*/
/**
* SECTION:element-alsamidisrc
+ * @title: alsamidisrc
* @see_also: #GstPushSrc
*
* The alsamidisrc element is an element that fetches ALSA MIDI sequencer
*
* It can be used to generate notes from a MIDI input device.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch -v alsamidisrc ports=129:0 ! fluiddec ! audioconvert ! autoaudiosink
- * ]| This pipeline will listen for events from the sequencer device at port 129:0,
+ * ]|
+ * This pipeline will listen for events from the sequencer device at port 129:0,
* and generate notes using the fluiddec element.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-alsasink
+ * @title: alsasink
* @see_also: alsasrc
*
* This element renders audio samples using the ALSA audio API.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! autoaudiosink
- * ]| Play an Ogg/Vorbis file and output audio via ALSA.
- * </refsect2>
+ * ]|
+ *
+ * Play an Ogg/Vorbis file and output audio via ALSA.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-alsasrc
+ * @title: alsasrc
* @see_also: alsasink
*
* This element reads data from an audio card using the ALSA API.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v alsasrc ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
- * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
- * </refsect2>
+ * ]|
+ * Record from a sound card using ALSA and encode to Ogg/Vorbis.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-oggdemux
+ * @title: oggdemux
* @see_also: <link linkend="gst-plugins-base-plugins-oggmux">oggmux</link>
*
* This element demuxes ogg files into their encoded audio and video components.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=test.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink
- * ]| Decodes a vorbis audio stream stored inside an ogg container and plays it.
- * </refsect2>
+ * ]|
+ * Decodes a vorbis audio stream stored inside an ogg container and plays it.
+ *
*/
/**
* SECTION:element-oggmux
+ * @title: oggmux
* @see_also: <link linkend="gst-plugins-base-plugins-oggdemux">oggdemux</link>
*
* This element merges streams (audio and video) into ogg files.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 v4l2src num-buffers=500 ! video/x-raw,width=320,height=240 ! videoconvert ! videorate ! theoraenc ! oggmux ! filesink location=video.ogg
- * ]| Encodes a video stream captured from a v4l2-compatible camera to Ogg/Theora
+ * ]|
+ * Encodes a video stream captured from a v4l2-compatible camera to Ogg/Theora
* (the encoding will stop automatically after 500 frames)
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/* make sure at least one buffer is queued on all pads, two if possible
- *
+ *
* if pad->buffer == NULL, pad->next_buffer != NULL, then
* we do not know if the buffer is the last or not
* if pad->buffer != NULL, pad->next_buffer != NULL, then
* pad->buffer is not the last buffer for the pad
* if pad->buffer != NULL, pad->next_buffer == NULL, then
* pad->buffer if the last buffer for the pad
- *
+ *
* returns a pointer to an oggpad that holds the best buffer, or
* NULL when no pad was usable. "best" means the buffer marked
* with the lowest timestamp. If best->buffer == NULL then either
* page that allows decoders to identify the type of the stream.
* After that we need to write out all extra info for the decoders.
* In the case of a codec that also needs data as configuration, we can
- * find that info in the streamcaps.
+ * find that info in the streamcaps.
* After writing the headers we must start a new page for the data.
*/
static GstFlowReturn
}
/* This function is called when there is data on all pads.
- *
+ *
* It finds a pad to pull on, this is done by looking at the buffers
* to decide which one to use, and using the 'oldest' one first. It then calls
* gst_ogg_mux_process_best_pad() to process as much data as possible.
- *
+ *
* If all the pads have received EOS, it flushes out all data by continually
* getting the best pad and calling gst_ogg_mux_process_best_pad() until they
* are all empty, and then sends EOS.
/**
* SECTION:element-opusdec
+ * @title: opusdec
* @see_also: opusenc, oggdemux
*
* This element decodes a OPUS stream to raw integer audio.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=opus.ogg ! oggdemux ! opusdec ! audioconvert ! audioresample ! alsasink
- * ]| Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc.
- * </refsect2>
+ * ]|
+ * Decode an Ogg/Opus file. To create an Ogg/Opus file refer to the documentation of opusenc.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-opusenc
+ * @title: opusenc
* @see_also: opusdec, oggmux
*
* This element encodes raw audio to OPUS.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! opusenc ! oggmux ! filesink location=sine.ogg
- * ]| Encode a test sine signal to Ogg/OPUS.
- * </refsect2>
+ * ]|
+ * Encode a test sine signal to Ogg/OPUS.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-clockoverlay
+ * @title: clockoverlay
* @see_also: #GstBaseTextOverlay, #GstTimeOverlay
*
* This element overlays the current clock time on top of a video
* time is displayed in the top left corner of the picture, with some
* padding to the left and to the top.
*
- * <refsect2>
- * <title>Example launch lines</title>
+ * ## Example launch lines
* |[
* gst-launch-1.0 -v videotestsrc ! clockoverlay ! autovideosink
- * ]| Display the current wall clock time in the top left corner of the video picture
+ * ]|
+ * Display the current wall clock time in the top left corner of the video picture
* |[
* gst-launch-1.0 -v videotestsrc ! clockoverlay halignment=right valignment=bottom text="Edge City" shaded-background=true font-desc="Sans, 36" ! videoconvert ! autovideosink
- * ]| Another pipeline that displays the current time with some leading
+ * ]|
+ * Another pipeline that displays the current time with some leading
* text in the bottom right corner of the video picture, with the background
* of the text being shaded in order to make it more legible on top of a
* bright video background.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-textoverlay
+ * @title: textoverlay
* @see_also: #GstTextRender, #GstTextOverlay, #GstTimeOverlay, #GstSubParse
*
* This plugin renders text on top of a video stream. This can be either
* The text can contain newline characters and text wrapping is enabled by
* default.
*
- * <refsect2>
- * <title>Example launch lines</title>
+ * ## Example launch lines
* |[
* gst-launch-1.0 -v gst-launch-1.0 videotestsrc ! textoverlay text="Room A" valignment=top halignment=left font-desc="Sans, 72" ! autovideosink
- * ]| Here is a simple pipeline that displays a static text in the top left
+ * ]|
+ * Here is a simple pipeline that displays a static text in the top left
* corner of the video picture
* |[
* gst-launch-1.0 -v filesrc location=subtitles.srt ! subparse ! txt. videotestsrc ! timeoverlay ! textoverlay name=txt shaded-background=yes ! autovideosink
- * ]| Here is another pipeline that displays subtitles from an .srt subtitle
+ * ]|
+ * Here is another pipeline that displays subtitles from an .srt subtitle
* file, centered at the bottom of the picture and with a rectangular shading
* around the text in the background:
- * <para>
+ *
* If you do not have such a subtitle file, create one looking like this
* in a text editor:
* |[
* Uh? What are you talking about?
* I don't understand (18-62s)
* ]|
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-textrender
+ * @title: textrender
* @see_also: #GstTextOverlay
*
* This plugin renders text received on the text sink pad to a video
* buffer (retaining the alpha channel), so it can later be overlayed
* on top of video streams using other elements.
*
- * The text can contain newline characters. (FIXME: What about text
+ * The text can contain newline characters. (FIXME: What about text
* wrapping? It does not make sense in this context)
*
- * <refsect2>
- * <title>Example launch lines</title>
+ * ## Example launch lines
* |[
* gst-launch-1.0 -v filesrc location=subtitles.srt ! subparse ! textrender ! videoconvert ! autovideosink
* ]|
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-timeoverlay
+ * @title: timeoverlay
* @see_also: #GstBaseTextOverlay, #GstClockOverlay
*
* This element overlays the buffer time stamps of a video stream on
* time stamp is displayed in the top left corner of the picture, with some
* padding to the left and to the top.
*
- * <refsect2>
* |[
* gst-launch-1.0 -v videotestsrc ! timeoverlay ! autovideosink
- * ]| Display the time stamps in the top left corner of the video picture.
+ * ]|
+ * Display the time stamps in the top left corner of the video picture.
* |[
* gst-launch-1.0 -v videotestsrc ! timeoverlay halignment=right valignment=bottom text="Stream time:" shaded-background=true font-desc="Sans, 24" ! autovideosink
- * ]| Another pipeline that displays the time stamps with some leading
+ * ]|
+ * Another pipeline that displays the time stamps with some leading
* text in the bottom right corner of the video picture, with the background
* of the text being shaded in order to make it more legible on top of a
* bright video background.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-theoradec
+ * @title: theoradec
* @see_also: theoraenc, oggdemux
*
* This element decodes theora streams into raw video
* video codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
* Foundation</ulink>, based on the VP3 codec.
*
- * <refsect2>
- * <title>Example pipeline</title>
+ * ## Example pipeline
* |[
* gst-launch-1.0 -v filesrc location=videotestsrc.ogg ! oggdemux ! theoradec ! videoconvert ! videoscale ! autovideosink
- * ]| This example pipeline will decode an ogg stream and decodes the theora video in it.
+ * ]|
+ * This example pipeline will decode an ogg stream and decodes the theora video in it.
* Refer to the theoraenc example to create the ogg file.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-theoraenc
+ * @title: theoraenc
* @see_also: theoradec, oggmux
*
* This element encodes raw video into a Theora stream.
* A videorate element is often required in front of theoraenc, especially
* when transcoding and when putting Theora into the Ogg container.
*
- * <refsect2>
- * <title>Example pipeline</title>
+ * ## Example pipeline
* |[
* gst-launch-1.0 -v videotestsrc num-buffers=500 ! video/x-raw,width=1280,height=720 ! queue ! progressreport ! theoraenc ! oggmux ! filesink location=videotestsrc.ogg
- * ]| This example pipeline will encode a test video source to theora muxed in an
+ * ]|
+ * This example pipeline will encode a test video source to theora muxed in an
* ogg container. Refer to the theoradec documentation to decode the create
* stream.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-theoraparse
+ * @title: theoraparse
* @see_also: theoradec, oggdemux, vorbisparse
*
* The theoraparse element will parse the header packets of the Theora
* offsetting all buffers that it outputs by a specified amount, and updating
* that offset from the value array whenever a keyframe is processed.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=video.ogg ! oggdemux ! theoraparse ! fakesink
- * ]| This pipeline shows that the streamheader is set in the caps, and that each
+ * ]|
+ * This pipeline shows that the streamheader is set in the caps, and that each
* buffer has the timestamp, duration, offset, and offset_end set.
* |[
* gst-launch-1.0 filesrc location=video.ogg ! oggdemux ! theoraparse \
* ! oggmux ! filesink location=video-remuxed.ogg
- * ]| This pipeline shows remuxing. video-remuxed.ogg might not be exactly the same
+ * ]|
+ * This pipeline shows remuxing. video-remuxed.ogg might not be exactly the same
* as video.ogg, but they should produce exactly the same decoded data.
- * </refsect2>
+ *
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
/**
* SECTION:element-vorbisdec
+ * @title: vorbisdec
* @see_also: vorbisenc, oggdemux
*
* This element decodes a Vorbis stream to raw float audio.
* Foundation</ulink>. As it outputs raw float audio you will often need to
* put an audioconvert element after it.
*
- *
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink
- * ]| Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
- * </refsect2>
+ * ]|
+ * Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-vorbisenc
+ * @title: vorbisenc
* @see_also: vorbisdec, oggmux
*
* This element encodes raw float audio into a Vorbis stream.
* audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
* Foundation</ulink>.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! vorbisenc ! oggmux ! filesink location=sine.ogg
- * ]| Encode a test sine signal to Ogg/Vorbis. Note that the resulting file
+ * ]|
+ * Encode a test sine signal to Ogg/Vorbis. Note that the resulting file
* will be really small because a sine signal compresses very well.
* |[
* gst-launch-1.0 -v autoaudiosrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
- * ]| Record from a sound card and encode to Ogg/Vorbis.
- * </refsect2>
+ * ]|
+ * Record from a sound card and encode to Ogg/Vorbis.
+ *
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
/**
* SECTION:element-vorbisparse
+ * @title: vorbisparse
* @see_also: vorbisdec, oggdemux, theoraparse
*
* The vorbisparse element will parse the header packets of the Vorbis
* vorbisparse outputs have all of the metadata that oggmux expects to receive,
* which allows you to (for example) remux an ogg/vorbis file.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=sine.ogg ! oggdemux ! vorbisparse ! fakesink
- * ]| This pipeline shows that the streamheader is set in the caps, and that each
+ * ]|
+ * This pipeline shows that the streamheader is set in the caps, and that each
* buffer has the timestamp, duration, offset, and offset_end set.
* |[
* gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisparse \
* ! oggmux ! filesink location=sine-remuxed.ogg
- * ]| This pipeline shows remuxing. sine-remuxed.ogg might not be exactly the same
+ * ]|
+ * This pipeline shows remuxing. sine-remuxed.ogg might not be exactly the same
* as sine.ogg, but they should produce exactly the same decoded data.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-vorbistag
+ * @title: vorbistag
* @see_also: #oggdemux, #oggmux, #vorbisparse, #GstTagSetter
*
* The vorbistags element can change the tag contained within a raw
* automatically (and merged according to the merge mode set via the tag
* setter interface).
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=foo.ogg ! oggdemux ! vorbistag ! oggmux ! filesink location=bar.ogg
- * ]| This element is not useful with gst-launch-1.0, because it does not support
+ * ]|
+ * This element is not useful with gst-launch-1.0, because it does not support
* setting the tags on a #GstTagSetter interface. Conceptually, the element
* will usually be used in this order though.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstdmabuf
+ * @title: GstDmaBufAllocator
* @short_description: Memory wrapper for Linux dmabuf memory
* @see_also: #GstMemory
*
/**
* SECTION:gstfdmemory
+ * @title: GstFdAllocator
* @short_description: Memory wrapper for fd backed memory
* @see_also: #GstMemory
*
*/
/**
* SECTION:gstappsink
+ * @title: GstAppSink
* @short_description: Easy way for applications to extract samples from a
* pipeline
* @see_also: #GstSample, #GstBaseSink, appsrc
*/
/**
* SECTION:gstappsrc
+ * @title: GstAppSrc
* @short_description: Easy way for applications to inject buffers into a
* pipeline
* @see_also: #GstBaseSrc, appsink
*/
/**
* SECTION:gstaudiochannels
+ * @title: Audio-channels
* @short_description: Support library for audio channel handling
*
* This library contains some helper functions for multichannel audio.
/**
* SECTION:audioconverter
+ * @title: GstAudioConverter
* @short_description: Generic audio conversion
*
- * <refsect2>
- * <para>
* This object is used to convert audio samples from one format to another.
* The object can perform conversion of:
- * <itemizedlist>
- * <listitem><para>
- * audio format with optional dithering and noise shaping
- * </para></listitem>
- * <listitem><para>
- * audio samplerate
- * </para></listitem>
- * <listitem><para>
- * audio channels and channel layout
- * </para></listitem>
- * </para>
- * </refsect2>
+ *
+ * * audio format with optional dithering and noise shaping
+ *
+ * * audio samplerate
+ *
+ * * audio channels and channel layout
+ *
*/
#ifndef GST_DISABLE_GST_DEBUG
}
/**
- * gst_audio_converter_supports_inplace
+ * gst_audio_converter_supports_inplace:
* @convert: a #GstAudioConverter
*
* Returns whether the audio converter can perform the conversion in-place.
/**
* SECTION:gstaudioresampler
+ * @title: GstAudioResampler
* @short_description: Utility structure for resampler information
*
* #GstAudioResampler is a structure which holds the information
typedef struct _GstAudioResampler GstAudioResampler;
/**
- * GST_AUDIO_RESAMPLER_OPT_CUTOFF
+ * GST_AUDIO_RESAMPLER_OPT_CUTOFF:
*
* G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_CUTOFF "GstAudioResampler.cutoff"
/**
- * GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUTATION
+ * GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUTATION:
*
* G_TYPE_DOUBLE, stopband attenuation in debibels. The attenutation
* after the stopband for the kaiser window. 85 dB is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION "GstAudioResampler.stop-attenutation"
/**
- * GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH
+ * GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH:
*
* G_TYPE_DOUBLE, transition bandwidth. The width of the
* transition band for the kaiser window. 0.087 is the default.
*/
#define GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION "GstAudioResampler.filter-interpolation"
/**
- * GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE
+ * GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE:
*
* G_TYPE_UINT, oversampling to use when interpolating filters
* 8 is the default.
*/
/**
* SECTION:gstaudio
+ * @title: GstAudio
* @short_description: Support library for audio elements
*
* This library contains some helper functions for audio elements.
* @segment: Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
* the buffer should be clipped.
* @rate: sample rate.
- * @bpf: size of one audio frame in bytes. This is the size of one sample *
+ * @bpf: size of one audio frame in bytes. This is the size of one sample *
* number of channels.
*
* Clip the buffer to the given %GstSegment.
/**
* SECTION:gstaudiobasesink
+ * @title: GstAudioBaseSink
* @short_description: Base class for audio sinks
* @see_also: #GstAudioSink, #GstAudioRingBuffer.
*
/**
* SECTION:gstaudiobasesrc
+ * @title: GstAudioBaseSrc
* @short_description: Base class for audio sources
* @see_also: #GstAudioSrc, #GstAudioRingBuffer.
*
/**
* SECTION:gstaudiocdsrc
+ * @title: GstAudioCdSrc
* @short_description: Base class for Audio CD sources
*
- * <para>
* Provides a base class for CD digital audio (CDDA) sources, which handles
* things like seeking, querying, discid calculation, tags, and buffer
* timestamping.
- * </para>
- * <refsect2>
- * <title>Using GstAudioCdSrc-based elements in applications</title>
- * <para>
+ *
+ * ## Using GstAudioCdSrc-based elements in applications
+ *
* GstAudioCdSrc registers two #GstFormat<!-- -->s of its own, namely
* the "track" format and the "sector" format. Applications will usually
* only find the "track" format interesting. You can retrieve that #GstFormat
* for use in seek events or queries with gst_format_get_by_nick("track").
- * </para>
- * <para>
+ *
* In order to query the number of tracks, for example, an application would
* set the CDDA source element to READY or PAUSED state and then query the
* the number of tracks via gst_element_query_duration() using the track
* format acquired above. Applications can query the currently playing track
* in the same way.
- * </para>
- * <para>
+ *
* Alternatively, applications may retrieve the currently playing track and
* the total number of tracks from the taglist that will posted on the bus
* whenever the CD is opened or the currently playing track changes. The
* taglist will contain GST_TAG_TRACK_NUMBER and GST_TAG_TRACK_COUNT tags.
- * </para>
- * <para>
+ *
* Applications playing back CD audio using playbin and cdda://n URIs should
* issue a seek command in track format to change between tracks, rather than
* setting a new cdda://n+1 URI on playbin (as setting a new URI on playbin
* involves closing and re-opening the CD device, which is much much slower).
