[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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