}
static void
+gst_rtp_src_handle_message (GstBin * bin, GstMessage * message)
+{
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_STREAM_START:
+ case GST_MESSAGE_EOS:
+ /* drop stream-start & eos from our internal udp sink(s);
+ https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1368 */
+ gst_message_unref (message);
+ break;
+ default:
+ GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+ break;
+ }
+}
+
+static void
gst_rtp_src_class_init (GstRtpSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+ GstBinClass *gstbin_class = GST_BIN_CLASS (klass);
gobject_class->set_property = gst_rtp_src_set_property;
gobject_class->get_property = gst_rtp_src_get_property;
gobject_class->finalize = gst_rtp_src_finalize;
gstelement_class->change_state = gst_rtp_src_change_state;
+ gstbin_class->handle_message = gst_rtp_src_handle_message;
/**
* GstRtpSrc:uri: