iter(*mediaSession);
    MediaSubsession* subsession;
    unsigned desiredReceiveBufferSize;
    while ((subsession = iter.next()) != NULL) {
      // Ignore any subsession that's not audio or video:
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	if (audiofound) {
	  fprintf(stderr, "Additional subsession \"audio/%s\" skipped\n", subsession->codecName());
	  continue;
	}
	desiredReceiveBufferSize = 100000;
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	if (videofound) {
	  fprintf(stderr, "Additional subsession \"video/%s\" skipped\n", subsession->codecName());
	  continue;
	}
	desiredReceiveBufferSize = 2000000;
      } else {
	continue;
      }

      if (rtsp_port)
          subsession->setClientPortNum (rtsp_port);

      if (!subsession->initiate()) {
	fprintf(stderr, "Failed to initiate \"%s/%s\" RTP subsession: %s\n", subsession->mediumName(), subsession->codecName(), env->getResultMsg());
      } else {
	fprintf(stderr, "Initiated \"%s/%s\" RTP subsession on port %d\n", subsession->mediumName(), subsession->codecName(), subsession->clientPortNum());

	// Set the OS's socket receive buffer sufficiently large to avoid
	// incoming packets getting dropped between successive reads from this
	// subsession's demuxer.  Depending on the bitrate(s) that you expect,
	// you may wish to tweak the "desiredReceiveBufferSize" values above.
	int rtpSocketNum = subsession->rtpSource()->RTPgs()->socketNum();
	int receiveBufferSize
	  = increaseReceiveBufferTo(*env, rtpSocketNum,
				    desiredReceiveBufferSize);
	if (verbose > 0) {
	  fprintf(stderr, "Increased %s socket receive buffer to %d bytes \n",
		  subsession->mediumName(), receiveBufferSize);
	}

	if (rtspClient != NULL) {
	  // Issue a RTSP "SETUP" command on the chosen subsession:
	  if (!rtspClient->setupMediaSubsession(*subsession, False,
						rtsp_transport_tcp)) break;
	  if (!strcmp(subsession->mediumName(), "audio"))
	    audiofound = 1;
	  if (!strcmp(subsession->mediumName(), "video"))
            videofound = 1;
	}
      }
    }

    if (rtspClient != NULL) {
      // Issue a RTSP aggregate "PLAY" command on the whole session:
      if (!rtspClient->playMediaSession(*mediaSession)) break;
    } else if (sipClient != NULL) {
      sipClient->sendACK(); // to start the stream flowing
    }

    // Now that the session is ready to be read, do additional
    // MPlayer codec-specific initialization on each subsession:
    iter.reset();
    while ((subsession = iter.next()) != NULL) {
      if (subsession->readSource() == NULL) continue; // not reading this

      unsigned flags = 0;
      if (strcmp(subsession->mediumName(), "audio") == 0) {
	rtpState->audioBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "audio");
	rtpState->audioBufferQueue->otherQueue = &(rtpState->videoBufferQueue);
	rtpCodecInitialize_audio(demuxer, subsession, flags);
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	rtpState->videoBufferQueue
	  = new ReadBufferQueue(subsession, demuxer, "video");
	rtpState->videoBufferQueue->otherQueue = &(rtpState->audioBufferQueue);
	rtpCodecInitialize_video(demuxer, subsession, flags);
      }
      rtpState->flags |= flags;
    }
    success = True;
  } while (0);
  if (!success) return NULL; // an error occurred