- * </para>
- * <refsect2>
- * </refsect2>
- * <title>Tags and meta-information</title>
- * <para>
+ *
+ * ## Tags and meta-information
+ *
* CDDA sources will automatically emit a number of tags, details about which
* can be found in the libgsttag documentation. Those tags are:
* #GST_TAG_CDDA_CDDB_DISCID, #GST_TAG_CDDA_CDDB_DISCID_FULL,
* #GST_TAG_CDDA_MUSICBRAINZ_DISCID, #GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL,
* among others.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Tracks and Table of Contents (TOC)</title>
- * <para>
+ *
+ * ## Tracks and Table of Contents (TOC)
+ *
* Applications will be informed of the available tracks via a TOC message
* on the pipeline's #GstBus. The #GstToc will contain a #GstTocEntry for
* each track, with information about each track. The duration for each
* track can be retrieved via the #GST_TAG_DURATION tag from each entry's
* tag list, or calculated via gst_toc_entry_get_start_stop_times().
* The track entries in the TOC will be sorted by track number.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstaudioclock
+ * @title: GstAudioClock
* @short_description: Helper object for implementing audio clocks
* @see_also: #GstAudioBaseSink, #GstSystemClock
*
/**
* SECTION:gstaudiodecoder
+ * @title: GstAudioDecoder
* @short_description: Base class for audio decoders
* @see_also: #GstBaseTransform
*
* raw audio samples.
*
* GstAudioDecoder and subclass should cooperate as follows.
- * <orderedlist>
- * <listitem>
- * <itemizedlist><title>Configuration</title>
- * <listitem><para>
- * Initially, GstAudioDecoder calls @start when the decoder element
+ *
+ * ## Configuration
+ *
+ * * Initially, GstAudioDecoder calls @start when the decoder element
* is activated, which allows subclass to perform any global setup.
* Base class (context) parameters can already be set according to subclass
* capabilities (or possibly upon receive more information in subsequent
* @set_format).
- * </para></listitem>
- * <listitem><para>
- * GstAudioDecoder calls @set_format to inform subclass of the format
+ * * GstAudioDecoder calls @set_format to inform subclass of the format
* of input audio data that it is about to receive.
* While unlikely, it might be called more than once, if changing input
* parameters require reconfiguration.
- * </para></listitem>
- * <listitem><para>
- * GstAudioDecoder calls @stop at end of all processing.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
+ * * GstAudioDecoder calls @stop at end of all processing.
+ *
* As of configuration stage, and throughout processing, GstAudioDecoder
* provides various (context) parameters, e.g. describing the format of
* output audio data (valid when output caps have been set) or current parsing state.
* Conversely, subclass can and should configure context to inform
* base class of its expectation w.r.t. buffer handling.
- * <listitem>
- * <itemizedlist>
- * <title>Data processing</title>
- * <listitem><para>
- * Base class gathers input data, and optionally allows subclass
+ *
+ * ## Data processing
+ * * Base class gathers input data, and optionally allows subclass
* to parse this into subsequently manageable (as defined by subclass)
* chunks. Such chunks are subsequently referred to as 'frames',
* though they may or may not correspond to 1 (or more) audio format frame.
- * </para></listitem>
- * <listitem><para>
- * Input frame is provided to subclass' @handle_frame.
- * </para></listitem>
- * <listitem><para>
- * If codec processing results in decoded data, subclass should call
+ * * Input frame is provided to subclass' @handle_frame.
+ * * If codec processing results in decoded data, subclass should call
* @gst_audio_decoder_finish_frame to have decoded data pushed
* downstream.
- * </para></listitem>
- * <listitem><para>
- * Just prior to actually pushing a buffer downstream,
+ * * Just prior to actually pushing a buffer downstream,
* it is passed to @pre_push. Subclass should either use this callback
* to arrange for additional downstream pushing or otherwise ensure such
* custom pushing occurs after at least a method call has finished since
* setting src pad caps.
- * </para></listitem>
- * <listitem><para>
- * During the parsing process GstAudioDecoderClass will handle both
+ * * During the parsing process GstAudioDecoderClass will handle both
* srcpad and sinkpad events. Sink events will be passed to subclass
* if @event callback has been provided.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist><title>Shutdown phase</title>
- * <listitem><para>
- * GstAudioDecoder class calls @stop to inform the subclass that data
+ *
+ * ## Shutdown phase
+ *
+ * * GstAudioDecoder class calls @stop to inform the subclass that data
* parsing will be stopped.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * </orderedlist>
*
* Subclass is responsible for providing pad template caps for
* source and sink pads. The pads need to be named "sink" and "src". It also
* bitrates.
*
* Things that subclass need to take care of:
- * <itemizedlist>
- * <listitem><para>Provide pad templates</para></listitem>
- * <listitem><para>
- * Set source pad caps when appropriate
- * </para></listitem>
- * <listitem><para>
- * Set user-configurable properties to sane defaults for format and
+ *
+ * * Provide pad templates
+ * * Set source pad caps when appropriate
+ * * Set user-configurable properties to sane defaults for format and
* implementing codec at hand, and convey some subclass capabilities and
* expectations in context.
- * </para></listitem>
- * <listitem><para>
- * Accept data in @handle_frame and provide encoded results to
+ *
+ * * Accept data in @handle_frame and provide encoded results to
* @gst_audio_decoder_finish_frame. If it is prepared to perform
* PLC, it should also accept NULL data in @handle_frame and provide for
* data for indicated duration.
- * </para></listitem>
- * </itemizedlist>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstaudioencoder
+ * @title: GstAudioEncoder
* @short_description: Base class for audio encoders
* @see_also: #GstBaseTransform
*
* encoded audio data.
*
* GstAudioEncoder and subclass should cooperate as follows.
- * <orderedlist>
- * <listitem>
- * <itemizedlist><title>Configuration</title>
- * <listitem><para>
- * Initially, GstAudioEncoder calls @start when the encoder element
+ *
+ * ## Configuration
+ *
+ * * Initially, GstAudioEncoder calls @start when the encoder element
* is activated, which allows subclass to perform any global setup.
- * </para></listitem>
- * <listitem><para>
- * GstAudioEncoder calls @set_format to inform subclass of the format
+ *
+ * * GstAudioEncoder calls @set_format to inform subclass of the format
* of input audio data that it is about to receive. Subclass should
* setup for encoding and configure various base class parameters
* appropriately, notably those directing desired input data handling.
* While unlikely, it might be called more than once, if changing input
* parameters require reconfiguration.
- * </para></listitem>
- * <listitem><para>
- * GstAudioEncoder calls @stop at end of all processing.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
+ *
+ * * GstAudioEncoder calls @stop at end of all processing.
+ *
* As of configuration stage, and throughout processing, GstAudioEncoder
* maintains various parameters that provide required context,
* e.g. describing the format of input audio data.
* Conversely, subclass can and should configure these context parameters
* to inform base class of its expectation w.r.t. buffer handling.
- * <listitem>
- * <itemizedlist>
- * <title>Data processing</title>
- * <listitem><para>
- * Base class gathers input sample data (as directed by the context's
+ *
+ * ## Data processing
+ *
+ * * Base class gathers input sample data (as directed by the context's
* frame_samples and frame_max) and provides this to subclass' @handle_frame.
- * </para></listitem>
- * <listitem><para>
- * If codec processing results in encoded data, subclass should call
+ * * If codec processing results in encoded data, subclass should call
* gst_audio_encoder_finish_frame() to have encoded data pushed
* downstream. Alternatively, it might also call
* gst_audio_encoder_finish_frame() (with a NULL buffer and some number of
* dropped samples) to indicate dropped (non-encoded) samples.
- * </para></listitem>
- * <listitem><para>
- * Just prior to actually pushing a buffer downstream,
+ * * Just prior to actually pushing a buffer downstream,
* it is passed to @pre_push.
- * </para></listitem>
- * <listitem><para>
- * During the parsing process GstAudioEncoderClass will handle both
+ * * During the parsing process GstAudioEncoderClass will handle both
* srcpad and sinkpad events. Sink events will be passed to subclass
* if @event callback has been provided.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist><title>Shutdown phase</title>
- * <listitem><para>
- * GstAudioEncoder class calls @stop to inform the subclass that data
+ *
+ * ## Shutdown phase
+ *
+ * * GstAudioEncoder class calls @stop to inform the subclass that data
* parsing will be stopped.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * </orderedlist>
*
* Subclass is responsible for providing pad template caps for
* source and sink pads. The pads need to be named "sink" and "src". It also
* by same sample count and sample rate).
*
* Things that subclass need to take care of:
- * <itemizedlist>
- * <listitem><para>Provide pad templates</para></listitem>
- * <listitem><para>
- * Set source pad caps when appropriate
- * </para></listitem>
- * <listitem><para>
- * Inform base class of buffer processing needs using context's
+ *
+ * * Provide pad templates
+ * * Set source pad caps when appropriate
+ * * Inform base class of buffer processing needs using context's
* frame_samples and frame_bytes.
- * </para></listitem>
- * <listitem><para>
- * Set user-configurable properties to sane defaults for format and
+ * * Set user-configurable properties to sane defaults for format and
* implementing codec at hand, e.g. those controlling timestamp behaviour
* and discontinuity processing.
- * </para></listitem>
- * <listitem><para>
- * Accept data in @handle_frame and provide encoded results to
+ * * Accept data in @handle_frame and provide encoded results to
* gst_audio_encoder_finish_frame().
- * </para></listitem>
- * </itemizedlist>
*
*/
/**
* SECTION:gstaudiofilter
+ * @title: GstAudioFilter
* @short_description: Base class for simple audio filters
*
* #GstAudioFilter is a #GstBaseTransform<!-- -->-derived base class for simple audio
/**
* SECTION:gstaudioiec61937
+ * @title: GstAudio IEC61937
* @short_description: Utility functions for IEC 61937 payloading
*
* This module contains some helper functions for encapsulating various
/**
* SECTION:gstaudiometa
+ * @title: GstAudioDownmixMeta
* @short_description: Buffer metadata for audio downmix matrix handling
*
* #GstAudioDownmixMeta defines an audio downmix matrix to be send along with
/**
* SECTION:gstaudioringbuffer
+ * @title: GstAudioRingBuffer
* @short_description: Base class for audio ringbuffer implementations
* @see_also: #GstAudioBaseSink, #GstAudioSink
*
- * <refsect2>
- * <para>
* This object is the base class for audio ringbuffers used by the base
* audio source and sink classes.
- * </para>
- * <para>
+ *
* The ringbuffer abstracts a circular buffer of data. One reader and
* one writer can operate on the data from different threads in a lockfree
* manner. The base class is sufficiently flexible to be used as an
* abstraction for DMA based ringbuffers as well as a pure software
* implementations.
- * </para>
- * </refsect2>
+ *
*/
#include <string.h>
/**
* SECTION:gstaudiosink
+ * @title: GstAudioSink
* @short_description: Simple base class for audio sinks
* @see_also: #GstAudioBaseSink, #GstAudioRingBuffer, #GstAudioSink.
*
* This is the most simple base class for audio sinks that only requires
* subclasses to implement a set of simple functions:
*
- * <variablelist>
- * <varlistentry>
- * <term>open()</term>
- * <listitem><para>Open the device.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>prepare()</term>
- * <listitem><para>Configure the device with the specified format.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>write()</term>
- * <listitem><para>Write samples to the device.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>reset()</term>
- * <listitem><para>Unblock writes and flush the device.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>delay()</term>
- * <listitem><para>Get the number of samples written but not yet played
- * by the device.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>unprepare()</term>
- * <listitem><para>Undo operations done by prepare.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>close()</term>
- * <listitem><para>Close the device.</para></listitem>
- * </varlistentry>
- * </variablelist>
+ * * `open()` :Open the device.
+ *
+ * * `prepare()` :Configure the device with the specified format.
+ *
+ * * `write()` :Write samples to the device.
+ *
+ * * `reset()` :Unblock writes and flush the device.
+ *
+ * * `delay()` :Get the number of samples written but not yet played
+ * by the device.
+ *
+ * * `unprepare()` :Undo operations done by prepare.
+ *
+ * * `close()` :Close the device.
*
* All scheduling of samples and timestamps is done in this base class
* together with #GstAudioBaseSink using a default implementation of a
/**
* SECTION:gstaudiosrc
+ * @title: GstAudioSrc
* @short_description: Simple base class for audio sources
* @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer, #GstAudioSrc.
*
* This is the most simple base class for audio sources that only requires
* subclasses to implement a set of simple functions:
*
- * <variablelist>
- * <varlistentry>
- * <term>open()</term>
- * <listitem><para>Open the device.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>prepare()</term>
- * <listitem><para>Configure the device with the specified format.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>read()</term>
- * <listitem><para>Read samples from the device.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>reset()</term>
- * <listitem><para>Unblock reads and flush the device.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>delay()</term>
- * <listitem><para>Get the number of samples in the device but not yet read.
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>unprepare()</term>
- * <listitem><para>Undo operations done by prepare.</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>close()</term>
- * <listitem><para>Close the device.</para></listitem>
- * </varlistentry>
- * </variablelist>
+ * * `open()` :Open the device.
+ * * `prepare()` :Configure the device with the specified format.
+ * * `read()` :Read samples from the device.
+ * * `reset()` :Unblock reads and flush the device.
+ * * `delay()` :Get the number of samples in the device but not yet read.
+ * * `unprepare()` :Undo operations done by prepare.
+ * * `close()` :Close the device.
*
* All scheduling of samples and timestamps is done in this base class
* together with #GstAudioBaseSrc using a default implementation of a
/**
* SECTION:gststreamvolume
+ * @title: GstStreamVolume
* @short_description: Interface for elements that provide a stream volume
*
- * <refsect2>
- * <para>
* This interface is implemented by elements that provide a stream volume. Examples for
* such elements are #volume and #playbin.
- * </para>
- * <para>
+ *
* Applications can use this interface to get or set the current stream volume. For this
* the "volume" #GObject property can be used or the helper functions gst_stream_volume_set_volume()
* and gst_stream_volume_get_volume(). This volume is always a linear factor, i.e. 0.0 is muted
*
* Separate from the volume the stream can also be muted by the "mute" #GObject property or
* gst_stream_volume_set_mute() and gst_stream_volume_get_mute().
- * </para>
- * <para>
+ *
* Elements that provide some kind of stream volume should implement the "volume" and
* "mute" #GObject properties and handle setting and getting of them properly.
* The volume property is defined to be a linear volume factor.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstfft
+ * @title: GstFFT
* @short_description: General FFT functions and declarations
- *
+ *
* This library includes general definitions and functions, useful for
* all typed FFT classes.
*/
/**
* SECTION:gstfftf32
+ * @title: GstFFTF32
* @short_description: FFT functions for 32 bit float samples
*
* #GstFFTF32 provides a FFT implementation and related functions for
/**
* SECTION:gstfftf64
+ * @title: GstFFTF64
* @short_description: FFT functions for 64 bit float samples
*
* #GstFFTF64 provides a FFT implementation and related functions for
/**
* SECTION:gstffts16
+ * @title: GstFFTS16
* @short_description: FFT functions for signed 16 bit integer samples
*
* #GstFFTS16 provides a FFT implementation and related functions for
/**
* SECTION:gstffts32
+ * @title: GstFFTS32
* @short_description: FFT functions for signed 32 bit integer samples
*
* #GstFFTS32 provides a FFT implementation and related functions for
/**
* SECTION:gstpbutilscodecutils
+ * @title: Codec utilities
* @short_description: Miscellaneous codec-specific utility functions
*
- * <refsect2>
- * <para>
* Provides codec-specific ulility functions such as functions to provide the
* codec profile and level in human-readable string form from header data.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
* determined using the AudioObjectType field which is in the first 5 bits of
* @audio_config.
*
- * <note>
- * HE-AAC support has not yet been implemented.
- * </note>
+ * > HE-AAC support has not yet been implemented.
*
* Returns: The profile as a const string and %NULL if the profile could not be
* determined.
* The @audio_config parameter follows the following format, starting from the
* most significant bit of the first byte:
*
- * <itemizedlist>
- * <listitem><para>
- * Bit 0:4 contains the AudioObjectType
- * </para></listitem>
- * <listitem><para>
- * Bit 5:8 contains the sample frequency index (if this is 0xf, then the
- * next 24 bits define the actual sample frequency, and subsequent
- * fields are appropriately shifted).
- * </para></listitem>
- * <listitem><para>
- * Bit 9:12 contains the channel configuration
- * </para></listitem>
- * </itemizedlist>
- *
- * <note>
- * HE-AAC support has not yet been implemented.
- * </note>
+ * * Bit 0:4 contains the AudioObjectType
+ * * Bit 5:8 contains the sample frequency index (if this is 0xf, then the
+ * next 24 bits define the actual sample frequency, and subsequent
+ * fields are appropriately shifted).
+ * * Bit 9:12 contains the channel configuration
+ *
+ * > HE-AAC support has not yet been implemented.
*
* Returns: The level as a const string and %NULL if the level could not be
* determined.
* as a bitstream here, with bit 0 being the most significant bit of the first
* byte.
*
- * <itemizedlist>
- * <listitem><para>Bit 0:7 - Profile indication</para></listitem>
- * <listitem><para>Bit 8 - constraint_set0_flag</para></listitem>
- * <listitem><para>Bit 9 - constraint_set1_flag</para></listitem>
- * <listitem><para>Bit 10 - constraint_set2_flag</para></listitem>
- * <listitem><para>Bit 11 - constraint_set3_flag</para></listitem>
- * <listitem><para>Bit 12 - constraint_set3_flag</para></listitem>
- * <listitem><para>Bit 13:15 - Reserved</para></listitem>
- * <listitem><para>Bit 16:24 - Level indication</para></listitem>
- * </itemizedlist>
+ * * Bit 0:7 - Profile indication
+ * * Bit 8 - constraint_set0_flag
+ * * Bit 9 - constraint_set1_flag
+ * * Bit 10 - constraint_set2_flag
+ * * Bit 11 - constraint_set3_flag
+ * * Bit 12 - constraint_set3_flag
+ * * Bit 13:15 - Reserved
+ * * Bit 16:24 - Level indication
*
* Returns: The profile as a const string, or %NULL if there is an error.
*/
* specification. The profile_tier_level is viewed as a bitstream here,
* with bit 0 being the most significant bit of the first byte.
*
- * <itemizedlist>
- * <listitem><para>Bit 0:1 - general_profile_space</para></listitem>
- * <listitem><para>Bit 2 - general_tier_flag</para></listitem>
- * <listitem><para>Bit 3:7 - general_profile_idc</para></listitem>
- * <listitem><para>Bit 8:39 - gernal_profile_compatibility_flags</para></listitem>
- * <listitem><para>Bit 40 - general_progressive_source_flag</para></listitem>
- * <listitem><para>Bit 41 - general_interlaced_source_flag</para></listitem>
- * <listitem><para>Bit 42 - general_non_packed_constraint_flag</para></listitem>
- * <listitem><para>Bit 43 - general_frame_only_constraint_flag</para></listitem>
- * <listitem><para>Bit 44:87 - general_reserved_zero_44bits</para></listitem>
- * <listitem><para>Bit 88:95 - general_level_idc</para></listitem>
- * </itemizedlist>
+ * * Bit 0:1 - general_profile_space
+ * * Bit 2 - general_tier_flag
+ * * Bit 3:7 - general_profile_idc
+ * * Bit 8:39 - gernal_profile_compatibility_flags
+ * * Bit 40 - general_progressive_source_flag
+ * * Bit 41 - general_interlaced_source_flag
+ * * Bit 42 - general_non_packed_constraint_flag
+ * * Bit 43 - general_frame_only_constraint_flag
+ * * Bit 44:87 - general_reserved_zero_44bits
+ * * Bit 88:95 - general_level_idc
*
* Returns: The profile as a const string, or %NULL if there is an error.
