i did it, the pad never is created :-(, but i get no message of warning or error neither. And on the list of signals of the gstrtpbin there is no "pad-added" signal, its normal to the signal dont be there? g_signal_connect (rtp_bin, "pad-added", G_CALLBACK (on_pad_added), rtp_decoder); <div class="gmail_quote">On Sat, May 9, 2009 at 3:55 AM, Aurelien Grimaud <<a href="mailto:gstelzz@yahoo.fr">gstelzz@yahoo.fr</a>> wrote: <blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">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 : <div class="im">> 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 > <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html</a> > i have read: > > "To use GstRtpBin </div>> <<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html#GstRtpBin" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html#GstRtpBin</a>> <div class="im">> 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 </div>> <<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpjitterbuffer.html#GstRtpJitterBuffer" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpjitterbuffer.html#GstRtpJitterBuffer</a>>. <div><div></div><div class="h5">> 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. > > <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html" target="_blank">http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-gstrtpbin.html</a> > */ > > 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; > } > </div></div>> ------------------------------------------------------------------------ > > ------------------------------------------------------------------------------ > The NEW KODAK i700 Series Scanners deliver under ANY circumstances! 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