diff options
| author | Stefan Kost <ensonic@users.sf.net> | 2009-01-28 12:29:42 +0200 |
|---|---|---|
| committer | Stefan Kost <ensonic@users.sf.net> | 2009-01-28 12:32:59 +0200 |
| commit | a99d3f8769ed3fd1266d5216ecefebfd1bdcf663 (patch) | |
| tree | 4a5cf5e0f2f44b1f9ccea5344c38ef98f0a92990 /gst | |
| parent | 00fdca0c14eb9a5fe6b8b9f2d5ce2313e3b32f23 (diff) | |
Update and add documentation for plugins with no deps (gst).
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
Diffstat (limited to 'gst')
59 files changed, 536 insertions, 649 deletions
diff --git a/gst/alpha/gstalphacolor.c b/gst/alpha/gstalphacolor.c index 6e628781e..26086ef65 100644 --- a/gst/alpha/gstalphacolor.c +++ b/gst/alpha/gstalphacolor.c @@ -20,12 +20,8 @@ /** * SECTION:element-alphacolor * - * <refsect2> - * <para> * The alphacolor element does memory-efficient (in-place) colourspace * conversion from RGBA to AYUV, preserving the alpha channel. - * </para> - * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/gst/apetag/gstapedemux.c b/gst/apetag/gstapedemux.c index 7adc2ccdc..fc9c3aae1 100644 --- a/gst/apetag/gstapedemux.c +++ b/gst/apetag/gstapedemux.c @@ -20,33 +20,26 @@ /** * SECTION:element-apedemux - * @short_description: reads tag information from APE tag data blocks and - * outputs them as GStreamer tag messages and events. * - * <refsect2> - * <para> * apedemux accepts data streams with APE tags at the start or at the end * (or both). The mime type of the data between the tag blocks is detected * using typefind functions, and the appropriate output mime type set on - * outgoing buffers. - * </para> - * <para> + * outgoing buffers. + * * The element is only able to read APE tags at the end of a stream from * a seekable stream, ie. when get_range mode is supported by the upstream * elements. If get_range operation is available, apedemux makes it available * downstream. This means that elements which require get_range mode, such as * wavparse or musepackdec, can operate on files containing APE tag * information. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch -t filesrc location=file.mpc ! apedemux ! fakesink - * </programlisting> - * This pipeline should read any available APE tag information and output it. + * ]| This pipeline should read any available APE tag information and output it. * The contents of the file inside the APE tag regions should be detected, and * the appropriate mime type set on buffers produced from apedemux. - * </para> * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c index ce4a0042b..da5412a48 100644 --- a/gst/audiofx/audioamplify.c +++ b/gst/audiofx/audioamplify.c @@ -21,19 +21,16 @@ /** * SECTION:element-audioamplify - * @short_description: Amplifies an audio stream with selectable clipping mode * - * <refsect2> * Amplifies an audio stream by a given factor and allows the selection of different clipping modes. * The difference between the clipping modes is best evaluated by testing. * <title>Example launch line</title> - * <para> - * <programlisting> + * <refsect2> + * |[ * gst-launch audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 method=wrap-negative ! alsasink * gst-launch audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 method=wrap-positive ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c index bf9c205ef..7ecb7956c 100644 --- a/gst/audiofx/audiochebband.c +++ b/gst/audiofx/audiochebband.c @@ -34,42 +34,35 @@ /** * SECTION:element-audiochebband - * @short_description: Chebyshev band pass and band reject filter * - * <refsect2> - * <para> * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency * band. The number of poles and the ripple parameter control the rolloff. - * </para> - * <para> + * * This element has the advantage over the windowed sinc bandpass and bandreject filter that it is * much faster and produces almost as good results. It's only disadvantages are the highly * non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel. - * </para> - * <para> + * * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. * some frequencies in the passband will be amplified by that value. A higher ripple value will allow * a faster rolloff. - * </para> - * <para> + * * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will * be at most this value. A lower ripple value will allow a faster rolloff. - * </para> - * <para> + * * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. - * </para> - * <para><note> + * + * <note> * Be warned that a too large number of poles can produce noise. The most poles are possible with * a cutoff frequency at a quarter of the sampling rate. - * </note></para> + * </note> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c index b4efbb3af..5d91909ef 100644 --- a/gst/audiofx/audiocheblimit.c +++ b/gst/audiofx/audiocheblimit.c @@ -30,42 +30,35 @@ /** * SECTION:element-audiocheblimit - * @short_description: Chebyshev low pass and high pass filter * - * <refsect2> - * <para> * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff. - * </para> - * <para> + * * This element has the advantage over the windowed sinc lowpass and highpass filter that it is * much faster and produces almost as good results. It's only disadvantages are the highly * non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel. - * </para> - * <para> + * * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. * some frequencies in the passband will be amplified by that value. A higher ripple value will allow * a faster rolloff. - * </para> - * <para> + * * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will * be at most this value. A lower ripple value will allow a faster rolloff. - * </para> - * <para> + * * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. * </para> - * <para><note> + * <note><para> * Be warned that a too large number of poles can produce noise. The most poles are possible with * a cutoff frequency at a quarter of the sampling rate. - * </note></para> - * <title>Example launch line</title> + * </para></note> * <para> - * <programlisting> + * <refsect2> + * <title>Example launch line</title> + * |[ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audiodynamic.c b/gst/audiofx/audiodynamic.c index 7a84fa810..240c270e7 100644 --- a/gst/audiofx/audiodynamic.c +++ b/gst/audiofx/audiodynamic.c @@ -20,21 +20,19 @@ /** * SECTION:element-audiodynamic - * @short_description: Compressor and Expander * - * <refsect2> * This element can act as a compressor or expander. A compressor changes the * amplitude of all samples above a specific threshold with a specific ratio, * a expander does the same for all samples below a specific threshold. If * soft-knee mode is selected the ratio is applied smoothly. + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink * gst-launch audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audioecho.c b/gst/audiofx/audioecho.c index dd209a235..6676d7919 100644 --- a/gst/audiofx/audioecho.c +++ b/gst/audiofx/audioecho.c @@ -20,24 +20,22 @@ /** * SECTION:element-audioecho + * @Since: 0.10.12 * - * <refsect2> * audioecho adds an echo or (simple) reverb effect to an audio stream. The echo * delay, intensity and the percentage of feedback can be configured. - * <para> + * * For getting an echo effect you have to set the delay to a larger value, * for example 200ms and more. Everything below will result in a simple * reverb effect, which results in a slightly metallic sounding. - * </para> - * <para> - * <programlisting> + * + * <refsect2> + * <title>Example launch line</title> + * |[ * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink - * </programlisting> - * </para> + * ]| * </refsect2> - * - * Since: 0.10.12 */ #ifdef HAVE_CONFIG_H diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c index 3ee4d832d..df4e2dcb5 100644 --- a/gst/audiofx/audiofirfilter.c +++ b/gst/audiofx/audiofirfilter.c @@ -21,31 +21,27 @@ /** * SECTION:element-audiofirfilter - * @short_description: Generic audio FIR filter * - * <refsect2> - * <para> * audiofirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Finite_impulse_response">FIR filter</ulink>. Before usage the * "kernel" property has to be set to the filter kernel that should be * used and the "latency" property has to be set to the latency (in samples) * that is introduced by the filter kernel. Setting a latency of n samples * will lead to the first n samples being dropped from the output and * n samples added to the end. - * </para> - * <para> + * * The filter kernel describes the impulse response of the filter. To * calculate the frequency response of the filter you have to calculate * the Fourier Transform of the impulse response. - * </para> - * <para> + * * To change the filter kernel whenever the sampling rate changes the * "rate-changed" signal can be used. This should be done for most * FIR filters as they're depending on the sampling rate. - * </para> + * + * <refsect2> * <title>Example application</title> - * <para> - * <include xmlns="http://www.w3.org/2003/XInclude" href="element-firfilter-example.xml" /> - * </para> + * |[ + * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/audiofx/firfilter-example.c" /> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c index 76112c6f1..1f0631220 100644 --- a/gst/audiofx/audioiirfilter.c +++ b/gst/audiofx/audioiirfilter.c @@ -21,27 +21,23 @@ /** * SECTION:element-audioiirfilter - * @short_description: Generic audio IIR filter * - * <refsect2> - * <para> * audioiirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Infinite_impulse_response">IIR filter</ulink>. Before usage the * "a" and "b" properties have to be set to the filter coefficients that * should be used. - * </para> - * <para> + * * The filter coefficients describe the numerator and denominator of the * transfer function. - * </para> - * <para> + * * To change the filter coefficients whenever the sampling rate changes the * "rate-changed" signal can be used. This should be done for most * IIR filters as they're depending on the sampling rate. - * </para> + * + * <refsect2> * <title>Example application</title> - * <para> - * <include xmlns="http://www.w3.org/2003/XInclude" href="element-iirfilter-example.xml" /> - * </para> + * |[ + * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/audiofx/iirfilter-example.c" /> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audioinvert.c b/gst/audiofx/audioinvert.c index 188793f50..a6911697f 100644 --- a/gst/audiofx/audioinvert.c +++ b/gst/audiofx/audioinvert.c @@ -21,20 +21,18 @@ /** * SECTION:element-audioinvert - * @short_description: Swaps upper and lower half of audio samples * - * <refsect2> * Swaps upper and lower half of audio samples. Mixing an inverted sample on top of * the original with a slight delay can produce effects that sound like resonance. * Creating a stereo sample from a mono source, with one channel inverted produces wide-stereo sounds. + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch audiotestsrc wave=saw ! audioinvert invert=0.4 ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert invert=0.4 ! alsasink * gst-launch audiotestsrc wave=saw ! audioconvert ! audioinvert invert=0.4 ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audiokaraoke.c b/gst/audiofx/audiokaraoke.c index ec5056810..fe34971e3 100644 --- a/gst/audiofx/audiokaraoke.c +++ b/gst/audiofx/audiokaraoke.c @@ -20,17 +20,15 @@ /** * SECTION:element-audiokaraoke - * @short_description: Voice removal element * - * <refsect2> * Remove the voice from audio by filtering the center channel. * This plugin is useful for karaoke applications. + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audiopanorama.c b/gst/audiofx/audiopanorama.c index e38f10d2a..3f57648dd 100644 --- a/gst/audiofx/audiopanorama.c +++ b/gst/audiofx/audiopanorama.c @@ -21,20 +21,18 @@ /** * SECTION:element-audiopanorama - * @short_description: audio stereo pan effect * - * <refsect2> * Stereo panorama effect with controllable pan position. One can choose between the default psychoacoustic panning method, * which keeps the same perceived loudness, and a simple panning method that just controls the volume on one channel. + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch audiotestsrc wave=saw ! audiopanorama panorama=-1.00 ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiopanorama panorama=-1.00 ! alsasink * gst-launch audiotestsrc wave=saw ! audioconvert ! audiopanorama panorama=-1.00 ! audioconvert ! alsasink * gst-launch audiotestsrc wave=saw ! audioconvert ! audiopanorama method=simple panorama=-0.50 ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index 69bf5c195..70c85b480 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -30,28 +30,23 @@ /** * SECTION:element-audiowsincband - * @short_description: Windowed Sinc band pass and band reject filter * - * <refsect2> - * <para> * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency * band. The length parameter controls the rolloff, the window parameter * controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit * worse stopband attenuation, the other way around for the Blackman window. - * </para> - * <para> + * * This element has the advantage over the Chebyshev bandpass and bandreject filter that it has * a much better rolloff when using a larger kernel size and almost linear phase. The only * disadvantage is the much slower execution time with larger kernels. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiosincband mode=band-pass lower-frequency=3000 upper-frequency=10000 length=501 window=blackman ! audioconvert ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsincband mode=band-reject lower-frequency=59 upper-frequency=61 length=10001 window=hamming ! audioconvert ! alsasink * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiowsincband mode=band-pass lower-frequency=1000 upper-frequency=2000 length=31 ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index 1f33ad2db..73bdbe505 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.c @@ -30,28 +30,23 @@ /** * SECTION:element-audiowsinclimit - * @short_description: Windowed Sinc low pass and high pass filter * - * <refsect2> - * <para> * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the * cutoff frequency (high-pass). The length parameter controls the rolloff, the window parameter * controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit * worse stopband attenuation, the other way around for the Blackman window. - * </para> - * <para> + * * This element has the advantage over the Chebyshev lowpass and highpass filter that it has * a much better rolloff when using a larger kernel size and almost linear phase. The only * disadvantage is the much slower execution time with larger kernels. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=501 ! audioconvert ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsinclimit mode=high-pass frequency=15000 length=501 ! audioconvert ! alsasink * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiowsinclimit mode=low-pass frequency=1000 length=10001 window=blackman ! audioconvert ! alsasink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/auparse/gstauparse.c b/gst/auparse/gstauparse.c index 1f0dccc06..14996ae19 100644 --- a/gst/auparse/gstauparse.c +++ b/gst/auparse/gstauparse.c @@ -20,13 +20,8 @@ /** * SECTION:element-auparse - * @short_description: .au file parser * - * <refsect2> - * <para> - * Parses .au files. - * </para> - * </refsect2> + * Parses .au files mostly originating from sun os based computers. */ #ifdef HAVE_CONFIG_H diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 9170935dd..cf7b0c125 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -22,24 +22,19 @@ /** * SECTION:element-avidemux * - * <refsect2> - * <para> * Demuxes an .avi file into raw or compressed audio and/or video streams. - * </para> - * <para> + * * This element supports both push and pull-based scheduling, depending on the * capabilities of the upstream elements. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=test.avi ! avidemux name=demux demux.audio_00 ! decodebin ! audioconvert ! audioresample ! autoaudiosink demux.video_00 ! queue ! decodebin ! ffmpegcolorspace ! videoscale ! autovideosink - * </programlisting> - * Play (parse and decode) an .avi file and try to output it to + * ]| Play (parse and decode) an .avi file and try to output it to * an automatically detected soundcard and videosink. If the AVI file contains * compressed audio or video data, this will only work if you have the * right decoder elements/plugins installed. - * </para> * </refsect2> * * Last reviewed on 2006-12-29 (0.10.6) diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index b432c6aa6..e648d3e43 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -28,42 +28,32 @@ /** * SECTION:element-avimux * - * <refsect2> - * <para> * Muxes raw or compressed audio and/or video streams into an AVI file. - * </para> - * <title>Example launch line</title> - * <para> - * (write everything in one line, without the backslash characters) - * <programlisting> + * + * <refsect2> + * <title>Example launch lines</title> + * <para>(write everything in one line, without the backslash characters)</para> + * |[ * gst-launch-0.10 videotestsrc num-buffers=250 \ * ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240,framerate=(fraction)25/1' \ * ! queue ! mux. \ * audiotestsrc num-buffers=440 ! audioconvert \ * ! 'audio/x-raw-int,rate=44100,channels=2' ! queue ! mux. \ * avimux name=mux ! filesink location=test.avi - * </programlisting> - * This will create an .AVI file containing an uncompressed video stream + * ]| This will create an .AVI file containing an uncompressed video stream * with a test picture and an uncompressed audio stream containing a * test sound. - * </para> - * <title>Another example launch line</title> - * <para> - * (write everything in one line, without the backslash characters) - * <programlisting> + * |[ * gst-launch-0.10 videotestsrc num-buffers=250 \ * ! 'video/x-raw-yuv,format=(fourcc)I420,width=320,height=240,framerate=(fraction)25/1' \ * ! xvidenc ! queue ! mux. \ * audiotestsrc num-buffers=440 ! audioconvert ! 