*
/**
* SECTION:gstpbutilsdescriptions
+ * @title: Descriptions
* @short_description: Provides human-readable descriptions for caps/codecs
* and encoder, decoder, URI source and URI sink elements
*
- * <refsect2>
- * <para>
* The above functions provide human-readable strings for media formats
* and decoder/demuxer/depayloader/encoder/muxer/payloader elements for use
* in error dialogs or other messages shown to users.
- * </para>
- * <para>
+ *
* gst_pb_utils_add_codec_description_to_tag_list() is a utility function
* for demuxer and decoder elements to add audio/video codec tags from a
* given (fixed) #GstCaps.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:encoding-profile
+ * @title: GstEncodingProfile
* @short_description: Encoding profile library
*
* Functions to create and handle encoding profiles.
* return (GstEncodingProfile*) prof;
*}
*
- *
* ]|
*
* # Example: Using an encoder preset with a profile
* return (GstEncodingProfile*) prof;
*}
*
- *
* ]|
*
* # Example: Listing categories, targets and profiles
*/
/**
* SECTION:gstaudiovisualizer
+ * @title: GstAudioVisualizer
*
* A baseclass for scopes (visualizers). It takes care of re-fitting the
* audio-rate to video-rate and handles renegotiation (downstream video size
/**
* SECTION:gstdiscoverer
+ * @title: GstDiscoverer
* @short_description: Utility for discovering information on URIs.
*
* The #GstDiscoverer is a utility object which allows to get as much
/**
* SECTION:gstpluginsbaseversion
+ * @title: Version
* @short_description: GStreamer gst-plugins-base libraries version macros.
*
* Use the GST_PLUGINS_BASE_VERSION_* macros e.g. to check what version of
/**
* SECTION:gstpbutilsinstallplugins
+ * @title: Install-plugins
* @short_description: Missing plugin installation support for applications
*
- * <refsect2>
- * <title>Overview</title>
- * <para>
+ * ## Overview
+ *
* Using this API, applications can request the installation of missing
- * GStreamer plugins. These may be missing decoders/demuxers or encoders/muxers
- * for a certain format, sources or sinks for a certain URI protocol
- * (e.g. 'http'), or certain elements known by their element factory name
- * ('audioresample').
- * </para>
- * <para>
+ * GStreamer plugins. These may be missing decoders/demuxers or
+ * encoders/muxers for a certain format, sources or sinks for a certain URI
+ * protocol (e.g. 'http'), or certain elements known by their element
+ * factory name ('audioresample').
+ *
* Whether plugin installation is supported or not depends on the operating
- * system and/or distribution in question. The vendor of the operating system
- * needs to make sure the necessary hooks and mechanisms are in place for
- * plugin installation to work. See below for more detailed information.
- * </para>
- * <para>
- * From the application perspective, plugin installation is usually triggered
- * either
- * <itemizedlist>
- * <listitem><para>
- * when the application itself has found that it wants or needs to install a
- * certain element
- * </para></listitem>
- * <listitem><para>
- * when the application has been notified by an element (such as playbin or
- * decodebin) that one or more plugins are missing <emphasis>and</emphasis>
- * the application has decided that it wants to install one or more of those
- * missing plugins
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * <title>Detail Strings</title>
- * <para>
- * The install functions in this section all take one or more 'detail strings'.
- * These detail strings contain information about the type of plugin that
- * needs to be installed (decoder, encoder, source, sink, or named element),
- * and some additional information such GStreamer version used and a
- * human-readable description of the component to install for user dialogs.
- * </para>
- * <para>
+ * system and/or distribution in question. The vendor of the operating
+ * system needs to make sure the necessary hooks and mechanisms are in
+ * place for plugin installation to work. See below for more detailed
+ * information.
+ *
+ * From the application perspective, plugin installation is usually
+ * triggered either
+ *
+ * - when the application itself has found that it wants or needs to
+ * install a certain element
+ * - when the application has been notified by an element (such as
+ * playbin or decodebin) that one or more plugins are missing *and* the
+ * application has decided that it wants to install one or more of
+ * those missing plugins
+ *
+ * The install functions in this section all take one or more 'detail
+ * strings'. These detail strings contain information about the type of
+ * plugin that needs to be installed (decoder, encoder, source, sink, or
+ * named element), and some additional information such GStreamer version
+ * used and a human-readable description of the component to install for
+ * user dialogs.
+ *
* Applications should not concern themselves with the composition of the
- * string itself. They should regard the string as if it was a shared secret
- * between GStreamer and the plugin installer application.
- * </para>
- * <para>
+ * string itself. They should regard the string as if it was a shared
+ * secret between GStreamer and the plugin installer application.
+ *
* Detail strings can be obtained using the function
- * gst_missing_plugin_message_get_installer_detail() on a missing-plugin
- * message. Such a message will either have been found by the application on
- * a pipeline's #GstBus, or the application will have created it itself using
- * gst_missing_element_message_new(), gst_missing_decoder_message_new(),
- * gst_missing_encoder_message_new(), gst_missing_uri_sink_message_new(), or
+ * gst_missing_plugin_message_get_installer_detail() on a
+ * missing-plugin message. Such a message will either have been found by
+ * the application on a pipeline's #GstBus, or the application will have
+ * created it itself using gst_missing_element_message_new(),
+ * gst_missing_decoder_message_new(),
+ * gst_missing_encoder_message_new(),
+ * gst_missing_uri_sink_message_new(), or
* gst_missing_uri_source_message_new().
- * </para>
- * <title>Plugin Installation from the Application Perspective</title>
- * <para>
- * For each GStreamer element/plugin/component that should be installed, the
- * application needs one of those 'installer detail' string mentioned in the
- * previous section. This string can be obtained, as already mentioned above,
- * from a missing-plugin message using the function
- * gst_missing_plugin_message_get_installer_detail(). The missing-plugin
- * message is either posted by another element and then found on the bus
- * by the application, or the application has created it itself as described
- * above.
- * </para>
- * <para>
+ *
+ * For each GStreamer element/plugin/component that should be installed,
+ * the application needs one of those 'installer detail' string mentioned
+ * in the previous section. This string can be obtained, as already
+ * mentioned above, from a missing-plugin message using the function
+ * gst_missing_plugin_message_get_installer_detail(). The
+ * missing-plugin message is either posted by another element and then
+ * found on the bus by the application, or the application has created it
+ * itself as described above.
+ *
* The application will then call gst_install_plugins_async(), passing a
* NULL-terminated array of installer detail strings, and a function that
* should be called when the installation of the plugins has finished
* (successfully or not). Optionally, a #GstInstallPluginsContext created
- * with gst_install_plugins_context_new() may be passed as well. This way
- * additional optional arguments like the application window's XID can be
- * passed to the external installer application.
- * </para>
- * <para>
+ * with gst_install_plugins_context_new() may be passed as well. This
+ * way additional optional arguments like the application window's XID can
+ * be passed to the external installer application.
+ *
* gst_install_plugins_async() will return almost immediately, with the
* return code indicating whether plugin installation was started or not.
* If the necessary hooks for plugin installation are in place and an
* external installer application has in fact been called, the passed in
- * function will be called with a result code as soon as the external installer
- * has finished. If the result code indicates that new plugins have been
- * installed, the application will want to call gst_update_registry() so the
- * run-time plugin registry is updated and the new plugins are made available
- * to the application.
- * <note>
- * A Gtk/GLib main loop must be running in order for the result function to
- * be called when the external installer has finished. If this is not the case,
- * make sure to regularly call
- * <programlisting>
- * g_main_context_iteration (NULL,FALSE);
- * </programlisting>
- * from your code.
- * </note>
- * </para>
- * <title>Plugin Installation from the Vendor/Distribution Perspective</title>
- * <para>
- * <emphasis>1. Installer hook</emphasis>
- * </para>
- * <para>
+ * function will be called with a result code as soon as the external
+ * installer has finished. If the result code indicates that new plugins
+ * have been installed, the application will want to call
+ * gst_update_registry() so the run-time plugin registry is updated and
+ * the new plugins are made available to the application.
+ *
+ * > A Gtk/GLib main loop must be running in order for the result function
+ * > to be called when the external installer has finished. If this is not
+ * > the case, make sure to regularly call in your code:
+ * >
+ * > g_main_context_iteration (NULL,FALSE);
+ *
+ * ## 1. Installer hook
+ *
* When GStreamer applications initiate plugin installation via
- * gst_install_plugins_async() or gst_install_plugins_sync(), a pre-defined
- * helper application will be called.
- * </para>
- * <para>
+ * gst_install_plugins_async() or gst_install_plugins_sync(), a
+ * pre-defined helper application will be called.
+ *
* The exact path of the helper application to be called is set at compile
- * time, usually by the <literal>./configure</literal> script based on the
- * install prefix. For a normal package build into the <literal>/usr</literal>
- * prefix, this will usually default to
- * <filename>/usr/libexec/gst-install-plugins-helper</filename> or
- * <filename>/usr/lib/gst-install-plugins-helper</filename>.
- * </para>
- * <para>
+ * time, usually by the `./configure` script based on the install prefix.
+ * For a normal package build into the `/usr` prefix, this will usually
+ * default to `/usr/libexec/gst-install-plugins-helper` or
+ * `/usr/lib/gst-install-plugins-helper`.
+ *
* Vendors/distros who want to support GStreamer plugin installation should
- * either provide such a helper script/application or use the
- * <literal>./configure</literal> option
- * <literal>--with-install-plugins-helper=/path/to/installer</literal> to
- * make GStreamer call an installer of their own directly.
- * </para>
- * <para>
- * It is strongly recommended that vendors provide a small helper application
- * as interlocutor to the real installer though, even more so if command line
- * argument munging is required to transform the command line arguments
- * passed by GStreamer to the helper application into arguments that are
- * understood by the real installer.
- * </para>
- * <para>
+ * either provide such a helper script/application or use the `./configure`
+ * option `--with-install-plugins-helper=/path/to/installer` to make
+ * GStreamer call an installer of their own directly.
+ *
+ * It is strongly recommended that vendors provide a small helper
+ * application as interlocutor to the real installer though, even more so
+ * if command line argument munging is required to transform the command
+ * line arguments passed by GStreamer to the helper application into
+ * arguments that are understood by the real installer.
+ *
* The helper application path defined at compile time can be overriden at
- * runtime by setting the <envar>GST_INSTALL_PLUGINS_HELPER</envar>
- * environment variable. This can be useful for testing/debugging purposes.
- * </para>
- * <para>
- * <emphasis>2. Arguments passed to the install helper</emphasis>
- * </para>
- * <para>
- * GStreamer will pass the following arguments to the install helper (this is
- * in addition to the path of the executable itself, which is by convention
- * argv[0]):
- * <itemizedlist>
- * <listitem><para>
- * none to many optional arguments in the form of
- * <literal>--foo-bar=val</literal>. Example:
- * <literal>--transient-for=XID</literal> where XID is the X Window ID of
- * the main window of the calling application (so the installer can make
- * itself transient to that window). Unknown optional arguments should
- * be ignored by the installer.
- * </para></listitem>
- * <listitem><para>
- * one 'installer detail string' argument for each plugin to be installed;
- * these strings will have a <literal>gstreamer</literal> prefix; the
- * exact format of the detail string is explained below
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * <para>
- * <emphasis>3. Detail string describing the missing plugin</emphasis>
- * </para>
- * <para>
- * The string is in UTF-8 encoding and is made up of several fields, separated
- * by '|' characters (but neither the first nor the last character is a '|').
- * The fields are:
- * <itemizedlist>
- * <listitem><para>
- * plugin system identifier, ie. "gstreamer"
- * </para><para>
- * This identifier determines the format of the rest of the detail string.
- * Automatic plugin installers should not process detail strings with
- * unknown identifiers. This allows other plugin-based libraries to use
- * the same mechanism for their automatic plugin installation needs, or
- * for the format to be changed should it turn out to be insufficient.
- * </para></listitem>
- * <listitem><para>
- * plugin system version, e.g. "0.10"
- * </para><para>
- * This is required so that when there is a GStreamer-0.12 or GStreamer-1.0
- * at some point in future, the different major versions can still co-exist
- * and use the same plugin install mechanism in the same way.
- * </para></listitem>
- * <listitem><para>
- * application identifier, e.g. "totem"
- * </para><para>
- * This may also be in the form of "pid/12345" if the program name can't
- * be obtained for some reason.
- * </para></listitem>
- * <listitem><para>
- * human-readable localised description of the required component,
- * e.g. "Vorbis audio decoder"
- * </para></listitem>
- * <listitem><para>
- * identifier string for the required component (see below for details about
- * how to map this to the package/plugin that needs installing), e.g.
- * <itemizedlist>
- * <listitem><para>
- * urisource-$(PROTOCOL_REQUIRED), e.g. urisource-http or urisource-mms
- * </para></listitem>
- * <listitem><para>
- * element-$(ELEMENT_REQUIRED), e.g. element-videoconvert
- * </para></listitem>
- * <listitem><para>
- * decoder-$(CAPS_REQUIRED), e.g. (do read below for more details!):
- * <itemizedlist>
- * <listitem><para>decoder-audio/x-vorbis</para></listitem>
- * <listitem><para>decoder-application/ogg</para></listitem>
- * <listitem><para>decoder-audio/mpeg, mpegversion=(int)4</para></listitem>
- * <listitem><para>decoder-video/mpeg, systemstream=(boolean)true, mpegversion=(int)2</para></listitem>
- </itemizedlist>
- * </para></listitem>
- * <listitem><para>
- * encoder-$(CAPS_REQUIRED), e.g. encoder-audio/x-vorbis
- * </para></listitem>
- * </itemizedlist>
- * </para></listitem>
- * <listitem><para>
- * optional further fields not yet specified
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * <para>
- * An entire ID string might then look like this, for example:
- * <literal>
- * gstreamer|0.10|totem|Vorbis audio decoder|decoder-audio/x-vorbis
- * </literal>
- * </para>
- * <para>
- * Plugin installers parsing this ID string should expect further fields also
- * separated by '|' symbols and either ignore them, warn the user, or error
- * out when encountering them.
- * </para>
- * <para>
- * Those unfamiliar with the GStreamer 'caps' system should note a few things
- * about the caps string used in the above decoder/encoder case:
- * <itemizedlist>
- * <listitem><para>
- * the first part ("video/mpeg") of the caps string is a GStreamer media
- * type and <emphasis>not</emphasis> a MIME type. Wherever possible, the
- * GStreamer media type will be the same as the corresponding MIME type,
- * but often it is not.
- * </para></listitem>
- * <listitem><para>
- * a caps string may or may not have additional comma-separated fields
- * of various types (as seen in the examples above)
- * </para></listitem>
- * <listitem><para>
- * the caps string of a 'required' component (as above) will always have
- * fields with fixed values, whereas an introspected string (see below)
- * may have fields with non-fixed values. Compare for example:
- * <itemizedlist>
- * <listitem><para>
- * <literal>audio/mpeg, mpegversion=(int)4</literal> vs.
- * <literal>audio/mpeg, mpegversion=(int){2, 4}</literal>
- * </para></listitem>
- * <listitem><para>
- * <literal>video/mpeg, mpegversion=(int)2</literal> vs.
- * <literal>video/mpeg, systemstream=(boolean){ true, false}, mpegversion=(int)[1, 2]</literal>
- * </para></listitem>
- * </itemizedlist>
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * <para>
- * <emphasis>4. Exit codes the installer should return</emphasis>
- * </para>
- * <para>
- * The installer should return one of the following exit codes when it exits:
- * <itemizedlist>
- * <listitem><para>
- * 0 if all of the requested plugins could be installed
+ * runtime by setting the GST_INSTALL_PLUGINS_HELPER environment
+ * variable. This can be useful for testing/debugging purposes.
+ *
+ * ## 2. Arguments passed to the install helper
+ *
+ * GStreamer will pass the following arguments to the install helper (this
+ * is in addition to the path of the executable itself, which is by
+ * convention argv[0]):
+ *
+ * - none to many optional arguments in the form of `--foo-bar=val`.
+ * Example: `--transient-for=XID` where XID is the X Window ID of the
+ * main window of the calling application (so the installer can make
+ * itself transient to that window). Unknown optional arguments should
+ * be ignored by the installer.
+ *
+ * - one 'installer detail string' argument for each plugin to be
+ * installed; these strings will have a `gstreamer` prefix; the exact
+ * format of the detail string is explained below
+ *
+ * ## 3. Detail string describing the missing plugin
+ *
+ * The string is in UTF-8 encoding and is made up of several fields,
+ * separated by '|' characters (but neither the first nor the last
+ * character is a '|'). The fields are:
+ *
+ * - plugin system identifier, ie. "gstreamer"
+ * This identifier determines the format of the rest of the detail
+ * string. Automatic plugin installers should not process detail
+ * strings with unknown identifiers. This allows other plugin-based
+ * libraries to use the same mechanism for their automatic plugin
+ * installation needs, or for the format to be changed should it turn
+ * out to be insufficient.
+ * - plugin system version, e.g. "0.10"
+ * This is required so that when there is a GStreamer-0.12 or
+ * GStreamer-1.0 at some point in future, the different major versions
+ * can still co-exist and use the same plugin install mechanism in the
+ * same way.
+ * - application identifier, e.g. "totem"
+ * This may also be in the form of "pid/12345" if the program name
+ * can't be obtained for some reason.
+ * - human-readable localised description of the required component, e.g.
+ * "Vorbis audio decoder"
+ * - identifier string for the required component (see below for details
+ * about how to map this to the package/plugin that needs installing),
+ * e.g.
+ * - urisource-$(PROTOCOL_REQUIRED), e.g. urisource-http or
+ * urisource-mms
+ * - element-$(ELEMENT_REQUIRED), e.g. element-videoconvert
+ * - decoder-$(CAPS_REQUIRED), e.g. (do read below for more
+ * details!):
+ * - decoder-audio/x-vorbis
+ * - decoder-application/ogg
+ * - decoder-audio/mpeg, mpegversion=(int)4
+ * - decoder-video/mpeg, systemstream=(boolean)true,
+ * mpegversion=(int)2
+ * - encoder-$(CAPS_REQUIRED), e.g. encoder-audio/x-vorbis
+ * - optional further fields not yet specified
+ *
+ * An entire ID string might then look like this, for example: `
+ * gstreamer|0.10|totem|Vorbis audio decoder|decoder-audio/x-vorbis`
+ *
+ * Plugin installers parsing this ID string should expect further fields
+ * also separated by '|' symbols and either ignore them, warn the user, or
+ * error out when encountering them.
