diff options
Diffstat (limited to 'ext')
| -rw-r--r-- | ext/amrwb/gstamrwbdec.c | 13 | ||||
| -rw-r--r-- | ext/amrwb/gstamrwbenc.c | 13 | ||||
| -rw-r--r-- | ext/amrwb/gstamrwbparse.c | 11 | ||||
| -rw-r--r-- | ext/dc1394/gstdc1394.c | 13 | ||||
| -rw-r--r-- | ext/directfb/dfbvideosink.c | 25 | ||||
| -rw-r--r-- | ext/ivorbis/vorbisdec.c | 13 | ||||
| -rw-r--r-- | ext/jack/gstjackaudiosink.c | 40 | ||||
| -rw-r--r-- | ext/mpeg2enc/gstmpeg2enc.cc | 28 | ||||
| -rw-r--r-- | ext/mplex/gstmplex.cc | 12 | ||||
| -rw-r--r-- | ext/musicbrainz/gsttrm.c | 22 | ||||
| -rw-r--r-- | ext/mythtv/gstmythtvsrc.c | 27 | ||||
| -rw-r--r-- | ext/theora/theoradec.c | 12 | ||||
| -rw-r--r-- | ext/timidity/gsttimidity.c | 12 | ||||
| -rw-r--r-- | ext/timidity/gstwildmidi.c | 12 |
14 files changed, 93 insertions, 160 deletions
diff --git a/ext/amrwb/gstamrwbdec.c b/ext/amrwb/gstamrwbdec.c index 525b49c3..54e8a830 100644 --- a/ext/amrwb/gstamrwbdec.c +++ b/ext/amrwb/gstamrwbdec.c @@ -21,17 +21,14 @@ * SECTION:element-amrwbdec * @see_also: #GstAmrwbEnc, #GstAmrwbParse * - * <refsect2> - * <para> - * This is an AMR wideband decoder based on the + * AMR wideband decoder based on the * <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=abc.amr ! amrwbparse ! amrwbdec ! audioresample ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/ext/amrwb/gstamrwbenc.c b/ext/amrwb/gstamrwbenc.c index 33b3dc73..6551b67b 100644 --- a/ext/amrwb/gstamrwbenc.c +++ b/ext/amrwb/gstamrwbenc.c @@ -21,17 +21,14 @@ * SECTION:element-amrwbenc * @see_also: #GstAmrwbDec, #GstAmrwbParse * - * <refsect2> - * <para> - * This is an AMR wideband encoder based on the + * AMR wideband encoder based on the * <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrwbenc ! filesink location=abc.amr - * </programlisting> - * </para> + * ]| * Please not that the above stream misses the header, that is needed to play * the stream. * </refsect2> diff --git a/ext/amrwb/gstamrwbparse.c b/ext/amrwb/gstamrwbparse.c index f7a7e274..cd162d58 100644 --- a/ext/amrwb/gstamrwbparse.c +++ b/ext/amrwb/gstamrwbparse.c @@ -21,16 +21,13 @@ * SECTION:element-amrwbparse * @see_also: #GstAmrwbDec, #GstAmrwbEnc * - * <refsect2> - * <para> * This is an AMR wideband parser. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=abc.amr ! amrwbparse ! amrwbdec ! audioresample ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/ext/dc1394/gstdc1394.c b/ext/dc1394/gstdc1394.c index 7eb26607..7cae5bdf 100644 --- a/ext/dc1394/gstdc1394.c +++ b/ext/dc1394/gstdc1394.c @@ -21,17 +21,14 @@ /** * SECTION:element-dc1394 * - * <refsect2> - * <title>Example launch line</title> - * <para> * Source for IIDC (Instrumentation & Industrial Digital Camera) firewire * cameras. - * </para> - * <para> - * <programlisting> + * + * <refsect2> + * <title>Example launch line</title> + * |[ * gst-launch -v dc1394 camera-number=0 ! xvimagesink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/ext/directfb/dfbvideosink.c b/ext/directfb/dfbvideosink.c index bc715a59..6251803e 100644 --- a/ext/directfb/dfbvideosink.c +++ b/ext/directfb/dfbvideosink.c @@ -20,8 +20,6 @@ /** * SECTION:element-dfbvideosink * - * <refsect2> - * <para> * DfbVideoSink renders video frames using the * <ulink url="http://www.directfb.org/">DirectFB</ulink> library. * Rendering can happen in two different modes : @@ -47,19 +45,19 @@ * <listitem> * <para> * Embedded: this mode will render video frames in a - * <link linkend="GstDfbVideoSink--surface">surface</link> provided by the + * #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 * <ulink url="http://www.directfb.org/">DirectFB</ulink> applications. * </para> * <para> * When using this mode the element just renders to the - * <link linkend="GstDfbVideoSink--surface">surface</link> provided by the + * #GstDfbVideoSink:surface provided by the * application, that means it won't handle navigation events and won't resize - * the <link linkend="GstDfbVideoSink--surface">surface</link> to fit video + * the #GstDfbVideoSink:surface to fit video * frames geometry. Application has to implement the necessary code to grab * informations about the negotiated geometry and resize there - * <link linkend="GstDfbVideoSink--surface">surface</link> accordingly. + * #GstDfbVideoSink:surface accordingly. * </para> * </listitem> * </itemizedlist> @@ -67,20 +65,19 @@ * optimize memory allocation time and handle reverse negotiation. Indeed if * you insert an element like videoscale in the pipeline the video sink will * negotiate with it to try get a scaled video for either the fullscreen layout - * or the application provided external - * <link linkend="GstDfbVideoSink--surface">surface</link>. - * </para> + * or the application provided external #GstDfbVideoSink:surface. + * + * <refsect2> * <title>Example application</title> * <para> * <include xmlns="http://www.w3.org/2003/XInclude" href="element-dfb-example.xml" /> * </para> + * </refsect2> + * <refsect2> * <title>Example pipelines</title> - * <para> - * Here is a test pipeline to test the colorbalance interface : - * <programlisting> + * |[ * gst-launch -v videotestsrc ! dfbvideosink hue=20000 saturation=40000 brightness=25000 - * </programlisting> - * </para> + * ]| test the colorbalance interface implementation in dfbvideosink * </refsect2> */ diff --git a/ext/ivorbis/vorbisdec.c b/ext/ivorbis/vorbisdec.c index 75e05336..325c46a3 100644 --- a/ext/ivorbis/vorbisdec.c +++ b/ext/ivorbis/vorbisdec.c @@ -24,20 +24,17 @@ * SECTION:element-ivorbisdec * @see_also: vorbisenc, oggdemux * - * <refsect2> - * <para> * This element decodes a Vorbis stream to raw int audio. * <ulink url="http://www.vorbis.com/">Vorbis</ulink> is a royalty-free * audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org * Foundation</ulink>. - * </para> + * + * <refsect2> * <title>Example pipelines</title> - * <para> - * <programlisting> + * |[ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink - * </programlisting> - * Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc. - * </para> + * ]| Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the + * documentation of vorbisenc. * </refsect2> * * Last reviewed on 2006-03-01 (0.10.4) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 5710eda6..05571b2c 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -23,41 +23,33 @@ * SECTION:element-jackaudiosink * @see_also: #GstBaseAudioSink, #GstRingBuffer * - * <refsect2> - * <para> * A Sink that outputs data to Jack ports. - * </para> - * <para> + * * It will create N Jack ports named out_<name>_<num> where * <name> is the element name and <num> is starting from 1. * Each port corresponds to a gstreamer channel. - * </para> - * <para> + * * The samplerate as exposed on the caps is always the same as the samplerate of * the jack server. - * </para> - * <para> - * When the ::connect property is set to auto, this element will try to connect - * each output port to a random physical jack input pin. In this mode, the sink - * will expose the number of physical channels on its pad caps. - * </para> - * <para> - * When the ::connect property is set to none, the element will accept any - * number of input channels and will create (but not connect) an output port for - * each channel. - * </para> - * <para> + * + * When the #GstJackAudioSink:connect property is set to auto, this element + * will try to connect each output port to a random physical jack input pin. In + * this mode, the sink will expose the number of physical channels on its pad + * caps. + * + * When the #GstJackAudioSink:connect property is set to none, the element will + * accept any number of input channels and will create (but not connect) an + * output port for each channel. + * * The element will generate an error when the Jack server is shut down when it * was PAUSED or PLAYING. This element does not support dynamic rate and buffer * size changes at runtime. