From a99d3f8769ed3fd1266d5216ecefebfd1bdcf663 Mon Sep 17 00:00:00 2001
From: Stefan Kost <ensonic@users.sf.net>
Date: Wed, 28 Jan 2009 12:29:42 +0200
Subject: 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.
---
 gst/rtsp/gstrtpdec.c  | 12 ++++--------
 gst/rtsp/gstrtspsrc.c | 27 ++++++++++-----------------
 2 files changed, 14 insertions(+), 25 deletions(-)

(limited to 'gst/rtsp')

diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c
index 0adfc7aa..c6a942e4 100644
--- a/gst/rtsp/gstrtpdec.c
+++ b/gst/rtsp/gstrtpdec.c
@@ -44,11 +44,7 @@
 /**
  * SECTION:element-rtpdec
  *
- * <refsect2>
- * <para>
  * A simple RTP session manager used internally by rtspsrc.
- * </para>
- * </refsect2>
  *
  * Last reviewed on 2006-06-20 (0.10.4)
  */
@@ -252,8 +248,8 @@ gst_rtp_dec_marshal_BOXED__UINT_UINT (GClosure * closure,
     data2 = closure->data;
   }
   callback =
-      (GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data : cc->
-      callback);
+      (GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data :
+      cc->callback);
 
   v_return = callback (data1,
       g_marshal_value_peek_uint (param_values + 1),
@@ -285,8 +281,8 @@ gst_rtp_dec_marshal_VOID__UINT_UINT (GClosure * closure,
     data2 = closure->data;
   }
   callback =
-      (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data : cc->
-      callback);
+      (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data :
+      cc->callback);
 
   callback (data1,
       g_marshal_value_peek_uint (param_values + 1),
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)
-- 
cgit