+ *
+ * Those unfamiliar with the GStreamer 'caps' system should note a few
+ * things about the caps string used in the above decoder/encoder case:
+ *
+ * - the first part ("video/mpeg") of the caps string is a GStreamer
+ * media type and *not* a MIME type. Wherever possible, the GStreamer
+ * media type will be the same as the corresponding MIME type, but
+ * often it is not.
+ * - a caps string may or may not have additional comma-separated fields
+ * of various types (as seen in the examples above)
+ * - the caps string of a 'required' component (as above) will always
+ * have fields with fixed values, whereas an introspected string (see
+ * below) may have fields with non-fixed values. Compare for example:
+ * - `audio/mpeg, mpegversion=(int)4` vs.
+ * `audio/mpeg, mpegversion=(int){2, 4}`
+ * - `video/mpeg, mpegversion=(int)2` vs.
+ * `video/mpeg, systemstream=(boolean){ true, false}, mpegversion=(int)[1, 2]`
+ *
+ * ## 4. Exit codes the installer should return
+ *
+ * The installer should return one of the following exit codes when it
+ * exits:
+ *
+ * - 0 if all of the requested plugins could be installed
* (#GST_INSTALL_PLUGINS_SUCCESS)
- * </para></listitem>
- * <listitem><para>
- * 1 if no appropriate installation candidate for any of the requested
- * plugins could be found. Only return this if nothing has been installed
- * (#GST_INSTALL_PLUGINS_NOT_FOUND)
- * </para></listitem>
- * <listitem><para>
- * 2 if an error occured during the installation. The application will
+ * - 1 if no appropriate installation candidate for any of the requested
+ * plugins could be found. Only return this if nothing has been
+ * installed (#GST_INSTALL_PLUGINS_NOT_FOUND)
+ * - 2 if an error occured during the installation. The application will
* assume that the user will already have seen an error message by the
* installer in this case and will usually not show another one
* (#GST_INSTALL_PLUGINS_ERROR)
- * </para></listitem>
- * <listitem><para>
- * 3 if some of the requested plugins could be installed, but not all
+ * - 3 if some of the requested plugins could be installed, but not all
* (#GST_INSTALL_PLUGINS_PARTIAL_SUCCESS)
- * </para></listitem>
- * <listitem><para>
- * 4 if the user aborted the installation (#GST_INSTALL_PLUGINS_USER_ABORT)
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * <para>
- * <emphasis>5. How to map the required detail string to packages</emphasis>
- * </para>
- * <para>
+ * - 4 if the user aborted the installation
+ * (#GST_INSTALL_PLUGINS_USER_ABORT)
+ *
+ * ## 5. How to map the required detail string to packages
+ *
* It is up to the vendor to find mechanism to map required components from
* the detail string to the actual packages/plugins to install. This could
- * be a hardcoded list of mappings, for example, or be part of the packaging
- * system metadata.
- * </para>
- * <para>
+ * be a hardcoded list of mappings, for example, or be part of the
+ * packaging system metadata.
+ *
* GStreamer plugin files can be introspected for this information. The
- * <literal>gst-inspect</literal> utility has a special command line option
- * that will output information similar to what is required. For example
- * <command>
+ * `gst-inspect` utility has a special command line option that will output
+ * information similar to what is required. For example `
* $ gst-inspect-1.0 --print-plugin-auto-install-info /path/to/libgstvorbis.so
- * </command>
* should output something along the lines of
- * <computeroutput>
- * decoder-audio/x-vorbis
- * element-vorbisdec
- * element-vorbisenc
- * element-vorbisparse
- * element-vorbistag
- * encoder-audio/x-vorbis
- * </computeroutput>
- * Note that in the encoder and decoder case the introspected caps can be more
- * complex with additional fields, e.g.
- * <literal>audio/mpeg,mpegversion=(int){2,4}</literal>, so they will not
- * always exactly match the caps wanted by the application. It is up to the
- * installer to deal with this (either by doing proper caps intersection using
- * the GStreamer #GstCaps API, or by only taking into account the media type).
- * </para>
- * <para>
+ * `decoder-audio/x-vorbis`, `element-vorbisdec` `element-vorbisenc`
+ * `element-vorbisparse`, `element-vorbistag`, `encoder-audio/x-vorbis`
+ *
+ * Note that in the encoder and decoder case the introspected caps can be
+ * more complex with additional fields, e.g.
+ * `audio/mpeg,mpegversion=(int){2,4}`, so they will not always exactly
+ * match the caps wanted by the application. It is up to the installer to
+ * deal with this (either by doing proper caps intersection using the
+ * GStreamer #GstCaps API, or by only taking into account the media type).
+ *
* Another potential source of problems are plugins such as ladspa or
* libvisual where the list of elements depends on the installed
- * ladspa/libvisual plugins at the time. This is also up to the distribution
- * to handle (but usually not relevant for playback applications).
- * </para>
- * </refsect2>
+ * ladspa/libvisual plugins at the time. This is also up to the
+ * distribution to handle (but usually not relevant for playback
+ * applications).
*/
#ifdef HAVE_CONFIG_H
*
* GTK+/GNOME applications should be able to create a startup notification ID
* like this:
- * <programlisting>
+ * |[
* timestamp = gtk_get_current_event_time ();
* startup_id = g_strdup_printf ("_TIME%u", timestamp);
* ...
- * </programlisting>
+ * ]|
*
* Since: 1.6
*/
*
* Gtk+/Gnome application should be able to obtain the XID of the top-level
* window like this:
- * <programlisting>
+ * |[
* ##include <gtk/gtk.h>
* ##ifdef GDK_WINDOWING_X11
* ##include <gdk/gdkx.h>
* xid = GDK_WINDOW_XWINDOW (GTK_WIDGET (application_window)->window);
* ##endif
* ...
- * </programlisting>
+ * ]|
+ *
*/
void
gst_install_plugins_context_set_xid (GstInstallPluginsContext * ctx, guint xid)
* @ctx: (allow-none): a #GstInstallPluginsContext, or NULL
* @func: (scope async): the function to call when the installer program returns
* @user_data: (closure): the user data to pass to @func when called, or NULL
- *
+ *
* Requests plugin installation without blocking. Once the plugins have been
* installed or installation has failed, @func will be called with the result
* of the installation and your provided @user_data pointer.
* @details: (array zero-terminated=1) (transfer none): NULL-terminated array
* of installer string details
* @ctx: (allow-none): a #GstInstallPluginsContext, or NULL
- *
+ *
* Requests plugin installation and block until the plugins have been
* installed or installation has failed.
*
/**
* gst_install_plugins_return_get_name:
* @ret: the return status code
- *
+ *
* Convenience function to return the descriptive string associated
* with a status code. This function returns English strings and
* should not be used for user messages. It is here only to assist
/**
* gst_install_plugins_installation_in_progress:
- *
+ *
* Checks whether plugin installation (initiated by this application only)
* is currently in progress.
*
/**
* gst_install_plugins_supported:
- *
+ *
* Checks whether plugin installation is likely to be supported by the
* current environment. This currently only checks whether the helper script
* that is to be provided by the distribution or operating system vendor
/**
* SECTION:gstpbutilsmissingplugins
+ * @title: Missing plugins
* @short_description: Create, recognise and parse missing-plugins messages
*
- * <refsect2>
- * <para>
* Functions to create, recognise and parse missing-plugins messages for
* applications and elements.
- * </para>
- * <para>
+ *
* Missing-plugin messages are posted on the bus by elements like decodebin
* or playbin if they can't find an appropriate source element or decoder
* element. The application can use these messages for two things:
- * <itemizedlist>
- * <listitem><para>
- * concise error/problem reporting to the user mentioning what exactly
+ *
+ * * concise error/problem reporting to the user mentioning what exactly
* is missing, see gst_missing_plugin_message_get_description()
- * </para></listitem>
- * <listitem><para>
- * initiate installation of missing plugins, see
+ *
+ * * initiate installation of missing plugins, see
* gst_missing_plugin_message_get_installer_detail() and
* gst_install_plugins_async()
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * <para>
+ *
* Applications may also create missing-plugin messages themselves to install
* required elements that are missing, using the install mechanism mentioned
* above.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
* Returns an opaque string containing all the details about the missing
* element to be passed to an external installer called via
* gst_install_plugins_async() or gst_install_plugins_sync().
- *
+ *
* This function is mainly for applications that call external plugin
* installation mechanisms using one of the two above-mentioned functions.
*
* Returns an opaque string containing all the details about the missing
* element to be passed to an external installer called via
* gst_install_plugins_async() or gst_install_plugins_sync().
- *
+ *
* This function is mainly for applications that call external plugin
* installation mechanisms using one of the two above-mentioned functions in
* the case where the application knows exactly what kind of plugin it is
* Returns an opaque string containing all the details about the missing
* element to be passed to an external installer called via
* gst_install_plugins_async() or gst_install_plugins_sync().
- *
+ *
* This function is mainly for applications that call external plugin
* installation mechanisms using one of the two above-mentioned functions in
* the case where the application knows exactly what kind of plugin it is
* Returns an opaque string containing all the details about the missing
* element to be passed to an external installer called via
* gst_install_plugins_async() or gst_install_plugins_sync().
- *
+ *
* This function is mainly for applications that call external plugin
* installation mechanisms using one of the two above-mentioned functions in
* the case where the application knows exactly what kind of plugin it is
* Returns an opaque string containing all the details about the missing
* element to be passed to an external installer called via
* gst_install_plugins_async() or gst_install_plugins_sync().
- *
+ *
* This function is mainly for applications that call external plugin
* installation mechanisms using one of the two above-mentioned functions in
* the case where the application knows exactly what kind of plugin it is
* Returns an opaque string containing all the details about the missing
* element to be passed to an external installer called via
* gst_install_plugins_async() or gst_install_plugins_sync().
- *
+ *
* This function is mainly for applications that call external plugin
* installation mechanisms using one of the two above-mentioned functions in
* the case where the application knows exactly what kind of plugin it is
/**
* SECTION:gstpbutils
+ * @title: Pbutils
* @short_description: General Application and Plugin Utility Library
*
- * <refsect2>
- * <para>
* libgstpbutils is a general utility library for plugins and applications.
* It currently provides the
* following:
- * </para>
- * <itemizedlist>
- * <listitem>
- * <para>
- * human-readable description strings of codecs, elements, sources, decoders,
+ *
+ * * human-readable description strings of codecs, elements, sources, decoders,
* encoders, or sinks from decoder/encoder caps, element names, or protocol
* names.
- * </para>
- * </listitem>
- * <listitem>
- * <para>
- * support for applications to initiate installation of missing plugins (if
+ *
+ * * support for applications to initiate installation of missing plugins (if
* this is supported by the distribution or operating system used)
- * </para>
- * </listitem>
- * <listitem>
- * <para>
- * API for GStreamer elements to create missing-plugin messages in order to
+ *
+ * * API for GStreamer elements to create missing-plugin messages in order to
* communicate to the application that a certain type of plugin is missing
* (decoder, encoder, URI protocol source, URI protocol sink, named element)
- * </para>
- * </listitem>
- * <listitem>
- * <para>
- * API for applications to recognise and handle missing-plugin messages
- * </para>
- * </listitem>
- * </itemizedlist>
- * <title>Linking to this library</title>
- * <para>
+ *
+ * * API for applications to recognise and handle missing-plugin messages
+ *
+ * ## Linking to this library
+ *
* You should obtain the required CFLAGS and LIBS using pkg-config on the
* gstreamer-plugins-base-0.10 module. You will then also need to add
* '-lgstpbutils-0.10' manually to your LIBS line.
- * </para>
- * <title>Library initialisation</title>
- * <para>
+ *
+ * ## Library initialisation
+ *
* Before using any of its functions, applications and plugins must call
* gst_pb_utils_init() to initialise the library.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
* containing extradata for this particular stream (e.g.
* palette, codec initialization data).
*
- * Parses a video stream´s strf structure plus optionally some
+ * Parses a video stream's strf structure plus optionally some
* extradata from input data. This function takes ownership of @buf.
*
* Returns: TRUE if parsing succeeded, otherwise FALSE. The stream
* containing extradata for this particular stream (e.g.
* codec initialization data).
*
- * Parses an audio stream´s strf structure plus optionally some
+ * Parses an audio stream's strf structure plus optionally some
* extradata from input data. This function takes ownership of @buf.
* use.
*
*/
/**
* SECTION:gstriff
+ * @title: Riff utilities
* @short_description: Riff fileformat utillity functions.
*
* A collection of functions to handle riff base files, such as avi, wav and
/**
* SECTION:gstrtcpbuffer
+ * @title: GstRTCPBuffer
* @short_description: Helper methods for dealing with RTCP buffers
* @see_also: #GstRTPBasePayload, #GstRTPBaseDepayload, #gstrtpbuffer
*
* Note: The API in this module is not yet declared stable.
*
- * <refsect2>
- * <para>
- * The GstRTPCBuffer helper functions makes it easy to parse and create regular
+ * The GstRTPCBuffer helper functions makes it easy to parse and create regular
* #GstBuffer objects that contain compound RTCP packets. These buffers are typically
* of 'application/x-rtcp' #GstCaps.
- * </para>
- * <para>
+ *
* An RTCP buffer consists of 1 or more #GstRTCPPacket structures that you can
* retrieve with gst_rtcp_buffer_get_first_packet(). #GstRTCPPacket acts as a pointer
* into the RTCP buffer; you can move to the next packet with
* gst_rtcp_packet_move_to_next().
- * </para>
- * </refsect2>
+ *
*/
#include <string.h>
* @type: the #GstRTCPType of the new packet
* @packet: pointer to new packet
*
- * Add a new packet of @type to @rtcp. @packet will point to the newly created
+ * Add a new packet of @type to @rtcp. @packet will point to the newly created
* packet.
*
* Returns: %TRUE if the packet could be created. This function returns %FALSE
* gst_rtcp_packet_get_length:
* @packet: a valid #GstRTCPPacket
*
- * Get the length field of @packet. This is the length of the packet in
+ * Get the length field of @packet. This is the length of the packet in
* 32-bit words minus one.
*
* Returns: The length field of @packet.
/**
* gst_rtcp_packet_sr_set_sender_info:
* @packet: a valid SR #GstRTCPPacket
- * @ssrc: the SSRC
+ * @ssrc: the SSRC
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
/**
* SECTION:gstrtpbaseaudiopayload
+ * @title: GstRTPBaseAudioPayload
* @short_description: Base class for audio RTP payloader
*
* Provides a base class for audio RTP payloaders for frame or sample based
* sent in a last RTP packet. In the case of frame based codecs, the resulting
* RTP packets always contain full frames.
*
- * <refsect2>
- * <title>Usage</title>
- * <para>
+ * ## Usage
+ *
* To use this base class, your child element needs to call either
* gst_rtp_base_audio_payload_set_frame_based() or
* gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the
* must set any variables or call/override any functions required by that base
* class. The child element does not need to override any other functions
* specific to GstRTPBaseAudioPayload.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstrtpbasedepayload
+ * @title: GstRTPBaseDepayload
* @short_description: Base class for RTP depayloader
*
* Provides a base class for RTP depayloaders
* application/x-rtp-depayload-stats containing the following fields relating to
* the last processed buffer and current state of the stream being depayloaded:
*
- * <variablelist>
- * <varlistentry>
- * <term>clock-rate</term>
- * <listitem><para>#G_TYPE_UINT, clock-rate of the
- * stream</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>npt-start</term>
- * <listitem><para>#G_TYPE_UINT64, time of playback start
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>npt-stop</term>
- * <listitem><para>#G_TYPE_UINT64, time of playback stop
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>play-speed</term>
- * <listitem><para>#G_TYPE_DOUBLE, the playback speed
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>play-scale</term>
- * <listitem><para>#G_TYPE_DOUBLE, the playback scale
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>running-time-dts</term>
- * <listitem><para>#G_TYPE_UINT64, the last running-time of the
+ * * `clock-rate`: #G_TYPE_UINT, clock-rate of the stream
+ * * `npt-start`: #G_TYPE_UINT64, time of playback start
+ * * `npt-stop`: #G_TYPE_UINT64, time of playback stop
+ * * `play-speed`: #G_TYPE_DOUBLE, the playback speed
+ * * `play-scale`: #G_TYPE_DOUBLE, the playback scale
+ * * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the
* last DTS
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>running-time-pts</term>
- * <listitem><para>#G_TYPE_UINT64, the last running-time of the
+ * * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the
* last PTS
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>seqnum</term>
- * <listitem><para>#G_TYPE_UINT, the last seen seqnum
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>timestamp</term>
- * <listitem><para>#G_TYPE_UINT, the last seen RTP timestamp
- * </para></listitem>
- * </varlistentry>
- * </variablelist>
+ * * `seqnum`: #G_TYPE_UINT, the last seen seqnum
+ * * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp
**/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS,
g_param_spec_boxed ("stats", "Statistics", "Various statistics",
/**
* SECTION:gstrtpbasepayload
+ * @title: GstRTPBasePayload
* @short_description: Base class for RTP payloader
*
* Provides a base class for RTP payloaders
* application/x-rtp-payload-stats containing the following fields relating to
* the last processed buffer and current state of the stream being payloaded:
*
- * <variablelist>
- * <varlistentry>
- * <term>clock-rate</term>
- * <listitem><para>#G_TYPE_UINT, clock-rate of the
- * stream</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>running-time</term>
- * <listitem><para>#G_TYPE_UINT64, running time
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>seqnum</term>
- * <listitem><para>#G_TYPE_UINT, sequence number, same as
- * #GstRTPBasePayload:seqnum</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>timestamp</term>
- * <listitem><para>#G_TYPE_UINT, RTP timestamp, same as
- * #GstRTPBasePayload:timestamp</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>ssrc</term>
- * <listitem><para>#G_TYPE_UINT, The SSRC in use
- * </para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>pt</term>
- * <listitem><para>#G_TYPE_UINT, The Payload type in use, same as
- * #GstRTPBasePayload:pt</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>seqnum-offset</term>
- * <listitem><para>#G_TYPE_UINT, The current offset added to the
- * seqnum</para></listitem>
- * </varlistentry>
- * <varlistentry>
- * <term>timestamp-offset</term>
- * <listitem><para>#G_TYPE_UINT, The current offset added to the
- * timestamp</para></listitem>
- * </varlistentry>
- * </variablelist>
+ * * `clock-rate` :#G_TYPE_UINT, clock-rate of the stream
+ * * `running-time` :#G_TYPE_UINT64, running time
+ * * `seqnum` :#G_TYPE_UINT, sequence number, same as #GstRTPBasePayload:seqnum
+ * * `timestamp` :#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestamp
+ * * `ssrc` :#G_TYPE_UINT, The SSRC in use
+ * * `pt` :#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload:pt
+ * * `seqnum-offset` :#G_TYPE_UINT, The current offset added to the seqnum
+ * * `timestamp-offset` :#G_TYPE_UINT, The current offset added to the timestamp
**/
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_STATS,
g_param_spec_boxed ("stats", "Statistics", "Various statistics",
/**
* SECTION:gstrtpbuffer
+ * @title: GstRTPBuffer
* @short_description: Helper methods for dealing with RTP buffers
* @see_also: #GstRTPBasePayload, #GstRTPBaseDepayload, gstrtcpbuffer
*
- * <refsect2>
- * <para>
- * The GstRTPBuffer helper functions makes it easy to parse and create regular
+ * The GstRTPBuffer helper functions makes it easy to parse and create regular
* #GstBuffer objects that contain RTP payloads. These buffers are typically of
* 'application/x-rtp' #GstCaps.