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch audiotestsrc ! jackaudiosink - * </programlisting> - * Play a sine wave to using jack. - * </para> + * ]| Play a sine wave to using jack. * </refsect2> * * Last reviewed on 2006-11-30 (0.10.4) diff --git a/ext/mpeg2enc/gstmpeg2enc.cc b/ext/mpeg2enc/gstmpeg2enc.cc index 24a582f5..87902da9 100644 --- a/ext/mpeg2enc/gstmpeg2enc.cc +++ b/ext/mpeg2enc/gstmpeg2enc.cc @@ -24,40 +24,32 @@ * SECTION:element-mpeg2enc * @see_also: mpeg2dec * - * <refsect2> - * <para> * This element encodes raw video into an MPEG ?? stream using the * <ulink url="http://mjpeg.sourceforge.net/">mjpegtools</ulink> library. * Documentation on MPEG encoding in general can be found in the * <ulink url="https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776#s7">MJPEG Howto</ulink> * and on the various available parameters in the documentation * of the mpeg2enc tool in particular, which shares options with this element. - * </para> + * + * <refsect2> * <title>Example pipeline</title> - * <para> - * <programlisting> + * |[ * gst-launch-0.10 videotestsrc num-buffers=1000 ! mpeg2enc ! filesink location=videotestsrc.m1v - * </programlisting> - * This example pipeline will encode a test video source to a an - * MPEG1 elementary stream (with Generic MPEG1 profile). - * </para> + * ]| This example pipeline will encode a test video source to a an MPEG1 + * elementary stream (with Generic MPEG1 profile). * <para> - * Likely, the <link linkend="GstMpeg2enc--format">format</link> property + * Likely, the #GstMpeg2enc:format property * is most important, as it selects the type of MPEG stream that is produced. * In particular, default property values are dependent on the format, * and can even be forcibly restrained to certain pre-sets (and thereby ignored). * Note that the (S)VCD profiles also restrict the image size, so some scaling * may be needed to accomodate this. The so-called generic profiles (as used * in the example above) allow most parameters to be adjusted. - * <programlisting> - * gst-launch-0.10 videotestsrc num-buffers=1000 ! videoscale \ - * ! mpeg2enc format=1 norm=p ! filesink location=videotestsrc.m1v - * </programlisting> - * (write everything in one line, without the backslash characters) - * This will produce an MPEG1 profile stream according to VCD2.0 specifications - * for PAL <link linkend="GstMpeg2enc--norm">norm</link> (as the image height - * is dependent on video norm). * </para> + * |[ + * gst-launch-0.10 videotestsrc num-buffers=1000 ! videoscale ! mpeg2enc format=1 norm=p ! filesink location=videotestsrc.m1v + * ]| This will produce an MPEG1 profile stream according to VCD2.0 specifications + * for PAL #GstMpeg2enc:norm (as the image height is dependent on video norm). * </refsect2> */ diff --git a/ext/mplex/gstmplex.cc b/ext/mplex/gstmplex.cc index e9535fe8..297bdcdd 100644 --- a/ext/mplex/gstmplex.cc +++ b/ext/mplex/gstmplex.cc @@ -24,8 +24,6 @@ * SECTION:element-mplex * @see_also: mpeg2enc * - * <refsect2> - * <para> * 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 <ulink url="http://mjpeg.sourceforge.net/">mjpegtools</ulink> library. @@ -33,15 +31,13 @@ * <ulink url="https://sourceforge.net/docman/display_doc.php?docid=3456&group_id=5776#s7">MJPEG