No "pad-added" signal in rtspsrc

[Francisco Velázquez]

Hello,

I am trying to create a rtsp client using the rtspsrc element, but I’m failing to have it working. The problem I see is that the “pad-added” signal is never triggered. The code is:

#include <stdio.h>
#include <gst/gst.h>
#include <string.h>

GST_DEBUG_CATEGORY_STATIC (my_category);
#define GST_CAT_DEFAULT my_category

/* Structure to contain all our information, so we can pass it around */
typedef struct _CustomData{
  GMainLoop *main_loop;
  GstElement *pipeline, *rtspsrc, *rtpamrdepay, *amrnbdec, *pulsesink, *rtpvp8depay, *vp8dec, *videoconvert, *ximagesink;
}CustomData;

static void new_pad_cb (GstElement *rtspsrc, GstPad* pad, CustomData *data){
  GST_INFO("New pad in rtspsrc added!");
  
  gchar *dynamic_pad_name;
  
  dynamic_pad_name = gst_pad_get_name (pad);
  
  if(gst_element_link_pads(data->rtspsrc, dynamic_pad_name, data->rtpamrdepay, "sink")){
    GST_INFO("Pad for audio linked");
    g_free (dynamic_pad_name);
    return;
  }
  else if(gst_element_link_pads(data->rtspsrc, dynamic_pad_name, data->rtpvp8depay, "sink")){
    GST_INFO("Pad for video linked");
    g_free (dynamic_pad_name);
    return;
  }
  g_free (dynamic_pad_name);
}

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

  /* Initialize our data structure */
  CustomData data;
  memset (&data, 0, sizeof (data));

  gst_init (&argc, &argv);
  
  GST_DEBUG_CATEGORY_INIT (my_category, "my_code", 0, "This is the debug category for my code.");

  /* Generic elements */
  data.pipeline = gst_pipeline_new("pipeline");
  data.rtspsrc = gst_element_factory_make("rtspsrc", "rtspsrc");

  /* listen for newly created pads in rtpbin */
  g_signal_connect (data.rtspsrc, "pad-added", G_CALLBACK (new_pad_cb), &data);

  g_object_set(G_OBJECT (data.rtspsrc), "location", "rtsp://127.0.0.1:8554/test", NULL);

  /* Audio elements */
  data.rtpamrdepay = gst_element_factory_make("rtpamrdepay", "rtpamrdepay");
  data.amrnbdec = gst_element_factory_make("amrnbdec", "amrnbdec");
  data.pulsesink = gst_element_factory_make("pulsesink", "pulsesink");
  
  /* Video elements */
  data.rtpvp8depay = gst_element_factory_make("rtpvp8depay", "rtpvp8depay");
  data.vp8dec = gst_element_factory_make("vp8dec", "vp8dec");
  data.videoconvert = gst_element_factory_make("videoconvert", "videoconvert");
  data.ximagesink = gst_element_factory_make("ximagesink", "ximagesink");
  
  gst_bin_add_many(GST_BIN(data.pipeline), data.rtpamrdepay, data.amrnbdec, data.pulsesink, data.rtpvp8depay, data.vp8dec, data.videoconvert, data.ximagesink, NULL);
  
  /* Linking audio elements */
  gst_element_link_many(data.rtpamrdepay, data.amrnbdec, data.pulsesink, NULL);
  
  /* Linding video elements */
  gst_element_link_many(data.rtpvp8depay, data.vp8dec, data.videoconvert, data.ximagesink, NULL);
  
  gst_element_set_state(data.pipeline, GST_STATE_PLAYING);
  
  GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (data.pipeline), GST_DEBUG_GRAPH_SHOW_ALL ,"rtcp_client");
  
  /* Create a GLib Main Loop and set it to run */
  data.main_loop = g_main_loop_new (NULL, FALSE);
  g_main_loop_run (data.main_loop);
  
  gst_element_set_state (data.pipeline, GST_STATE_NULL);
  
  return 0;
}

 This pipeline using gst-launch-1.0 works:

gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/test <rtsp://127.0.0.1:8554/test> name=src src. ! rtpamrdepay ! amrnbdec ! autoaudiosink src. ! rtpvp8depay ! vp8dec ! videoconvert ! ximagesink

This means that my rtsp server is up and running, and producing an appropriate stream for this client.

Log from rtspsrc and attached logs for *:9.

$ GST_DEBUG=rtspsrc:9 ./rtsp_client
0:00:00.012373298 21227       0xa2ec00 DEBUG                rtspsrc gstrtspsrc.c:8087:gst_rtspsrc_uri_set_uri:<rtspsrc> parsing URI
0:00:00.012954043 21227       0xa2ec00 DEBUG                rtspsrc gstrtspsrc.c:8094:gst_rtspsrc_uri_set_uri:<rtspsrc> configuring URI
0:00:00.013509915 21227       0xa2ec00 DEBUG                rtspsrc gstrtspsrc.c:8110:gst_rtspsrc_uri_set_uri:<rtspsrc> set uri: rtsp://127.0.0.1:8554/test
0:00:00.013924933 21227       0xa2ec00 DEBUG                rtspsrc gstrtspsrc.c:8112:gst_rtspsrc_uri_set_uri:<rtspsrc> request uri is: rtsp://127.0.0.1:8554/test



Thanks for any hint on what I'm doing wrong.

Best regards,

Francisco

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