[gst-devel] Problem using gstrtpbin

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Im trying to do a rtp stream sending data and another side receiving the
data, the part that sends the data is working fine, but the part that
receives is giving me a lot of trouble. At
http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.htmli
have read:

"To use GstRtpBin<http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html#GstRtpBin>as
an RTP receiver, request a recv_rtp_sink_%
d pad. The session number must be specified in the pad name. Data received
on the recv_rtp_sink_%d pad will be processed in the gstrtpsession manager
and after being validated forwarded on GstRtpsSrcDemux element. Each RTP
stream is demuxed based on the SSRC and send to a
GstRtpJitterBuffer<http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpjitterbuffer.html#GstRtpJitterBuffer>.
After the packets are released from the jitterbuffer, they will be forwarded
to a GstRtpsSrcDemux element. The GstRtpsSrcDemux element will demux the
packets based on the payload type and will create a unique pad
recv_rtp_src_%d_%d_%d on gstrtpbin with the session number, SSRC and payload
type respectively as the pad name. "

on my application i cant get the recv_rtp_src_%d_%d_%d,  i already tried on
a lot of ways, my last shot was try to iterate over all the pads on the bin
and try to conect, i discovered that the src pad never shows up. No error is
given. I can get the on-new-ssrc signal...and other signals as
on-ssrc-validated...
but on all this signals the recv_rtp_src_%d_%d_%d is not created yet, i also
tried to get the "on-pad-added" signal but this signal never happens.

My problem is, when the recv_rtp_src_%d_%d_%d is created. When i iterate
over the pads i always get a
** (teste_rtp:9516): DEBUG: GstRtpBin has [0] src pads

here goes the source code, is a little messy because im all day trying a lot
of different ways to do this. And i get no error message.

#include <gst/gst.h>
#include <glib.h>

#define PORTA_UDP_ENTRADA 5000

static gboolean
bus_call (GstBus     *bus,
          GstMessage *msg,
          gpointer    data)
{
  GMainLoop *loop = (GMainLoop *) data;

  switch (GST_MESSAGE_TYPE (msg)) {

    case GST_MESSAGE_EOS:
      g_print ("End of stream\n");
      g_main_loop_quit (loop);
      break;

    case GST_MESSAGE_ERROR: {
      gchar  *debug;
      GError *error;

      gst_message_parse_error (msg, &error, &debug);
      g_free (debug);

      g_printerr ("Error: %s\n", error->message);
      g_error_free (error);

      g_main_loop_quit (loop);
      break;
    }
    default:
      g_print("Tipo da mensagem [%d], Nome da mensagem [%s]\n",
GST_MESSAGE_TYPE (msg), GST_MESSAGE_TYPE_NAME(msg));
      break;
  }

  return TRUE;
}

static void
on_new_ssrc (GstElement* gstrtpbin,
                   guint session,
                   guint ssrc,
                   gpointer data)
{
  GstPad* sinkpad;
  GstPad* srcpad[1];
  GstElement* decoder = (GstElement *) data;
  GstIterator* iter;
  gint done, linked, iter_count;

  g_print ("New session stabilished, linking gstrtpbin session src pad to
the rtp_decoder\n");

  sinkpad = gst_element_get_static_pad(decoder, "sink");
  // TODO Esta dificil de pegar o pad src do gstrtpbin que eh criado ao
iniciar uma sessao nova.
  if(!sinkpad){
      g_warning("Error getting rtp_decoder sink pad");
      return;
  }
  /*
     unique pad recv_rtp_src_%d_%d_%d on gstrtpbin with the session number,
SSRC and payload type respectively as the pad name.

http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html
  */

  iter = gst_element_iterate_src_pads(gstrtpbin);
  if(!iter){
      g_warning("Error getting gstrtpbin pads iterator");
      return;
  }

  done = FALSE;
  linked = FALSE;
  iter_count = 0;

  while (!done) {
      switch (gst_iterator_next (iter, (gpointer *) srcpad)) {
          case GST_ITERATOR_OK:
              if(gst_pad_link (*srcpad, sinkpad) != GST_PAD_LINK_OK){
                  g_warning("Error linking gstrtpbin pad[%s] to rtp_decoder
pad[%s]", gst_pad_get_name(*srcpad), gst_pad_get_name(sinkpad));
              }else{
                  g_warning("Linked gstrtpbin pad[%s] to rtp_decoder pad[%s]
with success", gst_pad_get_name(*srcpad), gst_pad_get_name(sinkpad));
                  linked = TRUE;
              }
              iter_count++;
              gst_object_unref (*srcpad);
          break;
          case GST_ITERATOR_RESYNC:
              gst_iterator_resync (iter);
          break;
          case GST_ITERATOR_ERROR:
              done = TRUE;
          break;
          case GST_ITERATOR_DONE:
              done = TRUE;
          break;
      }
   }
  if(!linked){
      g_warning("failed to found a valid recv_src_pad on gstrtpbin");
  }
  g_debug("GstRtpBin has [%d] src pads", iter_count);

  gst_iterator_free (iter);
  gst_object_unref (sinkpad);
}

static void
on_pad_added (GstElement *element,
              GstPad     *pad,
              gpointer    data)
{
  GstPad *sinkpad;
  GstElement *decoder = (GstElement *) data;

