mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 10:11:08 +00:00
docstrings: port ulinks to markdown links
This commit is contained in:
Mathieu Duponchelle
parent
c5738c6125
commit
42adb02a10
30 changed files with 74 additions and 65 deletions
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@ -27,8 +27,8 @@
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*
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* The chromaprint element calculates an acoustic fingerprint for an
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* audio stream which can be used to identify a song and look up
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* further metadata from the <ulink url="http://acoustid.org/">Acoustid</ulink>
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* and Musicbrainz databases.
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* further metadata from the [Acoustid](http://acoustid.org/) and Musicbrainz
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* databases.
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*
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* ## Example launch line
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* |[
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@ -23,19 +23,21 @@
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* @title: dfbvideosink
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*
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* DfbVideoSink renders video frames using the
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* <ulink url="http://www.directfb.org/">DirectFB</ulink> library.
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* [DirectFB](http://www.directfb.org/) library.
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* Rendering can happen in two different modes :
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*
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* * Standalone: this mode will take complete control of the monitor forcing
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* <ulink url="http://www.directfb.org/">DirectFB</ulink> to fullscreen layout.
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* DirectFB to fullscreen layout.
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*
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* This is convenient to test using the gst-launch-1.0 command line tool or
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* other simple applications. It is possible to interrupt playback while
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* being in this mode by pressing the Escape key.
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* This mode handles navigation events for every input device supported by
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* the <ulink url="http://www.directfb.org/">DirectFB</ulink> library, it will
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* look for available video modes in the fb.modes file and try to switch
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* the framebuffer video mode to the most suitable one. Depending on
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* hardware acceleration capabilities the element will handle scaling or not.
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* the DirectFB library, it will look for available video modes in the fb.modes
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* file and try to switch the framebuffer video mode to the most suitable one.
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* Depending on hardware acceleration capabilities the element will handle
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* scaling or not.
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*
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* If no acceleration is available it will do clipping or centering of the
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* video frames respecting the original aspect ratio.
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*
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@ -43,7 +45,8 @@
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* #GstDfbVideoSink:surface provided by the
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* application developer. This is a more advanced usage of the element and
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* it is required to integrate video playback in existing
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* <ulink url="http://www.directfb.org/">DirectFB</ulink> applications.
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* DirectFB applications.
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*
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* When using this mode the element just renders to the
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* #GstDfbVideoSink:surface provided by the
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* application, that means it won't handle navigation events and won't resize
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@ -25,7 +25,7 @@
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* @see_also: timidity, wildmidi
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*
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* This element renders midi-events as audio streams using
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* <ulink url="http://fluidsynth.sourceforge.net//">Fluidsynth</ulink>.
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* [Fluidsynth](http://fluidsynth.sourceforge.net/).
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* It offers better sound quality compared to the timidity or wildmidi element.
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*
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* ## Example pipeline
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@ -48,8 +48,9 @@
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* @title: katedec
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* @see_also: oggdemux
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*
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* This element decodes Kate streams
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* <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
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* This element decodes Kate streams.
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*
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* [Kate](http://libkate.googlecode.com/) is a free codec
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* for text based data, such as subtitles. Any number of kate streams can be
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* embedded in an Ogg stream.
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*
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@ -49,10 +49,11 @@
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* @title: kateenc
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* @see_also: oggmux
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*
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* This element encodes Kate streams
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* <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
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* for text based data, such as subtitles. Any number of kate streams can be
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* embedded in an Ogg stream.
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* This element encodes Kate streams.
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*
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* [Kate](http://libkate.googlecode.com/) is a free codec for text based data,
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* such as subtitles. Any number of kate streams can be embedded in an Ogg
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* stream.
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*
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* libkate (see above url) is needed to build this plugin.
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*
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@ -48,12 +48,12 @@
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* @title: tiger
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* @see_also: katedec
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*
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* This element decodes and renders Kate streams
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* <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
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* for text based data, such as subtitles. Any number of kate streams can be
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* embedded in an Ogg stream.
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* This element decodes and renders Kate streams.
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* [Kate](http://libkate.googlecode.com/) is a free codec for text based data,
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* such as subtitles. Any number of kate streams can be embedded in an Ogg
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* stream.
