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authorStefan Kost <ensonic@users.sf.net>2009-01-28 12:29:42 +0200
committerStefan Kost <ensonic@users.sf.net>2009-01-28 12:32:59 +0200
commita99d3f8769ed3fd1266d5216ecefebfd1bdcf663 (patch)
tree4a5cf5e0f2f44b1f9ccea5344c38ef98f0a92990 /gst/rtsp/gstrtspsrc.c
parent00fdca0c14eb9a5fe6b8b9f2d5ce2313e3b32f23 (diff)
Update and add documentation for plugins with no deps (gst).
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
Diffstat (limited to 'gst/rtsp/gstrtspsrc.c')
-rw-r--r--gst/rtsp/gstrtspsrc.c27
1 files changed, 10 insertions, 17 deletions
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index 79455cd1..e677c991 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -43,43 +43,36 @@
/**
* SECTION:element-rtspsrc
*
- * <refsect2>
- * <para>
* Makes a connection to an RTSP server and read the data.
* rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
* RealMedia/Quicktime/Microsoft extensions.
- * </para>
- * <para>
+ *
* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
* default rtspsrc will negotiate a connection in the following order:
* UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
- * protocols can be controlled with the "protocols" property.
- * </para>
- * <para>
+ * protocols can be controlled with the #GstRTSPSrc:protocols property.
+ *
* rtspsrc currently understands SDP as the format of the session description.
* For each stream listed in the SDP a new rtp_stream%d pad will be created
* with caps derived from the SDP media description. This is a caps of mime type
* "application/x-rtp" that can be connected to any available RTP depayloader
* element.
- * </para>
- * <para>
+ *
* rtspsrc will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal,
* packet reordering along with providing a clock for the pipeline.
* This feature is currently fully implemented with the gstrtpbin in the
* gst-plugins-bad module.
- * </para>
- * <para>
+ *
* rtspsrc acts like a live source and will therefore only generate data in the
* PLAYING state.
- * </para>
+ *
+ * <refsect2>
* <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
- * </programlisting>
- * Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.
- * </para>
+ * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
+ * fakesink.
* </refsect2>
*
* Last reviewed on 2006-08-18 (0.10.5)