From c49cf83ee352a5317f264b60d66f3317605a4e3e Mon Sep 17 00:00:00 2001 From: Stefan Kost Date: Fri, 13 Jun 2008 06:57:21 +0000 Subject: [PATCH] Add missing elements to docs. Restore alphabetical order in section file. Document mad (it was included in docs alrea... 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. --- ChangeLog | 23 +++++++++ .../gst-plugins-ugly-plugins-docs.sgml | 4 ++ .../gst-plugins-ugly-plugins-sections.txt | 50 +++++++++---------- ext/a52dec/gsta52dec.c | 19 ++----- ext/amrnb/amrnbdec.c | 13 ++--- ext/amrnb/amrnbenc.c | 15 +++--- ext/amrnb/amrnbparse.c | 11 ++-- ext/lame/gstlame.c | 48 ++++++------------ ext/mad/gstmad.c | 14 ++++++ ext/sidplay/gstsiddec.cc | 20 +++----- gst/asfdemux/gstrtspwms.c | 4 -- gst/mpegaudioparse/gstxingmux.c | 14 ++---- gst/realmedia/rademux.c | 19 +++---- gst/realmedia/rdtmanager.c | 5 +- gst/realmedia/rtspreal.c | 4 -- gst/synaesthesia/gstsynaesthesia.c | 10 ++-- 16 files changed, 124 insertions(+), 149 deletions(-) diff --git a/ChangeLog b/ChangeLog index 5b71115ac5..6067ecce0c 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,26 @@ +2008-06-13 Stefan Kost + + * 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 * ext/lame/gstlame.c: diff --git a/docs/plugins/gst-plugins-ugly-plugins-docs.sgml b/docs/plugins/gst-plugins-ugly-plugins-docs.sgml index fb21f97dfc..376cd9249e 100644 --- a/docs/plugins/gst-plugins-ugly-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-ugly-plugins-docs.sgml @@ -18,6 +18,10 @@ + + + + diff --git a/docs/plugins/gst-plugins-ugly-plugins-sections.txt b/docs/plugins/gst-plugins-ugly-plugins-sections.txt index ad59628d95..f9d619509f 100644 --- a/docs/plugins/gst-plugins-ugly-plugins-sections.txt +++ b/docs/plugins/gst-plugins-ugly-plugins-sections.txt @@ -54,6 +54,20 @@ GST_TYPE_AMRNBPARSE gst_amrnbparse_get_type +
+element-lame +lame +GstLame + +GstLameClass +GST_LAME +GST_LAME_CLASS +GST_IS_LAME +GST_IS_LAME_CLASS +GST_TYPE_LAME +gst_lame_get_type +
+
element-mad mad @@ -72,17 +86,18 @@ gst_mad_tag_list_to_id3_tag
-element-lame -lame -GstLame +element-rademux +rademux +GstRealAudioDemux -GstLameClass -GST_LAME -GST_LAME_CLASS -GST_IS_LAME -GST_IS_LAME_CLASS -GST_TYPE_LAME -gst_lame_get_type +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
@@ -101,21 +116,6 @@ gst_rdt_manager_get_type gst_rdt_manager_plugin_init
-
-element-rademux -rademux -GstRealAudioDemux - -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 -
-
element-rtspreal rtspreal diff --git a/ext/a52dec/gsta52dec.c b/ext/a52dec/gsta52dec.c index a942b4bc6f..062defc63c 100644 --- a/ext/a52dec/gsta52dec.c +++ b/ext/a52dec/gsta52dec.c @@ -20,25 +20,16 @@ /** * SECTION:element-a52dec * - * - * * Dolby Digital (AC-3) audio decoder. - * - * + * * * Example launch line - * - * + * |[ * gst-launch dvdreadsrc title=1 ! dvddemux ! a52dec ! audioresample ! audioconvert ! alsasink - * - * Play audio track from a dvd. - * - * - * + * ]| Play audio track from a dvd. + * |[ * gst-launch filesrc location=abc.ac3 ! a52dec ! audioresample ! audioconvert ! alsasink - * - * Decode a stand alone file and play it. - * + * ]| Decode a stand alone file and play it. * */ diff --git a/ext/amrnb/amrnbdec.c b/ext/amrnb/amrnbdec.c index 1f6ff87681..2a3b1ce134 100644 --- a/ext/amrnb/amrnbdec.c +++ b/ext/amrnb/amrnbdec.c @@ -21,17 +21,14 @@ * SECTION:element-amrnbdec * @see_also: #GstAmrnbEnc, #GstAmrnbParse * - * - * - * This is an AMR narrowband decoder based on the + * AMR narrowband decoder based on the * reference codec implementation. - * + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink - * - * + * ]| * */ diff --git a/ext/amrnb/amrnbenc.c b/ext/amrnb/amrnbenc.c index 41e8c12129..fa5d24e1e2 