- * </para>
- * </refsect2>
+ *
*/
#include "gstrtpbuffer.h"
* @rtp: the RTP packet
*
* Check if the extension bit is set on the RTP packet in @buffer.
- *
+ *
* Returns: TRUE if @buffer has the extension bit set.
*/
gboolean
*
* If @buffer did not contain an extension, this function will return %FALSE
* with @bits, @data and @wordlen unchanged.
- *
+ *
* Returns: TRUE if @buffer had the extension bit set.
*/
gboolean
* @rtp: the RTP packet
*
* Get the SSRC of the RTP packet in @buffer.
- *
+ *
* Returns: the SSRC of @buffer in host order.
*/
guint32
* @rtp: the RTP packet
*
* Get the CSRC count of the RTP packet in @buffer.
- *
+ *
* Returns: the CSRC count of @buffer.
*/
guint8
* @idx: the index of the CSRC to get
*
* Get the CSRC at index @idx in @buffer.
- *
+ *
* Returns: the CSRC at index @idx in host order.
*/
guint32
/**
* SECTION:gstrtphdrext
+ * @title: GstRtphdrext
* @short_description: Helper methods for dealing with RTP header extensions
* @see_also: #GstRTPBasePayload, #GstRTPBaseDepayload, gstrtpbuffer
*
- * <refsect2>
- * <para>
- * </para>
- * </refsect2>
*/
#include "gstrtphdrext.h"
/**
* SECTION:gstrtppayloads
+ * @title: GstRTPPayloadInfo
* @short_description: Helper methods for dealing with RTP payloads
* @see_also: gstrtpbuffer
*
- * <refsect2>
- * <para>
* The GstRTPPayloads helper functions makes it easy to deal with static and dynamic
- * payloads. Its main purpose is to retrieve properties such as the default clock-rate
+ * payloads. Its main purpose is to retrieve properties such as the default clock-rate
* and get session bandwidth information.
- * </para>
- * </refsect2>
+ *
*/
#include <string.h>
* @GST_RTP_PAYLOAD_MP2T: MPEG-2 transport stream (RFC 2250)
* @GST_RTP_PAYLOAD_H263: Video H263 (RFC 2190)
*
- *
* Standard predefined fixed payload types.
*
* The official list is at:
/**
* SECTION:gstrtspconnection
+ * @title: GstRTSPConnection
* @short_description: manage RTSP connections
* @see_also: gstrtspurl
*
/**
* SECTION:gstrtspdefs
+ * @title: GstRtspdefs
* @short_description: common RTSP defines
* @see_also: gstrtspurl, gstrtspconnection
- *
- * Provides common defines for the RTSP library.
+ *
+ * Provides common defines for the RTSP library.
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstrtspextension
+ * @title: GstRTSPExtension
* @short_description: Interface for extending RTSP protocols
*
- * <refsect2>
- * <para>
* This interface is implemented e.g. by the Windows Media Streaming RTSP
* exentension (rtspwms) and the RealMedia RTSP extension (rtspreal).
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstrtspmessage
+ * @title: GstRTSPMessage
* @short_description: RTSP messages
* @see_also: gstrtspconnection
- *
+ *
* Provides methods for creating and parsing request, response and data messages.
*/
* @msg: a #GstRTSPMessage
*
* Unset the contents of @msg so that it becomes an uninitialized
- * #GstRTSPMessage again. This function is mostly used in combination with
+ * #GstRTSPMessage again. This function is mostly used in combination with
* gst_rtsp_message_init_request(), gst_rtsp_message_init_response() and
* gst_rtsp_message_init_data() on stack allocated #GstRTSPMessage structures.
*
/**
* SECTION:gstrtsprange
+ * @title: GstRTSPTimeRange
* @short_description: dealing with time ranges
- *
+ *
* Provides helper functions to deal with time ranges.
*/
/**
* SECTION:gstrtsptransport
+ * @title: GstRTSPRange
* @short_description: dealing with RTSP transports
- *
+ *
* Provides helper functions to deal with RTSP transport strings.
*/
* Allocate a new initialized #GstRTSPTransport. Use gst_rtsp_transport_free()
* after usage.
*
- * Returns: a #GstRTSPResult.
+ * Returns: a #GstRTSPResult.
*/
GstRTSPResult
gst_rtsp_transport_new (GstRTSPTransport ** transport)
*
* Initialize @transport so that it can be used.
*
- * Returns: #GST_RTSP_OK.
+ * Returns: #GST_RTSP_OK.
*/
GstRTSPResult
gst_rtsp_transport_init (GstRTSPTransport * transport)
* @manager will contain an element name or #NULL when no manager is
* needed/available for @trans.
*
- * Returns: #GST_RTSP_OK.
+ * Returns: #GST_RTSP_OK.
*/
GstRTSPResult
gst_rtsp_transport_get_manager (GstRTSPTransMode trans, const gchar ** manager,
/**
* SECTION:gstrtspurl
+ * @title: GstRTSPUrl
* @short_description: handling RTSP urls
- *
+ *
* Provides helper functions to handle RTSP urls.
*/
* gst_rtsp_url_get_request_uri:
* @url: a #GstRTSPUrl
*
- * Get a newly allocated string describing the request URI for @url.
+ * Get a newly allocated string describing the request URI for @url.
*
* Returns: a string with the request URI. g_free() after usage.
*/
/**
* SECTION:gstmikey
+ * @title: GstMIKEYMessage
* @short_description: Helper methods for dealing with MIKEY messages
*
- * <refsect2>
- * <para>
* The GstMIKEY helper functions makes it easy to parse and create MIKEY
* messages.
- * </para>
- * </refsect2>
*
* Since: 1.4
*/
/**
* SECTION:gstsdpmessage
+ * @title: GstSDPMessage
* @short_description: Helper methods for dealing with SDP messages
*
- * <refsect2>
- * <para>
* The GstSDPMessage helper functions makes it easy to parse and create SDP
* messages.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gsttagexif
+ * @title: GstExiftag
* @short_description: tag mappings and support functions for plugins
* dealing with exif tags
* @see_also: #GstTagList
/**
* SECTION:gsttagid3
+ * @title: ID3 tag utils
* @short_description: tag mappings and support functions for plugins
* dealing with ID3v1 and ID3v2 tags
* @see_also: #GstTagList
- *
- * <refsect2>
- * <para>
+ *
* Contains various utility functions for plugins to parse or create
* ID3 tags and map ID3v2 identifiers to and from GStreamer identifiers.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* gst_tag_id3_genre_count:
*
- * Gets the number of ID3v1 genres that can be identified. Winamp genres are
+ * Gets the number of ID3v1 genres that can be identified. Winamp genres are
* included.
*
* Returns: the number of ID3v1 genres that can be identified
/**
* SECTION:gsttagdemux
+ * @title: GstTagDemux
* @see_also: GstApeDemux, GstID3Demux
* @short_description: Base class for demuxing tags that are in chunks
* directly at the beginning or at the end of a file
- *
- * <refsect2>
- * <para>
+ *
* Provides a base class for demuxing tags at the beginning or end of a
* stream and handles things like typefinding, querying, seeking, and
* different modes of operation (chain-based, pull_range-based, and providing
* there was no tag at all. Also, once the tag has been parsed, GstTagDemux
* will try to determine the media type of the resulting stream and add a
* source pad with the appropriate caps in order to facilitate auto-plugging.
- * </para>
- * <title>Deriving from GstTagDemux</title>
- * <para>
+ *
+ * ## Deriving from GstTagDemux
+ *
* Subclasses have to do four things:
- * <itemizedlist>
- * <listitem><para>
- * In their base init function, they must add a pad template for the sink
- * pad to the element class, describing the media type they can parse in
- * the caps of the pad template.
- * </para></listitem>
- * <listitem><para>
- * In their class init function, they must override
- * GST_TAG_DEMUX_CLASS(demux_klass)->identify_tag with their own identify
- * function.
- * </para></listitem>
- * <listitem><para>
- * In their class init function, they must override
+ *
+ * * In their base init function, they must add a pad template for the sink
+ * pad to the element class, describing the media type they can parse in
+ * the caps of the pad template.
+ * * In their class init function, they must override
+ * GST_TAG_DEMUX_CLASS(demux_klass)->identify_tag with their own identify
+ * function.
+ * * In their class init function, they must override
* GST_TAG_DEMUX_CLASS(demux_klass)->parse_tag with their own parse
* function.
- * </para></listitem>
- * <listitem><para>
- * In their class init function, they must also set
- * GST_TAG_DEMUX_CLASS(demux_klass)->min_start_size and/or
+ * * In their class init function, they must also set
+ * GST_TAG_DEMUX_CLASS(demux_klass)->min_start_size and/or
* GST_TAG_DEMUX_CLASS(demux_klass)->min_end_size to the minimum size required
* for the identify function to decide whether the stream has a supported tag
* or not. A class parsing ID3v1 tags, for example, would set min_end_size to
* 128 bytes.
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * </refsect2>
*/
#ifdef HAVE_CONFIG_H
GList *pending_events;
};
-/* Require at least 8kB of data before we attempt typefind.
+/* Require at least 8kB of data before we attempt typefind.
* Seems a decent value based on test files
- * 40kB is massive overkill for the maximum, I think, but it
+ * 40kB is massive overkill for the maximum, I think, but it
* doesn't do any harm (tpm: increased to 64kB after watching
* typefinding fail on a wavpack file that needed 42kB to succeed) */
#define TYPE_FIND_MIN_SIZE 8192
g_assert (gst_buffer_is_writable (collect));
- /* If we receive a buffer that's from the middle of the file,
+ /* If we receive a buffer that's from the middle of the file,
* we can't read tags so move to typefinding */
if (GST_BUFFER_OFFSET_IS_VALID (collect) && GST_BUFFER_OFFSET (collect) != 0) {
GST_DEBUG_OBJECT (demux, "Received buffer from non-zero offset %"
/* 1: */
/* If we can activate pull_range upstream, then read any end and start
- * tags, otherwise activate in push mode and the chain function will
+ * tags, otherwise activate in push mode and the chain function will
* collect buffers, read the start tag and output a buffer to end
* preroll.
*/
if (ret != GST_FLOW_OK)
return ret;
- /* Adjust offset and length of the request to trim off tag information.
+ /* Adjust offset and length of the request to trim off tag information.
* For the returned buffer, adjust the output offset to match what downstream
* should see */
in_offset = offset + demux->priv->strip_start;
/**
* SECTION:gsttagmux
+ * @title: GstTagMux
* @see_also: GstApeMux, GstId3Mux
* @short_description: Base class for adding tags that are in one single chunk
* directly at the beginning or at the end of a file
*
- * <refsect2>
- * <para>
* Provides a base class for adding tags at the beginning or end of a
* stream.
- * </para>
- * <title>Deriving from GstTagMux</title>
- * <para>
+ *
+ * ## Deriving from GstTagMux
+ *
* Subclasses have to do the following things:
- * <itemizedlist>
- * <listitem><para>
- * In their base init function, they must add pad templates for the sink
- * pad and the source pad to the element class, describing the media type
- * they accept and output in the caps of the pad template.
- * </para></listitem>
- * <listitem><para>
- * In their class init function, they must override the
- * GST_TAG_MUX_CLASS(mux_klass)->render_start_tag and/or
- * GST_TAG_MUX_CLASS(mux_klass)->render_end_tag vfuncs and set up a render
- * function.
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * </refsect2>
+ *
+ * * In their base init function, they must add pad templates for the sink
+ * pad and the source pad to the element class, describing the media type
+ * they accept and output in the caps of the pad template.
+ * * In their class init function, they must override the
+ * GST_TAG_MUX_CLASS(mux_klass)->render_start_tag and/or
+ * GST_TAG_MUX_CLASS(mux_klass)->render_end_tag vfuncs and set up a render
+ * function.
+ *
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
/**
* SECTION:gsttagvorbis
+ * @title: GstVorbisTag
* @short_description: tag mappings and support functions for plugins
* dealing with vorbiscomments
* @see_also: #GstTagList
*
- * <refsect2>
- * <para>
* Contains various utility functions for plugins to parse or create
* vorbiscomments and map them to and from #GstTagList<!-- -->s.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gsttagxmp
+ * @title: GstXmptag
* @short_description: tag mappings and support functions for plugins
* dealing with xmp packets
* @see_also: #GstTagList
/**
* SECTION:gsttaglanguagecodes
+ * @title: ISO-639 lang mappings
* @short_description: mappings for ISO-639 language codes and names
* @see_also: #GstTagList
*
- * <refsect2>
- * <para>
* Provides helper functions to convert between the various ISO-639 language
* codes, and to map language codes to language names.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gsttaglicenses
+ * @title: Licenses
* @short_description: utility functions for Creative Commons licenses
* @see_also: #GstTagList
*
/**
* SECTION:gsttag
+ * @title: Tags
* @short_description: additional tag definitions for plugins and applications
* @see_also: #GstTagList
- *
- * <refsect2>
- * <para>
+ *
* Contains additional standardized GStreamer tag definitions for plugins
* and applications, and functions to register them with the GStreamer
* tag system.
- * </para>
- * </refsect2>
+ *
*/
#ifndef GST_DISABLE_GST_DEBUG
/**
* SECTION:gsttagxmpwriter
+ * @title: GstTagXmpWriter
* @short_description: Interface for elements that provide XMP serialization
*
- * <refsect2>
- * <para>
* This interface is implemented by elements that are able to do XMP serialization. Examples for
* such elements are #jifmux and #qtmux.
- * </para>
- * <para>
+ *
* Applications can use this interface to configure which XMP schemas should be used when serializing
* tags into XMP. Schemas are represented by their names, a full list of the supported schemas can be
* obtained from gst_tag_xmp_list_schemas(). By default, all schemas are used.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:gstcolorbalance
+ * @title: GstColorBalance
* @short_description: Interface for adjusting color balance settings
*
- * <refsect2><para>
* This interface is implemented by elements which can perform some color
* balance operation on video frames they process. For example, modifying
* the brightness, contrast, hue or saturation.
- * </para><para>
+ *
* Example elements are 'xvimagesink' and 'colorbalance'
- * </para>
- * </refsect2>
+ *
*/
/* FIXME 0.11: check if we need to add API for sometimes-supportedness
*
* Sets the current value of the channel to the passed value, which must
* be between min_value and max_value.
- *
+ *
* See Also: The #GstColorBalanceChannel.min_value and
* #GstColorBalanceChannel.max_value members of the
* #GstColorBalanceChannel object.
*
* Retrieve the current value of the indicated channel, between min_value
* and max_value.
- *
+ *
* See Also: The #GstColorBalanceChannel.min_value and
* #GstColorBalanceChannel.max_value members of the
* #GstColorBalanceChannel object.
- *
+ *
* Returns: The current value of the channel.
*/
gint
/**
* SECTION:gstcolorbalancechannel
+ * @title: GstColorBalanceChannel
* @short_description: Object representing a channel from the #GstColorBalance
* interface.
*
- * <refsect2><para>The #GstColorBalanceChannel object represents a parameter
+ * The #GstColorBalanceChannel object represents a parameter
* for modifying the color balance implemented by an element providing the
* #GstColorBalance interface. For example, Hue or Saturation.
- * </para></refsect2>
+ *
*/
enum
}
/**
- * gst_buffer_add_video_affine_transformation_meta
+ * gst_buffer_add_video_affine_transformation_meta:
* @buffer: a #GstBuffer
*
* Attaches GstVideoAffineTransformationMeta metadata to @buffer with
/**
* SECTION:gstvideodecoder
+ * @title: GstVideoDecoder
* @short_description: Base class for video decoders
* @see_also:
*
*
* The GstVideoDecoder base class and derived subclasses should cooperate as
* follows:
- * <orderedlist>
- * <listitem>
- * <itemizedlist><title>Configuration</title>
- * <listitem><para>
- * Initially, GstVideoDecoder calls @start when the decoder element
+ *
+ * ## Configuration
+ *
+ * * Initially, GstVideoDecoder calls @start when the decoder element
* is activated, which allows the subclass to perform any global setup.
- * </para></listitem>
- * <listitem><para>
- * GstVideoDecoder calls @set_format to inform the subclass of caps
+ *
+ * * GstVideoDecoder calls @set_format to inform the subclass of caps
* describing input video data that it is about to receive, including
* possibly configuration data.
* While unlikely, it might be called more than once, if changing input
* parameters require reconfiguration.
- * </para></listitem>
- * <listitem><para>
- * Incoming data buffers are processed as needed, described in Data
+ *
+ * * Incoming data buffers are processed as needed, described in Data
* Processing below.
- * </para></listitem>
- * <listitem><para>
- * GstVideoDecoder calls @stop at end of all processing.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist>
- * <title>Data processing</title>
- * <listitem><para>
- * The base class gathers input data, and optionally allows subclass
+ *
+ * * GstVideoDecoder calls @stop at end of all processing.
+ *
+ * ## Data processing
+ *
+ * * The base class gathers input data, and optionally allows subclass
* to parse this into subsequently manageable chunks, typically
* corresponding to and referred to as 'frames'.
- * </para></listitem>
- * <listitem><para>
- * Each input frame is provided in turn to the subclass' @handle_frame
+ *
+ * * Each input frame is provided in turn to the subclass' @handle_frame
* callback.
* The ownership of the frame is given to the @handle_frame callback.
- * </para></listitem>
- * <listitem><para>
- * If codec processing results in decoded data, the subclass should call
+ *
+ * * If codec processing results in decoded data, the subclass should call
* @gst_video_decoder_finish_frame to have decoded data pushed.