  // Hack: If audio and video are demuxed together on a single RTP stream,
  // then create a new "demuxer_t" structure to allow the higher-level
  // code to recognize this:
  if (demux_is_multiplexed_rtp_stream(demuxer)) {
    stream_t* s = new_ds_stream(demuxer->video);
    demuxer_t* od = demux_open(s, DEMUXER_TYPE_UNKNOWN,
			       audio_id, video_id, dvdsub_id, NULL);
    demuxer = new_demuxers_demuxer(od, od, od);
  }

  return demuxer;
}

extern "C" int demux_is_mpeg_rtp_stream(demuxer_t* demuxer) {
  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);

  return (rtpState->flags&RTPSTATE_IS_MPEG12_VIDEO) != 0;
}

extern "C" int demux_is_multiplexed_rtp_stream(demuxer_t* demuxer) {
  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);

  return (rtpState->flags&RTPSTATE_IS_MULTIPLEXED) != 0;
}

static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
				 Boolean mustGetNewData,
				 float& ptsBehind); // forward

extern "C" int demux_rtp_fill_buffer(demuxer_t* demuxer, demux_stream_t* ds) {
  // Get a filled-in "demux_packet" from the RTP source, and deliver it.
  // Note that this is called as a synchronous read operation, so it needs
  // to block in the (hopefully infrequent) case where no packet is
  // immediately available.

  while (1) {
    float ptsBehind;
    demux_packet_t* dp = getBuffer(demuxer, ds, False, ptsBehind); // blocking
    if (dp == NULL) return 0;

    if (demuxer->stream->eof) return 0; // source stream has closed down

    // Before using this packet, check to make sure that its presentation
    // time is not far behind the other stream (if any).  If it is,
    // then we discard this packet, and get another instead.  (The rest of
    // MPlayer doesn't always do a good job of synchronizing when the
    // audio and video streams get this far apart.)
    // (We don't do this when streaming over TCP, because then the audio and
    // video streams are interleaved.)
    // (Also, if the stream is *excessively* far behind, then we allow
    // the packet, because in this case it probably means that there was
    // an error in the source's timestamp synchronization.)
    const float ptsBehindThreshold = 1.0; // seconds
    const float ptsBehindLimit = 60.0; // seconds
    if (ptsBehind < ptsBehindThreshold ||
	ptsBehind > ptsBehindLimit ||
	rtsp_transport_tcp) { // packet's OK
      ds_add_packet(ds, dp);
      break;
    }

#ifdef DEBUG_PRINT_DISCARDED_PACKETS
    RTPState* rtpState = (RTPState*)(demuxer->priv);
    ReadBufferQueue* bufferQueue = ds == demuxer->video ? rtpState->videoBufferQueue : rtpState->audioBufferQueue;
    fprintf(stderr, "Discarding %s packet (%fs behind)\n", bufferQueue->tag(), ptsBehind);
#endif
    free_demux_packet(dp); // give back this packet, and get another one
  }

  return 1;
}

Boolean awaitRTPPacket(demuxer_t* demuxer, demux_stream_t* ds,
		       unsigned char*& packetData, unsigned& packetDataLen,
		       float& pts) {
  // Similar to "demux_rtp_fill_buffer()", except that the "demux_packet"
  // is not delivered to the "demux_stream".
  float ptsBehind;
  demux_packet_t* dp = getBuffer(demuxer, ds, True, ptsBehind); // blocking
  if (dp == NULL) return False;

  packetData = dp->buffer;
  packetDataLen = dp->len;
  pts = dp->pts;

  return True;
}

static void teardownRTSPorSIPSession(RTPState* rtpState); // forward

extern "C" void demux_close_rtp(demuxer_t* demuxer) {
  // Reclaim all RTP-related state:

  // Get the RTP state that was stored in the demuxer's 'priv' field:
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  if (rtpState == NULL) return;

  teardownRTSPorSIPSession(rtpState);

  UsageEnvironment* env = NULL;
  TaskScheduler* scheduler = NULL;
  if (rtpState->mediaSession != NULL) {
    env = &(rtpState->mediaSession->envir());
    scheduler = &(env->taskScheduler());
  }
  Medium::close(rtpState->mediaSession);
  Medium::close(rtpState->rtspClient);
  Medium::close(rtpState->sipClient);
  delete rtpState->audioBufferQueue;
  delete rtpState->videoBufferQueue;
  delete[] rtpState->sdpDescription;
  delete rtpState;
#ifdef CONFIG_FFMPEG
  av_freep(&avcctx);
#endif

  env->reclaim(); delete scheduler;
}

////////// Extra routines that help implement the above interface functions:

#define MAX_RTP_FRAME_SIZE 5000000
    // >= the largest conceivable frame composed from one or more RTP packets

static void afterReading(void* clientData, unsigned frameSize,
			 unsigned /*numTruncatedBytes*/,
			 struct timeval presentationTime,
			 unsigned /*durationInMicroseconds*/) {
  int headersize = 0;
  if (frameSize >= MAX_RTP_FRAME_SIZE) {
    fprintf(stderr, "Saw an input frame too large (>=%d).  Increase MAX_RTP_FRAME_SIZE in \"demux_rtp.cpp\".\n",
	    MAX_RTP_FRAME_SIZE);
  }
  ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
  demuxer_t* demuxer = bufferQueue->ourDemuxer();
  RTPState* rtpState = (RTPState*)(demuxer->priv);

  if (frameSize > 0) demuxer->stream->eof = 0;

  demux_packet_t* dp = bufferQueue->dp;

  if (bufferQueue->readSource()->isAMRAudioSource())
    headersize = 1;
  else if (bufferQueue == rtpState->videoBufferQueue &&
      ((sh_video_t*)demuxer->video->sh)->format == mmioFOURCC('H','2','6','4')) {
    dp->buffer[0]=0x00;
    dp->buffer[1]=0x00;
    dp->buffer[2]=0x01;
    headersize = 3;
  }

  resize_demux_packet(dp, frameSize + headersize);

  // Set the packet's presentation time stamp, depending on whether or
  // not our RTP source's timestamps have been synchronized yet:
  Boolean hasBeenSynchronized
    = bufferQueue->rtpSource()->hasBeenSynchronizedUsingRTCP();
  if (hasBeenSynchronized) {
    if (verbose > 0 && !bufferQueue->prevPacketWasSynchronized) {
      fprintf(stderr, "%s stream has been synchronized using RTCP \n",
	      bufferQueue->tag());
    }

    struct timeval* fst = &(rtpState->firstSyncTime); // abbrev
    if (fst->tv_sec == 0 && fst->tv_usec == 0) {
      *fst = presentationTime;
    }

    // For the "pts" field, use the time differential from the first
    // synchronized time, rather than absolute time, in order to avoid
    // round-off errors when converting to a float:
    dp->pts = presentationTime.tv_sec - fst->tv_sec
      + (presentationTime.tv_usec - fst->tv_usec)/1000000.0;
    bufferQueue->prevPacketPTS = dp->pts;
  } else {
    if (verbose > 0 && bufferQueue->prevPacketWasSynchronized) {
      fprintf(stderr, "%s stream is no longer RTCP-synchronized \n",
	      bufferQueue->tag());
    }

    // use the previous packet's "pts" once again:
    dp->pts = bufferQueue->prevPacketPTS;
  }
  bufferQueue->prevPacketWasSynchronized = hasBeenSynchronized;

  dp->pos = demuxer->filepos;
  demuxer->filepos += frameSize + headersize;

  // Signal any pending 'doEventLoop()' call on this queue:
  bufferQueue->blockingFlag = ~0;
}

static void onSourceClosure(void* clientData) {
  ReadBufferQueue* bufferQueue = (ReadBufferQueue*)clientData;
  demuxer_t* demuxer = bufferQueue->ourDemuxer();

  demuxer->stream->eof = 1;

