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Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs alrea...

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Original commit message from CVS:
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
* ext/a52dec/gsta52dec.c:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c:
* ext/amrnb/amrnbparse.c:
* ext/lame/gstlame.c:
* ext/mad/gstmad.c:
* ext/sidplay/gstsiddec.cc:
* gst/asfdemux/gstrtspwms.c:
* gst/mpegaudioparse/gstxingmux.c:
* gst/realmedia/rademux.c:
* gst/realmedia/rdtmanager.c:
* gst/realmedia/rtspreal.c:
* gst/synaesthesia/gstsynaesthesia.c:
Add missing elements to docs. Restore alphabetical order in section
file. Document mad (it was included in docs already).
Fix doc-markup: use convinience syntax for examples
(produces valid docbook), add several refsec2 when we have several
titles. Fix some types.
This commit is contained in:
Stefan Kost 2008-06-13 06:57:21 +00:00
parent 81e36c292e
commit c49cf83ee3
16 changed files with 124 additions and 149 deletions
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23
ChangeLog
View file

@ -1,3 +1,26 @@
2008-06-13 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
* ext/a52dec/gsta52dec.c:
* ext/amrnb/amrnbdec.c:
* ext/amrnb/amrnbenc.c:
* ext/amrnb/amrnbparse.c:
* ext/lame/gstlame.c:
* ext/mad/gstmad.c:
* ext/sidplay/gstsiddec.cc:
* gst/asfdemux/gstrtspwms.c:
* gst/mpegaudioparse/gstxingmux.c:
* gst/realmedia/rademux.c:
* gst/realmedia/rdtmanager.c:
* gst/realmedia/rtspreal.c:
* gst/synaesthesia/gstsynaesthesia.c:
Add missing elements to docs. Restore alphabetical order in section
file. Document mad (it was included in docs already).
Fix doc-markup: use convinience syntax for examples
(produces valid docbook), add several refsec2 when we have several
titles. Fix some types.
2008-06-13 Stefan Kost <ensonic@users.sf.net> 2008-06-13 Stefan Kost <ensonic@users.sf.net>
* ext/lame/gstlame.c: * ext/lame/gstlame.c:

4
docs/plugins/gst-plugins-ugly-plugins-docs.sgml
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@ -18,6 +18,10 @@
<xi:include href="xml/element-amrnbparse.xml" /> <xi:include href="xml/element-amrnbparse.xml" />
<xi:include href="xml/element-lame.xml" /> <xi:include href="xml/element-lame.xml" />
<xi:include href="xml/element-mad.xml" /> <xi:include href="xml/element-mad.xml" />
<xi:include href="xml/element-rademux.xml" />
<xi:include href="xml/element-rdtmanager.xml" />
<xi:include href="xml/element-rtspreal.xml" />
<xi:include href="xml/element-rtspwms.xml" />
<xi:include href="xml/element-siddec.xml" /> <xi:include href="xml/element-siddec.xml" />
<xi:include href="xml/element-synaesthesia.xml" /> <xi:include href="xml/element-synaesthesia.xml" />
<xi:include href="xml/element-xingmux.xml" /> <xi:include href="xml/element-xingmux.xml" />

