Final round of doc updates.

Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
This commit is contained in:
Stefan Kost 2008-06-16 07:30:34 +00:00
parent 68993006c3
commit b62583b216
12 changed files with 94 additions and 122 deletions

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@ -1,3 +1,18 @@
2008-06-16 Stefan Kost <ensonic@users.sf.net>
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
2008-06-16 Stefan Kost <ensonic@users.sf.net> 2008-06-16 Stefan Kost <ensonic@users.sf.net>
* gst/deinterlace/gstdeinterlace.c: * gst/deinterlace/gstdeinterlace.c:

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@ -320,7 +320,8 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
* GstRtpJitterBuffer::clear-pt-map: * GstRtpJitterBuffer::clear-pt-map:
* @buffer: the object which received the signal * @buffer: the object which received the signal
* *
* Invalidate the clock-rate as obtained with the ::request-pt-map signal. * Invalidate the clock-rate as obtained with the
* #GstRtpJitterBuffer::request-pt-map signal.
*/ */
gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] = gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass), g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),

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@ -21,23 +21,18 @@
/** /**
* SECTION:element-speed * SECTION:element-speed
* *
* <refsect2>
* <para>
* Plays an audio stream at a different speed. * Plays an audio stream at a different speed.
* </para> *
* <para>
* Do not use this element. Either use the 'pitch' element, or do a seek with * Do not use this element. Either use the 'pitch' element, or do a seek with
* a non-1.0 rate parameter, this will have the same effect as using the speed * a non-1.0 rate parameter, this will have the same effect as using the speed
* element (but relies on the decoder/demuxer to handle this correctly, also * element (but relies on the decoder/demuxer to handle this correctly, also
* requires a fairly up-to-date gst-plugins-base, as of February 2007). * requires a fairly up-to-date gst-plugins-base, as of February 2007).
* </para> *
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! speed speed=1.5 ! audioconvert ! audioresample ! autoaudiosink * gst-launch filesrc location=test.ogg ! decodebin ! audioconvert ! speed speed=1.5 ! audioconvert ! audioresample ! autoaudiosink
* </programlisting> * ]| Plays an .ogg file at 1.5x speed.
* Plays an .ogg file at 1.5x speed.
* </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-02-26 (0.10.4.1) * Last reviewed on 2007-02-26 (0.10.4.1)

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@ -22,19 +22,16 @@
/** /**
* SECTION:element-speexresample * SECTION:element-speexresample
* *
* <refsect2>
* speexresample resamples raw audio buffers to different sample rates using * speexresample resamples raw audio buffers to different sample rates using
* a configurable windowing function to enhance quality. * a configurable windowing function to enhance quality.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! speexresample ! audio/x-raw-int, rate=8000 ! alsasink
* </programlisting>
* Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* </para>
* </refsect2>
* *
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! speexresample ! audio/x-raw-int, rate=8000 ! alsasink
* ]| Decode an Ogg/Vorbis downsample to 8Khz and play sound through alsa.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* </refsect2>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

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@ -20,11 +20,10 @@
/** /**
* SECTION:element-videoanalyse * SECTION:element-videoanalyse
* *
* <refsect2> * This plugin analyses every video frame and if the #GstVideoAnalyse:message
* <para> * property is #TRUE, posts an element message with video statistics called
* This plugin analyses every video frame and if the <link * <classname>&quot;GstVideoAnalyse&quot;</classname>.
* linkend="GstVideoAnalyse--message">message property</link> is #TRUE, posts an element *
* message with video statistics called <classname>&quot;GstVideoAnalyse&quot;</classname>.
* The message's structure contains these fields: * The message's structure contains these fields:
* <itemizedlist> * <itemizedlist>
* <listitem> * <listitem>
@ -69,15 +68,13 @@
* the brightness variance of the frame. * the brightness variance of the frame.
* </para> * </para>
* </listitem> * </listitem>
* </itemizedlist> * </itemizedlist>
* </para> *
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch -m videotestsrc ! videoanalyse ! ffmpegcolorspace ! ximagesink * gst-launch -m videotestsrc ! videoanalyse ! ffmpegcolorspace ! ximagesink
* </programlisting> * ]| This pipeline emits messages to the console for each frame that has been analysed.
* This pipeline emits messages to the console for each frame that has been analysed.
* </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-05-30 (0.10.5) * Last reviewed on 2007-05-30 (0.10.5)

