gstreamer app stumbles with trace/bpt trap:5

aft aftnix at
Wed Feb 12 02:19:07 PST 2014

This seemingly innocent code fails to run on Max OSX 10.8

    #include <string.h>
    #include <math.h>

    #include <gst/gst.h>

    static void
    print_source_stats (GObject *source)
        GstStructure *stats;
        gchar *str;

        g_object_get(source, "stats", &stats, NULL);
        str = gst_structure_to_string (stats);
        g_print("source stat :%s\n", str);



    static gboolean
    print_stats (GstElement *rtpbin)
        GObject *session;
        GValueArray *arr;
        GValue *val;
        guint i;


        g_signal_emit_by_name(rtpbin, "get-internal-session",0, &session);
        g_object_get(session, "sources", &arr, NULL);

        for (i=0; i <arr->n_values; i++) {
            GObject *sources;
            val = g_value_array_get_nth(arr, i);
            sources = g_value_get_object(val);

        return TRUE;


     Build a pipeline equivalent to :

     gst-launch -v rtpbin name=rtpbin \
      audiotestsrc ! audioconvert ! audioresample ! alawenc !
rtppcmapay ! rtpbin.send_rtp_src_0 !
            rtpbin.send_rtp_sink_0 ! udpsink port=5002 host=

    int main(int argc, char **argv) {
        GstElement *audiosrc, *audioconv, *audiores, *audiopay, *audioenc;
        GstElement *rtpbin, *rtpsink;
        GstElement *pipeline;
        GMainLoop *loop;
        GstPad *srcpad, *sinkpad;

        gst_init(&argc, &argv);

        pipeline = gst_pipeline_new (NULL);

        audiosrc = gst_element_factory_make ("audiotestsrc", "audiosrc");
        g_assert (audiosrc);
        audioconv = gst_element_factory_make ("audioconvert", "audioconv");
        g_assert (audioconv);
        audiores = gst_element_factory_make ("audioresample", "audiores");
        g_assert (audiores);
        /* the encoding and payloading */
        audioenc = gst_element_factory_make ("alawenc", "audioenc");
        g_assert (audioenc);
        audiopay = gst_element_factory_make ("rtppcmapay", "audiopay");
        g_assert (audiopay);
        /* add capture and payloading to the pipeline and link */
        gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores,
                          audioenc, audiopay, NULL);

        if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc,
                                    audiopay, NULL)) {
            g_error ("Failed to link audiosrc, audioconv, audioresample, "
                     "audio encoder and audio payloader");
        /* the rtpbin element */
        rtpbin = gst_element_factory_make ("rtpbin", "rtpbin");
        g_assert (rtpbin);

        gst_bin_add (GST_BIN (pipeline), rtpbin);
        rtpsink = gst_element_factory_make ("udpsink", "rtpsink");
        g_assert (rtpsink);
        g_object_set (rtpsink, "port", 5002, "host", "", NULL);
        sinkpad = gst_element_get_request_pad (rtpbin, "send_rtp_sink_0");
        srcpad = gst_element_get_static_pad (audiopay, "src");
        if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
            g_error ("Failed to link audio payloader to rtpbin");
        gst_object_unref (srcpad);
        /* get the RTP srcpad that was created when we requested the
sinkpad above and
         * link it to the rtpsink sinkpad*/
        srcpad = gst_element_get_static_pad (rtpbin, "send_rtp_src_0");
        sinkpad = gst_element_get_static_pad (rtpsink, "sink");
        if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
            g_error ("Failed to link rtpbin to rtpsink");
        gst_object_unref (srcpad);
        gst_object_unref (sinkpad);

        /* set the pipeline to playing */
        g_print ("starting sender pipeline\n");
        gst_element_set_state (pipeline, GST_STATE_PLAYING);

        /* print stats every second */
        g_timeout_add_seconds (1, (GSourceFunc) print_stats, rtpbin);

        /* we need to run a GLib main loop to get the messages */
        loop = g_main_loop_new (NULL, FALSE);
        g_main_loop_run (loop);

        g_print ("stopping sender pipeline\n");
        gst_element_set_state (pipeline, GST_STATE_NULL);

        return 0;

The error generated does not seems like a fault in code itself, it
seems to be a "dynamic library" related issue :

    $./rtpSender --gst-debug-level=3

    ** (rtpSender:1360): ERROR **: Failed to link rtpbin to rtpsink
    Trace/BPT trap: 5

The backtrace generated :

    (lldb) bt
    * thread #1: tid = 0x12d62, 0x000000010065dcb2
libglib-2.0.0.dylib`g_logv + 786, queue = ', stop
reason = EXC_BREAKPOINT (code=EXC_I386_BPT, subcode=0x0)
        frame #0: 0x000000010065dcb2 libglib-2.0.0.dylib`g_logv + 786
        frame #1: 0x00000001000019a6
rtpSender`g_error(format=0x0000000100001e5d) + 358 at gmessages.h:196
        frame #2: 0x0000000100001788 rtpSender`main(argc=1,
argv=0x00007fff5fbff8b8) + 1160 at main.c:106
        frame #3: 0x00007fff87c097e1 libdyld.dylib`start + 1

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