* downstream. Otherwise, the subclass must call
* @gst_video_decoder_drop_frame, to allow the base class to do timestamp
* and offset tracking, and possibly to requeue the frame for a later
* attempt in the case of reverse playback.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist><title>Shutdown phase</title>
- * <listitem><para>
- * The GstVideoDecoder class calls @stop to inform the subclass that data
+ *
+ * ## Shutdown phase
+ *
+ * * The GstVideoDecoder class calls @stop to inform the subclass that data
* parsing will be stopped.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist><title>Additional Notes</title>
- * <listitem>
- * <itemizedlist><title>Seeking/Flushing</title>
- * <listitem><para>
- * When the pipeline is seeked or otherwise flushed, the subclass is
- * informed via a call to its @reset callback, with the hard parameter
- * set to true. This indicates the subclass should drop any internal data
- * queues and timestamps and prepare for a fresh set of buffers to arrive
- * for parsing and decoding.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist><title>End Of Stream</title>
- * <listitem><para>
- * At end-of-stream, the subclass @parse function may be called some final
- * times with the at_eos parameter set to true, indicating that the element
- * should not expect any more data to be arriving, and it should parse and
- * remaining frames and call gst_video_decoder_have_frame() if possible.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * </itemizedlist>
- * </listitem>
- * </orderedlist>
+ *
+ * ## Additional Notes
+ *
+ * * Seeking/Flushing
+ *
+ * * When the pipeline is seeked or otherwise flushed, the subclass is
+ * informed via a call to its @reset callback, with the hard parameter
+ * set to true. This indicates the subclass should drop any internal data
+ * queues and timestamps and prepare for a fresh set of buffers to arrive
+ * for parsing and decoding.
+ *
+ * * End Of Stream
+ *
+ * * At end-of-stream, the subclass @parse function may be called some final
+ * times with the at_eos parameter set to true, indicating that the element
+ * should not expect any more data to be arriving, and it should parse and
+ * remaining frames and call gst_video_decoder_have_frame() if possible.
*
* The subclass is responsible for providing pad template caps for
* source and sink pads. The pads need to be named "sink" and "src". It also
* incoming data.
*
* The bare minimum that a functional subclass needs to implement is:
- * <itemizedlist>
- * <listitem><para>Provide pad templates</para></listitem>
- * <listitem><para>
- * Inform the base class of output caps via
+ *
+ * * Provide pad templates
+ * * Inform the base class of output caps via
* @gst_video_decoder_set_output_state
- * </para></listitem>
- * <listitem><para>
- * Parse input data, if it is not considered packetized from upstream
+ *
+ * * Parse input data, if it is not considered packetized from upstream
* Data will be provided to @parse which should invoke
* @gst_video_decoder_add_to_frame and @gst_video_decoder_have_frame to
* separate the data belonging to each video frame.
- * </para></listitem>
- * <listitem><para>
- * Accept data in @handle_frame and provide decoded results to
+ *
+ * * Accept data in @handle_frame and provide decoded results to
* @gst_video_decoder_finish_frame, or call @gst_video_decoder_drop_frame.
- * </para></listitem>
- * </itemizedlist>
*/
#ifdef HAVE_CONFIG_H
}
/* Pass the frame in priv->current_frame through the
- * handle_frame() callback for decoding and passing to gvd_finish_frame(),
+ * handle_frame() callback for decoding and passing to gvd_finish_frame(),
* or dropping by passing to gvd_drop_frame() */
static GstFlowReturn
gst_video_decoder_decode_frame (GstVideoDecoder * decoder,
decoder_class = GST_VIDEO_DECODER_GET_CLASS (decoder);
- /* FIXME : This should only have to be checked once (either the subclass has an
+ /* FIXME : This should only have to be checked once (either the subclass has an
* implementation, or it doesn't) */
g_return_val_if_fail (decoder_class->handle_frame != NULL, GST_FLOW_ERROR);
* @frame_number: system_frame_number of a frame
*
* Get a pending unfinished #GstVideoCodecFrame
- *
+ *
* Returns: (transfer full): pending unfinished #GstVideoCodecFrame identified by @frame_number.
*/
GstVideoCodecFrame *
* @decoder: a #GstVideoDecoder
*
* Get all pending unfinished #GstVideoCodecFrame
- *
+ *
* Returns: (transfer full) (element-type GstVideoCodecFrame): pending unfinished #GstVideoCodecFrame.
*/
GList *
/**
* SECTION:gstvideoencoder
+ * @title: GstVideoEncoder
* @short_description: Base class for video encoders
* @see_also:
*
* encoded video data.
*
* GstVideoEncoder and subclass should cooperate as follows.
- * <orderedlist>
- * <listitem>
- * <itemizedlist><title>Configuration</title>
- * <listitem><para>
- * Initially, GstVideoEncoder calls @start when the encoder element
+ *
+ * ## Configuration
+ *
+ * * Initially, GstVideoEncoder calls @start when the encoder element
* is activated, which allows subclass to perform any global setup.
- * </para></listitem>
- * <listitem><para>
- * GstVideoEncoder calls @set_format to inform subclass of the format
+ * * GstVideoEncoder calls @set_format to inform subclass of the format
* of input video data that it is about to receive. Subclass should
* setup for encoding and configure base class as appropriate
* (e.g. latency). While unlikely, it might be called more than once,
* if changing input parameters require reconfiguration. Baseclass
* will ensure that processing of current configuration is finished.
- * </para></listitem>
- * <listitem><para>
- * GstVideoEncoder calls @stop at end of all processing.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist>
- * <title>Data processing</title>
- * <listitem><para>
- * Base class collects input data and metadata into a frame and hands
+ * * GstVideoEncoder calls @stop at end of all processing.
+ *
+ * ## Data processing
+ *
+ * * Base class collects input data and metadata into a frame and hands
* this to subclass' @handle_frame.
- * </para></listitem>
- * <listitem><para>
- * If codec processing results in encoded data, subclass should call
+ *
+ * * If codec processing results in encoded data, subclass should call
* @gst_video_encoder_finish_frame to have encoded data pushed
* downstream.
- * </para></listitem>
- * <listitem><para>
- * If implemented, baseclass calls subclass @pre_push just prior to
+ *
+ * * If implemented, baseclass calls subclass @pre_push just prior to
* pushing to allow subclasses to modify some metadata on the buffer.
* If it returns GST_FLOW_OK, the buffer is pushed downstream.
- * </para></listitem>
- * <listitem><para>
- * GstVideoEncoderClass will handle both srcpad and sinkpad events.
+ *
+ * * GstVideoEncoderClass will handle both srcpad and sinkpad events.
* Sink events will be passed to subclass if @event callback has been
* provided.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * <listitem>
- * <itemizedlist><title>Shutdown phase</title>
- * <listitem><para>
- * GstVideoEncoder class calls @stop to inform the subclass that data
+ *
+ * ## Shutdown phase
+ *
+ * * GstVideoEncoder class calls @stop to inform the subclass that data
* parsing will be stopped.
- * </para></listitem>
- * </itemizedlist>
- * </listitem>
- * </orderedlist>
*
* Subclass is responsible for providing pad template caps for
* source and sink pads. The pads need to be named "sink" and "src". It should
* @gst_video_encoder_finish_frame.
*
* Things that subclass need to take care of:
- * <itemizedlist>
- * <listitem><para>Provide pad templates</para></listitem>
- * <listitem><para>
- * Provide source pad caps before pushing the first buffer
- * </para></listitem>
- * <listitem><para>
- * Accept data in @handle_frame and provide encoded results to
+ *
+ * * Provide pad templates
+ * * Provide source pad caps before pushing the first buffer
+ * * Accept data in @handle_frame and provide encoded results to
* @gst_video_encoder_finish_frame.
- * </para></listitem>
- * </itemizedlist>
*
*/
/**
* gst_video_encoder_finish_frame:
* @encoder: a #GstVideoEncoder
- * @frame: (transfer full): an encoded #GstVideoCodecFrame
+ * @frame: (transfer full): an encoded #GstVideoCodecFrame
*
* @frame must have a valid encoded data buffer, whose metadata fields
* are then appropriately set according to frame data or no buffer at
* @frame_number: system_frame_number of a frame
*
* Get a pending unfinished #GstVideoCodecFrame
- *
+ *
* Returns: (transfer full): pending unfinished #GstVideoCodecFrame identified by @frame_number.
*/
GstVideoCodecFrame *
* @encoder: a #GstVideoEncoder
*
* Get all pending unfinished #GstVideoCodecFrame
- *
+ *
* Returns: (transfer full) (element-type GstVideoCodecFrame): pending unfinished #GstVideoCodecFrame.
*/
GList *
/**
* SECTION:gstvideofilter
+ * @title: GstVideoFilter
* @short_description: Base class for video filters
- *
- * <refsect2>
- * <para>
+ *
* Provides useful functions and a base class for video filters.
- * </para>
- * <para>
+ *
* The videofilter will by default enable QoS on the parent GstBaseTransform
* to implement frame dropping.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
* @GST_VIDEO_GL_TEXTURE_ORIENTATION_X_NORMAL_Y_FLIP: Bottom line first in memory, left row first
* @GST_VIDEO_GL_TEXTURE_ORIENTATION_X_FLIP_Y_NORMAL: Top line first in memory, right row first
* @GST_VIDEO_GL_TEXTURE_ORIENTATION_X_FLIP_Y_FLIP: Bottom line first in memory, right row first
- *
+ *
* The orientation of the GL texture.
*/
typedef enum
/**
* SECTION:gstvideopool
+ * @title: GstVideoBufferPool
* @short_description: GstBufferPool for raw video buffers
* @see_also: #GstBufferPool
*
/**
* SECTION:gstvideosink
+ * @title: GstVideoSink
* @short_description: Base class for video sinks
- *
- * <refsect2>
- * <para>
- * Provides useful functions and a base class for video sinks.
- * </para>
- * <para>
+ *
+ * Provides useful functions and a base class for video sinks.
+ *
* GstVideoSink will configure the default base sink to drop frames that
* arrive later than 20ms as this is considered the default threshold for
* observing out-of-sync frames.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
* @dst: the #GstVideoRectangle describing the destination area
* @result: a pointer to a #GstVideoRectangle which will receive the result area
* @scaling: a #gboolean indicating if scaling should be applied or not
- *
+ *
* Takes @src rectangle and position it at the center of @dst rectangle with or
* without @scaling. It handles clipping if the @src rectangle is bigger than
* the @dst one and @scaling is set to FALSE.
* @parent_class: the parent class structure
* @show_frame: render a video frame. Maps to #GstBaseSinkClass.render() and
* #GstBaseSinkClass.preroll() vfuncs. Rendering during preroll will be
- * suppressed if the #GstVideoSink:show-preroll-frame property is set to
+ * suppressed if the #GstVideoSink:show-preroll-frame property is set to
* %FALSE.
*
* The video sink class structure. Derived classes should override the
/**
* SECTION:gstnavigation
+ * @title: GstNavigation
* @short_description: Interface for creating, sending and parsing navigation
* events.
*
* receiving navigation related bus events. One main usecase is DVD menu navigation.
*
* The main parts of the API are:
- * <itemizedlist>
- * <listitem>
- * <para>
- * The GstNavigation interface, implemented by elements which provide an application
- * with the ability to create and inject navigation events into the pipeline.
- * </para>
- * </listitem>
- * <listitem>
- * <para>
- * GstNavigation event handling API. GstNavigation events are created in response to
- * calls on a GstNavigation interface implementation, and sent in the pipeline. Upstream
- * elements can use the navigation event API functions to parse the contents of received
- * messages.
- * </para>
- * </listitem>
- * <listitem>
- * <para>
- * GstNavigation message handling API. GstNavigation messages may be sent on the message
- * bus to inform applications of navigation related changes in the pipeline, such as the
- * mouse moving over a clickable region, or the set of available angles changing.
- * </para><para>
+ *
+ * * The GstNavigation interface, implemented by elements which provide an application
+ * with the ability to create and inject navigation events into the pipeline.
+ * * GstNavigation event handling API. GstNavigation events are created in response to
+ * calls on a GstNavigation interface implementation, and sent in the pipeline. Upstream
+ * elements can use the navigation event API functions to parse the contents of received
+ * messages.
+ *
+ * * GstNavigation message handling API. GstNavigation messages may be sent on the message
+ * bus to inform applications of navigation related changes in the pipeline, such as the
+ * mouse moving over a clickable region, or the set of available angles changing.
+ *
* The GstNavigation message functions provide functions for creating and parsing
* custom bus messages for signaling GstNavigation changes.
- * </para>
- * </listitem>
- * </itemizedlist>
+ *
*/
#ifdef HAVE_CONFIG_H
* event.
* @y: Pointer to a gdouble to receive the y coordinate of the mouse button
* event.
- *
+ *
* Retrieve the details of either a #GstNavigation mouse button press event or
* a mouse button release event. Determine which type the event is using
* gst_navigation_event_get_type() to retrieve the #GstNavigationEventType.
/**
* SECTION:gstvideochroma
+ * @title: GstVideoChromaResample
* @short_description: Functions and utility object for operating on chroma video planes
*
* The functions gst_video_chroma_from_string() and gst_video_chroma_to_string() convert
/**
* SECTION:videoconverter
+ * @title: GstVideoConverter
* @short_description: Generic video conversion
*
- * <refsect2>
- * <para>
* This object is used to convert video frames from one format to another.
* The object can perform conversion of:
- * <itemizedlist>
- * <listitem><para>
- * video format
- * </para></listitem>
- * <listitem><para>
- * video colorspace
- * </para></listitem>
- * <listitem><para>
- * chroma-siting
- * </para></listitem>
- * <listitem><para>
- * video size
- * </para></listitem>
- * </para>
- * </refsect2>
+ *
+ * * video format
+ * * video colorspace
+ * * chroma-siting
+ * * video size
+ *
*/
/*
/**
* SECTION:gstvideodither
+ * @title: GstVideoDither
* @short_description: Utility object for dithering and quantizing lines of video
*
* GstVideoDither provides implementations of several dithering algorithms
* @count: integer that can be used to number key units
*
* Creates a new upstream force key unit event. An upstream force key unit event
- * can be sent to request upstream elements to produce a key unit.
+ * can be sent to request upstream elements to produce a key unit.
*
* @running_time can be set to request a new key unit at a specific
* running_time. If set to GST_CLOCK_TIME_NONE, upstream elements will produce a
/**
* SECTION:gstvideooverlaycomposition
+ * @title: GstVideoOverlayRectangle
* @short_description: Video Buffer Overlay Compositions (Subtitles, Logos)
*
- * <refsect2>
- * <para>
* Functions to create and handle overlay compositions on video buffers.
- * </para>
- * <para>
+ *
* An overlay composition describes one or more overlay rectangles to be
* blended on top of a video buffer.
- * </para>
- * <para>
+ *
* This API serves two main purposes:
- * <itemizedlist>
- * <listitem>
- * it can be used to attach overlay information (subtitles or logos)
- * to non-raw video buffers such as GL/VAAPI/VDPAU surfaces. The actual
- * blending of the overlay can then be done by e.g. the video sink that
- * processes these non-raw buffers.
- * </listitem>
- * <listitem>
- * it can also be used to blend overlay rectangles on top of raw video
- * buffers, thus consolidating blending functionality for raw video in
- * one place.
- * </listitem>
+ *
+ * * it can be used to attach overlay information (subtitles or logos)
+ * to non-raw video buffers such as GL/VAAPI/VDPAU surfaces. The actual
+ * blending of the overlay can then be done by e.g. the video sink that
+ * processes these non-raw buffers.
+ *
+ * * it can also be used to blend overlay rectangles on top of raw video
+ * buffers, thus consolidating blending functionality for raw video in
+ * one place.
+ *
* Together, this allows existing overlay elements to easily handle raw
* and non-raw video as input in without major changes (once the overlays
* have been put into a #GstOverlayComposition object anyway) - for raw
* video the overlay can just use the blending function to blend the data
* on top of the video, and for surface buffers it can just attach them to
* the buffer and let the sink render the overlays.
- * </itemizedlist>
- * </para>
- * </refsect2>
+ *
*/
/* TODO:
/**
* SECTION:gstvideoresampler
+ * @title: GstVideoResampler
* @short_description: Utility structure for resampler information
*
* #GstVideoResampler is a structure which holds the information
/**
* SECTION:gstvideoscaler
+ * @title: GstVideoScaler
* @short_description: Utility object for rescaling video frames
*
* #GstVideoScaler is a utility object for rescaling and resampling
/**
* SECTION:gstvideo
+ * @title: GstVideoAlignment
* @short_description: Support library for video operations
*
- * <refsect2>
- * <para>
* This library contains some helper functions and includes the
* videosink and videofilter base classes.
- * </para>
- * </refsect2>
+ *
*/
/**
/**
* SECTION:gstvideodirection
+ * @title: GstVideoDirection
* @short_description: Interface for elements providing video
* rotation and flipping controls
*
/**
* SECTION:gstvideoorientation
+ * @title: GstVideoOrientation
* @short_description: Interface for elements providing video orientation
* controls
*
*/
/**
* SECTION:gstvideooverlay
+ * @title: GstVideoOverlay
* @short_description: Interface for setting/getting a window system resource
* on elements supporting it to configure a window into which to render a
* video.
*
- * <refsect2>
- * <para>
* The #GstVideoOverlay interface is used for 2 main purposes :
- * <itemizedlist>
- * <listitem>
- * <para>
- * To get a grab on the Window where the video sink element is going to render.
- * This is achieved by either being informed about the Window identifier that
- * the video sink element generated, or by forcing the video sink element to use
- * a specific Window identifier for rendering.
- * </para>
- * </listitem>
- * <listitem>
- * <para>
- * To force a redrawing of the latest video frame the video sink element
- * displayed on the Window. Indeed if the #GstPipeline is in #GST_STATE_PAUSED
- * state, moving the Window around will damage its content. Application
- * developers will want to handle the Expose events themselves and force the
- * video sink element to refresh the Window's content.
- * </para>
- * </listitem>
- * </itemizedlist>
- * </para>
- * <para>
+ *
+ * * To get a grab on the Window where the video sink element is going to render.
+ * This is achieved by either being informed about the Window identifier that
+ * the video sink element generated, or by forcing the video sink element to use
+ * a specific Window identifier for rendering.
+ * * To force a redrawing of the latest video frame the video sink element
+ * displayed on the Window. Indeed if the #GstPipeline is in #GST_STATE_PAUSED
+ * state, moving the Window around will damage its content. Application
+ * developers will want to handle the Expose events themselves and force the
+ * video sink element to refresh the Window's content.
+ *
* Using the Window created by the video sink is probably the simplest scenario,
* in some cases, though, it might not be flexible enough for application
* developers if they need to catch events such as mouse moves and button
* clicks.
- * </para>
- * <para>
+ *
* Setting a specific Window identifier on the video sink element is the most
* flexible solution but it has some issues. Indeed the application needs to set
* its Window identifier at the right time to avoid internal Window creation
* ...
* }
* ]|
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Two basic usage scenarios</title>
- * <para>
+ *
+ * ## Two basic usage scenarios
+ *
* There are two basic usage scenarios: in the simplest case, the application
* uses #playbin or #plasink or knows exactly what particular element is used
* for video output, which is usually the case when the application creates
* As #playbin and #playsink implement the video overlay interface and proxy
* it transparently to the actual video sink even if it is created later, this
* case also applies when using these elements.