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@ -19,29 +19,24 @@
/** /**
* SECTION:element-videodetect * SECTION:element-videodetect
* @see_also: #GstVideoMark
* *
* <refsect2> * This plugin detects #GstVideoDetect:pattern-count squares in the bottom left
* <para> * corner of the video frames. The squares have a width and height of
* This plugin detects ::pattern-count squares in the bottom left corner of * respectively #GstVideoDetect:pattern-width and #GstVideoDetect:pattern-height.
* the video frames. The squares have a width and height of respectively * Even squares must be black and odd squares must be white.
* ::pattern-width and ::patern-height. Even squares must be black and odd *
* squares must be white. * When the pattern has been found, #GstVideoDetect:pattern-data-count squares
* </para> * after the pattern squares are read as a bitarray. White squares represent a 1
* <para> * bit and black squares a 0 bit. The bitarray will will included in the element
* When the pattern has been found, ::pattern-data-count squares after the * message that is posted (see below).
* pattern squares are read as a bitarray. White squares represent a 1 bit and *
* black squares a 0 bit. The bitarray will will included in the element message
* that is posted (see below).
* </para>
* <para>
* After the pattern has been found and the data pattern has been read, an * After the pattern has been found and the data pattern has been read, an
* element message called <classname>&quot;GstVideoDetect&quot;</classname> will * element message called <classname>&quot;GstVideoDetect&quot;</classname> will
* be posted on the bus. If the pattern is no longer found in the frame, the * be posted on the bus. If the pattern is no longer found in the frame, the
* same element message is posted with the have-pattern field set to #FALSE. * same element message is posted with the have-pattern field set to #FALSE.
* The message is only posted if the <link linkend="GstVideoDetect--message">message * The message is only posted if the #GstVideoDetect:message property is #TRUE.
* property</link> is #TRUE. *
* </para>
* <para>
* The message's structure contains these fields: * The message's structure contains these fields:
* <itemizedlist> * <itemizedlist>
* <listitem> * <listitem>
@ -89,13 +84,12 @@
* </para> * </para>
* </listitem> * </listitem>
* </itemizedlist> * </itemizedlist>
* </para> *
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch videotestsrc ! videodetect ! ffmpegcolorspace ! ximagesink * gst-launch videotestsrc ! videodetect ! ffmpegcolorspace ! ximagesink
* </programlisting> * ]|
* </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-05-30 (0.10.5) * Last reviewed on 2007-05-30 (0.10.5)

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@ -19,30 +19,26 @@
/** /**
* SECTION:element-videomark * SECTION:element-videomark
* @see_also: #GstVideoDetect
* *
* This plugin produces #GstVideoMark::pattern-count squares in the bottom left
* corner of the video frames. The squares have a width and height of
* respectively #GstVideoMark:pattern-width and #GstVideoMark:pattern-height.
* Even squares will be black and odd squares will be white.
*
* After writing the pattern, #GstVideoMark:pattern-data-count squares after the
* pattern squares are produced as the bitarray given in
* #GstVideoMark:pattern-data. 1 bits will produce white squares and 0 bits will
* produce black squares.
*
* The element can be enabled with the #GstVideoMark:enabled property. It is
* mostly used together with the #GstVideoDetect plugin.
*
* <refsect2> * <refsect2>
* <para>
* This plugin produces ::pattern-count squares in the bottom left corner of
* the video frames. The squares have a width and height of respectively
* ::pattern-width and ::patern-height. Even squares will be black and odd
* squares will be white.
* </para>
* <para>
* After writing the pattern, ::pattern-data-count squares after the
* pattern squares are produced as the bitarray given in ::pattern-data. 1 bits
* will produce white squares and 0 bits will produce black squares.
* </para>
* <para>
* The element can be enabled with the ::enabled property. It is mostly used
* together with the videodetect plugin.
* </para>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch videotestsrc ! videomark ! ximagesink * gst-launch videotestsrc ! videomark ! ximagesink
* </programlisting> * ]| Add the default black/white squares at the bottom left of the video frames.
* Add the default black/white squares at the bottom left of the video frames.
* </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-06-01 (0.10.6) * Last reviewed on 2007-06-01 (0.10.6)

