view libmpdemux/demux_rtp.cpp @ 18001:a2683ee7cb5a

fix descrambling of asf file, where signed buffer could cause erroneous values to be filled in descrable variables, add misssing check for one of these variables and restore sign of these variables as insurance that these checks will work even in such case.

////////// Routines (with C-linkage) that interface between "MPlayer"
////////// and the "LIVE555 Streaming Media" libraries:

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 "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();

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:
typedef 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);
  }
}

int rtspStreamOverTCP = 0; 
int rtsp_port = 0; 

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) {
	rtspClient = RTSPClient::createNew(*env, verbose, "MPlayer");
	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;

    // 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) {
	desiredReceiveBufferSize = 100000;
      } else if (strcmp(subsession->mediumName(), "video") == 0) {
	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,
						rtspStreamOverTCP)) break;
	}
      }
    }

    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 ||
	rtspStreamOverTCP) { // 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;
}

Boolean insertRTPData(demuxer_t* demuxer, demux_stream_t* ds,
		      unsigned char* data, unsigned dataLen) {
  // Begin by finding the buffer queue that we want to add data to.
  // (Get this from the RTP state, which we stored in
  //  the demuxer's 'priv' field)
  RTPState* rtpState = (RTPState*)(demuxer->priv);
  ReadBufferQueue* bufferQueue = NULL;
  if (ds == demuxer->video) {
    bufferQueue = rtpState->videoBufferQueue;
  } else if (ds == demuxer->audio) {
    bufferQueue = rtpState->audioBufferQueue;
  } else {
    fprintf(stderr, "(demux_rtp)insertRTPData: internal error: unknown stream\n");
    return False;
  }

  if (data == NULL || dataLen == 0) return False;

  demux_packet_t* dp = new_demux_packet(dataLen);
  if (dp == NULL) return False;

  // Copy our data into the buffer, and save it:
  memmove(dp->buffer, data, dataLen);
  dp->len = dataLen;
  dp->pts = 0;
  bufferQueue->savePendingBuffer(dp);
  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;

  env->reclaim(); delete scheduler;
}

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

#define MAX_RTP_FRAME_SIZE 50000
    // >= 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*/) {
  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;
  dp->len = frameSize;

  // 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;

  // 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;
  if (ds == demuxer->video) {
    bufferQueue = rtpState->videoBufferQueue;
  } else if (ds == demuxer->audio) {
    bufferQueue = rtpState->audioBufferQueue;
  } 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;
  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:
  dp = new_demux_packet(MAX_RTP_FRAME_SIZE);
  bufferQueue->dp = dp;
  if (dp == NULL) return NULL;

  // Schedule the read operation:
  bufferQueue->blockingFlag = 0;
  bufferQueue->readSource()->getNextFrame(dp->buffer, MAX_RTP_FRAME_SIZE,
					  afterReading, bufferQueue,
					  onSourceClosure, bufferQueue);
  // Block ourselves until data becomes available:
  TaskScheduler& scheduler
    = bufferQueue->readSource()->envir().taskScheduler();
  scheduler.doEventLoop(&bufferQueue->blockingFlag);

  // 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) {
    MediaSubsessionIterator iter(*mediaSession);
    MediaSubsession* subsession;

    while ((subsession = iter.next()) != NULL) {
      rtpState->rtspClient->teardownMediaSubsession(*subsession);
    }
  } 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), pendingDPHead(NULL), pendingDPTail(NULL),
    fReadSource(subsession == NULL ? NULL : subsession->readSource()),
    fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
    fOurDemuxer(demuxer), fTag(strdup(tag)) {
} 

ReadBufferQueue::~ReadBufferQueue() {
  delete 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;
}


demuxer_desc_t demuxer_desc_rtp = {
  "LIVE555 RTP demuxer",
  "rtp",
  "",
  "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,
  NULL
};