100644 --- a/ext/amrnb/amrnbenc.c +++ b/ext/amrnb/amrnbenc.c @@ -21,18 +21,15 @@ * SECTION:element-amrnbenc * @see_also: #GstAmrnbDec, #GstAmrnbParse * - * - * - * This is an AMR narrowband encoder based on the + * AMR narrowband encoder based on the * reference codec implementation. - * + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! filesink location=abc.amr - * - * - * Please not that the above stream misses the header, that is needed to play + * ]| + * Please note that the above stream misses the header, that is needed to play * the stream. * */ diff --git a/ext/amrnb/amrnbparse.c b/ext/amrnb/amrnbparse.c index c91ebedab3..8de80f5c26 100644 --- a/ext/amrnb/amrnbparse.c +++ b/ext/amrnb/amrnbparse.c @@ -21,16 +21,13 @@ * SECTION:element-amrnbparse * @see_also: #GstAmrnbDec, #GstAmrnbEnc * + * AMR narrowband bitstream parser. + * * - * - * This is an AMR narrowband parser. - * * Example launch line - * - * + * |[ * gst-launch filesrc location=abc.amr ! amrnbparse ! amrnbdec ! audioresample ! audioconvert ! alsasink - * - * + * ]| * */ diff --git a/ext/lame/gstlame.c b/ext/lame/gstlame.c index 99d6ea4dee..abcef003eb 100644 --- a/ext/lame/gstlame.c +++ b/ext/lame/gstlame.c @@ -23,61 +23,45 @@ * SECTION:element-lame * @see_also: mad, vorbisenc * - * - * * This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream. * Note that MP3 is not * a free format, there are licensing and patent issues to take into * consideration. See Ogg/Vorbis * for a royalty free (and often higher quality) alternative. - * + * + * * Output sample rate - * * 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. * 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 * particular sample rate. - * + * + * * Writing metadata (tags) - * * Whilst the lame encoder element does claim to implement the GstTagSetter * interface, it does so only for backwards compatibility reasons. Tag writing * 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 * xingmux element to add XING headers to your VBR mp3 file. - * + * + * * Example pipelines - * - * Encode a test sine signal to MP3. - * - * + * |[ * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lame ! filesink location=sine.mp3 - * - * - * Record from a sound card using ALSA and encode to MP3 - * - * + * ]| Encode a test sine signal to MP3. + * |[ * gst-launch -v alsasrc ! audioconvert ! lame bitrate=192 ! filesink location=alsasrc.mp3 - * - * - * Transcode from a .wav file to MP3 (the id3v2mux element is optional): - * - * + * ]| Record from a sound card using ALSA and encode to MP3 + * |[ * gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lame bitrate=192 ! id3v2mux ! filesink location=music.mp3 - * - * - * Encode Audio CD track 5 to MP3: - * - * + * ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional) + * |[ * gst-launch -v cdda://5 ! audioconvert ! lame bitrate=192 ! filesink location=track5.mp3 - * - * - * Encode to a fixed sample rate: - * - * + * ]| Encode Audio CD track 5 to 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 - * + * ]| Encode to a fixed sample rate * * * Last reviewed on 2007-07-24 (0.10.7) diff --git a/ext/mad/gstmad.c b/ext/mad/gstmad.c index d5379047c5..e9bb077ced 100644 --- a/ext/mad/gstmad.c +++ b/ext/mad/gstmad.c @@ -17,6 +17,20 @@ * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-mad + * @see_also: lame + * + * MP3 audio decoder. + * + * + * Example pipelines + * |[ + * gst-launch filesrc location=music.mp3 ! mad ! audioconvert ! audioresample ! autoaudiosink + * ]| Decode the mp3 file and play + * + */ + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/ext/sidplay/gstsiddec.cc b/ext/sidplay/gstsiddec.cc index 94f0749def..0b2c8f7bee 100644 --- a/ext/sidplay/gstsiddec.cc +++ b/ext/sidplay/gstsiddec.cc @@ -21,26 +21,20 @@ /** * SECTION:element-siddec * - * - * * 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 - * MOS 6581 sound chip. - * - * + * MOS 6581 sound chip. + * * This plugin will first load the complete program into memory before starting * the emulator and producing output. - * - * + * * Seeking is not (and cannot be) implemented. - * + * + * * Example pipelines - * - * + * |[ * gst-launch -v filesrc location=Hawkeye.sid ! siddec ! audioconvert ! alsasink - * - * Decode a sid file and play back the audio using alsasink. - * + * ]| Decode a sid file and play back the audio using alsasink. * * * Last reviewed on 2006-12-30 (0.10.5) diff --git a/gst/asfdemux/gstrtspwms.c b/gst/asfdemux/gstrtspwms.c index ba6426b01b..4f79568227 100644 --- a/gst/asfdemux/gstrtspwms.c +++ b/gst/asfdemux/gstrtspwms.c @@ -21,11 +21,7 @@ /** * SECTION:element-rtspwms * - * - * * A WMS RTSP extension - * - * * * Last reviewed on 2007-07-25 (0.10.14) */ diff --git a/gst/mpegaudioparse/gstxingmux.c b/gst/mpegaudioparse/gstxingmux.c index 99a3d2e1fb..58c0860ba5 100644 --- a/gst/mpegaudioparse/gstxingmux.c +++ b/gst/mpegaudioparse/gstxingmux.c @@ -24,23 +24,19 @@ /** * SECTION:element-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 * seeking on VBR MP3 files. - * - * + * * This element will remove any existing Xing, LAME or VBRI headers from the beginning of the file. - * + * + * * Example launch line - * - * + * |[ * 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 ! mp3parse ! xingmux ! filesink location=test2.mp3 - * - * + * ]| * */ diff --git a/gst/realmedia/rademux.c b/gst/realmedia/rademux.c index 71a0a8c0cc..5c368cf038 100644 --- a/gst/realmedia/rademux.c +++ b/gst/realmedia/rademux.c @@ -20,25 +20,18 @@ /** * SECTION:element-rademux * - * - * * Demuxes/parses a RealAudio (.ra) file or stream into compressed audio. - * + * + * * Example launch line - * - * + * |[ * gst-launch filesrc location=interview.ra ! rademux ! ffdec_real_288 ! audioconvert ! audioresample ! alsasink - * - * 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. - * - * - * + * |[ * gst-launch gnomevfssrc location=http://www.example.org/interview.ra ! rademux ! a52dec ! audioconvert ! audioresample ! alsasink - * - * 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. - * * * * Last reviewed on 2006-10-24 (0.10.5) diff --git a/gst/realmedia/rdtmanager.c b/gst/realmedia/rdtmanager.c index f75b353c2b..aad041fe6e 100644 --- a/gst/realmedia/rdtmanager.c +++ b/gst/realmedia/rdtmanager.c @@ -43,12 +43,9 @@ /** * SECTION:element-rdtmanager + * @see_also: GstRtspSrc * - * - * * A simple RTP session manager used internally by rtspsrc. - * - * * * Last reviewed on 2006-06-20 (0.10.4) */ diff --git a/gst/realmedia/rtspreal.c b/gst/realmedia/rtspreal.c index 9361963deb..ec93903fdf 100644 --- a/gst/realmedia/rtspreal.c +++ b/gst/realmedia/rtspreal.c @@ -21,11 +21,7 @@ /** * SECTION:element-rtspreal * - * - * * A RealMedia RTSP extension - * - * * * Last reviewed on 2007-07-25 (0.10.14) */ diff --git a/gst/synaesthesia/gstsynaesthesia.c b/gst/synaesthesia/gstsynaesthesia.c index 2c8f007048..953df7e9db 100644 --- a/gst/synaesthesia/gstsynaesthesia.c +++ b/gst/synaesthesia/gstsynaesthesia.c @@ -21,18 +21,14 @@ * SECTION:element-synaesthesia * @see_also: goom * - * - * * Synaesthesia is an audio visualisation element. It creates glitter and * pulsating fog based on the incomming audio signal. - * + * * Example launch line - * - * + * |[ * gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ximagesink * gst-launch -v audiotestsrc ! audioconvert ! synaesthesia ! ffmpegcolorspace ! xvimagesink - * - * + * ]| * */