- * </para>
- * <para>
+ *
* In the other and more common case, the application does not know in advance
* what GStreamer video sink element will be used for video output. This is
* usually the case when an element such as #autovideosink is used.
* posts a prepare-window-handle message, and that is also why this message needs
* to be handled in a sync bus handler which will be called from the streaming
* thread directly (because the video sink will need an answer right then).
- * </para>
- * <para>
+ *
* As response to the prepare-window-handle element message in the bus sync
* handler, the application may use gst_video_overlay_set_window_handle() to tell
* the video sink to render onto an existing window surface. At this point the
* Gtk+ 2.18 and later, which is likely to cause problems when called from a
* sync handler; see below for a better approach without GDK_WINDOW_XID()
* used in the callback).
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>GstVideoOverlay and Gtk+</title>
- * <para>
+ *
+ * ## GstVideoOverlay and Gtk+
+ *
* |[
* #include <gst/video/videooverlay.h>
* #include <gtk/gtk.h>
* ...
* }
* ]|
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>GstVideoOverlay and Qt</title>
- * <para>
+ *
+ * ## GstVideoOverlay and Qt
+ *
* |[
* #include <glib.h>
* #include <gst/gst.h>
* return ret;
* }
* ]|
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
*/
/**
* SECTION:element-adder
+ * @title: adder
*
* The adder allows to mix several streams into one by adding the data.
* Mixed data is clamped to the min/max values of the data format.
* audio mixing element: It will sync input streams correctly and also handle
* live inputs properly.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc freq=100 ! adder name=mix ! audioconvert ! autoaudiosink audiotestsrc freq=500 ! mix.
- * ]| This pipeline produces two sine waves mixed together.
- * </refsect2>
+ * ]|
+ * This pipeline produces two sine waves mixed together.
+ *
*/
/* Element-Checklist-Version: 5 */
*/
/**
* SECTION:element-appsrc
+ * @title: appsrc
*
* The appsrc element can be used by applications to insert data into a
* GStreamer pipeline. Unlike most GStreamer elements, Appsrc provides
*/
/**
* SECTION:element-appsink
+ * @title: appsink
*
* Appsink is a sink plugin that supports many different methods for making
* the application get a handle on the GStreamer data in a pipeline. Unlike
/**
* SECTION:element-audioconvert
+ * @title: audioconvert
*
* Audioconvert converts raw audio buffers between various possible formats.
* It supports integer to float conversion, width/depth conversion,
* signedness and endianness conversion and channel transformations
* (ie. upmixing and downmixing), as well as dithering and noise-shaping.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 -v -m audiotestsrc ! audioconvert ! audio/x-raw,format=S8,channels=2 ! level ! fakesink silent=TRUE
- * ]| This pipeline converts audio to 8-bit. The level element shows that
+ * ]|
+ * This pipeline converts audio to 8-bit. The level element shows that
* the output levels still match the one for a sine wave.
* |[
* gst-launch-1.0 -v -m uridecodebin uri=file:///path/to/audio.flac ! audioconvert ! vorbisenc ! oggmux ! filesink location=audio.ogg
- * ]| The vorbis encoder takes float audio data instead of the integer data
+ * ]|
+ * The vorbis encoder takes float audio data instead of the integer data
* output by most other audio elements. This pipeline decodes a FLAC audio file
* (or any other audio file for which decoders are installed) and re-encodes
* it into an Ogg/Vorbis audio file.
- * </refsect2>
+ *
*/
/*
/**
* SECTION:element-audiorate
+ * @title: audiorate
* @see_also: #GstVideoRate
*
* This element takes an incoming stream of timestamped raw audio frames and
* that the incoming data is then simply shifted (by less than the indicated
* tolerance) to a perfect time.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v autoaudiosrc ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav
- * ]| Capture audio from the sound card and turn it into a perfect stream
+ * ]|
+ * Capture audio from the sound card and turn it into a perfect stream
* for saving in a raw audio file.
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.file ! audiorate ! audioconvert ! wavenc ! filesink location=alsa.wav
- * ]| Decodes an audio file and transforms it into a perfect stream for saving
+ * ]|
+ * Decodes an audio file and transforms it into a perfect stream for saving
* in a raw audio WAV file. Without the audio rate, the timing might not be
* preserved correctly in the WAV file in case the decoded stream is jittery
* or there are samples missing.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-audioresample
+ * @title: audioresample
*
* audioresample resamples raw audio buffers to different sample rates using
* a configurable windowing function to enhance quality.
* to initialize when the element is created. A third mode exists, which uses the full table
* unless said table would become too large, in which case the interpolated one is used instead.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.ogg ! audioconvert ! audioresample ! audio/x-raw, rate=8000 ! autoaudiosink
- * ]| Decode an audio file and downsample it to 8Khz and play sound.
+ * ]|
+ * Decode an audio file and downsample it to 8Khz and play sound.
* To create the Ogg/Vorbis file refer to the documentation of vorbisenc.
* This assumes there is an audio sink that will accept/handle 8kHz audio.
- * </refsect2>
+ *
*/
/* TODO:
*/
/**
* SECTION:element-audiotestsrc
+ * @title: audiotestsrc
*
* AudioTestSrc can be used to generate basic audio signals. It support several
* different waveforms and allows to set the base frequency and volume.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc ! audioconvert ! autoaudiosink
- * ]| This pipeline produces a sine with default frequency, 440 Hz, and the
+ * ]|
+ * This pipeline produces a sine with default frequency, 440 Hz, and the
* default volume, 0.8 (relative to a maximum 1.0).
* |[
* gst-launch-1.0 audiotestsrc wave=2 freq=200 ! tee name=t ! queue ! audioconvert ! autoaudiosink t. ! queue ! audioconvert ! libvisual_lv_scope ! videoconvert ! autovideosink
- * ]| In this example a saw wave is generated. The wave is shown using a
+ * ]|
+ * In this example a saw wave is generated. The wave is shown using a
* scope visualizer from libvisual, allowing you to visually verify that
* the saw wave is correct.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-encodebin
+ * @title: encodebin
*
* EncodeBin provides a bin for encoding/muxing various streams according to
* a specified #GstEncodingProfile.
* provide it raw or pre-encoded streams of data in input and have your
* encoded/muxed/converted stream in output.
*
- * <refsect2>
- * <title>Features</title>
- * <itemizedlist>
- * <listitem>
- * Automatic encoder and muxer selection based on elements available on the
+ * ## Features
+ *
+ * * Automatic encoder and muxer selection based on elements available on the
* system.
- * </listitem>
- * <listitem>
- * Conversion of raw audio/video streams (scaling, framerate conversion,
+ *
+ * * Conversion of raw audio/video streams (scaling, framerate conversion,
* colorspace conversion, samplerate conversion) to conform to the profile
* output format.
- * </listitem>
- * <listitem>
- * Variable number of streams. If the presence property for a stream encoding
+ *
+ * * Variable number of streams. If the presence property for a stream encoding
* profile is 0, you can request any number of sink pads for it via the
* standard request pad gstreamer API or the #GstEncodeBin::request-pad action
* signal.
- * </listitem>
- * <listitem>
- * Avoid reencoding (passthrough). If the input stream is already encoded and is
+ *
+ * * Avoid reencoding (passthrough). If the input stream is already encoded and is
* compatible with what the #GstEncodingProfile expects, then the stream won't
* be re-encoded but just passed through downstream to the muxer or the output.
- * </listitem>
- * <listitem>
- * Mix pre-encoded and raw streams as input. In addition to the passthrough
+ *
+ * * Mix pre-encoded and raw streams as input. In addition to the passthrough
* feature above, you can feed both raw audio/video *AND* already-encoded data
* to a pad. #GstEncodeBin will take care of passing through the compatible
* segments and re-encoding the segments of media that need encoding.
- * </listitem>
- * <listitem>
- * Standard behaviour is to use a #GstEncodingContainerProfile to have both
+ *
+ * * Standard behaviour is to use a #GstEncodingContainerProfile to have both
* encoding and muxing performed. But you can also provide a single stream
* profile (like #GstEncodingAudioProfile) to only have the encoding done and
* handle the encoded output yourself.
- * </listitem>
- * <listitem>
- * Audio imperfection corrections. Incoming audio streams can have non perfect
+ *
+ * * Audio imperfection corrections. Incoming audio streams can have non perfect
* timestamps (jitter), like the streams coming from ASF files. #GstEncodeBin
* will automatically fix those imperfections for you. See
* #GstEncodeBin:audio-jitter-tolerance for more details.
- * </listitem>
- * <listitem>
- * Variable or Constant video framerate. If your #GstEncodingVideoProfile has
+ *
+ * * Variable or Constant video framerate. If your #GstEncodingVideoProfile has
* the variableframerate property deactivated (default), then the incoming
* raw video stream will be retimestampped in order to produce a constant
* framerate.
- * </listitem>
- * <listitem>
- * Cross-boundary re-encoding. When feeding compatible pre-encoded streams that
+ *
+ * * Cross-boundary re-encoding. When feeding compatible pre-encoded streams that
* fall on segment boundaries, and for supported formats (right now only H263),
* the GOP will be decoded/reencoded when needed to produce an encoded output
* that fits exactly within the request GstSegment.
- * </listitem>
- * <listitem>
- * Missing plugin support. If a #GstElement is missing to encode/mux to the
+ *
+ * * Missing plugin support. If a #GstElement is missing to encode/mux to the
* request profile formats, a missing-plugin #GstMessage will be posted on the
* #GstBus, allowing systems that support the missing-plugin system to offer the
* user a way to install the missing element.
- * </listitem>
- * </itemizedlist>
- * </refsect2>
+ *
*/
/**
* SECTION:element-giosink
+ * @title: giosink
* @see_also: #GstFileSink, #GstGnomeVFSSink, #GstGioSrc
*
* This plugin writes incoming data to a local or remote location specified
* on the bus if the target location is not mounted yet and needs to be
* mounted. This message can be used by application to mount the location
* and retry after the location was mounted successfully.
- *
- * <refsect2>
- * <title>Example pipelines</title>
+ *
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=input.xyz ! giosink location=file:///home/joe/out.xyz
- * ]| The above pipeline will simply copy a local file. Instead of giosink,
+ * ]|
+ * The above pipeline will simply copy a local file. Instead of giosink,
* we could just as well have used the filesink element here.
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/audio.file ! audioconvert ! flacenc ! giosink location=smb://othercomputer/foo.flac
- * ]| The above pipeline will re-encode an audio file into FLAC format and store
+ * ]|
+ * The above pipeline will re-encode an audio file into FLAC format and store
* it on a remote host using the Samba protocol.
* |[
* gst-launch-1.0 -v audiotestsrc num-buffers=100 ! vorbisenc ! oggmux ! giosink location=file:///home/foo/bar.ogg
- * ]| The above pipeline will encode a 440Hz sine wave to Ogg Vorbis and stores
+ * ]|
+ * The above pipeline will encode a 440Hz sine wave to Ogg Vorbis and stores
* it in the home directory of user foo.
- * </refsect2>
+ *
*/
/* FIXME: We would like to mount the enclosing volume of an URL
/**
* SECTION:element-giosrc
+ * @title: giosrc
* @see_also: #GstFileSrc, #GstGnomeVFSSrc, #GstGioSink
*
* This plugin reads data from a local or remote location specified
* message was received and gst_bus_set_flushing(bus, FALSE) after the
* mounting was successful.
*
- * <refsect2>
- * <title>Example launch lines</title>
+ * ## Example launch lines
* |[
* gst-launch-1.0 -v giosrc location=file:///home/joe/foo.xyz ! fakesink
- * ]| The above pipeline will simply read a local file and do nothing with the
+ * ]|
+ * The above pipeline will simply read a local file and do nothing with the
* data read. Instead of giosrc, we could just as well have used the
* filesrc element here.
* |[
* gst-launch-1.0 -v giosrc location=smb://othercomputer/foo.xyz ! filesink location=/home/joe/foo.xyz
- * ]| The above pipeline will copy a file from a remote host to the local file
+ * ]|
+ * The above pipeline will copy a file from a remote host to the local file
* system using the Samba protocol.
* |[
* gst-launch-1.0 -v giosrc location=smb://othercomputer/demo.mp3 ! decodebin ! audioconvert ! audioresample ! autoaudiosink
- * ]| The above pipeline will read and decode and play an mp3 file from a
+ * ]|
+ * The above pipeline will read and decode and play an mp3 file from a
* SAMBA server.
- * </refsect2>
+ *
*/
/* FIXME: We would like to mount the enclosing volume of an URL
/**
* GstGioSrc:file:
- *
+ *
* %GFile to read from.
*/
g_object_class_install_property (gobject_class, PROP_FILE,
/**
* SECTION:element-giostreamsink
+ * @title: giostreamsink
*
* This plugin writes incoming data to a custom GIO #GOutputStream.
*
* It can, for example, be used to write a stream to memory with a
* #GMemoryOuputStream or to write to a file with a #GFileOuputStream.
*
- * <refsect2>
- * <title>Example code</title>
- * <para>
+ * ## Example code
+ *
* The following example writes the received data to a #GMemoryOutputStream.
* |[
...
* ]|
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-giostreamsrc
+ * @title: giostreamsrc
*
* This plugin reads data from a custom GIO #GInputStream.
*
* #GMemoryInputStream or to read from a file with a
* #GFileInputStream.
*
- * <refsect2>
- * <title>Example code</title>
- * <para>
+ * ## Example code
+ *
* The following example reads data from a #GMemoryInputStream.
* |[
...
* ]|
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-decodebin
+ * @title: decodebin
*
* #GstBin that auto-magically constructs a decoding pipeline using available
* decoders and demuxers via auto-plugging.
* This signal is emitted whenever decodebin finds a new stream. It is
* emitted before looking for any elements that can handle that stream.
*
- * <note>
- * Invocation of signal handlers stops after the first signal handler
- * returns #FALSE. Signal handlers are invoked in the order they were
- * connected in.
- * </note>
+ * > Invocation of signal handlers stops after the first signal handler
+ * > returns #FALSE. Signal handlers are invoked in the order they were
+ * > connected in.
*
* Returns: #TRUE if you wish decodebin to look for elements that can
* handle the given @caps. If #FALSE, those caps will be considered as
* If this function returns an empty array, the pad will be considered as
* having an unhandled type media type.
*
- * <note>
- * Only the signal handler that is connected first will ever by invoked.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Only the signal handler that is connected first will ever by invoked.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: a #GValueArray* with a list of factories to try. The factories are
* by default tried in the returned order or based on the index returned by
* The callee should copy and modify @factories or return #NULL if the
* order should not change.
*
- * <note>
- * Invocation of signal handlers stops after one signal handler has
- * returned something else than #NULL. Signal handlers are invoked in
- * the order they were connected in.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Invocation of signal handlers stops after one signal handler has
+ * > returned something else than #NULL. Signal handlers are invoked in
+ * > the order they were connected in.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: A new sorted array of #GstElementFactory objects.
*/
* A value of #GST_AUTOPLUG_SELECT_SKIP will skip @factory and move to the
* next factory.
*
- * <note>
- * The signal handler will not be invoked if any of the previously
- * registered signal handlers (if any) return a value other than
- * GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
- * GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
- * registered next (again, if any) can override that decision.
- * </note>
+ * > The signal handler will not be invoked if any of the previously
+ * > registered signal handlers (if any) return a value other than
+ * > GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
+ * > GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
+ * > registered next (again, if any) can override that decision.
*
* Returns: a #GST_TYPE_AUTOPLUG_SELECT_RESULT that indicates the required
* operation. the default handler will always return
/**
* SECTION:element-decodebin3
+ * @title: decodebin3
*
* #GstBin that auto-magically constructs a decoding pipeline using available
* decoders and demuxers via auto-plugging. The output is raw audio, video
*
* decodebin3 differs from the previous decodebin (decodebin2) in important ways:
*
- * <itemizedlist>
- * <listitem>
- * supports publication and selection of stream information via
+ * * supports publication and selection of stream information via
* GstStreamCollection messages and #GST_EVENT_SELECT_STREAM events.
- * </listitem>
- * <listitem>
- * dynamically switches stream connections internally, and
+ *
+ * * dynamically switches stream connections internally, and
* reuses decoder elements when stream selections change, so that in
* the normal case it maintains 1 decoder of each type (video/audio/subtitle)
* and only creates new elements when streams change and an existing decoder
* is not capable of handling the new format.
- * </listitem>
- * <listitem>
- * supports multiple input pads for the parallel decoding of auxilliary streams
+ *
+ * * supports multiple input pads for the parallel decoding of auxilliary streams
* not muxed with the primary stream.
- * </listitem>
- * <listitem>
- * does not handle network stream buffering. decodebin3 expects that network stream
+ *
+ * * does not handle network stream buffering. decodebin3 expects that network stream
* buffering is handled upstream, before data is passed to it.
- * </listitem>
- * </itemizedlist>
*
* <emphasis>decodebin3 is still experimental API and a technology preview.
* Its behaviour and exposed API is subject to change.</emphasis>
* * STREAM_START :
* a new stream is starting => link it further if needed
*
- *
* 3) Gradual replacement
*
* If the caps change at any point in decodebin (input sink pad, demuxer output,
* b.1) The new CAPS are accepted, keep current configuration
* b.2) The new CAPS are not accepted, remove following elements then do a)
*
- *
- *
* Components:
*
* MultiQ Output
/**
* SECTION:element-parsebin
+ * @title: parsebin
*
* #GstBin that auto-magically constructs a parsing pipeline
* using available parsers and demuxers via auto-plugging.
* This signal is emitted whenever ParseBin finds a new stream. It is
* emitted before looking for any elements that can handle that stream.
*
- * <note>
- * Invocation of signal handlers stops after the first signal handler
- * returns #FALSE. Signal handlers are invoked in the order they were
- * connected in.
- * </note>
+ * > Invocation of signal handlers stops after the first signal handler
+ * > returns #FALSE. Signal handlers are invoked in the order they were
+ * > connected in.
*
* Returns: #TRUE if you wish ParseBin to look for elements that can
* handle the given @caps. If #FALSE, those caps will be considered as
* If this function returns an empty array, the pad will be considered as
* having an unhandled type media type.
*
- * <note>
- * Only the signal handler that is connected first will ever by invoked.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Only the signal handler that is connected first will ever by invoked.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: a #GValueArray* with a list of factories to try. The factories are
* by default tried in the returned order or based on the index returned by
* The callee should copy and modify @factories or return #NULL if the
* order should not change.
*
- * <note>
- * Invocation of signal handlers stops after one signal handler has
- * returned something else than #NULL. Signal handlers are invoked in
- * the order they were connected in.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Invocation of signal handlers stops after one signal handler has
+ * > returned something else than #NULL. Signal handlers are invoked in
+ * > the order they were connected in.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: A new sorted array of #GstElementFactory objects.
*/
* A value of #GST_AUTOPLUG_SELECT_SKIP will skip @factory and move to the
* next factory.
