[gst-devel] Problem using gstrtpbin
Tiago Katcipis
katcipis at inf.ufsc.br
Sat May 9 14:45:35 CEST 2009
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);*
On Sat, May 9, 2009 at 3:55 AM, Aurelien Grimaud <gstelzz at yahoo.fr> wrote:
> 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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> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > gstreamer-devel mailing list
> > gstreamer-devel at lists.sourceforge.net
> > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel
> >
>
>
>
> ------------------------------------------------------------------------------
> The NEW KODAK i700 Series Scanners deliver under ANY circumstances! Your
> production scanning environment may not be a perfect world - but thanks to
> Kodak, there's a perfect scanner to get the job done! With the NEW KODAK
> i700
> Series Scanner you'll get full speed at 300 dpi even with all image
> processing features enabled. http://p.sf.net/sfu/kodak-com
> _______________________________________________
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> gstreamer-devel at lists.sourceforge.net
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>
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