From: Mathieu Duponchelle Date: Fri, 23 Aug 2019 17:56:35 +0000 (+0200) Subject: docstrings: port ulinks to markdown links X-Git-Tag: 1.19.3~507^2~2937 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=42adb02a10c363bc83e2f35797203dbcde0fc879;p=platform%2Fupstream%2Fgstreamer.git docstrings: port ulinks to markdown links --- diff --git a/ext/chromaprint/gstchromaprint.c b/ext/chromaprint/gstchromaprint.c index 6ebadd31a4..bdfc8d1526 100644 --- a/ext/chromaprint/gstchromaprint.c +++ b/ext/chromaprint/gstchromaprint.c @@ -27,8 +27,8 @@ * * The chromaprint element calculates an acoustic fingerprint for an * audio stream which can be used to identify a song and look up - * further metadata from the Acoustid - * and Musicbrainz databases. + * further metadata from the [Acoustid](http://acoustid.org/) and Musicbrainz + * databases. * * ## Example launch line * |[ diff --git a/ext/directfb/dfbvideosink.c b/ext/directfb/dfbvideosink.c index 2313e9bd6b..2e44872c22 100644 --- a/ext/directfb/dfbvideosink.c +++ b/ext/directfb/dfbvideosink.c @@ -23,19 +23,21 @@ * @title: dfbvideosink * * DfbVideoSink renders video frames using the - * DirectFB library. + * [DirectFB](http://www.directfb.org/) library. * Rendering can happen in two different modes : * * * Standalone: this mode will take complete control of the monitor forcing - * DirectFB to fullscreen layout. + * DirectFB to fullscreen layout. + * * This is convenient to test using the gst-launch-1.0 command line tool or * other simple applications. It is possible to interrupt playback while * being in this mode by pressing the Escape key. * This mode handles navigation events for every input device supported by - * the DirectFB library, it will - * look for available video modes in the fb.modes file and try to switch - * the framebuffer video mode to the most suitable one. Depending on - * hardware acceleration capabilities the element will handle scaling or not. + * the DirectFB library, it will look for available video modes in the fb.modes + * file and try to switch the framebuffer video mode to the most suitable one. + * Depending on hardware acceleration capabilities the element will handle + * scaling or not. + * * If no acceleration is available it will do clipping or centering of the * video frames respecting the original aspect ratio. * @@ -43,7 +45,8 @@ * #GstDfbVideoSink:surface provided by the * application developer. This is a more advanced usage of the element and * it is required to integrate video playback in existing - * DirectFB applications. + * DirectFB applications. + * * When using this mode the element just renders to the * #GstDfbVideoSink:surface provided by the * application, that means it won't handle navigation events and won't resize diff --git a/ext/fluidsynth/gstfluiddec.c b/ext/fluidsynth/gstfluiddec.c index c9514dba08..48b53ca191 100644 --- a/ext/fluidsynth/gstfluiddec.c +++ b/ext/fluidsynth/gstfluiddec.c @@ -25,7 +25,7 @@ * @see_also: timidity, wildmidi * * This element renders midi-events as audio streams using - * Fluidsynth. + * [Fluidsynth](http://fluidsynth.sourceforge.net/). * It offers better sound quality compared to the timidity or wildmidi element. * * ## Example pipeline diff --git a/ext/kate/gstkatedec.c b/ext/kate/gstkatedec.c index 65b29bfdbe..a2cec0970c 100644 --- a/ext/kate/gstkatedec.c +++ b/ext/kate/gstkatedec.c @@ -48,8 +48,9 @@ * @title: katedec * @see_also: oggdemux * - * This element decodes Kate streams - * Kate is a free codec + * This element decodes Kate streams. + * + * [Kate](http://libkate.googlecode.com/) is a free codec * for text based