Howto</ulink> * and the man-page of the mplex tool documents the properties of this element, * which are shared with the mplex tool. - * </para> + * + * <refsect2> * <title>Example pipeline</title> - * <para> - * <programlisting> + * |[ * gst-launch -v videotestsrc num-buffers=1000 ! mpeg2enc ! mplex ! filesink location=videotestsrc.mpg - * </programlisting> - * This example pipeline will encode a test video source to a an + * ]| This example pipeline will encode a test video source to an * MPEG1 elementary stream and multiplexes this to an MPEG system stream. - * </para> * <para> * If several streams are being multiplexed, there should (as usual) be * a queue in each stream, and due to mplex' buffering the capacities of these diff --git a/ext/musicbrainz/gsttrm.c b/ext/musicbrainz/gsttrm.c index a4441f26..a4883b35 100644 --- a/ext/musicbrainz/gsttrm.c +++ b/ext/musicbrainz/gsttrm.c @@ -21,35 +21,27 @@ /** * SECTION:element-trm * - * <refsect2> - * <para> * GstTRM computes <ulink url="http://www.musicbrainz.org/">MusicBrainz</ulink> * TRM identifiers for audio streams using libmusicbrainz. - * </para> - * <para> + * * A TRM identifier is something like an 'acoustic fingerprint', the aim is * to uniquely identify the same song regardless of which source it comes from * or which audio format the stream is in. - * </para> - * <para> + * * The TRM element will collect about 30 seconds of audio and let * libmusicbrainz calculate a preliminary audio signature from that. That audio * signature will then be sent over the internet to a musicbrainz.org server * which will calculate the TRM for that signature. - * </para> - * <para> + * * The TRM element will post a tag message with a #GST_TAG_MUSICBRAINZ_TRMID * tag on the bus once the TRM has been calculated (and also send a tag event * with that information downstream). - * </para> + * + * <refsect2> * <title>Example pipeline</title> - * <para> - * Here is a test pipeline to test the TRM element: - * <programlisting> + * |[ * gst-launch -m filesrc location=somefile.ogg ! decodebin ! audioconvert ! trm ! fakesink - * </programlisting> - * this should (among many other things) print the tag message with the TRM ID. - * </para> + * ]| calculate the TRM and print the tag message with the TRM ID. * </refsect2> */ diff --git a/ext/mythtv/gstmythtvsrc.c b/ext/mythtv/gstmythtvsrc.c index 60a445f0..894d9b68 100644 --- a/ext/mythtv/gstmythtvsrc.c +++ b/ext/mythtv/gstmythtvsrc.c @@ -17,47 +17,38 @@ /** * SECTION:element-mythtvsrc + * @see_also: nuvdemux * - * <refsect2> - * <para> * MythTVSrc allows to access a remote MythTV backend streaming Video/Audio server, * and to render audio and video content through a TCP/IP connection to a specific * port on this server, and based on a known MythTV protocol that is based on * some message passing, such as REQUEST_BLOCK on a specified number of bytes, to get * some chunk of remote file data. * You should pass the information aboute the remote MythTV backend server - * through the <link linkend="GstMythTVSrc--location">location</link> property. - * </para> + * through the #GstMythtvSrc:location property. + * + * <refsect2> * <title>Examples</title> * <para> * If you want to get the LiveTV content (set channel, TV tuner, RemoteEncoder, - * Recorder), - * put the following URI: - * - * <programlisting> + * Recorder), use the following URI: + * <programlisting> * myth://xxx.xxx.xxx.xxx:6543/livetv?channel=BBC * </programlisting> * - * This URI will say to the gmyth library to configure the Recorder instance (used to - * change the channel, start the TV multimedia content transmition, etc.), using + * This URI will configure the Recorder instance (used to