docs: Port all docstring to gtk-doc markdown

ext/a52dec/gsta52dec.c

@@ -19,18 +19,18 @@
 
 /**
  * SECTION:element-a52dec
+ * @title: a52dec
  *
  * Dolby Digital (AC-3) audio decoder.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioconvert ! audioresample ! autoaudiosink
  * ]| Play audio part of a dvd title.
  * |[
  * gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
  * ]| Decode and play a stand alone AC-3 file.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

ext/amrnb/amrnbdec.c

@@ -19,17 +19,17 @@
 
 /**
  * SECTION:element-amrnbdec
+ * @title: amrnbdec
  * @see_also: #GstAmrnbEnc, #GstAmrParse
  *
  * AMR narrowband decoder based on the
  * <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrnbdec ! audioconvert ! audioresample ! autoaudiosink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

ext/amrnb/amrnbenc.c

@@ -19,19 +19,19 @@
 
 /**
  * SECTION:element-amrnbenc
+ * @title: amrnbenc
  * @see_also: #GstAmrnbDec, #GstAmrnbParse
  *
  * AMR narrowband encoder based on the
  * <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
  * ]|
  * Please note that the above stream misses the header, that is needed to play
  * the stream.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

ext/amrwbdec/amrwbdec.c

@@ -19,17 +19,17 @@
 
 /**
  * SECTION:element-amrwbdec
+ * @title: amrwbdec
  * @see_also: #GstAmrwbEnc
  *
  * AMR wideband decoder based on the
  * <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

ext/cdio/gstcdiocddasrc.c

@@ -19,38 +19,29 @@
 
 /**
  * SECTION:element-cdiocddasrc
+ * @title: cdiocddasrc
  * @see_also: GstCdParanoiaSrc, GstAudioCdSrc
  *
- * <refsect2>
- * <para>
  * cdiocddasrc reads and extracts raw audio from Audio CDs. It can operate
  * in one of two modes:
- * <itemizedlist>
+ *
- *  <listitem><para>
+ *  * treat each track as a separate stream, counting time from the start
- *    treat each track as a separate stream, counting time from the start
  *    of the track to the end of the track and posting EOS at the end of
  *    a track, or
- *  </para></listitem>
+ *  * treat the entire disc as one stream, counting time from the start of
- *  <listitem><para>
- *    treat the entire disc as one stream, counting time from the start of
  *    the first track to the end of the last track, posting EOS only at
  *    the end of the last track.
- *  </para></listitem>
+ *
- * </itemizedlist>
- * </para>
- * <para>
  * With a recent-enough version of libcdio, the element will extract
  * CD-TEXT if this is supported by the CD-drive and CD-TEXT information
  * is available on the CD. The information will be posted on the bus in
  * form of a tag message.
- * </para>
+ *
- * <para>
  * When opened, the element will also calculate a CDDB disc ID and a
  * MusicBrainz disc ID, which applications can use to query online
  * databases for artist/title information. These disc IDs will also be
  * posted on the bus as part of the tag messages.
- * </para>
+ *
- * <para>
  * cdiocddasrc supports the GstUriHandler interface, so applications can use
  * playbin with cdda://&lt;track-number&gt; URIs for playback (they will have
  * to connect to playbin's notify::source signal and set the device on the
@@ -58,16 +49,15 @@
  * Applications should use seeks in "track" format to switch between different
  * tracks of the same CD (passing a new cdda:// URI to playbin involves opening
  * and closing the CD device, which is much slower).
- * </para>
+ *
- * <title>Example launch line</title>
+ * ## Example launch line
- * <para>
+ *
- * <programlisting>
+ * |[
  * gst-launch-1.0 cdiocddasrc track=5 device=/dev/cdrom ! audioconvert ! vorbisenc ! oggmux ! filesink location=track5.ogg
- * </programlisting>
+ * ]|
  * This pipeline extracts track 5 of the audio CD and encodes it into an
  * Ogg/Vorbis file.
- * </para>
+ *
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H