data, such as subtitles. Any number of kate streams can be * embedded in an Ogg stream. * diff --git a/ext/kate/gstkateenc.c b/ext/kate/gstkateenc.c index a76800191a..d2cbeb0c73 100644 --- a/ext/kate/gstkateenc.c +++ b/ext/kate/gstkateenc.c @@ -49,10 +49,11 @@ * @title: kateenc * @see_also: oggmux * - * This element encodes Kate streams - * Kate is a free codec - * for text based data, such as subtitles. Any number of kate streams can be - * embedded in an Ogg stream. + * This element encodes Kate streams. + * + * [Kate](http://libkate.googlecode.com/) is a free codec for text based data, + * such as subtitles. Any number of kate streams can be embedded in an Ogg + * stream. * * libkate (see above url) is needed to build this plugin. * diff --git a/ext/kate/gstkatetiger.c b/ext/kate/gstkatetiger.c index 21970c9f6d..8e877e77fa 100644 --- a/ext/kate/gstkatetiger.c +++ b/ext/kate/gstkatetiger.c @@ -48,12 +48,12 @@ * @title: tiger * @see_also: katedec * - * This element decodes and renders Kate streams - * Kate is a free codec - * for text based data, such as subtitles. Any number of kate streams can be - * embedded in an Ogg stream. + * This element decodes and renders Kate streams. + * [Kate](http://libkate.googlecode.com/) is a free codec for text based data, + * such as subtitles. Any number of kate streams can be embedded in an Ogg + * stream. * - * libkate (see above url) and libtiger + * libkate (see above url) and [libtiger](http://libtiger.googlecode.com/) * are needed to build this element. * * ## Example pipeline diff --git a/ext/ladspa/gstladspa.c b/ext/ladspa/gstladspa.c index 2acd70f72a..92a6838a03 100644 --- a/ext/ladspa/gstladspa.c +++ b/ext/ladspa/gstladspa.c @@ -27,7 +27,8 @@ * @see_also: #GstAudioConvert #GstAudioResample, #GstAudioTestSrc, #GstAutoAudioSink * * The LADSPA (Linux Audio Developer's Simple Plugin API) element is a bridge - * for plugins using the LADSPA API. + * for plugins using the [LADSPA](http://www.ladspa.org/) API. + * * It scans all installed LADSPA plugins and registers them as gstreamer * elements. If available it can also parse LRDF files and use the metadata for * element classification. The functionality you get depends on the LADSPA plugins diff --git a/ext/lv2/gstlv2.c b/ext/lv2/gstlv2.c index 705c4d404a..60d502f41a 100644 --- a/ext/lv2/gstlv2.c +++ b/ext/lv2/gstlv2.c @@ -30,8 +30,8 @@ * a successor to LADSPA (Linux Audio Developer's Simple Plugin API). * * The LV2 element is a bridge for plugins using the - * LV2 API. It scans all - * installed LV2 plugins and registers them as gstreamer elements. + * [LV2](http://www.lv2plug.in/) API. It scans all installed LV2 plugins and + * registers them as gstreamer elements. */ #ifdef HAVE_CONFIG_H diff --git a/ext/modplug/gstmodplug.cc b/ext/modplug/gstmodplug.cc index a106f8c7f8..30c5952ab2 100644 --- a/ext/modplug/gstmodplug.cc +++ b/ext/modplug/gstmodplug.cc @@ -28,8 +28,8 @@ /** * SECTION:element-modplug * - * Modplug uses the modplug - * library to decode tracked music in the MOD/S3M/XM/IT and related formats. + * Modplug uses the [modplug](http://modplug-xmms.sourceforge.net/) library to + * decode tracked music in the MOD/S3M/XM/IT and related formats. * * ## Example pipeline * diff --git a/ext/mpeg2enc/gstmpeg2enc.cc b/ext/mpeg2enc/gstmpeg2enc.cc index a86804d7bd..9bec3ef077 100644 --- a/ext/mpeg2enc/gstmpeg2enc.cc +++ b/ext/mpeg2enc/gstmpeg2enc.cc @@ -25,9 +25,10 @@ * @see_also: mpeg2dec * * This element encodes raw video into an MPEG-1/2 elementary stream using the - * mjpegtools library. + * [mjpegtools](http://mjpeg.sourceforge.net/) library. + * * Documentation