50
docs/plugins/gst-plugins-ugly-plugins-sections.txt
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@ -54,6 +54,20 @@ GST_TYPE_AMRNBPARSE
gst_amrnbparse_get_type gst_amrnbparse_get_type
</SECTION> </SECTION>
<SECTION>
<FILE>element-lame</FILE>
<TITLE>lame</TITLE>
GstLame
<SUBSECTION Standard>
GstLameClass
GST_LAME
GST_LAME_CLASS
GST_IS_LAME
GST_IS_LAME_CLASS
GST_TYPE_LAME
gst_lame_get_type
</SECTION>
<SECTION> <SECTION>
<FILE>element-mad</FILE> <FILE>element-mad</FILE>
<TITLE>mad</TITLE> <TITLE>mad</TITLE>
@ -72,17 +86,18 @@ gst_mad_tag_list_to_id3_tag
</SECTION> </SECTION>
<SECTION> <SECTION>
<FILE>element-lame</FILE> <FILE>element-rademux</FILE>
<TITLE>lame</TITLE> <TITLE>rademux</TITLE>
GstLame GstRealAudioDemux
<SUBSECTION Standard> <SUBSECTION Standard>
GstLameClass GstRealAudioDemuxClass
GST_LAME GstRealAudioDemuxState
GST_LAME_CLASS GST_REAL_AUDIO_DEMUX
GST_IS_LAME GST_REAL_AUDIO_DEMUX_CLASS
GST_IS_LAME_CLASS GST_IS_REAL_AUDIO_DEMUX
GST_TYPE_LAME GST_IS_REAL_AUDIO_DEMUX_CLASS
gst_lame_get_type GST_TYPE_REAL_AUDIO_DEMUX
gst_real_audio_demux_get_type
</SECTION> </SECTION>
<SECTION> <SECTION>
@ -101,21 +116,6 @@ gst_rdt_manager_get_type
gst_rdt_manager_plugin_init gst_rdt_manager_plugin_init
</SECTION> </SECTION>
<SECTION>
<FILE>element-rademux</FILE>
<TITLE>rademux</TITLE>
GstRealAudioDemux
<SUBSECTION Standard>
GstRealAudioDemuxClass
GstRealAudioDemuxState
GST_REAL_AUDIO_DEMUX
GST_REAL_AUDIO_DEMUX_CLASS
GST_IS_REAL_AUDIO_DEMUX
GST_IS_REAL_AUDIO_DEMUX_CLASS
GST_TYPE_REAL_AUDIO_DEMUX
gst_real_audio_demux_get_type
</SECTION>
<SECTION> <SECTION>
<FILE>element-rtspreal</FILE> <FILE>element-rtspreal</FILE>
<TITLE>rtspreal</TITLE> <TITLE>rtspreal</TITLE>

19
ext/a52dec/gsta52dec.c
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@ -20,25 +20,16 @@
/** /**
* SECTION:element-a52dec * SECTION:element-a52dec
* *
* <refsect2>
* <para>
* Dolby Digital (AC-3) audio decoder. * Dolby Digital (AC-3) audio decoder.
* </para> *
* </refsect2>
* <refsect2> * <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch dvdreadsrc title=1 ! dvddemux ! a52dec ! audioresample ! audioconvert ! alsasink * gst-launch dvdreadsrc title=1 ! dvddemux ! a52dec ! audioresample ! audioconvert ! alsasink
* </programlisting> * ]| Play audio track from a dvd.
* Play audio track from a dvd. * |[
* </para>
* <para>
* <programlisting>
* gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink * gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink
* </programlisting> * ]| Decode a stand alone file and play it.
* Decode a stand alone file and play it.
* </para>
* </refsect2> * </refsect2>
*/ */

13
ext/amrnb/amrnbdec.c
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@ -21,17 +21,14 @@
* SECTION:element-amrnbdec * SECTION:element-amrnbdec
* @see_also: #GstAmrnbEnc, #GstAmrnbParse * @see_also: #GstAmrnbEnc, #GstAmrnbParse
* *
* <refsect2> * AMR narrowband decoder based on the
* <para>
* This is an AMR narrowband decoder based on the
* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>. * <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
* </para> *
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink * gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink
* </programlisting> * ]|
* </para>
* </refsect2> * </refsect2>
*/ */

15
ext/amrnb/amrnbenc.c
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@ -21,18 +21,15 @@
* SECTION:element-amrnbenc * SECTION:element-amrnbenc
* @see_also: #GstAmrnbDec, #GstAmrnbParse * @see_also: #GstAmrnbDec, #GstAmrnbParse
* *
* <refsect2> * AMR narrowband encoder based on the
* <para>
* This is an AMR narrowband encoder based on the
* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>. * <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
* </para> *
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr
* </programlisting> * ]|
* </para> * Please note that the above stream misses the header, that is needed to play
* Please not that the above stream misses the header, that is needed to play
* the stream. * the stream.
* </refsect2> * </refsect2>
*/ */

