Access rtpsession stats property

joakim joakim.magnusson at spiideo.com
Wed Jan 30 10:57:29 UTC 2019


Hi,
I'm building a pipeline in c with rtspsrc. I want to access the stats
property in the rtpsession object created internally with rtspsrc. Is this
possible? This is my code for now:

#include <gst/gst.h>
#include <glib.h>

typedef struct myDataTag {
    GstElement *pipeline;
    GstElement *source;
    GstElement *depayloader;
    GstElement *decoder;
    GstElement *sink;
} myData_t;

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:
            break;
    }

    return TRUE;
}


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 vorbis-decoder sink pad */
    g_print ("Dynamic pad created, linking demuxer/decoder\n");

    sinkpad = gst_element_get_static_pad (decoder, "sink");

    gst_pad_link (pad, sinkpad);

    gst_object_unref (sinkpad);
}

/* pad added handler */
static void pad_added_handler (GstElement *src, GstPad *new_pad, myData_t
*pThis) {
    GstPad *sink_pad = gst_element_get_static_pad (pThis->depayloader,
"sink");
    GstPadLinkReturn ret;
    GstCaps *new_pad_caps = NULL;
    GstStructure *new_pad_struct = NULL;
    const gchar *new_pad_type = NULL;

    g_print ("Received new pad '%s' from '%s':\n", GST_PAD_NAME (new_pad),
GST_ELEMENT_NAME (src));

    /* Check the new pad's name */
    if (!g_str_has_prefix (GST_PAD_NAME (new_pad), "recv_rtp_src_")) {
        g_print ("  It is not the right pad.  Need recv_rtp_src_.
Ignoring.\n");
        goto exit;
    }

    /* If our converter is already linked, we have nothing to do here */
    if (gst_pad_is_linked (sink_pad)) {
        g_print (" Sink pad from %s already linked. Ignoring.\n",
GST_ELEMENT_NAME (src));
        goto exit;
    }

    /* Check the new pad's type */
    new_pad_caps = gst_pad_query_caps (new_pad, NULL);
    new_pad_struct = gst_caps_get_structure (new_pad_caps, 0);
    new_pad_type = gst_structure_get_name (new_pad_struct);

    /* Attempt the link */
    ret = gst_pad_link (new_pad, sink_pad);
    if (GST_PAD_LINK_FAILED (ret)) {
        g_print ("  Type is '%s' but link failed.\n", new_pad_type);
    } else {
        g_print ("  Link succeeded (type '%s').\n", new_pad_type);
    }

exit:
    /* Unreference the new pad's caps, if we got them */
    if (new_pad_caps != NULL)
        gst_caps_unref (new_pad_caps);

    /* Unreference the sink pad */
    gst_object_unref (sink_pad);
}


int
main (int   argc,
      char *argv[])
{
    argv[0] = "gstreamer-test";
    argv[1] = "rtsp://ipadress/axis-media/media.amp";
    //argv[2] = "para2";

    myData_t appData;

    GMainLoop *loop;
    GstElement *rtpbin;

    GstBus *bus;
    guint bus_watch_id;
    GstStructure *stats;

    /* Initialisation */
    gst_init (&argc, &argv);

    loop = g_main_loop_new (NULL, FALSE);

    /* Check input arguments */
    if (argc != 2) {
        g_printerr ("Usage: %s <Ogg/Vorbis filename>\n", argv[0]);
        return -1;
    }

    /* Create gstreamer elements */
    appData.pipeline = gst_pipeline_new ("pipeline");
    appData.source = gst_element_factory_make ("rtspsrc", "rtspsrc");
    g_object_set (G_OBJECT (appData.source), "location",
"rtsp://root:spiideo2017@10.0.3.69/axis-media/media.amp", NULL);
    appData.depayloader = gst_element_factory_make
("rtph264depay","depayloader");
    appData.decoder = gst_element_factory_make ("avdec_h264",
"avh264decoder");
    appData.sink = gst_element_factory_make ("autovideosink", "sink");

    if (!appData.pipeline || !appData.source || !appData.depayloader ||
!appData.decoder || !appData.sink) {
        g_printerr ("One element could not be created. Exiting.\n");
        return -1;
    }

    g_signal_connect (appData.source, "pad-added", G_CALLBACK
(pad_added_handler), &appData);
    /* Set up the pipeline */

    /* we set the input filename to the source element */
    g_object_set (G_OBJECT (appData.source), "location", argv[1], NULL);

    /* we add a message handler */
    bus = gst_pipeline_get_bus (GST_PIPELINE (appData.pipeline));
    bus_watch_id = 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 (appData.pipeline),
                       appData.source, appData.depayloader, appData.decoder,
appData.sink, NULL);

    /* we link the elements together */
    /* file-source -> ogg-demuxer ~> vorbis-decoder -> converter ->
alsa-output */
    gst_element_link_many (appData.depayloader,
                           appData.decoder, appData.sink, NULL);
    g_signal_connect (appData.source, "pad-added", G_CALLBACK
(pad_added_handler), &appData);

    /* Set the pipeline to "playing" state*/
    g_print ("Now playing: %s\n", argv[1]);
    gst_element_set_state (appData.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 playback\n");
    gst_element_set_state (appData.pipeline, GST_STATE_NULL);

    g_print ("Deleting pipeline\n");
    gst_object_unref (GST_OBJECT (appData.pipeline));
    g_source_remove (bus_watch_id);
    g_main_loop_unref (loop);

    return 0;




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