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*
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* libkate (see above url) and <ulink url="http://libtiger.googlecode.com/">libtiger</ulink>
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* libkate (see above url) and [libtiger](http://libtiger.googlecode.com/)
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* are needed to build this element.
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*
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* ## Example pipeline
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@ -27,7 +27,8 @@
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* @see_also: #GstAudioConvert #GstAudioResample, #GstAudioTestSrc, #GstAutoAudioSink
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*
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* The LADSPA (Linux Audio Developer's Simple Plugin API) element is a bridge
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* for plugins using the <ulink url="http://www.ladspa.org/">LADSPA</ulink> API.
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* for plugins using the [LADSPA](http://www.ladspa.org/) API.
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*
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* It scans all installed LADSPA plugins and registers them as gstreamer
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* elements. If available it can also parse LRDF files and use the metadata for
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* element classification. The functionality you get depends on the LADSPA plugins
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@ -30,8 +30,8 @@
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* a successor to LADSPA (Linux Audio Developer's Simple Plugin API).
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*
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* The LV2 element is a bridge for plugins using the
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* <ulink url="http://www.lv2plug.in/">LV2</ulink> API. It scans all
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* installed LV2 plugins and registers them as gstreamer elements.
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* [LV2](http://www.lv2plug.in/) API. It scans all installed LV2 plugins and
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* registers them as gstreamer elements.
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*/
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#ifdef HAVE_CONFIG_H
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@ -28,8 +28,8 @@
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/**
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* SECTION:element-modplug
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*
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* Modplug uses the <ulink url="http://modplug-xmms.sourceforge.net/">modplug</ulink>
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* library to decode tracked music in the MOD/S3M/XM/IT and related formats.
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* Modplug uses the [modplug](http://modplug-xmms.sourceforge.net/) library to
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* decode tracked music in the MOD/S3M/XM/IT and related formats.
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*
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* ## Example pipeline
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*
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@ -25,9 +25,10 @@
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* @see_also: mpeg2dec
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*
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* This element encodes raw video into an MPEG-1/2 elementary stream using the
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* <ulink url="http://mjpeg.sourceforge.net/">mjpegtools</ulink> library.
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* [mjpegtools](http://mjpeg.sourceforge.net/) library.
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*
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* Documentation on MPEG encoding in general can be found in the
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* <ulink url="https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776">MJPEG Howto</ulink>
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* [MJPEG Howto](https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776)
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* and on the various available parameters in the documentation
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* of the mpeg2enc tool in particular, which shares options with this element.
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*
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*
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* This element is an audio/video multiplexer for MPEG-1/2 video streams
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* and (un)compressed audio streams such as AC3, MPEG layer I/II/III.
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* It is based on the <ulink url="http://mjpeg.sourceforge.net/">mjpegtools</ulink> library.
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* It is based on the [mjpegtools](http://mjpeg.sourceforge.net/) library.
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* Documentation on creating MPEG videos in general can be found in the
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* <ulink url="https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776">MJPEG Howto</ulink>
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* [MJPEG Howto](https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776)
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* and the man-page of the mplex tool documents the properties of this element,
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* which are shared with the mplex tool.
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*
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@ -23,8 +23,8 @@
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* @see_also: #GstOpenMptDec
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*
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* openmpdec decodes module music formats, such as S3M, MOD, XM, IT.
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* It uses the <ulink url="https://lib.openmpt.org">OpenMPT library</ulink>
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* for this purpose. It can be autoplugged and therefore works with decodebin.
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* It uses the [OpenMPT library](https://lib.openmpt.org) for this purpose.
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* It can be autoplugged and therefore works with decodebin.
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*
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* ## Example launch line
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*
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@ -23,7 +23,7 @@
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* SECTION:element-srtsink
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* @title: srtsink
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*
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* srtsink is a network sink that sends <ulink url="http://www.srtalliance.org/">SRT</ulink>
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* srtsink is a network sink that sends [SRT](http://www.srtalliance.org/)
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* packets to the network.
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*
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* ## Examples</title>
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@ -23,7 +23,7 @@
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* SECTION:element-srtsrc
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* @title: srtsrc
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*
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* srtsrc is a network source that reads <ulink url="http://www.srtalliance.org/">SRT</ulink>
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* srtsrc is a network source that reads [SRT](http://www.srtalliance.org/)
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* packets from the network.
