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| author | Stefan Kost <ensonic@users.sourceforge.net> | 2008-06-16 07:03:58 +0000 |
|---|---|---|
| committer | Stefan Kost <ensonic@users.sourceforge.net> | 2008-06-16 07:03:58 +0000 |
| commit | 68993006c374a92474db6e908f7d3c13981bfed9 (patch) | |
| tree | c4ff9daee109a4282c1bfffdbc0bed8e362a48ed /gst/rtpmanager/gstrtpjitterbuffer.c | |
| parent | bb7f93bd4eee6e78378e2157c6448e28dda717e1 (diff) | |
| download | gst-plugins-bad-68993006c374a92474db6e908f7d3c13981bfed9.tar.gz gst-plugins-bad-68993006c374a92474db6e908f7d3c13981bfed9.tar.bz2 gst-plugins-bad-68993006c374a92474db6e908f7d3c13981bfed9.zip | |
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.
Diffstat (limited to 'gst/rtpmanager/gstrtpjitterbuffer.c')
| -rw-r--r-- | gst/rtpmanager/gstrtpjitterbuffer.c | 32 |
1 files changed, 13 insertions, 19 deletions
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 * - * <refsect2> - * <para> * 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. - * </para> - * <para> - * This element acts as a live element and so adds ::latency to the pipeline. - * </para> - * <para> + * 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. - * </para> - * <para> + * 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. - * </para> + * + * <refsect2> * <title>Example pipelines</title> - * <para> - * <programlisting> + * |[ * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink - * </programlisting> - * 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. - * </para> * </refsect2> * * Last reviewed on 2007-05-28 (0.10.5) |