No clock rate in caps! dropping buffer

General Discussion
1

In discussion it was said
[bilboed]
You can only do that once your pad is active (i.e. just before you’re pushing data)
Sorry to be thick, but I dont understand that.
I have
GstElement *udpsrc_video = gst_element_factory_make (“udpsrc”, NULL);
g_object_set (udpsrc_video, “multicast-group”, “224.1.1.1”, NULL);
g_object_set (udpsrc_video, “auto-multicast”, TRUE, NULL);
g_object_set (udpsrc_video, “port”, 5002, NULL);
g_object_set (udpsrc_video, “close-socket”, FALSE, NULL);

and

GstElement *video_rtpjitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL);
g_object_set (video_rtpjitterbuffer, "drop-on-latency", TRUE, NULL);

Where would I set the caps? After running the pipeline?

2

I think you’ll have to provide more context of that discussion. The code you pasted doesn’t do anything with caps.

Typically you would set the “caps” property on udpsrc to rtp caps with a clock-rate field.

3

Thanks. Eventually I “cheated” and used gst_parse_launch ( )
which does work
but I was not able to do that with this code:
The QT includes are blanked !?!: let me fiddle

#include <QApplication
#include <QWidget
#include <QSettings
#include <QTimer
#include <gst/gst.h>
#include <gst/video/videooverlay.h>

int main (int argc, char *argv[])
{
    // Initialize Qt
    QApplication app (argc, argv);

    // Initialize GStreamer
    gst_init (&argc, &argv);

    // Create the pipeline
    GstElement *pipeline = gst_pipeline_new ("pipeline");

    // Create UDP sources for video and audio
    GstElement *udpsrc_video = gst_element_factory_make ("udpsrc", NULL);
    g_object_set (udpsrc_video, "multicast-group", "224.1.1.1", NULL);
    g_object_set (udpsrc_video, "auto-multicast", TRUE, NULL);
    g_object_set (udpsrc_video, "port", 5002, NULL);
    g_object_set (udpsrc_video, "close-socket", FALSE, NULL);

    GstElement *udpsrc_audio = gst_element_factory_make ("udpsrc", NULL);
    g_object_set (udpsrc_audio, "multicast-group", "224.1.1.1", NULL);
    g_object_set (udpsrc_audio, "auto-multicast", TRUE, NULL);
    g_object_set (udpsrc_audio, "port", 5000, NULL);
    g_object_set (udpsrc_audio, "caps", gst_caps_from_string ("application/x-rtp, media=(string)audio,"
            " clock-rate=(int)8000, encoding-name=(string)PCMU, payload=(int)0, ssrc=(guint)1350777638,"
            " clock-base=(guint)2942119800, seqnum-base=(guint)47141"), NULL);

    // Create video processing elements
    GstElement *video_rtpjitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL);
    g_object_set (video_rtpjitterbuffer, "drop-on-latency", TRUE, NULL);

//    g_object_set(video_rtpjitterbuffer, "caps", gst_caps_from_string("application/x-rtp, encoding-name=(string)H264, media=(string)video, clock-rate=(int)90000"), NULL);

    GstElement *video_rtph264depay = gst_element_factory_make ("rtph264depay", NULL);
    GstElement *video_h264parse = gst_element_factory_make ("h264parse", NULL);
    GstElement *video_avdec_h264 = gst_element_factory_make ("avdec_h264", NULL);
    GstElement *videosink = gst_element_factory_make ("glimagesink", NULL);

    // Create audio processing elements
    GstElement *audio_queue = gst_element_factory_make ("queue", NULL);
    GstElement *audio_rtppcmudepay = gst_element_factory_make ("rtppcmudepay", NULL);
    GstElement *audio_mulawdec = gst_element_factory_make ("mulawdec", NULL);
    GstElement *audio_audioconvert = gst_element_factory_make ("audioconvert", NULL);
    GstElement *audio_audioresample = gst_element_factory_make ("audioresample", NULL);
    GstElement *audio_autoaudiosink = gst_element_factory_make ("autoaudiosink", NULL);

    // Add elements to the pipeline
    gst_bin_add_many (GST_BIN (pipeline),
        udpsrc_video, video_rtpjitterbuffer, video_rtph264depay, video_h264parse, video_avdec_h264,
        videosink, udpsrc_audio, audio_queue, audio_rtppcmudepay, audio_mulawdec, audio_audioconvert,
        audio_audioresample, audio_autoaudiosink, NULL);

    // Link elements
    gst_element_link_many (udpsrc_video, video_rtpjitterbuffer, video_rtph264depay, video_h264parse, video_avdec_h264,
        videosink, NULL);
    gst_element_link_many (udpsrc_audio, audio_queue, audio_rtppcmudepay, audio_mulawdec, audio_audioconvert,
        audio_audioresample, audio_autoaudiosink, NULL);

    QWidget window;
    QSettings settings;
    //restoreGeometry (settings.value ("geometry").toByteArray ());
    window.resize (1200, 720);
    window.show ();
    WId xwinid = window.winId ();
    gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (videosink), xwinid);

    // run the pipeline

    GstStateChangeReturn sret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
    if (sret == GST_STATE_CHANGE_FAILURE) {
        gst_element_set_state (pipeline, GST_STATE_NULL);
        gst_object_unref (pipeline);
        // Exit application
        QTimer::singleShot (0, QApplication::activeWindow (), SLOT (quit ()));
    }

//GstCaps* caps = gst_caps_new_simple("application/x-rtp", "clock-rate", G_TYPE_INT, 90000, NULL);
//gst_pad_set_caps(filter->srcpad, caps);
//gst_caps_unref(caps);

    int ret = app.exec ();
    //settings.setValue ("geometry", saveGeometry ());
    window.hide ();
    gst_element_set_state (pipeline, GST_STATE_NULL);
    gst_object_unref (pipeline);

    // Run the Qt application
    return app.exec ();
}

An extra query: I’ve got a rtpjitterbuf on the video pipe; ought I do one on the audio pipe too?

Thanks
James