on MPEG encoding in general can be found in the - * MJPEG Howto + * [MJPEG Howto](https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776) * and on the various available parameters in the documentation * of the mpeg2enc tool in particular, which shares options with this element. * diff --git a/ext/mplex/gstmplex.cc b/ext/mplex/gstmplex.cc index f78e5bd340..5df12d6d03 100644 --- a/ext/mplex/gstmplex.cc +++ b/ext/mplex/gstmplex.cc @@ -26,9 +26,9 @@ * * This element is an audio/video multiplexer for MPEG-1/2 video streams * and (un)compressed audio streams such as AC3, MPEG layer I/II/III. - * It is based on the mjpegtools library. + * It is based on the [mjpegtools](http://mjpeg.sourceforge.net/) library. * Documentation on creating MPEG videos in general can be found in the - * MJPEG Howto + * [MJPEG Howto](https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776) * and the man-page of the mplex tool documents the properties of this element, * which are shared with the mplex tool. * diff --git a/ext/openmpt/gstopenmptdec.c b/ext/openmpt/gstopenmptdec.c index bc8c04fd58..9c84113158 100644 --- a/ext/openmpt/gstopenmptdec.c +++ b/ext/openmpt/gstopenmptdec.c @@ -23,8 +23,8 @@ * @see_also: #GstOpenMptDec * * openmpdec decodes module music formats, such as S3M, MOD, XM, IT. - * It uses the OpenMPT library - * for this purpose. It can be autoplugged and therefore works with decodebin. + * It uses the [OpenMPT library](https://lib.openmpt.org) for this purpose. + * It can be autoplugged and therefore works with decodebin. * * ## Example launch line * diff --git a/ext/srt/gstsrtsink.c b/ext/srt/gstsrtsink.c index 8d499f2fbb..591f414730 100644 --- a/ext/srt/gstsrtsink.c +++ b/ext/srt/gstsrtsink.c @@ -23,7 +23,7 @@ * SECTION:element-srtsink * @title: srtsink * - * srtsink is a network sink that sends SRT + * srtsink is a network sink that sends [SRT](http://www.srtalliance.org/) * packets to the network. * * ## Examples diff --git a/ext/srt/gstsrtsrc.c b/ext/srt/gstsrtsrc.c index cc52a60faa..59103fe39d 100644 --- a/ext/srt/gstsrtsrc.c +++ b/ext/srt/gstsrtsrc.c @@ -23,7 +23,7 @@ * SECTION:element-srtsrc * @title: srtsrc * - * srtsrc is a network source that reads SRT + * srtsrc is a network source that reads [SRT](http://www.srtalliance.org/) * packets from the network. * * ## Examples diff --git a/ext/voaacenc/gstvoaacenc.c b/ext/voaacenc/gstvoaacenc.c index 0580d27f00..91eacb8c9c 100644 --- a/ext/voaacenc/gstvoaacenc.c +++ b/ext/voaacenc/gstvoaacenc.c @@ -21,8 +21,9 @@ * SECTION:element-voaacenc * @title: voaacenc * - * AAC audio encoder based on vo-aacenc library - * vo-aacenc library source file. + * AAC audio encoder based on vo-aacenc library. + * + * [vo-aacenc library source file](http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/) * * ## Example launch line * |[ diff --git a/ext/voamrwbenc/gstvoamrwbenc.c b/ext/voamrwbenc/gstvoamrwbenc.c index c5eae31d7f..dfb997f454 100644 --- a/ext/voamrwbenc/gstvoamrwbenc.c +++ b/ext/voamrwbenc/gstvoamrwbenc.c @@ -23,7 +23,7 @@ * @see_also: #GstAmrWbDec, #GstAmrWbParse * * AMR wideband encoder based on the - * reference codec implementation. + * [reference codec implementation](http://www.penguin.cz/~utx/amr). * * ## Example launch line * |[ diff --git a/ext/wayland/gstwaylandsink.c b/ext/wayland/gstwaylandsink.c index 78dd294a01..bfc4eb2ebf 100644 --- a/ext/wayland/gstwaylandsink.c +++ b/ext/wayland/gstwaylandsink.c @@ -27,8 +27,9 @@ * * The waylandsink is creating its own window and render the decoded video frames to that. * Setup the Wayland environment as described