[gst-devel] Problem using gstrtpbin

Aurelien Grimaud gstelzz at yahoo.fr
Sat May 9 08:55:54 CEST 2009


You should add the pad-added signal on the rtpbin.
When it triggers, check the pad name to find out which pad it is.
If pad is a recv_rtp_src_%d_%d_%d, link your decoder and sink in the 
call back.

Aurelien

Tiago Katcipis a écrit :
> 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.html 
> i 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;
> }
>
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