'audio/x-raw-int,rate=44100,channels=2' \ * ! lame ! queue ! mux. \ * avimux name=mux ! filesink location=test.avi - * </programlisting> - * This will create an .AVI file containing the same test video and sound + * ]| This will create an .AVI file containing the same test video and sound * as above, only that both streams will be compressed this time. This will * only work if you have the necessary encoder elements installed of course. - * </para> * </refsect2> - * */ #ifdef HAVE_CONFIG_H diff --git a/gst/cutter/gstcutter.c b/gst/cutter/gstcutter.c index 9ebf95008..60ce3703c 100644 --- a/gst/cutter/gstcutter.c +++ b/gst/cutter/gstcutter.c @@ -18,6 +18,37 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-cutter + * + * Analyses the audio signal for periods of silence. The start and end of + * silence is signalled by bus messages named + * <classname>"cutter"</classname>. + * The message's structure contains two fields: + * <itemizedlist> + * <listitem> + * <para> + * #GstClockTime + * <classname>"timestamp"</classname>: + * the timestamp of the buffer that triggered the message. + * </para> + * </listitem> + * <listitem> + * <para> + * gboolean + * <classname>"above"</classname>: + * %TRUE for begin of silence and %FALSE for end of silence. + * </para> + * </listitem> + * </itemizedlist> + * + * <refsect2> + * <title>Example launch line</title> + * |[ + * gst-launch -m filesrc location=foo.ogg ! decodebin ! audioconvert ! cutter ! autoaudiosink + * ]| Show cut messages. + * </refsect2> + */ #ifdef HAVE_CONFIG_H #include "config.h" @@ -82,7 +113,7 @@ static void gst_cutter_get_property (GObject * object, guint prop_id, static GstFlowReturn gst_cutter_chain (GstPad * pad, GstBuffer * buffer); -void gst_cutter_get_caps (GstPad * pad, GstCutter * filter); +static gboolean gst_cutter_get_caps (GstPad * pad, GstCutter * filter); static void gst_cutter_base_init (gpointer g_class) @@ -224,8 +255,10 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) g_return_val_if_fail (filter != NULL, GST_FLOW_ERROR); g_return_val_if_fail (GST_IS_CUTTER (filter), GST_FLOW_ERROR); - if (!filter->have_caps) - gst_cutter_get_caps (pad, filter); + if (!filter->have_caps) { + if (!(gst_cutter_get_caps (pad, filter))) + return GST_FLOW_NOT_NEGOTIATED; + } in_data = (gint16 *) GST_BUFFER_DATA (buf); GST_LOG_OBJECT (filter, "length of prerec buffer: %" GST_TIME_FORMAT, @@ -257,12 +290,12 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) */ GST_LOG_OBJECT (filter, "buffer stats: NMS %f, RMS %f, audio length %f", NMS, RMS, - gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer (filter-> - sinkpad, buf))); + gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer + (filter->sinkpad, buf))); if (RMS < filter->threshold_level) filter->silent_run_length += - gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer (filter-> - sinkpad, buf)); + gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer + (filter->sinkpad, buf)); else { filter->silent_run_length = 0 * GST_SECOND; filter->silent = FALSE; @@ -306,15 +339,15 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) if (filter->silent) { filter->pre_buffer = g_list_append (filter->pre_buffer, buf); filter->pre_run_length += - gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer (filter-> - sinkpad, buf)); + gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer + (filter->sinkpad, buf)); while (filter->pre_run_length > filter->pre_length) { prebuf = (g_list_first (filter->pre_buffer))->data; g_assert (GST_IS_BUFFER (prebuf)); filter->pre_buffer = g_list_remove (filter->pre_buffer, prebuf); filter->pre_run_length -= - gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer (filter-> - sinkpad, prebuf)); + gst_guint64_to_gdouble (gst_audio_duration_from_pad_buffer + (filter->sinkpad, prebuf)); /* only pass buffers if we don't leak */ if (!filter->leaky) gst_pad_push (filter->srcpad, prebuf); @@ -327,6 +360,28 @@ gst_cutter_chain (GstPad * pad, GstBuffer * buf) return GST_FLOW_OK; } + +static gboolean +gst_cutter_get_caps (GstPad * pad, GstCutter * filter) +{ + GstCaps *caps; + GstStructure *structure; + + caps = gst_pad_get_caps (pad); + if (!caps) { + GST_INFO ("no caps on pad %s:%s", GST_DEBUG_PAD_NAME (pad)); + return FALSE; + } + structure = gst_caps_get_structure (caps, 0); + gst_structure_get_int (structure, "width", &filter->width); + filter->max_sample = 1 << (filter->width - 1); /* signed */ + filter->have_caps = TRUE; + + gst_caps_unref (caps); + return TRUE; +} + + static void gst_cutter_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) @@ -414,21 +469,3 @@ GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, "cutter", "Audio Cutter to split audio into non-silent bits", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); - - -void -gst_cutter_get_caps (GstPad * pad, GstCutter * filter) -{ - GstCaps *caps; - GstStructure *structure; - - caps = gst_pad_get_caps (pad); - /* FIXME : Please change this to a better warning method ! */ - g_assert (caps != NULL); - structure = gst_caps_get_structure (caps, 0); - gst_structure_get_int (structure, "width", &filter->width); - filter->max_sample = 1 << (filter->width - 1); /* signed */ - filter->have_caps = TRUE; - - gst_caps_unref (caps); -} diff --git a/gst/debug/gstpushfilesrc.c b/gst/debug/gstpushfilesrc.c index 7ef5259c1..47daa83e1 100644 --- a/gst/debug/gstpushfilesrc.c +++ b/gst/debug/gstpushfilesrc.c @@ -19,26 +19,21 @@ /** * SECTION:element-pushfilesrc - * @short_description: Works like a filesrc, but only push-based (for debugging) * @see_also: filesrc * - * <refsect2> - * <para> * This element is only useful for debugging purposes. It implements an URI * protocol handler for the 'pushfile' protocol and behaves like a file source * element that cannot be activated in pull-mode. This makes it very easy to * debug demuxers or decoders that can operate both pull and push-based in * connection with the playbin element (which creates a source based on the * URI passed). - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch -m playbin uri=pushfile:///home/you/some/file.ogg - * </programlisting> - * This plays back the given file using playbin, with the demuxer operating + * ]| This plays back the given file using playbin, with the demuxer operating * push-based. - * </para> * </refsect2> */ @@ -156,6 +151,7 @@ gst_push_file_src_uri_get_type (void) { return GST_URI_SRC; } + static gchar ** gst_push_file_src_uri_get_protocols (void) { @@ -163,6 +159,7 @@ gst_push_file_src_uri_get_protocols (void) return protocols; } + static const gchar * gst_push_file_src_uri_get_uri (GstURIHandler * handler) { diff --git a/gst/debug/gsttaginject.c b/gst/debug/gsttaginject.c index 840aa2a40..3429fb147 100644 --- a/gst/debug/gsttaginject.c +++ b/gst/debug/gsttaginject.c @@ -23,12 +23,16 @@ * * Element that injects new metadata tags, but passes incomming data through * unmodified. + * + * <refsect2> + * <title>Example launch lines</title> * |[ * gst-launch audiotestsrc num-buffers=100 ! taginject tags="title=testsrc,artist=gstreamer" ! vorbisenc ! oggmux ! filesink location=test.ogg * ]| set title and artist * |[ * gst-launch audiotestsrc num-buffers=100 ! taginject tags="keywords=\"testone,audio\",title=\"audio testtone\"" ! vorbisenc ! oggmux ! filesink location=test.ogg * ]| set keywords and title demonstrating quoting of special chars + * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/gst/debug/progressreport.c b/gst/debug/progressreport.c index e79f6fd93..8f2547b38 100644 --- a/gst/debug/progressreport.c +++ b/gst/debug/progressreport.c @@ -22,32 +22,26 @@ /** * SECTION:element-progressreport - * @short_description: Reports progress - * @see_also: * - * <refsect2> - * <para> * The progressreport element can be put into a pipeline to report progress, * which is done by doing upstream duration and position queries in regular * (real-time) intervals. Both the interval and the prefered query format - * can be specified via the "update-freq" and the "format" property. - * </para> - * <para> + * can be specified via the #GstProgressReport:update-freq and the + * #GstProgressReport:format property. + * * Element messages containing a "progress" structure are posted on the bus * whenever progress has been queried (since gst-plugins-good 0.10.6 only). - * </para> - * <para> + * * Since the element was originally designed for debugging purposes, it will * by default also print information about the current progress to the - * terminal. This can be prevented by setting the "silent" property to TRUE. - * </para> - * <para> + * terminal. This can be prevented by setting the #GstProgressReport:silent + * property to %TRUE. + * * This element is most useful in transcoding pipelines or other situations * where just querying the pipeline might not lead to the wanted result. For * progress in TIME format, the element is best placed in a 'raw stream' * section of the pipeline (or after any demuxers/decoders/parsers). - * </para> - * <para> + * * Three more things should be pointed out: firstly, the element will only * query progress when data flow happens. If data flow is stalled for some * reason, no progress messages will be posted. Secondly, there are other @@ -58,20 +52,15 @@ * take action when they receive an EOS message (since the progress reported * is in reference to an internal point of a pipeline and not the pipeline as * a whole). - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch -m filesrc location=foo.ogg ! decodebin ! progressreport update-freq=1 ! audioconvert ! audioresample ! autoaudiosink - * </programlisting> - * This shows a progress query where a duration is available. - * </para> - * <para> - * <programlisting> + * ]| This shows a progress query where a duration is available. + * |[ * gst-launch -m audiotestsrc ! progressreport update-freq=1 ! audioconvert ! autoaudiosink - * </programlisting> - * This shows a progress query where no duration is available. - * </para> + * ]| This shows a progress query where no duration is available. * </refsect2> */ diff --git a/gst/equalizer/gstiirequalizer10bands.c b/gst/equalizer/gstiirequalizer10bands.c index 833c80699..e56bf19ee 100644 --- a/gst/equalizer/gstiirequalizer10bands.c +++ b/gst/equalizer/gstiirequalizer10bands.c @@ -19,20 +19,15 @@ /** * SECTION:element-equalizer-10bands - * @short_description: 10-band equalizer * - * <refsect2> - * <para> * The 10 band equalizer element allows to change the gain of 10 equally distributed * frequency bands between 30 Hz and 15 kHz. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audioconvert ! equalizer-10bands band2=3.0 ! alsasink - * </programlisting> - * This raises the volume of the 3rd band which is at 119 Hz by 3 db. - * </para> + * ]| This raises the volume of the 3rd band which is at 119 Hz by 3 db. * </refsect2> */ diff --git a/gst/equalizer/gstiirequalizer3bands.c b/gst/equalizer/gstiirequalizer3bands.c index c550f7868..91129525c 100644 --- a/gst/equalizer/gstiirequalizer3bands.c +++ b/gst/equalizer/gstiirequalizer3bands.c @@ -19,20 +19,15 @@ /** * SECTION:element-equalizer-3bands - * @short_description: 3-band equalizer * - * <refsect2> - * <para> * The 3-band equalizer element allows to change the gain of a low frequency, * medium frequency and high frequency band. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audioconvert ! equalizer-3bands band1=6.0 ! alsasink - * </programlisting> - * This raises the volume of the 2nd band, which is at 1110 Hz, by 6 db. - * </para> + * ]| This raises the volume of the 2nd band, which is at 1110 Hz, by 6 db. * </refsect2> */ diff --git a/gst/equalizer/gstiirequalizernbands.c b/gst/equalizer/gstiirequalizernbands.c index 316cd0297..e2af57efe 100644 --- a/gst/equalizer/gstiirequalizernbands.c +++ b/gst/equalizer/gstiirequalizernbands.c @@ -20,67 +20,59 @@ /** * SECTION:element-equalizer-nbands - * @short_description: Fully parametric N-band equalizer * - * <refsect2> - * <para> * The n-band equalizer element is a fully parametric equalizer. It allows to * select between 1 and 64 bands and has properties on each band to change * the center frequency, band width and gain. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=song.ogg ! oggdemux ! vorbisdec ! audioconvert ! equalizer-nbands num-bands=15 band5::gain=6.0 ! alsasink - * </programlisting> - * This make the equalizer use 15 bands and raises the volume of the 5th band by 6 db. - * </para> + * ]| This make the equalizer use 15 bands and raises the volume of the 5th band by 6 db. + * </refsect2> + * <refsect2> * <title>Example code</title> - * <para> - * <programlisting> - -#include <gst/gst.h> - -... -typedef struct { - gfloat freq; - gfloat width; - gfloat gain; -} GstEqualizerBandState; - -... - - GstElement *equalizer; - GstObject *band; - gint i; - GstEqualizerBandState state[] = { - { 120.0, 50.0, - 3.0}, - { 500.0, 20.0, 12.0}, - {1503.0, 2.0, -20.0}, - {6000.0, 1000.0, 6.0}, - {3000.0, 120.0, 2.0} - }; - -... - - - equalizer = gst_element_factory_make ("equalizer-nbands", "equalizer"); - g_object_set (G_OBJECT (equalizer), "num-bands", 5, NULL); - -... - - for (i = 0; i < 5; i++) { - band = gst_child_proxy_get_child_by_index (GST_CHILD_PROXY (equalizer), i); - g_object_set (G_OBJECT (band), "freq", state[i].freq, - "bandwidth", state[i].width, - "gain", state[i].gain); - g_object_unref (G_OBJECT (band)); - } - -... - - * </programlisting> - * </para> + * |[ + * #include <gst/gst.h> + * + * ... + * typedef struct { + * gfloat freq; + * gfloat width; + * gfloat gain; + * } GstEqualizerBandState; + * + * ... + * + * GstElement *equalizer; + * GstObject *band; + * gint i; + * GstEqualizerBandState state[] = { + * { 120.0, 50.0, - 3.0}, + * { 500.0, 20.0, 12.0}, + * {1503.0, 2.0, -20.0}, + * {6000.0, 1000.0, 6.0}, + * {3000.0, 120.0, 2.0} + * }; + * + * ... + * + * equalizer = gst_element_factory_make ("equalizer-nbands", "equalizer"); + * g_object_set (G_OBJECT (equalizer), "num-bands", 5, NULL); + * + * ... + * + * for (i = 0; i < 5; i++) { + * band = gst_child_proxy_get_child_by_index (GST_CHILD_PROXY (equalizer), i); + * g_object_set (G_OBJECT (band), "freq", state[i].freq, + * "bandwidth", state[i].width, + * "gain", state[i].gain); + * g_object_unref (G_OBJECT (band)); + * } + * + * ... + * ]| * </refsect2> */ diff --git a/gst/flx/gstflxdec.c b/gst/flx/gstflxdec.c index 6f9cb7fee..8e93e98cd 100644 --- a/gst/flx/gstflxdec.c +++ b/gst/flx/gstflxdec.c @@ -19,10 +19,7 @@ /** * SECTION:element-flxdec * - * <refsect2> - * <para> * This element decodes fli/flc/flx-video into raw video - * </refsect2> */ /* * http://www.coolutils.com/Formats/FLI diff --git a/gst/goom/gstgoom.c b/gst/goom/gstgoom.c index 287d177ac..4c40c1b3d 100644 --- a/gst/goom/gstgoom.c +++ b/gst/goom/gstgoom.c @@ -22,17 +22,14 @@ * SECTION:element-goom * @see_also: synaesthesia * - * <refsect2> - * <para> * Goom is an audio visualisation element. It creates warping structures * based on the incoming audio signal. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch -v audiotestsrc ! goom ! ffmpegcolorspace ! xvimagesink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/goom2k1/gstgoom.c b/gst/goom2k1/gstgoom.c index bc6da3ceb..d01a88627 100644 --- a/gst/goom2k1/gstgoom.c +++ b/gst/goom2k1/gstgoom.c @@ -22,18 +22,15 @@ * SECTION:element-goom2k1 * @see_also: goom, synaesthesia * - * <refsect2> - * <para> * Goom2k1 is an audio visualisation element. It creates warping structures * based on the incomming audio signal. Goom2k1 is the older version of the * visualisation. Also available is goom2k4, with a different look. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch -v audiotestsrc ! goom2k1 ! ffmpegcolorspace ! xvimagesink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/icydemux/gsticydemux.c b/gst/icydemux/gsticydemux.c index 8ea4d1a5f..234f3f800 100644 --- a/gst/icydemux/gsticydemux.c +++ b/gst/icydemux/gsticydemux.c @@ -21,27 +21,21 @@ /** * SECTION:element-icydemux - * @short_description: reads tag information from an Icy (Icecast/Shoutcast) - * stream, outputting them as tag messages, and forwarding the enclosed data. * - * <refsect2> - * <para> * icydemux accepts data streams with ICY metadata at known intervals, as * transmitted from an upstream element (usually read as response headers from * an HTTP stream). The mime type of the data between the tag blocks is * detected using typefind functions, and the appropriate output mime type set * on outgoing buffers. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch souphttpsrc location=http://some.server/ iradio-mode=true ! icydemux ! fakesink -t - * </programlisting> - * This pipeline should read any available ICY tag information and output it. + * ]| This pipeline should read any available ICY tag information and output it. * The contents of the stream should be detected, and the appropriate mime * type set on buffers produced from icydemux. (Using gnomevfssrc, neonhttpsrc * or giosrc instead of souphttpsrc should also work.) - * </para> * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/gst/id3demux/gstid3demux.c b/gst/id3demux/gstid3demux.c index 14713756e..3e7a2b735 100644 --- a/gst/id3demux/gstid3demux.c +++ b/gst/id3demux/gstid3demux.c @@ -21,23 +21,28 @@ /** * SECTION:element-id3demux - * @short_description: reads tag information from ID3v1 and ID3v2 (<= 2.4.0) data blocks and outputs them as GStreamer tag messages and events. + * + * id3demux accepts data streams with either (or both) ID3v2 regions at the + * start, or ID3v1 at the end. The mime type of the data between the tag blocks + * is detected using typefind functions, and the appropriate output mime type + * set on outgoing buffers. + * + * The element is only able to read ID3v1 tags from a seekable stream, because + * they are at the end of the stream. That is, when get_range mode is supported + * by the upstream elements. If get_range operation is available, id3demux makes + * it available downstream. This means that elements which require get_range + * mode, such as wavparse, can operate on files containing ID3 tag information. + * + * This id3demux element replaced an older element with the same name which + * relied on libid3tag from the MAD project. * * <refsect2> - * <para> - * id3demux accepts data streams with either (or both) ID3v2 regions at the start, or ID3v1 at the end. The mime type of the data between the tag blocks is detected using typefind functions, and the appropriate output mime type set on outgoing buffers. - * </para><para> - * The element is only able to read ID3v1 tags from a seekable stream, because they are at the end of the stream. That is, when get_range mode is supported by the upstream elements. If get_range operation is available, id3demux makes it available downstream. This means that elements which require get_range mode, such as wavparse, can operate on files containing ID3 tag information. - * </para> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=file.mp3 ! id3demux ! fakesink -t - * </programlisting> - * This pipeline should read any available ID3 tag information and output it. The contents of the file inside the ID3 tag regions should be detected, and the appropriate mime type set on buffers produced from id3demux. - * </para><para> - * This id3demux element replaced an older element with the same name which relied on libid3tag from the MAD project. - * </para> + * ]| This pipeline should read any available ID3 tag information and output it. + * The contents of the file inside the ID3 tag regions should be detected, and + * the appropriate mime type set on buffers produced from id3demux. * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/gst/law/alaw-decode.c b/gst/law/alaw-decode.c index 6db91cfd8..9b990f7e1 100644 --- a/gst/law/alaw-decode.c +++ b/gst/law/alaw-decode.c @@ -16,6 +16,11 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-alawdec + * + * This element decodes alaw audio. Alaw coding is also known as G.711. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/law/alaw-encode.c b/gst/law/alaw-encode.c index 0a025fa29..ba8587a2d 100644 --- a/gst/law/alaw-encode.c +++ b/gst/law/alaw-encode.c @@ -16,6 +16,11 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-alawenc + * + * This element encode alaw audio. Alaw coding is also known as G.711. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/law/mulaw-decode.c b/gst/law/mulaw-decode.c index fb40f18bf..42b208f61 100644 --- a/gst/law/mulaw-decode.c +++ b/gst/law/mulaw-decode.c @@ -16,6 +16,11 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-mulawdec + * + * This element decodes mulaw audio. Mulaw coding is also known as G.711. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/law/mulaw-encode.c b/gst/law/mulaw-encode.c index c8c8c054c..f60e0b2e0 100644 --- a/gst/law/mulaw-encode.c +++ b/gst/law/mulaw-encode.c @@ -16,6 +16,11 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ +/** + * SECTION:element-mulawenc + * + * This element encode mulaw audio. Mulaw coding is also known as G.711. + */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/law/mulaw.c b/gst/law/mulaw.c index 8945a1012..3317e58fa 100644 --- a/gst/law/mulaw.c +++ b/gst/law/mulaw.c @@ -1,3 +1,21 @@ +/* GStreamer PCM/A-Law conversions + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2.1 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c index d4bdc4e0a..bf4d9628c 100644 --- a/gst/level/gstlevel.c +++ b/gst/level/gstlevel.c @@ -21,10 +21,7 @@ /** * SECTION:element-level - * @short_description: audio level analyzer * - * <refsect2> - * <para> * Level analyses incoming audio buffers and, if the #GstLevel:message property * is #TRUE, generates an element message named * <classname>"level"</classname>: @@ -95,11 +92,12 @@ * </para> * </listitem> * </itemizedlist> - * </para> + * + * <refsect2> * <title>Example application</title> - * <para> - * <include xmlns="http://www.w3.org/2003/XInclude" href="element-level-example.xml" /> - * </para> + * |[ + * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" /> + * ]| * </refsect2> */ diff --git a/gst/monoscope/gstmonoscope.c b/gst/monoscope/gstmonoscope.c index 5612e6e11..9cefea895 100644 --- a/gst/monoscope/gstmonoscope.c +++ b/gst/monoscope/gstmonoscope.c @@ -23,17 +23,14 @@ * SECTION:element-monoscope * @see_also: goom * - * <refsect2> - * <para> * Monoscope is an audio visualisation element. It creates a coloured * curve of the audio signal like on an oscilloscope. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch -v audiotestsrc ! audioconvert ! monoscope ! ffmpegcolorspace ! ximagesink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/multifile/gstmultifilesink.c b/gst/multifile/gstmultifilesink.c index 583b989c5..5edb3cd7f 100644 --- a/gst/multifile/gstmultifilesink.c +++ b/gst/multifile/gstmultifilesink.c @@ -23,13 +23,9 @@ */ /** * SECTION:element-multifilesink - * @short_description: Writes buffers to sequentially-named files * @see_also: #GstFileSrc * - * <para> * Write incoming data to a series of sequentially-named files. - * </para> - * */ #ifdef HAVE_CONFIG_H @@ -154,6 +150,7 @@ gst_multi_file_sink_set_location (GstMultiFileSink * sink, return TRUE; } + static void gst_multi_file_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) diff --git a/gst/multifile/gstmultifilesrc.c b/gst/multifile/gstmultifilesrc.c index ee39c5652..f6b55ac6b 100644 --- a/gst/multifile/gstmultifilesrc.c +++ b/gst/multifile/gstmultifilesrc.c @@ -20,29 +20,23 @@ */ /** * SECTION:element-multifilesrc - * @short_description: Read buffers from sequentially-named files * @see_also: #GstFileSrc * - * <refsect2> - * <para> * Reads buffers from sequentially named files. If used together with an image - * decoder, one needs to use the GstMultiFileSrc::caps property or a capsfilter + * decoder, one needs to use the #GstMultiFileSrc:caps property or a capsfilter * to force to caps containing a framerate. Otherwise image decoders send EOS * after the first picture. - * </para> + * + * File names are created by replacing "%%d" with the index using printf(). + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch multifilesrc location="img.%04d.png" index=0 caps="image/png,framerate=\(fraction\)12/1" ! \ * pngdec ! ffmpegcolorspace ! theoraenc ! oggmux ! \ * filesink location="images.ogg" - * </programlisting> - * This pipeline creates a video file "images.ogg" by joining multiple PNG + * ]| This pipeline creates a video file "images.ogg" by joining multiple PNG * files named img.0000.png, img.0001.png, etc. - * </para> - * <para> - * File names are created by replacing "%%d" with the index using printf(). - * </para> * </refsect2> */ @@ -232,6 +226,7 @@ gst_multi_file_src_set_location (GstMultiFileSrc * src, const gchar * location) return TRUE; } + static void gst_multi_file_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) diff --git a/gst/multipart/multipartdemux.c b/gst/multipart/multipartdemux.c index 3d0cfd72d..8a6be4434 100644 --- a/gst/multipart/multipartdemux.c +++ b/gst/multipart/multipartdemux.c @@ -22,34 +22,27 @@ /** * SECTION:element-multipartdemux - * @short_description: Demuxer that takes a multipart digital stream as input - * and demuxes one or many digital streams from it. * @see_also: #GstMultipartMux * - * <refsect2> - * <para> * MultipartDemux uses the Content-type field of incoming buffers to demux and * push data to dynamic source pads. Most of the time multipart streams are * sequential JPEG frames generated from a live source such as a network source * or a camera. - * </para> + * + * The output buffers of the multipartdemux typically have no timestamps and are + * usually played as fast as possible (at the rate that the source provides the + * data). + * + * the content in multipart files is separated with a boundary string that can + * be configured specifically with the #GstMultipartDemux:boundary property + * otherwise it will be autodetected. + * + * <refsect2> * <title>Sample pipelines</title> - * <para> - * Here is a simple pipeline to demux a multipart file muxed with - * #GstMultipartMux containing JPEG frames: - * <programlisting> + * |[ * gst-launch filesrc location=/tmp/test.multipart ! multipartdemux ! jpegdec ! ffmpegcolorspace ! ximagesink - * </programlisting> - * </para> - * <para> - * The output buffers of the multipartdemux typically have no timestamps and are usually - * played as fast as possible (at the rate that the source provides the data). - * </para> - * <para> - * the content in multipart files is separated with a boundary string that can be - * configured specifically with the "boundary" property otherwise it will be - * autodetected. - * </para> + * ]| a simple pipeline to demux a multipart file muxed with #GstMultipartMux + * containing JPEG frames. * </refsect2> */ diff --git a/gst/multipart/multipartmux.c b/gst/multipart/multipartmux.c index 45bc4bd7e..c1916a96d 100644 --- a/gst/multipart/multipartmux.c +++ b/gst/multipart/multipartmux.c @@ -19,23 +19,17 @@ /** * SECTION:element-multipartmux - * @short_description: Muxer that takes one or several digital streams - * and muxes them to a single multipart stream. * - * <refsect2> - * <para> * MultipartMux uses the #GstCaps of the sink pad as the Content-type field for * incoming buffers when muxing them to a multipart stream. Most of the time * multipart streams are sequential JPEG frames. - * </para> + * + * <refsect2> * <title>Sample pipelines</title> - * <para> - * Here is a simple pipeline to mux 5 JPEG frames per second into a multipart - * stream stored to a file : - * <programlisting> + * |[ * gst-launch videotestsrc ! video/x-raw-yuv, framerate=(fraction)5/1 ! jpegenc ! multipartmux ! filesink location=/tmp/test.multipart - * </programlisting> - * </para> + * ]| a simple pipeline to mux 5 JPEG frames per second into a multipart stream + * stored to a file. * </refsect2> */ diff --git a/gst/qtdemux/qtdemux.c b/gst/qtdemux/qtdemux.c index 3994b734d..540168d9f 100644 --- a/gst/qtdemux/qtdemux.c +++ b/gst/qtdemux/qtdemux.c @@ -23,24 +23,19 @@ /** * SECTION:element-qtdemux * - * <refsect2> - * <para> * Demuxes a .mov file into raw or compressed audio and/or video streams. - * </para> - * <para> + * * This element supports both push and pull-based scheduling, depending on the * capabilities of the upstream elements. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=test.mov ! qtdemux name=demux demux.audio_00 ! decodebin ! audioconvert ! audioresample ! autoaudiosink demux.video_00 ! queue ! decodebin ! ffmpegcolorspace ! videoscale ! autovideosink - * </programlisting> - * Play (parse and decode) a .mov file and try to output it to + * ]| Play (parse and decode) a .mov file and try to output it to * an automatically detected soundcard and videosink. If the MOV file contains * compressed audio or video data, this will only work if you have the * right decoder elements/plugins installed. - * </para> * </refsect2> * * Last reviewed on 2006-12-29 (0.10.5) diff --git a/gst/rtp/gstrtpjpegpay.c b/gst/rtp/gstrtpjpegpay.c index 1810eff8d..0de0ac0f9 100644 --- a/gst/rtp/gstrtpjpegpay.c +++ b/gst/rtp/gstrtpjpegpay.c @@ -21,7 +21,6 @@ /** * SECTION:rtpjpegpay - * @short_description: RTP payloader for JPEG pictures * * Payload encode JPEG pictures into RTP packets according to RFC 2435. * For detailed information see: http://www.rfc-editor.org/rfc/rfc2435.txt @@ -31,7 +30,6 @@ * the actual JPEG entropy scan. * * The payloader assumes that correct width and height is found in the caps. - * */ #ifdef HAVE_CONFIG_H @@ -74,7 +72,7 @@ GST_STATIC_PAD_TEMPLATE ("src", GST_DEBUG_CATEGORY_STATIC (rtpjpegpay_debug); #define GST_CAT_DEFAULT (rtpjpegpay_debug) -/** +/* * QUANT_PREFIX_LEN: * * Prefix length in the header before the quantization tables: @@ -82,13 +80,13 @@ GST_DEBUG_CATEGORY_STATIC (rtpjpegpay_debug); */ #define QUANT_PREFIX_LEN 3 -/** +/* * DEFAULT_JPEG_QUALITY: * */ #define DEFAULT_JPEG_QUALITY 255 -/** +/* * DEFAULT_JPEG_TYPE: * */ @@ -96,7 +94,7 @@ GST_DEBUG_CATEGORY_STATIC (rtpjpegpay_debug); typedef enum _RtpJpegMarker RtpJpegMarker; -/** +/* * RtpJpegMarker: * @JPEG_MARKER: Prefix for JPEG marker * @JPEG_MARKER_SOI: Start of Image marker @@ -140,7 +138,7 @@ enum typedef struct _RtpJpegHeader RtpJpegHeader; typedef struct _RtpQuantHeader RtpQuantHeader; -/** +/* * RtpJpegHeader: * @type_spec: type specific * @offset: fragment offset @@ -172,7 +170,7 @@ struct _RtpJpegHeader guint8 height; }; -/** +/* * RtpQuantHeader * @mbz: must be zero * @precision: specify size of quantization tables diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c index 0adfc7aa5..c6a942e4e 100644 --- a/gst/rtsp/gstrtpdec.c +++ b/gst/rtsp/gstrtpdec.c @@ -44,11 +44,7 @@ /** * SECTION:element-rtpdec * - * <refsect2> - * <para> * A simple RTP session manager used internally by rtspsrc. - * </para> - * </refsect2> * * Last reviewed on 2006-06-20 (0.10.4) */ @@ -252,8 +248,8 @@ gst_rtp_dec_marshal_BOXED__UINT_UINT (GClosure * closure, data2 = closure->data; } callback = - (GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data : cc-> - callback); + (GMarshalFunc_BOXED__UINT_UINT) (marshal_data ? marshal_data : + cc->callback); v_return = callback (data1, g_marshal_value_peek_uint (param_values + 1), @@ -285,8 +281,8 @@ gst_rtp_dec_marshal_VOID__UINT_UINT (GClosure * closure, data2 = closure->data; } callback = - (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data : cc-> - callback); + (GMarshalFunc_VOID__UINT_UINT) (marshal_data ? marshal_data : + cc->callback); callback (data1, g_marshal_value_peek_uint (param_values + 1), diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 79455cd17..e677c9918 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -43,43 +43,36 @@ /** * SECTION:element-rtspsrc * - * <refsect2> - * <para> * Makes a connection to an RTSP server and read the data. * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support * RealMedia/Quicktime/Microsoft extensions. - * </para> - * <para> + * * RTSP supports transport over TCP or UDP in unicast or multicast mode. By * default rtspsrc will negotiate a connection in the following order: * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed - * protocols can be controlled with the "protocols" property. - * </para> - * <para> + * protocols can be controlled with the #GstRTSPSrc:protocols property. + * * rtspsrc currently understands SDP as the format of the session description. * For each stream listed in the SDP a new rtp_stream%d pad will be created * with caps derived from the SDP media description. This is a caps of mime type * "application/x-rtp" that can be connected to any available RTP depayloader * element. - * </para> - * <para> + * * rtspsrc will internally instantiate an RTP session manager element * that will handle the RTCP messages to and from the server, jitter removal, * packet reordering along with providing a clock for the pipeline. * This feature is currently fully implemented with the gstrtpbin in the * gst-plugins-bad module. - * </para> - * <para> + * * rtspsrc acts like a live source and will therefore only generate data in the * PLAYING state. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch rtspsrc location=rtsp://some.server/url ! fakesink - * </programlisting> - * Establish a connection to an RTSP server and send the raw RTP packets to a fakesink. - * </para> + * ]| Establish a connection to an RTSP server and send the raw RTP packets to a + * fakesink. * </refsect2> * * Last reviewed on 2006-08-18 (0.10.5) diff --git a/gst/smpte/gstsmpte.c b/gst/smpte/gstsmpte.c index 91f6c2f8e..5a1a0871b 100644 --- a/gst/smpte/gstsmpte.c +++ b/gst/smpte/gstsmpte.c @@ -19,31 +19,23 @@ /** * SECTION:element-smpte - * @short_description: Takes to video frames and applies an SMPTE transition - * effect on them. * - * <refsect2> - * <para> * smpte can accept I420 video streams with the same width, height and * framerate. The two incomming buffers are blended together using an effect * specific alpha mask. - * </para> - * <para> - * The depth property defines the presision in bits of the mask. A higher - * presision will create a mask with smoother gradients in order to avoid + * + * The #GstSmpte:depth property defines the presision in bits of the mask. A + * higher presision will create a mask with smoother gradients in order to avoid * banding. - * </para> + * + * <refsect2> * <title>Sample pipelines</title> - * <para> - * Here is a pipeline to demonstrate the smpte transition : - * <programlisting> - * gst-launch -v videotestsrc pattern=1 ! smpte name=s border=20000 type=234 - * duration=2000000000 ! ffmpegcolorspace ! ximagesink videotestsrc ! s. - * </programlisting> - * This shows a pinwheel transition a from a snow videotestsrc to an smpte + * |[ + * gst-launch -v videotestsrc pattern=1 ! smpte name=s border=20000 type=234 duration=2000000000 ! ffmpegcolorspace ! ximagesink videotestsrc ! s. + * ]| A pipeline to demonstrate the smpte transition. + * It shows a pinwheel transition a from a snow videotestsrc to an smpte * pattern videotestsrc. The transition will take 2 seconds to complete. The * edges of the transition are smoothed with a 20000 big border. - * </para> * </refsect2> */ @@ -491,8 +483,8 @@ gst_smpte_collected (GstCollectPads * pads, GstSMPTE * smpte) GstCaps *caps; caps = - gst_caps_copy (gst_static_caps_get (&gst_smpte_src_template. - static_caps)); + gst_caps_copy (gst_static_caps_get + (&gst_smpte_src_template.static_caps)); gst_caps_set_simple (caps, "width", G_TYPE_INT, smpte->width, "height", G_TYPE_INT, smpte->height, "framerate", GST_TYPE_FRACTION, smpte->fps_num, smpte->fps_denom, NULL); diff --git a/gst/smpte/gstsmptealpha.c b/gst/smpte/gstsmptealpha.c index 95c8b2458..74f27ba29 100644 --- a/gst/smpte/gstsmptealpha.c +++ b/gst/smpte/gstsmptealpha.c @@ -19,26 +19,21 @@ /** * SECTION:element-smptealpha - * @short_description: Takes a video frames and applies an SMPTE transition - * effect on it in the alpha channel. * - * <refsect2> - * <para> * smptealpha can accept an I420 or AYUV video stream. An alpha channel is added - * using an effect specific SMPTE mask in the I420 input case. In the AYUV case, the - * alpha channel is modified using the effect specific SMPTE mask. - * </para> - * <para> - * The "position" property is a controllabe double between 0.0 and 1.0 that - * specifies the position in the transition. 0.0 is the start of the transition - * with the alpha channel to complete opaque where 1.0 has the alpha channel set - * to completely transparent. - * </para> - * <para> - * The depth property defines the precision in bits of the mask. A higher - * presision will create a mask with smoother gradients in order to avoid - * banding. - * </para> + * using an effect specific SMPTE mask in the I420 input case. In the AYUV case, + * the alpha channel is modified using the effect specific SMPTE mask. + * + * The #GstSmpteAlpha:position property is a controllabe double between 0.0 and + * 1.0 that specifies the position in the transition. 