No "pad-added" signal in rtspsrc

[Sérgio Agostinho]

Hey,

You're not adding the rstpsrc to the pipeline. You need to add the rtspsrc
to the pipeline, set the pipeline to play and only then add and link the
remainder elements once the rtspsrc source pads are created. If you try to
set the pipeline to play with elements unlinked it will fail to transition
to play.

Cheers,
Sérgio

2015-03-02 22:37 GMT+01:00 Francisco Velázquez <francisv at ifi.uio.no>:

> 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 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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>
>
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