  /* We can now link this pad with the converter sink pad */
  g_print ("Dynamic pad created, linking wavparser/converter\n");

  sinkpad = gst_element_get_static_pad (decoder, "sink");
  if(gst_pad_link (pad, sinkpad) != GST_PAD_LINK_OK){
      g_warning("Error linking recv_rtp_src pad to sinkpad");
  }
  gst_object_unref (sinkpad);
}

int
main (int   argc,
      char *argv[])
{
  GMainLoop *loop;

  GstElement *pipeline, *source, *rtp_bin, *rtp_decoder, *sink;
  GstPad *gstrtp_sink_pad;
  GstBus *bus;

  /* Initialisation */
  gst_init (&argc, &argv);

  loop = g_main_loop_new (NULL, FALSE);

  /* Create gstreamer elements */
  pipeline    = gst_pipeline_new ("audio-player");
  source      = gst_element_factory_make ("udpsrc","udp-source");
  rtp_bin     = gst_element_factory_make ("gstrtpbin", "gst_rtpbin");
  rtp_decoder = gst_element_factory_make ("rtpL16depay", "rtp_decoder");
  sink        = gst_element_factory_make ("filesink", "file-sink");

  if (!pipeline || !source || !sink || !rtp_decoder || !rtp_bin) {
    g_printerr ("One element could not be created. Exiting.\n");
    return -1;
  }

  gstrtp_sink_pad = gst_element_get_request_pad(rtp_bin, "recv_rtp_sink_0");
  if (!gstrtp_sink_pad) {
    g_printerr ("Sink pad could not be created. Exiting.\n");
    return -1;
  }

  /* Set up the pipeline */
  g_object_set (G_OBJECT (source), "port", PORTA_UDP_ENTRADA , NULL);
  g_object_set (G_OBJECT (sink), "location", "dados_recebidos_rtp" , NULL);

  /* we add a message handler */
  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
  gst_bus_add_watch (bus, bus_call, loop);
  gst_object_unref (bus);

  /* we add all elements into the pipeline */
  /* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */
  gst_bin_add_many (GST_BIN (pipeline),
                    source, sink, rtp_bin, rtp_decoder, NULL);

  /* we link the elements together */
  if(gst_pad_link(gst_element_get_static_pad(source, "src"),
gstrtp_sink_pad) != GST_PAD_LINK_OK){
      g_warning("Error linking source to the gstrtp_sink_pad");
      gst_object_unref (GST_OBJECT (pipeline));
      return 0;
  }

  /*
    After the packets are released from the jitterbuffer, they will be
forwarded to a GstRtpsSrcDemux element.
    The GstRtpsSrcDemux element will demux the packets based on the payload
type and will create a unique pad
    recv_rtp_src_%d_%d_%d on gstrtpbin with the session number, SSRC and
payload type respectively as the pad name.
    Because of that we have to dinamicaly link the src pads on runtime.
  */
  g_signal_connect (rtp_bin, "pad-added",   G_CALLBACK (on_pad_added),
rtp_decoder);
  g_signal_connect (rtp_bin, "on-new-ssrc", G_CALLBACK (on_new_ssrc),
rtp_decoder);

  if(!gst_element_link (rtp_decoder, sink)){
      g_warning("Error linking the rtp_decoder to the sink");
      gst_object_unref (GST_OBJECT (pipeline));
      return -1;
  }

  /* Set the pipeline to "playing" state*/
  g_print ("listening on port: %d\n", PORTA_UDP_ENTRADA);
  gst_element_set_state (pipeline, GST_STATE_PLAYING);

  /* Iterate */
  g_print ("Running...\n");
  g_main_loop_run (loop);

  /* Out of the main loop, clean up nicely */
  g_print ("Returned, stopping listening on port\n");
  gst_element_set_state (pipeline, GST_STATE_NULL);

  g_print ("Deleting pipeline\n");
  gst_object_unref (GST_OBJECT (pipeline));

  return 0;
}