11
ext/amrnb/amrnbparse.c
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@ -21,16 +21,13 @@
* SECTION:element-amrnbparse * SECTION:element-amrnbparse
* @see_also: #GstAmrnbDec, #GstAmrnbEnc * @see_also: #GstAmrnbDec, #GstAmrnbEnc
* *
* AMR narrowband bitstream parser.
*
* <refsect2> * <refsect2>
* <para>
* This is an AMR narrowband parser.
* </para>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink * gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink
* </programlisting> * ]|
* </para>
* </refsect2> * </refsect2>
*/ */

48
ext/lame/gstlame.c
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@ -23,61 +23,45 @@
* SECTION:element-lame * SECTION:element-lame
* @see_also: mad, vorbisenc * @see_also: mad, vorbisenc
* *
* <refsect2>
* <para>
* This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream. * This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream.
* Note that <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not * Note that <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not
* a free format, there are licensing and patent issues to take into * a free format, there are licensing and patent issues to take into
* consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink> * consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
* for a royalty free (and often higher quality) alternative. * for a royalty free (and often higher quality) alternative.
* </para> *
* <refsect2>
* <title>Output sample rate</title> * <title>Output sample rate</title>
* <para>
* If no fixed output sample rate is negotiated on the element's src pad, * If no fixed output sample rate is negotiated on the element's src pad,
* the element will choose an optimal sample rate to resample to internally. * the element will choose an optimal sample rate to resample to internally.
* For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will * For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
* get resampled to 32 KHz. Use filter caps on the src pad to force a * get resampled to 32 KHz. Use filter caps on the src pad to force a
* particular sample rate. * particular sample rate.
* </para> * </refsect2>
* <refsect2>
* <title>Writing metadata (tags)</title> * <title>Writing metadata (tags)</title>
* <para>
* Whilst the lame encoder element does claim to implement the GstTagSetter * Whilst the lame encoder element does claim to implement the GstTagSetter
* interface, it does so only for backwards compatibility reasons. Tag writing * interface, it does so only for backwards compatibility reasons. Tag writing
* has been removed from lame. Use external elements like id3v2mux or apev2mux * has been removed from lame. Use external elements like id3v2mux or apev2mux
* to add tags to your MP3 streams. The same goes for XING headers: use the * to add tags to your MP3 streams. The same goes for XING headers: use the
* xingmux element to add XING headers to your VBR mp3 file. * xingmux element to add XING headers to your VBR mp3 file.
* </para> * </refsect2>
* <refsect2>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * |[
* Encode a test sine signal to MP3.
* </para>
* <programlisting>
* gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lame ! filesink location=sine.mp3 * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lame ! filesink location=sine.mp3
* </programlisting> * ]| Encode a test sine signal to MP3.
* <para> * |[
* Record from a sound card using ALSA and encode to MP3
* </para>
* <programlisting>
* gst-launch -v alsasrc ! audioconvert ! lame bitrate=192 ! filesink location=alsasrc.mp3 * gst-launch -v alsasrc ! audioconvert ! lame bitrate=192 ! filesink location=alsasrc.mp3
* </programlisting> * ]| Record from a sound card using ALSA and encode to MP3
* <para> * |[
* Transcode from a .wav file to MP3 (the id3v2mux element is optional):
* </para>
* <programlisting>
* gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lame bitrate=192 ! id3v2mux ! filesink location=music.mp3 * gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lame bitrate=192 ! id3v2mux ! filesink location=music.mp3
* </programlisting> * ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional)
* <para> * |[
* Encode Audio CD track 5 to MP3:
* </para>
* <programlisting>
* gst-launch -v cdda://5 ! audioconvert ! lame bitrate=192 ! filesink location=track5.mp3 * gst-launch -v cdda://5 ! audioconvert ! lame bitrate=192 ! filesink location=track5.mp3
* </programlisting> * ]| Encode Audio CD track 5 to MP3
* <para> * |[
* Encode to a fixed sample rate:
* </para>
* <programlisting>
* gst-launch -v audiotestsrc num-buffers=10 ! audio/x-raw-int,rate=44100,channels=1 ! lame bitrate=48 mode=3 ! filesink location=test.mp3 * gst-launch -v audiotestsrc num-buffers=10 ! audio/x-raw-int,rate=44100,channels=1 ! lame bitrate=48 mode=3 ! filesink location=test.mp3
* </programlisting> * ]| Encode to a fixed sample rate
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-07-24 (0.10.7) * Last reviewed on 2007-07-24 (0.10.7)

