docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-amrwb.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-bayer.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdaudio.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-dvdspu.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-fbdevsink.xml:
* docs/plugins/inspect/plugin-festival.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-flvdemux.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
* docs/plugins/inspect/plugin-mpegtsparse.xml:
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-mve.xml:
* docs/plugins/inspect/plugin-mythtv.xml
* docs/plugins/inspect/plugin-nas.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
* docs/plugins/inspect/plugin-oss4.xml
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-real.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rfbsrc.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-sdp.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-sndfile.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-speexresample.xml:
* docs/plugins/inspect/plugin-stereo.xml:
* docs/plugins/inspect/plugin-subenc.xml
* docs/plugins/inspect/plugin-timidity.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-vcdsrc.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-vmnc.xml:
* docs/plugins/inspect/plugin-wildmidi.xml:
* docs/plugins/inspect/plugin-x264.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/dc1394/gstdc1394.c:
* ext/directfb/dfbvideosink.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mplex/gstmplex.cc:
* ext/musicbrainz/gsttrm.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst-libs/gst/app/gstappsink.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/dvdspu/gstdvdspu.c:
* gst/festival/gstfestival.c:
* gst/freeze/gstfreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c:
Add missing elements to docs. Fix doc-markup: use convinience syntax
for examples (produces valid docbook), add several refsec2 when we
have several titles. Fix some types.

8 files changed, 52 insertions, 81 deletions

diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c
index 73159b64..da4b978b 100644
--- a/gst/deinterlace/gstdeinterlace.c
+++ b/gst/deinterlace/gstdeinterlace.c
@@ -31,19 +31,16 @@
 /**
  * SECTION:element-deinterlace
  *
- * <refsect2>
- * <para>
  * Adaptively deinterlaces video frames by detecting interlacing artifacts.
  * An edge detection matrix is used, with a threshold value. Pixels detected
  * as 'interlaced' are replaced with pixels blended from the pixels above and
  * below.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
  * gst-launch -v videotestsrc ! deinterlace ! ffmpegcolorspace ! xvimagesink
- * </programlisting>
- * </para>
+ * ]|
  * </refsect2>
  */
 
diff --git a/gst/dvdspu/gstdvdspu.c b/gst/dvdspu/gstdvdspu.c
index 45b24d4f..c16743bd 100644
--- a/gst/dvdspu/gstdvdspu.c
+++ b/gst/dvdspu/gstdvdspu.c
@@ -16,20 +16,16 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
-/*
+/**
  * SECTION:element-dvdspu
  *
- * <refsect2>
- * <para>
  * DVD sub picture overlay element.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
  * FIXME: gst-launch ...
- * </programlisting>
- * FIXME: description for the sample launch pipeline
- * </para>
+ * ]| FIXME: description for the sample launch pipeline
  * </refsect2>
  */
 #ifdef HAVE_CONFIG_H
diff --git a/gst/festival/gstfestival.c b/gst/festival/gstfestival.c
index 7a147c95..f6f0a7b0 100644
--- a/gst/festival/gstfestival.c
+++ b/gst/festival/gstfestival.c
@@ -63,16 +63,15 @@
 /**
  * SECTION:element-festival
  * 
- * <refsect2>
- * <para>
  * This element connects to a
  * <ulink url="http://www.festvox.org/festival/index.html">festival</ulink> server 
  * process and uses it to synthesize speech.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example pipeline</title>
- * <programlisting>
+ * |[
  * echo "hi" | gst-launch fdsrc fd=0 ! festival ! wavparse ! audioconvert ! alsasink
- * </programlisting>
+ * ]|
  * </refsect2>
  */
 
diff --git a/gst/freeze/gstfreeze.c b/gst/freeze/gstfreeze.c
index 23c6efdf..d5549883 100644
--- a/gst/freeze/gstfreeze.c
+++ b/gst/freeze/gstfreeze.c
@@ -18,16 +18,14 @@
 /**
  * SECTION:element-freeze
  *
- * <refsect2>
- * <para>
  * Makes a stream from buffers of data.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example launch line</title>
  * <para>
- * <programlisting>
- * gst-launch-0.10 --gst-debug=*freeze*:5 -mvt filesrc location=gnome-home.png blocksize=4135 !  pngdec ! freeze ! ffmpegcolorspace ! xvimagesink 
- * </programlisting>
- * Produces a video stream from one picture.
+ * |[
+ * gst-launch-0.10 filesrc location=gnome-home.png blocksize=4135 !  pngdec ! freeze ! ffmpegcolorspace ! xvimagesink
+ * ]| Produces a video stream from one picture.
  * </para>
  * </refsect2>
  */
diff --git a/gst/interleave/deinterleave.c b/gst/interleave/deinterleave.c
index 6f2b0cef..4c81d39d 100644
--- a/gst/interleave/deinterleave.c
+++ b/gst/interleave/deinterleave.c
@@ -32,35 +32,29 @@
 