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*
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* ## Examples
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@ -21,8 +21,9 @@
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* SECTION:element-voaacenc
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* @title: voaacenc
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*
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* AAC audio encoder based on vo-aacenc library
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* <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
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* AAC audio encoder based on vo-aacenc library.
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*
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* [vo-aacenc library source file](http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/)
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*
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* ## Example launch line
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* |[
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@ -23,7 +23,7 @@
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* @see_also: #GstAmrWbDec, #GstAmrWbParse
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*
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* AMR wideband encoder based on the
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* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
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* [reference codec implementation](http://www.penguin.cz/~utx/amr).
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*
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* ## Example launch line
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* |[
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@ -27,8 +27,9 @@
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*
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* The waylandsink is creating its own window and render the decoded video frames to that.
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* Setup the Wayland environment as described in
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* <ulink url="http://wayland.freedesktop.org/building.html">Wayland</ulink> home page.
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* The current implementaion is based on weston compositor.
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* [Wayland](http://wayland.freedesktop.org/building.html) home page.
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*
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* The current implementation is based on weston compositor.
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*
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* ## Example pipelines
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* |[
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@ -23,7 +23,7 @@
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* @title: GstWebRTCDataChannel
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* @see_also: #GstWebRTCRTPTransceiver
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*
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* <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport</ulink>
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* <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport>
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*/
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#ifdef HAVE_CONFIG_H
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@ -23,8 +23,9 @@
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* @see_also: #GstWildmidiDec
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*
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* wildmididec decodes MIDI files.
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* It uses <ulink url="https://www.mindwerks.net/projects/wildmidi/">WildMidi</ulink>
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* for this purpose. It can be autoplugged and therefore works with decodebin.
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*
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* It uses [WildMidi](https://www.mindwerks.net/projects/wildmidi/) for this
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* purpose. It can be autoplugged and therefore works with decodebin.
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*
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* ## Example launch line
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*
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@ -23,7 +23,7 @@
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* @title: GstWebRTCDTLSTransport
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCICETransport
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*
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* <ulink url="https://www.w3.org/TR/webrtc/#rtcdtlstransport">https://www.w3.org/TR/webrtc/#rtcdtlstransport</ulink>
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* <https://www.w3.org/TR/webrtc/#rtcdtlstransport>
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*/
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#ifdef HAVE_CONFIG_H
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@ -23,7 +23,7 @@
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* @title: GstWebRTCICETransport
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCDTLSTransport
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*
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* <ulink url="https://www.w3.org/TR/webrtc/#rtcicetransport">https://www.w3.org/TR/webrtc/#rtcicetransport</ulink>
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* <https://www.w3.org/TR/webrtc/#rtcicetransport>
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*/
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#ifdef HAVE_CONFIG_H
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@ -22,7 +22,7 @@
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* @short_description: RTCSessionDescription object
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* @title: GstWebRTCSessionDescription
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*
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* <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
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* <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
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*/
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#ifdef HAVE_CONFIG_H
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@ -38,7 +38,7 @@ GType gst_webrtc_session_description_get_type (void);
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* @type: the #GstWebRTCSDPType of the description
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* @sdp: the #GstSDPMessage of the description
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*
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* See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink>
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* See <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class>
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*/
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struct _GstWebRTCSessionDescription
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{
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* @title: GstWebRTCRTPReceiver
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPTransceiver
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*
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* <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface</ulink>
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* <https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface>
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*/