14
ext/mad/gstmad.c
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@ -17,6 +17,20 @@
* Boston, MA 02111-1307, USA. * Boston, MA 02111-1307, USA.
*/ */
/**
* SECTION:element-mad
* @see_also: lame
*
* MP3 audio decoder.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch filesrc location=music.mp3 ! mad ! audioconvert ! audioresample ! autoaudiosink
* ]| Decode the mp3 file and play
* </refsect2>
*/
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
#include "config.h" #include "config.h"
#endif #endif

18
ext/sidplay/gstsiddec.cc
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@ -21,26 +21,20 @@
/** /**
* SECTION:element-siddec * SECTION:element-siddec
* *
* <refsect2>
* <para>
* This element decodes .sid files to raw audio. .sid files are in fact * This element decodes .sid files to raw audio. .sid files are in fact
* small Commodore 64 programs that are executed on an emulated 6502 CPU and a * small Commodore 64 programs that are executed on an emulated 6502 CPU and a
* MOS 6581 sound chip. * MOS 6581 sound chip.
* </para> *
* <para>
* This plugin will first load the complete program into memory before starting * This plugin will first load the complete program into memory before starting
* the emulator and producing output. * the emulator and producing output.
* </para> *
* <para>
* Seeking is not (and cannot be) implemented. * Seeking is not (and cannot be) implemented.
* </para> *
* <refsect2>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * |[
* <programlisting>
* gst-launch -v filesrc location=Hawkeye.sid ! siddec ! audioconvert ! alsasink * gst-launch -v filesrc location=Hawkeye.sid ! siddec ! audioconvert ! alsasink
* </programlisting> * ]| Decode a sid file and play back the audio using alsasink.
* Decode a sid file and play back the audio using alsasink.
* </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2006-12-30 (0.10.5) * Last reviewed on 2006-12-30 (0.10.5)

4
gst/asfdemux/gstrtspwms.c
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@ -21,11 +21,7 @@
/** /**
* SECTION:element-rtspwms * SECTION:element-rtspwms
* *
* <refsect2>
* <para>
* A WMS RTSP extension * A WMS RTSP extension
* </para>
* </refsect2>
* *
* Last reviewed on 2007-07-25 (0.10.14) * Last reviewed on 2007-07-25 (0.10.14)
*/ */

