From 138c2b7cf9b713b5d6e56ac7934e228703bfc9f5 Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Mon, 16 Jun 2008 07:03:58 +0000 Subject: gst/: More doc updates. More xrefs. Original commit message from CVS: * gst/deinterlace/gstdeinterlace.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/sdp/gstsdpdemux.c: More doc updates. More xrefs. --- gst/rtpmanager/gstrtpjitterbuffer.c | 32 +++++++++++++------------------- 1 file changed, 13 insertions(+), 19 deletions(-) (limited to 'gst/rtpmanager/gstrtpjitterbuffer.c') diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 8f876680..771684c8 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -26,34 +26,28 @@ /** * SECTION:element-gstrtpjitterbuffer * - * - * * This element reorders and removes duplicate RTP packets as they are received * from a network source. It will also wait for missing packets up to a - * configurable time limit using the ::latency property. Packets arriving too - * late are considered to be lost packets. - * - * - * This element acts as a live element and so adds ::latency to the pipeline. - * - * + * configurable time limit using the #GstRtpJitterBuffer:latency property. + * Packets arriving too late are considered to be lost packets. + * + * This element acts as a live element and so adds #GstRtpJitterBuffer:latency + * to the pipeline. + * * The element needs the clock-rate of the RTP payload in order to estimate the * delay. This information is obtained either from the caps on the sink pad or, - * when no caps are present, from the ::request-pt-map signal. To clear the - * previous pt-map use the ::clear-pt-map signal. - * - * + * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal. + * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal. + * * This element will automatically be used inside gstrtpbin. - * + * + * * Example pipelines - * - * + * |[ * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink - * - * Connect to a streaming server and decode the MPEG video. The jitterbuffer is + * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is * inserted into the pipeline to smooth out network jitter and to reorder the * out-of-order RTP packets. - * * * * Last reviewed on 2007-05-28 (0.10.5) -- cgit