  // Signal any pending 'doEventLoop()' call on this queue:
  bufferQueue->blockingFlag = ~0;
}

static demux_packet_t* getBuffer(demuxer_t* demuxer, demux_stream_t* ds,
				 Boolean mustGetNewData,
				 float& ptsBehind) {
  // Begin by finding the buffer queue that we want to read from:
  // (Get this from the RTP state, which we stored in
  //  the demuxer's 'priv' field)
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  ReadBufferQueue* bufferQueue = NULL;
  int headersize = 0;
  TaskToken task;

  if (demuxer->stream->eof) return NULL;

  if (ds == demuxer->video) {
    bufferQueue = rtpState->videoBufferQueue;
    if (((sh_video_t*)ds->sh)->format == mmioFOURCC('H','2','6','4'))
      headersize = 3;
  } else if (ds == demuxer->audio) {
    bufferQueue = rtpState->audioBufferQueue;
    if (bufferQueue->readSource()->isAMRAudioSource())
      headersize = 1;
  } else {
    fprintf(stderr, "(demux_rtp)getBuffer: internal error: unknown stream\n");
    return NULL;
  }

  if (bufferQueue == NULL || bufferQueue->readSource() == NULL) {
    fprintf(stderr, "(demux_rtp)getBuffer failed: no appropriate RTP subsession has been set up\n");
    return NULL;
  }

  demux_packet_t* dp = NULL;
  if (!mustGetNewData) {
    // Check whether we have a previously-saved buffer that we can use:
    dp = bufferQueue->getPendingBuffer();
    if (dp != NULL) {
      ptsBehind = 0.0; // so that we always accept this data
      return dp;
    }
  }

  // Allocate a new packet buffer, and arrange to read into it:
    if (!bufferQueue->nextpacket) {
  dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
  bufferQueue->dp = dp;
  if (dp == NULL) return NULL;
    }

#ifdef CONFIG_FFMPEG
  extern AVCodecParserContext * h264parserctx;
  int consumed, poutbuf_size = 1;
  const uint8_t *poutbuf = NULL;
  float lastpts = 0.0;

  do {
    if (!bufferQueue->nextpacket) {
#endif
  // Schedule the read operation:
  bufferQueue->blockingFlag = 0;
  bufferQueue->readSource()->getNextFrame(&dp->buffer[headersize], MAX_RTP_FRAME_SIZE - headersize,
					  afterReading, bufferQueue,
					  onSourceClosure, bufferQueue);
  // Block ourselves until data becomes available:
  TaskScheduler& scheduler
    = bufferQueue->readSource()->envir().taskScheduler();
  int delay = 10000000;
  if (bufferQueue->prevPacketPTS * 1.05 > rtpState->mediaSession->playEndTime())
    delay /= 10;
  task = scheduler.scheduleDelayedTask(delay, onSourceClosure, bufferQueue);
  scheduler.doEventLoop(&bufferQueue->blockingFlag);
  scheduler.unscheduleDelayedTask(task);
  if (demuxer->stream->eof) {
    free_demux_packet(dp);
    return NULL;
  }

  if (headersize == 1) // amr
    dp->buffer[0] =
        ((AMRAudioSource*)bufferQueue->readSource())->lastFrameHeader();
#ifdef CONFIG_FFMPEG
    } else {
      bufferQueue->dp = dp = bufferQueue->nextpacket;
      bufferQueue->nextpacket = NULL;
    }
    if (headersize == 3 && h264parserctx) { // h264
      consumed = h264parserctx->parser->parser_parse(h264parserctx,
                               avcctx,
                               &poutbuf, &poutbuf_size,
                               dp->buffer, dp->len);

      if (!consumed && !poutbuf_size)
        return NULL;

      if (!poutbuf_size) {
        lastpts=dp->pts;
        free_demux_packet(dp);
        bufferQueue->dp = dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
      } else {
        bufferQueue->nextpacket = dp;
        bufferQueue->dp = dp = new_demux_packet(poutbuf_size);
        memcpy(dp->buffer, poutbuf, poutbuf_size);
        dp->pts=lastpts;
      }
    }
  } while (!poutbuf_size);
#endif

  // Set the "ptsBehind" result parameter:
  if (bufferQueue->prevPacketPTS != 0.0
      && bufferQueue->prevPacketWasSynchronized
      && *(bufferQueue->otherQueue) != NULL
      && (*(bufferQueue->otherQueue))->prevPacketPTS != 0.0
      && (*(bufferQueue->otherQueue))->prevPacketWasSynchronized) {
    ptsBehind = (*(bufferQueue->otherQueue))->prevPacketPTS
		 - bufferQueue->prevPacketPTS;
  } else {
    ptsBehind = 0.0;
  }

  if (mustGetNewData) {
    // Save this buffer for future reads:
    bufferQueue->savePendingBuffer(dp);
  }

  return dp;
}

static void teardownRTSPorSIPSession(RTPState* rtpState) {
  MediaSession* mediaSession = rtpState->mediaSession;
  if (mediaSession == NULL) return;
  if (rtpState->rtspClient != NULL) {
    rtpState->rtspClient->teardownMediaSession(*mediaSession);
  } else if (rtpState->sipClient != NULL) {
    rtpState->sipClient->sendBYE();
  }
}

////////// "ReadBuffer" and "ReadBufferQueue" implementation:

ReadBufferQueue::ReadBufferQueue(MediaSubsession* subsession,
				 demuxer_t* demuxer, char const* tag)
  : prevPacketWasSynchronized(False), prevPacketPTS(0.0), otherQueue(NULL),
    dp(NULL), nextpacket(NULL),
    pendingDPHead(NULL), pendingDPTail(NULL),
    fReadSource(subsession == NULL ? NULL : subsession->readSource()),
    fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
    fOurDemuxer(demuxer), fTag(strdup(tag)) {
}

ReadBufferQueue::~ReadBufferQueue() {
  free((void *)fTag);

  // Free any pending buffers (that never got delivered):
  demux_packet_t* dp = pendingDPHead;
  while (dp != NULL) {
    demux_packet_t* dpNext = dp->next;
    dp->next = NULL;
    free_demux_packet(dp);
    dp = dpNext;
  }
}

void ReadBufferQueue::savePendingBuffer(demux_packet_t* dp) {
  // Keep this buffer around, until MPlayer asks for it later:
  if (pendingDPTail == NULL) {
    pendingDPHead = pendingDPTail = dp;
  } else {
    pendingDPTail->next = dp;
    pendingDPTail = dp;
  }
  dp->next = NULL;
}

demux_packet_t* ReadBufferQueue::getPendingBuffer() {
  demux_packet_t* dp = pendingDPHead;
  if (dp != NULL) {
    pendingDPHead = dp->next;
    if (pendingDPHead == NULL) pendingDPTail = NULL;

    dp->next = NULL;
  }

  return dp;
}

static int demux_rtp_control(struct demuxer *demuxer, int cmd, void *arg) {
  double endpts = ((RTPState*)demuxer->priv)->mediaSession->playEndTime();

  switch(cmd) {
    case DEMUXER_CTRL_GET_TIME_LENGTH:
      if (endpts <= 0)
        return DEMUXER_CTRL_DONTKNOW;
      *((double *)arg) = endpts;
      return DEMUXER_CTRL_OK;

    case DEMUXER_CTRL_GET_PERCENT_POS:
      if (endpts <= 0)
        return DEMUXER_CTRL_DONTKNOW;
      *((int *)arg) = (int)(((RTPState*)demuxer->priv)->videoBufferQueue->prevPacketPTS*100/endpts);
      return DEMUXER_CTRL_OK;

    default:
      return DEMUXER_CTRL_NOTIMPL;
    }
}

demuxer_desc_t demuxer_desc_rtp = {
  "LIVE555 RTP demuxer",
  "live555",
  "",
  "Ross Finlayson",
  "requires LIVE555 Streaming Media library",
  DEMUXER_TYPE_RTP,
  0, // no autodetect
  NULL,
  demux_rtp_fill_buffer,
  demux_open_rtp,
  demux_close_rtp,
  NULL,
  demux_rtp_control
};