 /**
  * SECTION:element-deinterleave
+ * @see_also: interleave
  *
- * <refsect2>
- * <para>
  * Splits one interleaved multichannel audio stream into many mono audio streams.
- * </para>
- * <para>
+ * 
  * This element handles all raw audio formats and supports changing the input caps as long as
  * all downstream elements can handle the new caps and the number of channels and the channel
  * positions stay the same. This restriction will be removed in later versions by adding or
  * removing some source pads as required.
- * </para>
- * <para>
+ * 
  * In most cases a queue and an audioconvert element should be added after each source pad
  * before further processing of the audio data.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
  * gst-launch-0.10 filesrc location=/path/to/file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2 ! deinterleave name=d  d.src0 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel1.ogg  d.src1 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel2.ogg
- * </programlisting>
- * Decodes an MP3 file and encodes the left and right channel into separate Ogg Vorbis files.
- * </para>
- * <para>
- * <programlisting>
+ * ]| Decodes an MP3 file and encodes the left and right channel into separate
+ * Ogg Vorbis files.
+ * |[
  * gst-launch-0.10 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2" ! deinterleave name=d  interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav    d.src0 ! queue ! audioconvert ! i.sink1    d.src1 ! queue ! audioconvert ! i.sink0
- * </programlisting>
- * Decodes and deinterleaves a Stereo MP3 file into separate channels and then interleaves the channels
- * again to a WAV file with the channel with the channels exchanged.
- * </para>
+ * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and
+ * then interleaves the channels again to a WAV file with the channel with the
+ * channels exchanged.
  * </refsect2>
  */
 
diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c
index 76843e45..831e928f 100644
--- a/gst/interleave/interleave.c
+++ b/gst/interleave/interleave.c
@@ -30,38 +30,30 @@
 
 /**
  * SECTION:element-interleave
+ * @see_also: deinterleave
  *
- * <refsect2>
- * <para>
  * Merges separate mono inputs into one interleaved stream.
- * </para>
- * <para>
+ * 
  * This element handles all raw floating point sample formats and all signed integer sample formats. The first
  * caps on one of the sinkpads will set the caps of the output so usually an audioconvert element should be
  * placed before every sinkpad of interleave.
- * </para>
- * <para>
+ * 
  * It's possible to change the number of channels while the pipeline is running by adding or removing
  * some of the request pads but this will change the caps of the output buffers. Changing the input
  * caps is _not_ supported yet.
- * </para>
- * <para>
+ * 
  * The channel number of every sinkpad in the out can be retrieved from the "channel" property of the pad.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
  * gst-launch-0.10 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2" ! deinterleave name=d  interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav    d.src0 ! queue ! audioconvert ! i.sink1    d.src1 ! queue ! audioconvert ! i.sink0
- * </programlisting>
- * Decodes and deinterleaves a Stereo MP3 file into separate channels and then interleaves the channels
- * again to a WAV file with the channel with the channels exchanged.
- * </para>
- * <para>
- * <programlisting>
+ * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and
+ * then interleaves the channels again to a WAV file with the channel with the
+ * channels exchanged.
+ * |[
  * gst-launch-0.10 interleave name=i ! audioconvert ! wavenc ! filesink location=file.wav  filesrc location=file1.wav ! decodebin ! audioconvert ! "audio/x-raw-int,channels=1" ! queue ! i.sink0   filesrc location=file2.wav ! decodebin ! audioconvert ! "audio/x-raw-int,channels=1" ! queue ! i.sink1
- * </programlisting>
- * Interleaves two Mono WAV files to a single Stereo WAV file.
- * </para>
+ * ]| Interleaves two Mono WAV files to a single Stereo WAV file.
  * </refsect2>
  */
 
diff --git a/gst/modplug/gstmodplug.cc b/gst/modplug/gstmodplug.cc
index fea87230..6f6e1869 100644
--- a/gst/modplug/gstmodplug.cc
+++ b/gst/modplug/gstmodplug.cc
@@ -28,15 +28,14 @@
 /**
  * SECTION:element-modplug
  * 
- * <refsect2>
- * <para>
  * Modplug uses the <ulink url="http://modplug-xmms.sourceforge.net/">modplug</ulink>
  * library to decode tracked music in the MOD/S3M/XM/IT and related formats.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example pipeline</title>
- * <programlisting>
+ * |[
  * gst-launch -v filesrc location=1990s-nostalgia.xm ! modplug ! audioconvert ! alsasink
- * </programlisting>
+ * ]| Play a FastTracker xm file.
  * </refsect2>
  */
 
diff --git a/gst/nuvdemux/gstnuvdemux.c b/gst/nuvdemux/gstnuvdemux.c
index 70f4f496..27fa3967 100644
--- a/gst/nuvdemux/gstnuvdemux.c
+++ b/gst/nuvdemux/gstnuvdemux.c
@@ -21,25 +21,21 @@
 
 /**
  * SECTION:element-nuvdemux
+ * @see_also: mythtvsrc
  *
- * <refsect2>
- * <para>
  * Demuxes MythTVs NuppelVideo .nuv file into raw or compressed audio and/or
  * video streams.
- * </para>
- * <para>
+ * 
  * This element currently only supports pull-based scheduling.
- * </para>
+ * 
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
- * <programlisting>
+ * |[
  * gst-launch filesrc test.nuv ! nuvdemux name=demux  demux.audio_00 ! decodebin ! audioconvert ! audioresample ! autoaudiosink   demux.video_00 ! queue ! decodebin ! ffmpegcolorspace ! videoscale ! autovideosink
- * </programlisting>
- * Play (parse and decode) an .nuv file and try to output it to
+ * ]| Play (parse and decode) an .nuv file and try to output it to
  * an automatically detected soundcard and videosink. If the NUV file contains
  * compressed audio or video data, this will only work if you have the
  * right decoder elements/plugins installed.
- * </para>
  * </refsect2>
  *
  */