ext/x264/gstx264enc.c

@@ -22,6 +22,7 @@
 
 /**
  * SECTION:element-x264enc
+ * @title: x264enc
  * @see_also: faac
  *
  * This element encodes raw video into H264 compressed data,
@@ -49,21 +50,19 @@
  * applied, followed by the user-set properties, fast first pass restrictions and
  * finally the profile restrictions.
  *
- * <note>Some settings, including the default settings, may lead to quite
+ * > Some settings, including the default settings, may lead to quite
- * some latency (i.e. frame buffering) in the encoder. This may cause problems
+ * > some latency (i.e. frame buffering) in the encoder. This may cause problems
- * with pipeline stalling in non-trivial pipelines, because the encoder latency
+ * > with pipeline stalling in non-trivial pipelines, because the encoder latency
- * is often considerably higher than the default size of a simple queue
+ * > is often considerably higher than the default size of a simple queue
- * element. Such problems are caused by one of the queues in the other
+ * > element. Such problems are caused by one of the queues in the other
- * non-x264enc streams/branches filling up and blocking upstream. They can
+ * > non-x264enc streams/branches filling up and blocking upstream. They can
- * be fixed by relaxing the default time/size/buffer limits on the queue
+ * > be fixed by relaxing the default time/size/buffer limits on the queue
- * elements in the non-x264 branches, or using a (single) multiqueue element
+ * > elements in the non-x264 branches, or using a (single) multiqueue element
- * for all branches. Also see the last example below. You can also work around
+ * > for all branches. Also see the last example below. You can also work around
- * this problem by setting the tune=zerolatency property, but this will affect
+ * > this problem by setting the tune=zerolatency property, but this will affect
- * overall encoding quality so may not be appropriate for your use case.
+ * > overall encoding quality so may not be appropriate for your use case.
- * </note>
  *
- * <refsect2>
+ * ## Example pipeline
- * <title>Example pipeline</title>
  * |[
  * gst-launch-1.0 -v videotestsrc num-buffers=1000 ! x264enc qp-min=18 ! \
  *   avimux ! filesink location=videotestsrc.avi
@@ -92,7 +91,7 @@
  * specific settings are needed in this case to avoid pipeline stalling.
  * Depending on goals and context, other approaches are possible, e.g.
  * tune=zerolatency might be configured, or queue sizes increased.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

gst/asfdemux/gstasfdemux.h

@@ -70,7 +70,7 @@ enum _GstASF3DMode
   GST_ASF_3D_SIDE_BY_SIDE_HALF_RL = 0x02,
   GST_ASF_3D_TOP_AND_BOTTOM_HALF_LR = 0x03,
   GST_ASF_3D_TOP_AND_BOTTOM_HALF_RL = 0x04,
-  GST_ASF_3D_DUAL_STREAM = 0x0D,                  /**< Full format*/
+  GST_ASF_3D_DUAL_STREAM = 0x0D,                  /*< Full format*/
 };
 
 typedef struct

gst/asfdemux/gstrtspwms.c

@@ -20,6 +20,7 @@
 
 /**
  * SECTION:element-rtspwms
+ * @title: rtspwms
  *
  * A WMS RTSP extension
  */

gst/realmedia/gstrdtbuffer.h

@@ -26,7 +26,6 @@
 #include <gst/gst.h>
 
 G_BEGIN_DECLS
-
 /**
  * GstRDTType:
  * @GST_RDT_TYPE_INVALID:

gst/realmedia/rademux.c

@@ -19,11 +19,11 @@
 
 /**
  * SECTION:element-rademux
+ * @title: rademux
  *
  * Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=interview.ra ! rademux ! avdec_real_288 ! audioconvert ! audioresample ! autoaudiosink
  * ]| Read a RealAudio file and decode it and output it to the soundcard using
@@ -32,7 +32,7 @@
  * gst-launch-1.0 souphttpsrc location=http://www.example.org/interview.ra ! rademux ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
  * ]| Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
  * and output it to the soundcard.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

gst/realmedia/rdtmanager.c

@@ -44,6 +44,7 @@
 
 /**
  * SECTION:element-rdtmanager
+ * @title: rdtmanager
  * @see_also: GstRtspSrc
  *
  * A simple RTP session manager used internally by rtspsrc.

gst/realmedia/rtspreal.c

@@ -20,6 +20,7 @@
 
 /**
  * SECTION:element-rtspreal
+ * @title: rtspreal
  *
  * A RealMedia RTSP extension
  */

gst/xingmux/gstxingmux.c

@@ -23,6 +23,7 @@
 
 /**
  * SECTION:element-xingmux
+ * @title: xingmux
  *
  * xingmux adds a Xing header to MP3 files. This contains information about the duration and size
  * of the file and a seek table and is very useful for getting an almost correct duration and better
@@ -30,14 +31,13 @@
  *
  * This element will remove any existing Xing, LAME or VBRI headers from the beginning of the file.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 audiotestsrc num-buffers=1000 ! audioconvert ! lamemp3enc ! xingmux ! filesink location=test.mp3
  * gst-launch-1.0 filesrc location=test.mp3 ! xingmux ! filesink location=test2.mp3
  * gst-launch-1.0 filesrc location=test.mp3 ! mp3parse ! xingmux ! filesink location=test2.mp3
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H