[MPlayer-dev-eng] [PATCH] Audio/video synchronization for RTP streams

Ross Finlayson finlayson at live.com
Wed Sep 25 21:35:45 CEST 2002


Index: libmpdemux/demux_rtp.cpp
===================================================================
RCS file: /cvsroot/mplayer/main/libmpdemux/demux_rtp.cpp,v
retrieving revision 1.3
diff -u -r1.3 demux_rtp.cpp
--- libmpdemux/demux_rtp.cpp	29 Aug 2002 20:19:33 -0000	1.3
+++ libmpdemux/demux_rtp.cpp	25 Sep 2002 18:23:49 -0000
@@ -68,6 +68,7 @@
    ReadBuffer* dequeue();

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

@@ -77,6 +78,7 @@
    unsigned counter; // used for debugging
  private:
    FramedSource* fReadSource;
+  RTPSource* fRTPSource;
    demuxer_t* fOurDemuxer;
    char const* fTag; // used for debugging
  };
@@ -90,6 +92,7 @@
    ReadBufferQueue* audioBufferQueue;
    ReadBufferQueue* videoBufferQueue;
    int isMPEG; // TRUE for MPEG audio, video, or transport streams
+  struct timeval firstSyncTime;
  };

  extern "C" void demux_open_rtp(demuxer_t* demuxer) {
@@ -252,6 +255,7 @@
      rtpState->videoBufferQueue
        = new ReadBufferQueue(videoSubsession, demuxer, "video");
      rtpState->isMPEG = isMPEG;
+    rtpState->firstSyncTime.tv_sec = rtpState->firstSyncTime.tv_usec = 0;

      demuxer->priv = rtpState;
    } while (0);
@@ -368,16 +372,38 @@
  }

  static void afterReading(void* clientData, unsigned frameSize,
-			 struct timeval /*presentationTime*/) {
+			 struct timeval presentationTime) {
    ReadBuffer* readBuffer = (ReadBuffer*)clientData;
    ReadBufferQueue* bufferQueue = readBuffer->ourQueue();
    demuxer_t* demuxer = bufferQueue->ourDemuxer();
+  RTPState* rtpState = (RTPState*)(demuxer->priv);

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

    demux_packet_t* dp = readBuffer->dp();
    dp->len = frameSize;
-  dp->pts = 0;
+
+  // 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) {
+      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;
+    } else {
+      dp->pts = 0.0;
+    }
+  }
+
    dp->pos = demuxer->filepos;
    demuxer->filepos += frameSize;
    if (!readBuffer->enqueue()) {
@@ -441,6 +467,7 @@
  				 demuxer_t* demuxer, char const* tag)
    : head(NULL), tail(NULL), counter(0),
      fReadSource(subsession == NULL ? NULL : subsession->readSource()),
+    fRTPSource(subsession == NULL ? NULL : subsession->rtpSource()),
      fOurDemuxer(demuxer), fTag(strdup(tag)) {
  }
  




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