in - * Wayland home page. - * The current implementaion is based on weston compositor. + * [Wayland](http://wayland.freedesktop.org/building.html) home page. + * + * The current implementation is based on weston compositor. * * ## Example pipelines * |[ diff --git a/ext/webrtc/webrtcdatachannel.c b/ext/webrtc/webrtcdatachannel.c index a4d49db145..693e626cd7 100644 --- a/ext/webrtc/webrtcdatachannel.c +++ b/ext/webrtc/webrtcdatachannel.c @@ -23,7 +23,7 @@ * @title: GstWebRTCDataChannel * @see_also: #GstWebRTCRTPTransceiver * - * http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport + * */ #ifdef HAVE_CONFIG_H diff --git a/ext/wildmidi/gstwildmididec.c b/ext/wildmidi/gstwildmididec.c index ea88f74294..60ae3b3dea 100644 --- a/ext/wildmidi/gstwildmididec.c +++ b/ext/wildmidi/gstwildmididec.c @@ -23,8 +23,9 @@ * @see_also: #GstWildmidiDec * * wildmididec decodes MIDI files. - * It uses WildMidi - * for this purpose. It can be autoplugged and therefore works with decodebin. + * + * It uses [WildMidi](https://www.mindwerks.net/projects/wildmidi/) for this + * purpose. It can be autoplugged and therefore works with decodebin. * * ## Example launch line * diff --git a/gst-libs/gst/webrtc/dtlstransport.c b/gst-libs/gst/webrtc/dtlstransport.c index c3b2d519d5..ea7671fb25 100644 --- a/gst-libs/gst/webrtc/dtlstransport.c +++ b/gst-libs/gst/webrtc/dtlstransport.c @@ -23,7 +23,7 @@ * @title: GstWebRTCDTLSTransport * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCICETransport * - * https://www.w3.org/TR/webrtc/#rtcdtlstransport + * */ #ifdef HAVE_CONFIG_H diff --git a/gst-libs/gst/webrtc/icetransport.c b/gst-libs/gst/webrtc/icetransport.c index e6f44378fe..d7e77d90f2 100644 --- a/gst-libs/gst/webrtc/icetransport.c +++ b/gst-libs/gst/webrtc/icetransport.c @@ -23,7 +23,7 @@ * @title: GstWebRTCICETransport * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver, #GstWebRTCDTLSTransport * - * https://www.w3.org/TR/webrtc/#rtcicetransport + * */ #ifdef HAVE_CONFIG_H diff --git a/gst-libs/gst/webrtc/rtcsessiondescription.c b/gst-libs/gst/webrtc/rtcsessiondescription.c index af5cd1c0d7..abdf5ca920 100644 --- a/gst-libs/gst/webrtc/rtcsessiondescription.c +++ b/gst-libs/gst/webrtc/rtcsessiondescription.c @@ -22,7 +22,7 @@ * @short_description: RTCSessionDescription object * @title: GstWebRTCSessionDescription * - * https://www.w3.org/TR/webrtc/#rtcsessiondescription-class + * */ #ifdef HAVE_CONFIG_H diff --git a/gst-libs/gst/webrtc/rtcsessiondescription.h b/gst-libs/gst/webrtc/rtcsessiondescription.h index 375642e767..5308c549a1 100644 --- a/gst-libs/gst/webrtc/rtcsessiondescription.h +++ b/gst-libs/gst/webrtc/rtcsessiondescription.h @@ -38,7 +38,7 @@ GType gst_webrtc_session_description_get_type (void); * @type: the #GstWebRTCSDPType of the description * @sdp: the #GstSDPMessage of the description * - * See https://www.w3.org/TR/webrtc/#rtcsessiondescription-class + * See */ struct _GstWebRTCSessionDescription { diff --git a/gst-libs/gst/webrtc/rtpreceiver.c b/gst-libs/gst/webrtc/rtpreceiver.c index f21d77ef13..768e9876d3 100644 --- a/gst-libs/gst/webrtc/rtpreceiver.c +++ b/gst-libs/gst/webrtc/rtpreceiver.c @@ -23,7 +23,7 @@ * @title: GstWebRTCRTPReceiver * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPTransceiver * - * https://www.w3.org/TR/webrtc/#rtcrtpreceiver-interface + * */ #ifdef HAVE_CONFIG_H diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c index da743f32d1..3a8a9044f0 100644 --- a/gst-libs/gst/webrtc/rtpsender.c +++ b/gst-libs/gst/webrtc/rtpsender.c @@ -23,7 +23,7 @@ * @title: GstWebRTCRTPSender * @see_also: #GstWebRTCRTPReceiver, #GstWebRTCRTPTransceiver * - * https://www.w3.org/TR/webrtc/#rtcrtpsender-interface + * */ #ifdef HAVE_CONFIG_H diff --git a/gst-libs/gst/webrtc/rtptransceiver.c b/gst-libs/gst/webrtc/rtptransceiver.c index 8ea85f1686..08019462ad 100644 --- a/gst-libs/gst/webrtc/rtptransceiver.c +++ b/gst-libs/gst/webrtc/rtptransceiver.c @@ -23,7 +23,7 @@ * @title: GstWebRTCRTPTransceiver * @see_also: #GstWebRTCRTPSender, #GstWebRTCRTPReceiver * - * https://www.w3.org/TR/webrtc/#rtcrtptransceiver-interface + * */ #ifdef HAVE_CONFIG_H diff --git a/gst-libs/gst/webrtc/webrtc_fwd.h b/gst-libs/gst/webrtc/webrtc_fwd.h index 07d9b39ec2..61c1aca9e5 100644 --- a/gst-libs/gst/webrtc/webrtc_fwd.h +++ b/gst-libs/gst/webrtc/webrtc_fwd.h @@ -82,7 +82,7 @@ typedef enum /*< underscore_name=gst_webrtc_dtls_transport_state >*/ * @GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering * @GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete * - * See http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate + * See */ typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/ { @@ -101,7 +101,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_gathering_state >*/ * @GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected * @GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed * - * See http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate + * See */ typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/ { @@ -123,7 +123,7 @@ typedef enum /*< underscore_name=gst_webrtc_ice_connection_state >*/ * @GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer * @GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer * - * See http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate + * See */ typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/ { @@ -144,7 +144,7 @@ typedef enum /*< underscore_name=gst_webrtc_signaling_state >*/ * @GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed * @GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed * - * See http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate + * See */ 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 http://w3c.github.io/webrtc-pc/#rtcsdptype + * See */ 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 http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate + * See * * 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 http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype + * See * * 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 http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate + * See * * Since: 1.16 */ diff --git a/gst/accurip/gstaccurip.c b/gst/accurip/gstaccurip.c index 05578d4e18..a3d4877342 100644 --- a/gst/accurip/gstaccurip.c +++ b/gst/accurip/gstaccurip.c @@ -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 - * AccurateRip. 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 * |[ diff --git a/gst/festival/gstfestival.c b/gst/festival/gstfestival.c index e3206ef768..fbe9ecdae9 100644 --- a/gst/festival/gstfestival.c +++ b/gst/festival/gstfestival.c @@ -65,9 +65,9 @@ * @title: festival * * This element connects to a - * festival - * 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 * |[ diff --git a/gst/pcapparse/gstpcapparse.c b/gst/pcapparse/gstpcapparse.c index 8a60cdad6c..aab2b8a1f5 100644 --- a/gst/pcapparse/gstpcapparse.c +++ b/gst/pcapparse/gstpcapparse.c @@ -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 libpcap file - * format. + * The supported data format is the classical + * [libpcap file format](https://wiki.wireshark.org/Development/LibpcapFileFormat) * * ## Example pipelines * |[