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@ -61,28 +61,20 @@ static GstElementDetails dvbsrc_details = {
/** /**
* SECTION:element-dvbsrc * SECTION:element-dvbsrc
* *
* <refsect2>
* dvbsrc can be used to capture video from DVB cards, DVB-T, DVB-S or DVB-T. * dvbsrc can be used to capture video from DVB cards, DVB-T, DVB-S or DVB-T.
* <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8MHz frequency=514000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! flutsdemux crc-check=false name=demux ! queue max-size-buffers=0 max-size-time=0 ! flumpeg2vdec ! xvimagesink sync=false demux. ! queue max-size-buffers=0 max-size-time=0 ! flump3dec ! alsasink sync=false * gst-launch dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8MHz frequency=514000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 ! flutsdemux crc-check=false name=demux ! queue max-size-buffers=0 max-size-time=0 ! flumpeg2vdec ! xvimagesink sync=false demux. ! queue max-size-buffers=0 max-size-time=0 ! flump3dec ! alsasink sync=false
* </programlisting> * ]| Captures a full transport stream from dvb card 0 that is a DVB-T card at tuned frequency 514000000 with other parameters as seen in the
* This pipeline captures a full transport stream from dvb card 0 that is a DVB-T card at tuned frequency 514000000 with other parameters as seen in the
* pipeline and outputs the first tv program on the transport stream. The reason the sinks have to be set to have sync=false is due to bug #340482. * pipeline and outputs the first tv program on the transport stream. The reason the sinks have to be set to have sync=false is due to bug #340482.
* </para> * |[
* <para>
* <programlisting>
* gst-launch dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=514000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 pids=256:257 ! flutsdemux crc-check=false name=demux es-pids=256:257 ! queue max-size-buffers=0 max-size-time=0 ! flumpeg2vdec ! xvimagesink sync=false demux. ! queue max-size-buffers=0 max-size-time=0 ! flump3dec ! alsasink sync=false * gst-launch dvbsrc modulation="QAM 64" trans-mode=8k bandwidth=8 frequency=514000000 code-rate-lp=AUTO code-rate-hp=2/3 guard=4 hierarchy=0 pids=256:257 ! flutsdemux crc-check=false name=demux es-pids=256:257 ! queue max-size-buffers=0 max-size-time=0 ! flumpeg2vdec ! xvimagesink sync=false demux. ! queue max-size-buffers=0 max-size-time=0 ! flump3dec ! alsasink sync=false
* </programlisting> * ]| Captures a partial transport stream from dvb card 0 that is a DVB-T card for a program at tuned frequency 514000000 and pids of 256:257 with other parameters as seen in the pipeline and outputs the program with the pids 256 and 257. The reason the sinks have to be set to
* This pipeline captures a partial transport stream from dvb card 0 that is a DVB-T card for a program at tuned frequency 514000000 and pids of 256:257 with other parameters as seen in the pipeline and outputs the program with the pids 256 and 257. The reason the sinks have to be set to * have sync=false is due to bug #340482 (FIXME: wrong bug number?).
* have sync=false is due to bug #340482. * |[
* </para>
* <para>
* <programlisting>
* gst-launch dvbsrc polarity="h" frequency=11302000 srate=27500 diseqc-src=0 pids=102:103 ! queue max-size-buffers=0 max-size-time=0 ! flumpeg2vdec ! xvimagesink sync=false demux. ! queue max-size-buffers=0 max-size-time=0 ! flump3dec ! alsasink sync=false * gst-launch dvbsrc polarity="h" frequency=11302000 srate=27500 diseqc-src=0 pids=102:103 ! queue max-size-buffers=0 max-size-time=0 ! flumpeg2vdec ! xvimagesink sync=false demux. ! queue max-size-buffers=0 max-size-time=0 ! flump3dec ! alsasink sync=false
* </programlisting> * ]| Captures a partial transport stream from dvb card 0 that is a DVB-S card for a program at tuned frequency 11302000 Hz, symbol rate of 27500 kHz and pids of 256:257 and outputs the program with the pids 256 and 257. The reason the sinks have to be set to have sync=false is due to bug #340482.
* This pipeline captures a partial transport stream from dvb card 0 that is a DVB-S card for a program at tuned frequency 11302000 Hz, symbol rate of 27500 kHz and pids of 256:257 and outputs the program with the pids 256 and 257. The reason the sinks have to be set to have sync=false is due to bug #340482.
* </para> * </para>
* </refsect2> * </refsect2>
*/ */

