GstRtpBuffer and GstBaseTransform Filter

[Mandar Joshi]

Hello everyone,
I would like to write an element that adds
gst_rtp_buffer_add_extension_onebyte_header to GstRtpBuffer.

I am able to add a 4 byte data in a GstBaseTransform element and
receive it over RTSP using the following pipelines

------------------------------------------------------------------------------------------------------------
./test-launch "( audiotestsrc ! audioconvert ! opusenc !
mywriteelement ! rtpopuspay name=pay0  )"

gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/test !
rtpopusdepay ! myreadelement !  fakesink
-------------------------------------------------------------------------------------------------------------

For this, I am using the following in the transform function:


static GstFlowReturn gst_youraydio_identity_transform
(GstBaseTransform * trans, GstBuffer * in, GstBuffer *out) {
  GstMyWriteElement *filter = ...

  GstMapInfo map;

  if (gst_buffer_map (in, &map, GST_MAP_READ) == FALSE) {
    g_print ("gst_buffer_map failed\n");
    return GST_BASE_TRANSFORM_FLOW_DROPPED;
  }

  gst_buffer_fill(out, 0, map.data, map.size);
  gst_buffer_unmap (in, &map);

  gst_rtp_buffer_allocate_data (out, 4, 0, 0);
  guint8 misc_data[4] = { 1, 2, 3, 4 };
  GstRTPBuffer rtp = { NULL, };
  guint16 bits;
  guint size;
  gpointer pointer;
  gst_rtp_buffer_map (out, GST_MAP_READWRITE, &rtp);
  gst_rtp_buffer_add_extension_onebyte_header (&rtp, 5, misc_data, 4);
  gst_rtp_buffer_get_extension_onebyte_header (&rtp, 5, 0, &pointer, &size);
  guint8 *data = (guint8 *) pointer;
  printf ("Size: %d Data: %d %d %d %d\n", size, data[0], data[1],
data[2], data[3]);
  gst_rtp_buffer_unmap (&rtp);

  return GST_FLOW_OK;
}
------------------------------------------------------------------------------------------------------------

With the above code, I am able to receive the data added using
onbyte_header but the audio doesn't go through. I guess the problem
lies in my usage of gst_rtp_buffer_allocate_data which according to
the docs [1] says  "all previous memory in buffer will be freed". So,
if this is the case, how do I preserve data in the GstBuffer and add
custom data to GstRtpBuffer.
----------------------------------------------------------------------------------------------------------

I read ndufresne's post here
http://gstreamer-devel.966125.n4.nabble.com/streaming-video-metadata-td4669761.html
where he has used a pad probe. I can't use that approach because I
want to use a gst-rtsp-server factory to stream data. So, I would like
to write an element.gst

Any possible solutions?

Regards
Mandar Joshi

[1] https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-GstRTPBuffer.html#gst-rtp-buffer-allocate-data
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[Mandar Joshi]

Thank You ndufresne for your inputs on IRC

I tried using gst_pad_add_probe (..) on the src pad of the payloader
in gst-rtsp-server's factory bin but for some reason the pad probe
callback never gets called.

This is what I have so far
https://gist.github.com/mndar/267478e5b102bd1730a2ceb2631332b9

What I am trying to do in the program above is send a onebyte rtp
header every second after a rtsp client connects.

Could someone please have a look.

Thanks
Mandar Joshi
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[Mandar Joshi]

Thank you slomo and ndufresne.
Got it working with just one modification to the code above.
Connected to the media-constructed signal of the factory and got the
pay0 element there. Pad probe now works.

Posted the updated code in a comment to the above gist.

Thanks Again
Mandar Joshi
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