0.0 is the start of the + * transition with the alpha channel to complete opaque where 1.0 has the alpha + * channel set to completely transparent. + * + * The #GstSmpteAlpha:depth property defines the precision in bits of the mask. + * A higher presision will create a mask with smoother gradients in order to + * avoid banding. + * + * <refsect2> * <title>Sample pipelines</title> * <para> * Here is a pipeline to demonstrate the smpte transition : @@ -432,6 +427,7 @@ gst_smpte_alpha_do_ayuv (GstSMPTEAlpha * smpte, guint8 * in, guint8 * out, } } } + static void gst_smpte_alpha_do_i420 (GstSMPTEAlpha * smpte, guint8 * in, guint8 * out, GstMask * mask, gint width, gint height, gint border, gint pos) diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c index 15d3592e4..63675c177 100644 --- a/gst/spectrum/gstspectrum.c +++ b/gst/spectrum/gstspectrum.c @@ -20,17 +20,13 @@ */ /** * SECTION:element-spectrum - * @short_description: audio spectrum analyzer * - * <refsect2> - * <para> * The Spectrum element analyzes the frequency spectrum of an audio signal. * If the #GstSpectrum:message property is #TRUE, it sends analysis results as * application messages named * <classname>"spectrum"</classname> after each interval of time given * by the #GstSpectrum:interval property. - * </para> - * <para> + * * The message's structure contains some combination of these fields: * <itemizedlist> * <listitem> @@ -87,19 +83,15 @@ * </para> * </listitem> * </itemizedlist> - * </para> - * <para> - * This element cannot be used with the gst-launch command in a sensible way. - * This sample code demonstrates how to use it in an application. - * </para> + * + * <refsect2> * <title>Example application</title> - * <para> - * <include xmlns="http://www.w3.org/2003/XInclude" href="element-spectrum-example.xml" /> - * </para> - * <para> - * Last reviewed on 2009-01-14 (0.10.12) - * </para> + * |[ + * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" /> + * ]| * </refsect2> + * + * Last reviewed on 2009-01-14 (0.10.12) */ #ifdef HAVE_CONFIG_H diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c index 9483483ba..a4821ffe3 100644 --- a/gst/udp/gstmultiudpsink.c +++ b/gst/udp/gstmultiudpsink.c @@ -21,13 +21,9 @@ * SECTION:element-multiupdsink * @see_also: udpsink, multifdsink * - * <refsect2> - * <para> * multiudpsink is a network sink that sends UDP packets to multiple * clients. * It can be combined with rtp payload encoders to implement RTP streaming. - * </para> - * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/gst/udp/gstudpsink.c b/gst/udp/gstudpsink.c index f808c3b51..a9666ed07 100644 --- a/gst/udp/gstudpsink.c +++ b/gst/udp/gstudpsink.c @@ -16,7 +16,20 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ - +/** + * SECTION:element-udpsink + * @see_also: udpsrc, multifdsink + * + * udpsink is a network sink that sends UDP packets to the network. + * It can be combined with RTP payloaders to implement RTP streaming. + * + * <refsect2> + * <title>Examples</title> + * |[ + * gst-launch -v audiotestsrc ! udpsink + * ]| + * </refsect2> + */ #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index b16a0521c..0da694610 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -22,72 +22,46 @@ * SECTION:element-udpsrc * @see_also: udpsink, multifdsink * - * <refsect2> - * <para> * udpsrc is a network source that reads UDP packets from the network. * It can be combined with RTP depayloaders to implement RTP streaming. - * </para> - * <title>Examples</title> - * <para> - * Here is a simple pipeline to read from the default port and dump the udp packets. - * <programlisting> - * gst-launch -v udpsrc ! fakesink dump=1 - * </programlisting> - * To actually generate udp packets on the default port one can use the - * udpsink element. When running the following pipeline in another terminal, the - * above mentioned pipeline should dump data packets to the console. - * <programlisting> - * gst-launch -v audiotestsrc ! udpsink - * </programlisting> - * </para> - * <para> + * * The udpsrc element supports automatic port allocation by setting the - * "port" property to 0. the following pipeline reads UDP from a free port. - * <programlisting> - * gst-launch -v udpsrc port=0 ! fakesink - * </programlisting> - * After setting the udpsrc to PAUSED, the allocated port can be obtained by - * reading the port property. - * </para> - * <para> - * udpsrc can read from multicast groups by setting the multicast_group property - * to the IP address of the multicast group. - * </para> - * <para> - * Alternatively one can provide a custom socket to udpsrc with the "sockfd" property, - * udpsrc will then not allocate a socket itself but use the provided one. - * </para> - * <para> - * The "caps" property is mainly used to give a type to the UDP packet so that they - * can be autoplugged in GStreamer pipelines. This is very usefull for RTP - * implementations where the contents of the UDP packets is transfered out-of-bounds - * using SDP or other means. - * </para> - * <para> - * The "buffer" property is used to change the default kernel buffer sizes used for - * receiving packets. The buffer size may be increased for high-volume connections, - * or may be decreased to limit the possible backlog of incoming data. - * The system places an absolute limit on these values, on Linux, for example, the - * default buffer size is typically 50K and can be increased to maximally 100K. - * </para> - * <para> - * The "skip-first-bytes" property is used to strip off an arbitrary number of - * bytes from the start of the raw udp packet and can be used to strip off - * proprietary header, for example. - * </para> - * <para> - * The udpsrc is always a live source. It does however not provide a GstClock, this - * is left for upstream elements such as an RTP session manager or demuxer (such - * as an MPEG demuxer). As with all live sources, the captured buffers will have - * their timestamp set to the current running time of the pipeline. - * </para> - * <para> - * udpsrc implements a GstURIHandler interface that handles udp://host:port type - * URIs. - * </para> - * <para> - * If the <link linkend="GstUDPSrc--timeout">timeout property</link> is set to a - * value bigger than 0, udpsrc will generate an element message named + * #GstUDPSrc:port property to 0. After setting the udpsrc to PAUSED, the + * allocated port can be obtained by reading the port property. + * + * udpsrc can read from multicast groups by setting the #GstUDPSrc:multicast_group + * property to the IP address of the multicast group. + * + * Alternatively one can provide a custom socket to udpsrc with the #GstUDPSrc:sockfd + * property, udpsrc will then not allocate a socket itself but use the provided + * one. + * + * The #GstUDPSrc:caps property is mainly used to give a type to the UDP packet + * so that they can be autoplugged in GStreamer pipelines. This is very usefull + * for RTP implementations where the contents of the UDP packets is transfered + * out-of-bounds using SDP or other means. + * + * The #GstUDPSrc:buffer property is used to change the default kernel buffer + * sizes used for receiving packets. The buffer size may be increased for + * high-volume connections, or may be decreased to limit the possible backlog of + * incoming data. The system places an absolute limit on these values, on Linux, + * for example, the default buffer size is typically 50K and can be increased to + * maximally 100K. + * + * The #GstUDPSrc:skip-first-bytes property is used to strip off an arbitrary + * number of bytes from the start of the raw udp packet and can be used to strip + * off proprietary header, for example. + * + * The udpsrc is always a live source. It does however not provide a #GstClock, + * this is left for upstream elements such as an RTP session manager or demuxer + * (such as an MPEG demuxer). As with all live sources, the captured buffers + * will have their timestamp set to the current running time of the pipeline. + * + * udpsrc implements a #GstURIHandler interface that handles udp://host:port + * type URIs. + * + * If the #GstUDPSrc:timeout property is set to a value bigger than 0, udpsrc + * will generate an element message named * <classname>"GstUDPSrcTimeout"</classname> * if no data was recieved in the given timeout. * The message's structure contains one field: @@ -105,15 +79,28 @@ * </para> * <para> * A custom file descriptor can be configured with the - * <link linkend="GstUDPSrc--sockfd">sockfd property</link>. The socket will be - * closed when setting the element to READY by default. This behaviour can be - * overriden with the <link linkend="GstUDPSrc--closefd">closefd property</link>, - * in which case the application is responsible for closing the file descriptor. - * </para> - * <para> - * Last reviewed on 2007-09-20 (0.10.7) - * </para> + * #GstUDPSrc:sockfd property. The socket will be closed when setting the + * element to READY by default. This behaviour can be + * overriden with the #GstUDPSrc:closefd property, in which case the application + * is responsible for closing the file descriptor. + * + * <refsect2> + * <title>Examples</title> + * |[ + * gst-launch -v udpsrc ! fakesink dump=1 + * ]| A pipeline to read from the default port and dump the udp packets. + * To actually generate udp packets on the default port one can use the + * udpsink element. When running the following pipeline in another terminal, the + * above mentioned pipeline should dump data packets to the console. + * |[ + * gst-launch -v audiotestsrc ! udpsink + * ]| + * |[ + * gst-launch -v udpsrc port=0 ! fakesink + * ]| read udp packets from a free port. * </refsect2> + * + * Last reviewed on 2007-09-20 (0.10.7) */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/videobox/gstvideobox.c b/gst/videobox/gstvideobox.c index 0e6e112c2..c515634d6 100644 --- a/gst/videobox/gstvideobox.c +++ b/gst/videobox/gstvideobox.c @@ -16,7 +16,25 @@ * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ - +/** + * SECTION:element-videobox + * @see_also: #GstVideoCrop + * + * This plugin crops or enlarges the image. It takes 4 values as input, a + * top, bottom, left and right offset. Positive values will crop that much + * pixels from the respective border of the image, negative values will add + * that much pixels. When pixels are added, you can specify their color. + * Some predefined colors are usable with an enum property. + * + * The plugin is alpha channel aware and will try to negotiate with a format + * that supports alpha channels first. When alpha channel is active two + * other properties, alpha and border_alpha can be used to set the alpha + * values of the inner picture and the border respectively. an alpha value of + * 0.0 means total transparency, 1.0 is opaque. + * + * The videobox plugin has many uses such as doing a mosaic of pictures, + * letterboxing video, cutting out pieces of video, picture in picture, etc.. + */ #ifdef HAVE_CONFIG_H #include "config.h" #endif @@ -1147,6 +1165,7 @@ invalid_format: } } +/* FIXME: 0.11 merge with videocrop plugin */ static gboolean plugin_init (GstPlugin * plugin) { diff --git a/gst/videocrop/gstaspectratiocrop.c b/gst/videocrop/gstaspectratiocrop.c index 28ca397b4..cd59323ad 100644 --- a/gst/videocrop/gstaspectratiocrop.c +++ b/gst/videocrop/gstaspectratiocrop.c @@ -21,14 +21,16 @@ * SECTION:element-aspectratiocrop * @see_also: #GstVideoCrop * - * This element crops video frames to a specified aspect-ratio. + * This element crops video frames to a specified #GstAspectRatioCrop:aspect-ratio. * - * If the aspect-ratio is already correct, the element will operate in pass-through mode. + * If the aspect-ratio is already correct, the element will operate + * in pass-through mode. * * <refsect2> * <title>Example launch line</title> - * |[gst-launch -v videotestsrc ! video/x-raw-rgb,height=640,width=480 ! aspectratiocrop aspect-ratio=16/9 ! ximagesink]| - * This pipeline generates a videostream in 4/3 and crops it to 16/9. + * |[ + * gst-launch -v videotestsrc ! video/x-raw-rgb,height=640,width=480 ! aspectratiocrop aspect-ratio=16/9 ! ximagesink + * ]| This pipeline generates a videostream in 4/3 and crops it to 16/9. * </refsect2> */ diff --git a/gst/videocrop/gstvideocrop.c b/gst/videocrop/gstvideocrop.c index 0d0ebfedb..607d92483 100644 --- a/gst/videocrop/gstvideocrop.c +++ b/gst/videocrop/gstvideocrop.c @@ -21,35 +21,29 @@ * SECTION:element-videocrop * @see_also: #GstVideoBox * - * <refsect2> - * <para> * This element crops video frames, meaning it can remove parts of the * picture on the left, right, top or bottom of the picture and output * a smaller picture than the input picture, with the unwanted parts at the * border removed. - * </para> - * <para> + * * The videocrop element is similar to the videobox element, but its main * goal is to support a multitude of formats as efficiently as possible. * Unlike videbox, it cannot add borders to the picture and unlike videbox * it will always output images in exactly the same format as the input image. - * </para> - * <para> + * * If there is nothing to crop, the element will operate in pass-through mode. - * </para> - * <para> + * * Note that no special efforts are made to handle chroma-subsampled formats * in the case of odd-valued cropping and compensate for sub-unit chroma plane - * shifts for such formats in the case where the "left" or "top" property is - * set to an odd number. This doesn't matter for most use cases, but it might - * matter for yours. - * </para> + * shifts for such formats in the case where the #GstVideoCrop:left or + * #GstVideoCrop:top property is set to an odd number. This doesn't matter for + * most use cases, but it might matter for yours. + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch -v videotestsrc ! videocrop top=42 left=1 right=4 bottom=0 ! ximagesink - * </programlisting> - * </para> + * ]| * </refsect2> */ diff --git a/gst/videofilter/gstgamma.c b/gst/videofilter/gstgamma.c index b1342687f..e94de1832 100644 --- a/gst/videofilter/gstgamma.c +++ b/gst/videofilter/gstgamma.c @@ -30,17 +30,13 @@ /** * SECTION:element-gamma * - * <refsect2> - * <para> * Performs gamma correction on a video stream. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch videotestsrc ! gamma gamma=2.0 ! ffmpegcolorspace ! ximagesink - * </programlisting> - * This pipeline will make the image "brighter". - * </para> + * ]| This pipeline will make the image "brighter". * </refsect2> */ diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c index 2464ee055..80b503df8 100644 --- a/gst/videofilter/gstvideobalance.c +++ b/gst/videofilter/gstvideobalance.c @@ -26,17 +26,14 @@ /** * SECTION:element-videobalance * - * <refsect2> - * <para> * Adjusts brightness, contrast, hue, saturation on a video stream. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch videotestsrc ! videobalance saturation=0.0 ! ffmpegcolorspace ! ximagesink - * </programlisting> - * This pipeline converts the image to black and white by setting the saturation to 0.0. - * </para> + * ]| This pipeline converts the image to black and white by setting the + * saturation to 0.0. * </refsect2> * * Last reviewed on 2006-03-03 (0.10.3) diff --git a/gst/videofilter/gstvideoflip.c b/gst/videofilter/gstvideoflip.c index 336e05387..051805c5a 100644 --- a/gst/videofilter/gstvideoflip.c +++ b/gst/videofilter/gstvideoflip.c @@ -25,17 +25,13 @@ /** * SECTION:element-videoflip * - * <refsect2> - * <para> * Flips and rotates video. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch videotestsrc ! videoflip method=clockwise ! ffmpegcolorspace ! ximagesink - * </programlisting> - * This pipeline flips the test image 90 degrees clockwise. - * </para> + * ]| This pipeline flips the test image 90 degrees clockwise. * </refsect2> * * Last reviewed on 2006-03-03 (0.10.3) diff --git a/gst/videomixer/videomixer.c b/gst/videomixer/videomixer.c index 46dc58296..00572b23f 100644 --- a/gst/videomixer/videomixer.c +++ b/gst/videomixer/videomixer.c @@ -19,33 +19,26 @@ /** * SECTION:element-videomixer - * @short_description: Takes several AYUV video streams as input and mixes them - * together. * - * <refsect2> - * <para> * Videomixer can only accept AYUV video streams. For each of the requested * sink pads it will compare the incoming geometry and framerate to define the * output parameters. Indeed output video frames will have the geometry of the * biggest incoming video stream and the framerate of the fastest incoming one. - * </para> - * <para> + * * Individual parameters for each input stream can be configured on the * #GstVideoMixerPad. - * </para> + * + * <refsect2> * <title>Sample pipelines</title> - * <para> - * Here is a pipeline to demonstrate videomixer used together with videobox : - * <programlisting> + * |[ * gst-launch videotestsrc pattern=1 ! video/x-raw-yuv, framerate=\(fraction\)10/1, width=100, height=100 ! videobox border-alpha=0 alpha=0.5 top=-70 bottom=-70 right=-220 ! videomixer name=mix ! ffmpegcolorspace ! xvimagesink videotestsrc ! video/x-raw-yuv, framerate=\(fraction\)5/1, width=320, height=240 ! alpha alpha=0.7 ! mix. - * </programlisting> + * ]| A pipeline to demonstrate videomixer used together with videobox. * This should show a 320x240 pixels video test source with some transparency * showing the background checker pattern. Another video test source with just * the snow pattern of 100x100 pixels is overlayed on top of the first one on * the left vertically centered with a small transparency showing the first * video test source behind and the checker pattern under it. Note that the * framerate of the output video is 10 frames per second. - * </para> * </refsect2> */ diff --git a/gst/wavenc/gstwavenc.c b/gst/wavenc/gstwavenc.c index d563b7b1a..aa988fd83 100644 --- a/gst/wavenc/gstwavenc.c +++ b/gst/wavenc/gstwavenc.c @@ -19,7 +19,12 @@ * Boston, MA 02111-1307, USA. * */ - +/** + * SECTION:element-wavenc + * + * Format a audio stream into the wav format. + * + */ #ifdef HAVE_CONFIG_H #include "config.h" #endif diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index b2f498071..266c430fa 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -22,28 +22,20 @@ /** * SECTION:element-wavparse * - * <refsect2> - * <para> * Parse a .wav file into raw or compressed audio. - * </para> - * <para> + * * Wavparse supports both push and pull mode operations, making it possible to * stream from a network source. - * </para> + * + * <refsect2> * <title>Example launch line</title> - * <para> - * <programlisting> + * |[ * gst-launch filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink - * </programlisting> - * Read a wav file and output to the soundcard using the ALSA element. The + * ]| Read a wav file and output to the soundcard using the ALSA element. The * wav file is assumed to contain raw uncompressed samples. - * </para> - * <para> - * <programlisting> + * |[ * gst-launch gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink - * </programlisting> - * Stream data from a network url. - * </para> + * ]| Stream data from a network url. * </refsect2> * * Last reviewed on 2007-02-14 (0.10.6) |