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@ -20,13 +20,12 @@
/** /**
* SECTION:element-oss4mixer * SECTION:element-oss4mixer
* *
* <refsect2>
* <para>
* This element lets you adjust sound input and output levels with the * This element lets you adjust sound input and output levels with the
* Open Sound System (OSS) version 4. It supports the GstMixer interface, which * Open Sound System (OSS) version 4. It supports the GstMixer interface, which
* can be used to obtain a list of available mixer tracks. Set the mixer * can be used to obtain a list of available mixer tracks. Set the mixer
* element to READY state before using the GstMixer interface on it. * element to READY state before using the GstMixer interface on it.
* </para> *
* <refsect2>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * <para>
* oss4mixer can&apos;t be used in a sensible way in gst-launch. * oss4mixer can&apos;t be used in a sensible way in gst-launch.

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@ -19,32 +19,24 @@
/** /**
* SECTION:element-oss4sink * SECTION:element-oss4sink
* *
* <refsect2>
* <para>
* This element lets you output sound using the Open Sound System (OSS) * This element lets you output sound using the Open Sound System (OSS)
* version 4. * version 4.
* </para> *
* <para>
* Note that you should almost always use generic audio conversion elements * Note that you should almost always use generic audio conversion elements
* like audioconvert and audioresample in front of an audiosink to make sure * like audioconvert and audioresample in front of an audiosink to make sure
* your pipeline works under all circumstances (those conversion elements will * your pipeline works under all circumstances (those conversion elements will
* act in passthrough-mode if no conversion is necessary). * act in passthrough-mode if no conversion is necessary).
* </para> *
* <refsect2>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * |[
* <programlisting>
* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! oss4sink * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.1 ! oss4sink
* </programlisting> * ]| will output a sine wave (continuous beep sound) to your sound card (with
* will output a sine wave (continuous beep sound) to your sound card (with
* a very low volume as precaution). * a very low volume as precaution).
* </para> * |[
* <para>
* <programlisting>
* gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! oss4sink * gst-launch -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! oss4sink
* </programlisting> * ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System
* will play an Ogg/Vorbis audio file and output it using the Open Sound System
* version 4. * version 4.
* </para>
* </refsect2> * </refsect2>
* *
* Since: 0.10.7 * Since: 0.10.7

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@ -20,20 +20,16 @@
/** /**
* SECTION:element-oss4src * SECTION:element-oss4src
* *
* <refsect2>
* <para>
* This element lets you record sound using the Open Sound System (OSS) * This element lets you record sound using the Open Sound System (OSS)
* version 4. * version 4.
* </para> *
* <refsect2>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * |[
* <programlisting>
* gst-launch -v oss4src ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg * gst-launch -v oss4src ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg
* </programlisting> * ]| will record sound from your sound card using OSS4 and encode it to an
* will record sound from your sound card using OSS4 and encode it to an
* Ogg/Vorbis file (this will only work if your mixer settings are right * Ogg/Vorbis file (this will only work if your mixer settings are right
* and the right inputs areenabled etc.) * and the right inputs areenabled etc.)
* </para>
* </refsect2> * </refsect2>
* *
* Since: 0.10.7 * Since: 0.10.7

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@ -22,11 +22,9 @@
* *
* <refsect2> * <refsect2>
* <title>Example launch line</title> * <title>Example launch line</title>
* <para> * |[
* <programlisting>
* gst-launch-0.10 -v wininetsrc location="http://71.83.57.210:9000" ! application/x-icy,metadata-interval=0 ! icydemux ! mad ! audioconvert ! directsoundsink * gst-launch-0.10 -v wininetsrc location="http://71.83.57.210:9000" ! application/x-icy,metadata-interval=0 ! icydemux ! mad ! audioconvert ! directsoundsink
* </programlisting> * ]| receive mp3 audio over http and play it back.
* </para>
* </refsect2> * </refsect2>
*/ */