change the channel, + * start the TV multimedia content transmition, etc.), using * the IP address (xxx.xxx.xxx.xxx) and port number (6543) of the MythTV backend * server, and setting the channel name to "BBC". * * To get a already recorded the MythTV NUV file, put the following URI: - * * <programlisting> * myth://xxx.xxx.xxx.xxx:6543/filename.nuv * </programlisting> - * - * This URI will say to the gmyth library to configure the Recorder instance (used to - * change the channel, start the TV multimedia content transmition, etc.), using - * the IP address (xxx.xxx.xxx.xxx) and port number (6543) of the MythTV backend - * server, and setting the channel name to "BBC". * * Another possible way to use the LiveTV content, and just in the case you want to * use the mysql database, put the location URI in the following format: - * - * <programlisting> + * <programlisting> * myth://mythtv:mythtv@xxx.xxx.xxx.xxx:6543/?mythconverg&channel=9 * </programlisting> * diff --git a/ext/theora/theoradec.c b/ext/theora/theoradec.c index 4a46c826..afb0841e 100644 --- a/ext/theora/theoradec.c +++ b/ext/theora/theoradec.c @@ -22,21 +22,17 @@ * SECTION:element-theoradecexp * @see_also: theoradec, theoraenc, oggdemux * - * <refsect2> - * <para> * This element decodes theora streams into raw video using the theora-exp * decoder * <ulink url="http://www.theora.org/">Theora</ulink> is a royalty-free * video codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org * Foundation</ulink>, based on the VP3 codec. - * </para> - * <para> - * </para> + * + * <refsect2> * <title>Example pipeline</title> - * <programlisting> + * |[ * gst-launch -v filesrc location=videotestsrc.ogg ! oggdemux ! theoraexpdec ! xvimagesink - * </programlisting> - * This example pipeline will demux an ogg stream and decode the theora video, + * ]| This example pipeline will demux an ogg stream and decode the theora video, * displaying it on screen. * </refsect2> */ diff --git a/ext/timidity/gsttimidity.c b/ext/timidity/gsttimidity.c index 7dd89e90..09c2e778 100644 --- a/ext/timidity/gsttimidity.c +++ b/ext/timidity/gsttimidity.c @@ -23,18 +23,14 @@ * SECTION:element-timidity * @see_also: wildmidi * - * <refsect2> - * <para> * This element renders midi-files as audio streams using * <ulink url="http://timidity.sourceforge.net/">Timidity</ulink>. - * </para> - * <para> - * </para> + * + * <refsect2> * <title>Example pipeline</title> - * <programlisting> + * |[ * gst-launch filesrc location=song.mid ! timidity ! alsasink - * </programlisting> - * This example pipeline will parse the midi and render to raw audio which is + * ]| This example pipeline will parse the midi and render to raw audio which is * played via alsa. * </refsect2> */ diff --git a/ext/timidity/gstwildmidi.c b/ext/timidity/gstwildmidi.c index cded0d77..b8f72aa8 100644 --- a/ext/timidity/gstwildmidi.c +++ b/ext/timidity/gstwildmidi.c @@ -23,21 +23,17 @@ * SECTION:element-wildmidi * @see_also: timidity * - * <refsect2> - * <para> * This element renders midi-files as audio streams using * <ulink url="http://wildmidi.sourceforge.net//">Wildmidi</ulink>. * It offers better sound quality compared to the timidity element. Wildmidi * uses the same sound-patches as timidity (it tries the path in $WILDMIDI_CFG, * $HOME/.wildmidirc and /etc/wildmidi.cfg) - * </para> - * <para> - * </para> + * + * <refsect2> * <title>Example pipeline</title> - * <programlisting> + * |[ * gst-launch filesrc location=song.mid ! wildmidi ! alsasink - * </programlisting> - * This example pipeline will parse the midi and render to raw audio which is + * ]| This example pipeline will parse the midi and render to raw audio which is * played via alsa. * </refsect2> */ |