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#ifdef HAVE_CONFIG_H
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@ -23,7 +23,7 @@
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* @title: GstWebRTCRTPSender
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* @see_also: #GstWebRTCRTPReceiver, #GstWebRTCRTPTransceiver
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*
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* <ulink url="https://www.w3.org/TR/webrtc/#rtcrtpsender-interface">https://www.w3.org/TR/webrtc/#rtcrtpsender-interface</ulink>
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* <https://www.w3.org/TR/webrtc/#rtcrtpsender-interface>
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*/
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#ifdef HAVE_CONFIG_H
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@ -23,7 +23,7 @@
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* @title: GstWebRTCRTPTransceiver
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* @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver
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*
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* <ulink url="https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface">https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface</ulink>
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* <https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface>
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*/
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#ifdef HAVE_CONFIG_H
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@ -82,7 +82,7 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/
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* @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
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* @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
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*
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* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink>
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate>
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*/
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typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
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{
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@ -101,7 +101,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/
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* @GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
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* @GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
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*
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* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink>
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* See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate>
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*/
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typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
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{
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@ -123,7 +123,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/
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* @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
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* @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
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*
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* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink>
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* See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate>
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*/
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typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
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{
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@ -144,7 +144,7 @@ typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/
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* @GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
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||||
* @GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
|
||||
*
|
||||
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink>
|
||||
* See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate>
|
||||
*/
|
||||
typedef enum /*< underscore_name=gst_webrtc_peer_connection_state >*/
|
||||
{
|
||||
|
|
@ -185,7 +185,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_component >*/
|
|||
* @GST_WEBRTC_SDP_TYPE_ANSWER: answer
|
||||
* @GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
|
||||
*
|
||||
* See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink>
|
||||
* See <http://w3c.github.io/webrtc-pc/#rtcsdptype>
|
||||
*/
|
||||
typedef enum /*< underscore_name=gst_webrtc_sdp_type >*/
|
||||
{
|
||||
|
|
@ -282,7 +282,7 @@ typedef enum /*< underscore_name=gst_webrtc_fec_type >*/
|
|||
* GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
|
||||
* GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
|
||||
*
|
||||
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate</ulink>
|
||||
* See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate>
|
||||
*
|
||||
* Since: 1.16
|
||||
*/
|
||||
|
|
@ -301,7 +301,7 @@ typedef enum /*< underscore_name=gst_webrtc_sctp_transport_state >*/
|
|||
* GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
|
||||
* GST_WEBRTC_PRIORITY_TYPE_HIGH: high
|
||||
*
|
||||
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype</ulink>
|
||||
* See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype>
|
||||
*
|
||||
* Since: 1.16
|
||||
*/
|
||||
|
|
@ -321,7 +321,7 @@ typedef enum /*< underscore_name=gst_webrtc_priority_type >*/
|
|||
* GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
|
||||
* GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
|
||||
*
|
||||
* See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate</ulink>
|
||||
* See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate>
|
||||
*
|
||||
* Since: 1.16
|
||||
*/
|
||||
|
|
|
|||
|
|
@ -38,8 +38,8 @@
|
|||
*
|
||||
* The accurip element calculates a CRC for an audio stream which can be used
|
||||
* to match the audio stream to a database hosted on
|
||||
* <ulink url="http://accuraterip.com/">AccurateRip</ulink>. This database
|
||||
* is used to check for a CD rip accuracy.
|
||||
* [AccurateRip](http://accuraterip.com/). This database is used to check for a
|
||||
* CD rip accuracy.
|
||||
*
|
||||
* ## Example launch line
|
||||
* |[
|
||||
|
|
|
|||
|
|
@ -65,9 +65,9 @@
|
|||
* @title: festival
|
||||
*
|
||||
* This element connects to a
|
||||
* <ulink url="http://www.festvox.org/festival/index.html">festival</ulink>
|
||||
* server process and uses it to synthesize speech. Festival need to run already
|
||||
* in server mode, started as `festival --server`
|
||||
* [festival](http://www.festvox.org/festival/index.html) server process and
|
||||
* uses it to synthesize speech. Festival need to run already in server mode,
|
||||
* started as `festival --server`
|
||||
*
|
||||
* ## Example pipeline
|
||||
* |[
|
||||
|
|
|
|||
|
|
@ -26,9 +26,8 @@
|
|||
* #GstPcapParse:src-port and #GstPcapParse:dst-port to restrict which packets
|
||||
* should be included.
|
||||
*
|
||||
* The supported data format is the classical <ulink
|
||||
* url="https://wiki.wireshark.org/Development/LibpcapFileFormat">libpcap file
|
||||
* format</ulink>.
|
||||
* The supported data format is the classical
|
||||
* [libpcap file format](https://wiki.wireshark.org/Development/LibpcapFileFormat)
|
||||
*
|
||||
* ## Example pipelines
|
||||
* |[
|
||||
|
|
|
|||
Loading…
Reference in a new issue