How to correctly timestamp buffers in an appsrc element

[Nicolas Dufresne]

Le jeudi 26 novembre 2020 à 21:55 +0100, J. Krieg a écrit :
> Hello,
> 
> No ideas?
> Could anyone help please?
> Unfortunately I can’t figure out this by myself.

By default, appsrc uses an open segment (start=0, end=infinity). That means your
timestamp must match the running-time. The running time can be obtained like
this:

  clock = gst_pipeline_get_clock(pipeline)
  if (clock) {
  	time_now = gst_clock_get_time(clock)
  	rt_time = time_now - gst_element_get_base_time (GST_ELEMENT
(pipeline));
   } else {
        rt_time = GST_CLOCK_TIME_NONE; /* or 0 depending on your use case */
   }

If you have raw audio data, it might be easier to calculate the timestamp base
ont he data lenght, starting from zero. Of if your data isn't live, you might
also calculate timestamp using the framerate of a video (starting from 0 again).

> 
> Thank you very much.
> 
> Best Regards,
> Joerg
> 
> Am Fr., 20. Nov. 2020 um 15:47 Uhr schrieb J. Krieg <bellum07 at googlemail.com>:
> > 
> > Hello,
> > 
> > I’m currently working on an application to display live TV using
> > GStreamer on a Raspberry Pi 2B.
> > 
> > Therefore I use two appsrc elements (one for video and one for audio)
> > which are reading PES packets in 2 separate threads directly from the
> > V4L DVB demux device ‘/dev/dvb/adapter0/demux0’.
> > My current test pipelines are:
> > 
> > Video
> >   V4L DVB demux (DMX_OUT_TAP) -> appsrc ! h264parse ! v4l2h264dec !
> > queue ! kmssink
> > Audio
> >   V4L DVB demux (DMX_OUT_TAP) -> appsrc ! mpegaudioparse !
> > mpg123audiodec ! queue ! alsasink
> > 
> > I managed to get this working without timestamping the buffers at all
> > in both appsrc elements but then video and audio isn't synchronous.
> > 
> > I tried to implement timestamping the buffers according to
> > https://gstreamer.freedesktop.org/documentation/application-development/advanced/pipeline-manipulation.html?gi-language=c#inserting-data-with-appsrc
> > But when doing this I get slightly stuttering video and extremely
> > stuttering or no audio.
> > 
> > What I'm also struggling with is that in the link above the following is
> > said:
> > "In live mode, you should timestamp the buffers with the pipeline
> > running-time when the first byte of the buffer was captured before
> > feeding them to appsrc."
> > 
> > But according to my tests the pipeline only changes its state from
> > PAUSED to PLAYING (where the clock of the pipeline is only available)
> > after some captured buffers have already fed into the pipeline.
> > So how could the buffers be timestamped with the running time the very
> > first time before they have been put into the pipeline which is in a
> > PAUSED state to get video and audio synchronous?
> > 
> > What am I doing wrong?
> > Any help or pointing in the right direction would be really appreciated.
> > 
> > Thanks,
> > Joerg
> > 
> > Code:
> > // Create a new empty buffer
> > gbuffer = gst_buffer_new_allocate(NULL, rc, NULL);
> > 
> > // Timestamp buffer
> > if (((CustomData *)data)->pipelineclock) {
> >     pipeline_clock_time = gst_clock_get_time(((CustomData
> > *)data)->pipelineclock);
> >     pipeline_running_time = pipeline_clock_time - g_pipeline_base_time;
> >     GST_BUFFER_PTS(gbuffer) = pipeline_running_time;
> >     GST_BUFFER_DURATION(gbuffer) = pipeline_running_time -
> > g_last_pipeline_running_time_a;
> >     g_last_pipeline_running_time_a = pipeline_running_time;
> >     printf("*** DEBUG *** dmx_read_a | pipeline running timestamp for
> > audio is in ns: %lld\n", pipeline_running_time);
> > } else {
> >     printf("*** DEBUG *** dmx_read_a | Sorry, pipelineclock NOT
> > available...\n");
> >     GST_BUFFER_PTS(gbuffer) = GST_CLOCK_TIME_NONE;
> > }
> > 
> > // Fill data into buffer
> > bc = gst_buffer_fill(gbuffer, 0, buf, rc);
> > 
> > // Push the buffer into the appsrc
> > g_signal_emit_by_name (((CustomData *)data)->aappsrc, "push-buffer",
> > gbuffer, &rb);
> > 
> > // Free the buffer now that we are done with it
> > gst_buffer_unref (gbuffer);
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