14
gst/mpegaudioparse/gstxingmux.c
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@ -24,23 +24,19 @@
/** /**
* SECTION:element-xingmux * SECTION:element-xingmux
* *
* <refsect2>
* <para>
* xingmux adds a Xing header to MP3 files. This contains information about the duration and size * 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 * of the file and a seek table and is very useful for getting an almost correct duration and better
* seeking on VBR MP3 files. * seeking on VBR MP3 files.
* </para> *
* <para>
* This element will remove any existing Xing, LAME or VBRI headers from the beginning of the file. * This element will remove any existing Xing, LAME or VBRI headers from the beginning of the file.
* </para> *
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch audiotestsrc num-buffers=1000 ! audioconvert ! lame ! xingmux ! filesink location=test.mp3 * gst-launch audiotestsrc num-buffers=1000 ! audioconvert ! lame ! xingmux ! filesink location=test.mp3
* gst-launch filesrc location=test.mp3 ! xingmux ! filesink location=test2.mp3 * gst-launch filesrc location=test.mp3 ! xingmux ! filesink location=test2.mp3
* gst-launch filesrc location=test.mp3 ! mp3parse ! xingmux ! filesink location=test2.mp3 * gst-launch filesrc location=test.mp3 ! mp3parse ! xingmux ! filesink location=test2.mp3
* </programlisting> * ]|
* </para>
* </refsect2> * </refsect2>
*/ */

19
gst/realmedia/rademux.c
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@ -20,25 +20,18 @@
/** /**
* SECTION:element-rademux * SECTION:element-rademux
* *
* <refsect2>
* <para>
* Demuxes/parses a RealAudio (.ra) file or stream into compressed audio. * Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
* </para> *
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch filesrc location=interview.ra ! rademux ! ffdec_real_288 ! audioconvert ! audioresample ! alsasink * gst-launch filesrc location=interview.ra ! rademux ! ffdec_real_288 ! audioconvert ! audioresample ! alsasink
* </programlisting> * ]| Read a RealAudio file and decode it and output it to the soundcard using
* Read a RealAudio file and decode it and output it to the soundcard using
* the ALSA element. The .ra file is assumed to contain RealAudio version 2. * the ALSA element. The .ra file is assumed to contain RealAudio version 2.
* </para> * |[
* <para>
* <programlisting>
* gst-launch gnomevfssrc location=http://www.example.org/interview.ra ! rademux ! a52dec ! audioconvert ! audioresample ! alsasink * gst-launch gnomevfssrc location=http://www.example.org/interview.ra ! rademux ! a52dec ! audioconvert ! audioresample ! alsasink
* </programlisting> * ]| Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
* Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
* and output it to the soundcard using the ALSA element. * and output it to the soundcard using the ALSA element.
* </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2006-10-24 (0.10.5) * Last reviewed on 2006-10-24 (0.10.5)

5
gst/realmedia/rdtmanager.c
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@ -43,12 +43,9 @@
/** /**
* SECTION:element-rdtmanager * SECTION:element-rdtmanager
* @see_also: GstRtspSrc
* *
* <refsect2>
* <para>
* A simple RTP session manager used internally by rtspsrc. * A simple RTP session manager used internally by rtspsrc.
* </para>
* </refsect2>
* *
* Last reviewed on 2006-06-20 (0.10.4) * Last reviewed on 2006-06-20 (0.10.4)
*/ */

4
gst/realmedia/rtspreal.c
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@ -21,11 +21,7 @@
/** /**
* SECTION:element-rtspreal * SECTION:element-rtspreal
* *
* <refsect2>
* <para>
* A RealMedia RTSP extension * A RealMedia RTSP extension
* </para>
* </refsect2>
* *
* Last reviewed on 2007-07-25 (0.10.14) * Last reviewed on 2007-07-25 (0.10.14)
*/ */

10
gst/synaesthesia/gstsynaesthesia.c
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@ -21,18 +21,14 @@
* SECTION:element-synaesthesia * SECTION:element-synaesthesia
* @see_also: goom * @see_also: goom
* *
* <refsect2>
* <para>
* Synaesthesia is an audio visualisation element. It creates glitter and * Synaesthesia is an audio visualisation element. It creates glitter and
* pulsating fog based on the incomming audio signal. * pulsating fog based on the incomming audio signal.
* </para> * <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ximagesink * gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ximagesink
* gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ffmpegcolorspace ! xvimagesink * gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ffmpegcolorspace ! xvimagesink
* </programlisting> * ]|
* </para>
* </refsect2> * </refsect2>
*/ */