mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-23 02:01:12 +00:00
Code Issues 1 Projects Releases Wiki Activity

docs: Port all docstring to gtk-doc markdown

Browse source
This commit is contained in:
Thibault Saunier 2018-10-22 11:47:32 +02:00
parent e778c5594f
commit 2df9f6e5ce
13 changed files with 64 additions and 73 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

6
ext/a52dec/gsta52dec.c
View file

@ -19,18 +19,18 @@
/**
* SECTION:element-a52dec
* @title: a52dec
*
* Dolby Digital (AC-3) audio decoder.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* 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

6
ext/amrnb/amrnbdec.c
View file

@ -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>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrnbdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/amrnb/amrnbenc.c
View file

@ -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>
* <title>Example launch line</title>
* ## Example launch line
* |[
* 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

6
ext/amrwbdec/amrwbdec.c
View file

@ -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>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

36
ext/cdio/gstcdiocddasrc.c
View file

@ -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>
* <listitem><para>
* treat the entire disc as one stream, counting time from the start of
* * 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>
* <para>
* <programlisting>
*
* ## Example launch line
*
* |[
* 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

29
ext/x264/gstx264enc.c
View file

@ -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 latency (i.e. frame buffering) in the encoder. This may cause problems
* with pipeline stalling in non-trivial pipelines, because the encoder latency
* 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
* non-x264enc streams/branches filling up and blocking upstream. They can
* 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
* 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
* overall encoding quality so may not be appropriate for your use case.
* </note>
* > Some settings, including the default settings, may lead to quite
* > some latency (i.e. frame buffering) in the encoder. This may cause problems
* > with pipeline stalling in non-trivial pipelines, because the encoder latency
* > 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
* > non-x264enc streams/branches filling up and blocking upstream. They can
* > 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
* > 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
* > overall encoding quality so may not be appropriate for your use case.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* 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

2
gst/asfdemux/gstasfdemux.h
View file

@ -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

1
gst/asfdemux/gstrtspwms.c
View file

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

1
gst/realmedia/gstrdtbuffer.h
View file

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

6
gst/realmedia/rademux.c
View file

@ -19,11 +19,11 @@
/**
* SECTION:element-rademux
* @title: rademux
*
* Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* 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

1
gst/realmedia/rdtmanager.c
View file

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

1
gst/realmedia/rtspreal.c
View file

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

6
gst/xingmux/gstxingmux.c
View file

@ -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>
* <title>Example launch line</title>
* ## Example launch line
* |[
* 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