*
- * <note>
- * The signal handler will not be invoked if any of the previously
- * registered signal handlers (if any) return a value other than
- * GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
- * GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
- * registered next (again, if any) can override that decision.
- * </note>
+ * > The signal handler will not be invoked if any of the previously
+ * > registered signal handlers (if any) return a value other than
+ * > GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
+ * > GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
+ * > registered next (again, if any) can override that decision.
*
* Returns: a #GST_TYPE_AUTOPLUG_SELECT_RESULT that indicates the required
* operation. the default handler will always return
/**
* SECTION:element-playbin
+ * @title: playbin
*
* Playbin provides a stand-alone everything-in-one abstraction for an
* audio and/or video player.
*
* Playbin can handle both audio and video files and features
- * <itemizedlist>
- * <listitem>
- * automatic file type recognition and based on that automatic
+ *
+ * * automatic file type recognition and based on that automatic
* selection and usage of the right audio/video/subtitle demuxers/decoders
- * </listitem>
- * <listitem>
- * visualisations for audio files
- * </listitem>
- * <listitem>
- * subtitle support for video files. Subtitles can be store in external
+ * * visualisations for audio files
+ * * subtitle support for video files. Subtitles can be store in external
* files.
- * </listitem>
- * <listitem>
- * stream selection between different video/audio/subtitles streams
- * </listitem>
- * <listitem>
- * meta info (tag) extraction
- * </listitem>
- * <listitem>
- * easy access to the last video sample
- * </listitem>
- * <listitem>
- * buffering when playing streams over a network
- * </listitem>
- * <listitem>
- * volume control with mute option
- * </listitem>
- * </itemizedlist>
+ * * stream selection between different video/audio/subtitles streams
+ * * meta info (tag) extraction
+ * * easy access to the last video sample
+ * * buffering when playing streams over a network
+ * * volume control with mute option
+ *
+ * ## Usage
*
- * <refsect2>
- * <title>Usage</title>
- * <para>
* A playbin element can be created just like any other element using
* gst_element_factory_make(). The file/URI to play should be set via the #GstPlayBin:uri
* property. This must be an absolute URI, relative file paths are not allowed.
* via gst_element_query_position() and gst_element_query_duration() and
* setting the format passed to GST_FORMAT_TIME. If the query was successful,
* the duration or position will have been returned in units of nanoseconds.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Advanced Usage: specifying the audio and video sink</title>
- * <para>
+ *
+ * ## Advanced Usage: specifying the audio and video sink
+ *
* By default, if no audio sink or video sink has been specified via the
* #GstPlayBin:audio-sink or #GstPlayBin:video-sink property, playbin will use the autoaudiosink
* and autovideosink elements to find the first-best available output method.
* It is also possible to 'suppress' audio and/or video output by using
* 'fakesink' elements (or capture it from there using the fakesink element's
* "handoff" signal, which, nota bene, is fired from the streaming thread!).
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Retrieving Tags and Other Meta Data</title>
- * <para>
+ *
+ * ## Retrieving Tags and Other Meta Data
+ *
* Most of the common meta data (artist, title, etc.) can be retrieved by
* watching for TAG messages on the pipeline's bus (see above).
*
* Other more specific meta information like width/height/framerate of video
* streams or samplerate/number of channels of audio streams can be obtained
* from the negotiated caps on the sink pads of the sinks.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Buffering</title>
+ *
+ * ## Buffering
* Playbin handles buffering automatically for the most part, but applications
* need to handle parts of the buffering process as well. Whenever playbin is
* buffering, it will post BUFFERING messages on the bus with a percentage
* ...
* }
* ]|
+ *
* Note that applications should keep/set the pipeline in the PAUSED state when
* a BUFFERING message is received with a buffer percent value < 100 and set
* the pipeline back to PLAYING state when a BUFFERING message with a value
* of 100 percent is received (if PLAYING is the desired state, that is).
- * </refsect2>
- * <refsect2>
- * <title>Embedding the video window in your application</title>
+ *
+ * ## Embedding the video window in your application
* By default, playbin (or rather the video sinks used) will create their own
* window. Applications will usually want to force output to a window of their
* own, however. This can be done using the #GstVideoOverlay interface, which most
* video sinks implement. See the documentation there for more details.
- * </refsect2>
- * <refsect2>
- * <title>Specifying which CD/DVD device to use</title>
+ *
+ * ## Specifying which CD/DVD device to use
* The device to use for CDs/DVDs needs to be set on the source element
* playbin creates before it is opened. The most generic way of doing this
* is to connect to playbin's "source-setup" (or "notify::source") signal,
* elements involved if this will work or not. For example, for DVD menu
* playback, the following syntax might work (if the resindvd plugin is used):
* dvd://[/path/to/device]
- * </refsect2>
- * <refsect2>
- * <title>Handling redirects</title>
- * <para>
+ *
+ * ## Handling redirects
+ *
* Some elements may post 'redirect' messages on the bus to tell the
* application to open another location. These are element messages containing
* a structure named 'redirect' along with a 'new-location' field of string
* type. The new location may be a relative or an absolute URI. Examples
* for such redirects can be found in many quicktime movie trailers.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Examples</title>
+ *
+ * ## Examples
* |[
* gst-launch-1.0 -v playbin uri=file:///path/to/somefile.mp4
- * ]| This will play back the given AVI video file, given that the video and
+ * ]|
+ * This will play back the given AVI video file, given that the video and
* audio decoders required to decode the content are installed. Since no
* special audio sink or video sink is supplied (via playbin's audio-sink or
* video-sink properties) playbin will try to find a suitable audio and
* video sink automatically using the autoaudiosink and autovideosink elements.
* |[
* gst-launch-1.0 -v playbin uri=cdda://4
- * ]| This will play back track 4 on an audio CD in your disc drive (assuming
+ * ]|
+ * This will play back track 4 on an audio CD in your disc drive (assuming
* the drive is detected automatically by the plugin).
* |[
* gst-launch-1.0 -v playbin uri=dvd://
- * ]| This will play back the DVD in your disc drive (assuming
+ * ]|
+ * This will play back the DVD in your disc drive (assuming
* the drive is detected automatically by the plugin).
- * </refsect2>
+ *
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
ave_list = g_list_prepend (ave_list, NULL);
}
- /* if it is a decoder and we don't have a fixed sink, then find out
+ /* if it is a decoder and we don't have a fixed sink, then find out
* the matching audio/video sink from GstAVElements list */
for (l = ave_list; l; l = l->next) {
gboolean created_sink = FALSE;
}
/* remember the sink in the group now, the element is floating, we take
- * ownership now
+ * ownership now
*
* store the sink in the group, we will configure it later when we
* reconfigure the sink */
/**
* SECTION:element-playbin3
+ * @title: playbin3
*
* playbin3 provides a stand-alone everything-in-one abstraction for an
* audio and/or video player. It differs from the previous playbin (playbin2)
* Its behaviour and exposed API is subject to change.</emphasis>
*
* playbin3 can handle both audio and video files and features
- * <itemizedlist>
- * <listitem>
- * automatic file type recognition and based on that automatic
+ *
+ * * automatic file type recognition and based on that automatic
* selection and usage of the right audio/video/subtitle demuxers/decoders
- * </listitem>
- * <listitem>
- * auxilliary files - such as external subtitles and audio tracks
- * </listitem>
- * <listitem>
- * visualisations for audio files
- * </listitem>
- * <listitem>
- * subtitle support for video files. Subtitles can be store in external
- * files.
- * </listitem>
- * <listitem>
- * stream selection between different video/audio/subtitles streams
- * </listitem>
- * <listitem>
- * meta info (tag) extraction
- * </listitem>
- * <listitem>
- * easy access to the last video sample
- * </listitem>
- * <listitem>
- * buffering when playing streams over a network
- * </listitem>
- * <listitem>
- * volume control with mute option
- * </listitem>
- * </itemizedlist>
*
- * <refsect2>
- * <title>Usage</title>
- * <para>
+ * * auxilliary files - such as external subtitles and audio tracks
+ * * visualisations for audio files
+ * * subtitle support for video files. Subtitles can be store in external
+ * files.
+ * * stream selection between different video/audio/subtitles streams
+ * * meta info (tag) extraction
+ * * easy access to the last video sample
+ * * buffering when playing streams over a network
+ * * volume control with mute option
+ *
+ * ## Usage
+ *
* A playbin element can be created just like any other element using
* gst_element_factory_make(). The file/URI to play should be set via the #GstPlayBin3:uri
* property. This must be an absolute URI, relative file paths are not allowed.
* via gst_element_query_position() and gst_element_query_duration() and
* setting the format passed to GST_FORMAT_TIME. If the query was successful,
* the duration or position will have been returned in units of nanoseconds.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Advanced Usage: specifying the audio and video sink</title>
- * <para>
+ *
+ * ## Advanced Usage: specifying the audio and video sink
+ *
* By default, if no audio sink or video sink has been specified via the
* #GstPlayBin3:audio-sink or #GstPlayBin3:video-sink property, playbin3 will use the autoaudiosink
* and autovideosink elements to find the first-best available output method.
* It is also possible to 'suppress' audio and/or video output by using
* 'fakesink' elements (or capture it from there using the fakesink element's
* "handoff" signal, which, nota bene, is fired from the streaming thread!).
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Retrieving Tags and Other Meta Data</title>
- * <para>
+ *
+ * ## Retrieving Tags and Other Meta Data
+ *
* Most of the common meta data (artist, title, etc.) can be retrieved by
* watching for TAG messages on the pipeline's bus (see above).
*
* Other more specific meta information like width/height/framerate of video
* streams or samplerate/number of channels of audio streams can be obtained
* from the negotiated caps on the sink pads of the sinks.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Buffering</title>
+ *
+ * ## Buffering
* Playbin3 handles buffering automatically for the most part, but applications
* need to handle parts of the buffering process as well. Whenever playbin3 is
* buffering, it will post BUFFERING messages on the bus with a percentage
* ...
* }
* ]|
+ *
* Note that applications should keep/set the pipeline in the PAUSED state when
* a BUFFERING message is received with a buffer percent value < 100 and set
* the pipeline back to PLAYING state when a BUFFERING message with a value
* of 100 percent is received (if PLAYING is the desired state, that is).
- * </refsect2>
- * <refsect2>
- * <title>Embedding the video window in your application</title>
+ *
+ * ## Embedding the video window in your application
* By default, playbin3 (or rather the video sinks used) will create their own
* window. Applications will usually want to force output to a window of their
* own, however. This can be done using the #GstVideoOverlay interface, which most
* video sinks implement. See the documentation there for more details.
- * </refsect2>
- * <refsect2>
- * <title>Specifying which CD/DVD device to use</title>
+ *
+ * ## Specifying which CD/DVD device to use
* The device to use for CDs/DVDs needs to be set on the source element
* playbin3 creates before it is opened. The most generic way of doing this
* is to connect to playbin3's "source-setup" (or "notify::source") signal,
* elements involved if this will work or not. For example, for DVD menu
* playback, the following syntax might work (if the resindvd plugin is used):
* dvd://[/path/to/device]
- * </refsect2>
- * <refsect2>
- * <title>Handling redirects</title>
- * <para>
+ *
+ * ## Handling redirects
+ *
* Some elements may post 'redirect' messages on the bus to tell the
* application to open another location. These are element messages containing
* a structure named 'redirect' along with a 'new-location' field of string
* type. The new location may be a relative or an absolute URI. Examples
* for such redirects can be found in many quicktime movie trailers.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Examples</title>
+ *
+ * ## Examples
* |[
* gst-launch-1.0 -v playbin3 uri=file:///path/to/somefile.mp4
- * ]| This will play back the given AVI video file, given that the video and
+ * ]|
+ * This will play back the given AVI video file, given that the video and
* audio decoders required to decode the content are installed. Since no
* special audio sink or video sink is supplied (via playbin3's audio-sink or
* video-sink properties) playbin3 will try to find a suitable audio and
* video sink automatically using the autoaudiosink and autovideosink elements.
* |[
* gst-launch-1.0 -v playbin3 uri=cdda://4
- * ]| This will play back track 4 on an audio CD in your disc drive (assuming
+ * ]|
+ * This will play back track 4 on an audio CD in your disc drive (assuming
* the drive is detected automatically by the plugin).
* |[
* gst-launch-1.0 -v playbin3 uri=dvd://
- * ]| This will play back the DVD in your disc drive (assuming
+ * ]|
+ * This will play back the DVD in your disc drive (assuming
* the drive is detected automatically by the plugin).
- * </refsect2>
+ *
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
/**
* SECTION:element-subtitleoverlay
+ * @title: subtitleoverlay
*
* #GstBin that auto-magically overlays a video stream with subtitles by
* autoplugging the required elements.
* It supports raw, timestamped text, different textual subtitle formats and
* DVD subpicture subtitles.
*
- * <refsect2>
- * <title>Examples</title>
+ * ## Examples
* |[
* gst-launch-1.0 -v filesrc location=test.mkv ! matroskademux name=demux ! video/x-h264 ! queue ! decodebin ! subtitleoverlay name=overlay ! videoconvert ! autovideosink demux. ! subpicture/x-dvd ! queue ! overlay.
- * ]| This will play back the given Matroska file with h264 video and dvd subpicture style subtitles.
- * </refsect2>
+ * ]|
+ * This will play back the given Matroska file with h264 video and dvd subpicture style subtitles.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-uridecodebin
+ * @title: uridecodebin
*
* Decodes data from a URI into raw media. It selects a source element that can
* handle the given #GstURIDecodeBin:uri scheme and connects it to a decodebin.
* If set to %FALSE, then only the streams that can be decoded to the final
* caps (see 'caps' property) will have a pad exposed. Streams that do not
* match those caps but could have been decoded will not have decoder plugged
- * in internally and will not have a pad exposed.
+ * in internally and will not have a pad exposed.
*
* Since: 0.10.30
*/
* This signal is emitted whenever uridecodebin finds a new stream. It is
* emitted before looking for any elements that can handle that stream.
*
- * <note>
- * Invocation of signal handlers stops after the first signal handler
- * returns #FALSE. Signal handlers are invoked in the order they were
- * connected in.
- * </note>
+ * > Invocation of signal handlers stops after the first signal handler
+ * > returns #FALSE. Signal handlers are invoked in the order they were
+ * > connected in.
*
* Returns: #TRUE if you wish uridecodebin to look for elements that can
* handle the given @caps. If #FALSE, those caps will be considered as
* If this function returns an empty array, the pad will be considered as
* having an unhandled type media type.
*
- * <note>
- * Only the signal handler that is connected first will ever by invoked.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Only the signal handler that is connected first will ever by invoked.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: a #GValueArray* with a list of factories to try. The factories are
* by default tried in the returned order or based on the index returned by
* The callee should copy and modify @factories or return #NULL if the
* order should not change.
*
- * <note>
- * Invocation of signal handlers stops after one signal handler has
- * returned something else than #NULL. Signal handlers are invoked in
- * the order they were connected in.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Invocation of signal handlers stops after one signal handler has
+ * > returned something else than #NULL. Signal handlers are invoked in
+ * > the order they were connected in.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: A new sorted array of #GstElementFactory objects.
*
* A value of #GST_AUTOPLUG_SELECT_SKIP will skip @factory and move to the
* next factory.
*
- * <note>
- * The signal handler will not be invoked if any of the previously
- * registered signal handlers (if any) return a value other than
- * GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
- * GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
- * registered next (again, if any) can override that decision.
- * </note>
+ * > The signal handler will not be invoked if any of the previously
+ * > registered signal handlers (if any) return a value other than
+ * > GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
+ * > GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
+ * > registered next (again, if any) can override that decision.
*
* Returns: a #GST_TYPE_AUTOPLUG_SELECT_RESULT that indicates the required
* operation. The default handler will always return
/**
* SECTION:element-urisourcebin
+ * @title: urisourcebin
*
* urisourcebin is an element for accessing URIs in a uniform manner.
*
* This signal is emitted whenever urisourcebin finds a new stream. It is
* emitted before looking for any elements that can handle that stream.
*
- * <note>
- * Invocation of signal handlers stops after the first signal handler
- * returns #FALSE. Signal handlers are invoked in the order they were
- * connected in.
- * </note>
+ * > Invocation of signal handlers stops after the first signal handler
+ * > returns #FALSE. Signal handlers are invoked in the order they were
+ * > connected in.
*
* Returns: #TRUE if you wish urisourcebin to look for elements that can
* handle the given @caps. If #FALSE, those caps will be considered as
* If this function returns an empty array, the pad will be considered as
* having an unhandled type media type.
*
- * <note>
- * Only the signal handler that is connected first will ever by invoked.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Only the signal handler that is connected first will ever by invoked.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: a #GValueArray* with a list of factories to try. The factories are
* by default tried in the returned order or based on the index returned by
* The callee should copy and modify @factories or return #NULL if the
* order should not change.
*
- * <note>
- * Invocation of signal handlers stops after one signal handler has
- * returned something else than #NULL. Signal handlers are invoked in
- * the order they were connected in.
- * Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
- * signal, they will never be invoked!
- * </note>
+ * > Invocation of signal handlers stops after one signal handler has
+ * > returned something else than #NULL. Signal handlers are invoked in
+ * > the order they were connected in.
+ * > Don't connect signal handlers with the #G_CONNECT_AFTER flag to this
+ * > signal, they will never be invoked!
*
* Returns: A new sorted array of #GstElementFactory objects.
*
* A value of #GST_AUTOPLUG_SELECT_SKIP will skip @factory and move to the
* next factory.
*
- * <note>
- * The signal handler will not be invoked if any of the previously
- * registered signal handlers (if any) return a value other than
- * GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
- * GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
- * registered next (again, if any) can override that decision.
- * </note>
+ * > The signal handler will not be invoked if any of the previously
+ * > registered signal handlers (if any) return a value other than
+ * > GST_AUTOPLUG_SELECT_TRY. Which also means that if you return
+ * > GST_AUTOPLUG_SELECT_TRY from one signal handler, handlers that get
+ * > registered next (again, if any) can override that decision.
*
* Returns: a #GST_TYPE_AUTOPLUG_SELECT_RESULT that indicates the required
* operation. The default handler will always return
/**
* SECTION:element-rawaudioparse
+ * @title: rawaudioparse
*
* This element parses incoming data as raw audio samples and timestamps it.
* It also handles seek queries in said raw audio data, and ensures that output
* GStreamer positioning is used. This property is also useful for swapping left
* and right in a stereo signal for example.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 souphttpsrc http://my-dlna-server/track.l16 \
* rawaudioparse ! audioconvert ! audioresample ! autoaudiosink
- * ]| Receive L16 data from a DLNA server, parse and timestamp it with
+ * ]|
+ * Receive L16 data from a DLNA server, parse and timestamp it with
* rawaudioparse, and play it. use-sink-caps is set to true since souphttpsrc
* will set its source pad's caps to audio/x-unaligned-raw for the L16 stream.
* |[
* gst-launch-1.0 filesrc location=audio.raw ! rawaudioparse use-sink-caps=false \
* format=pcm pcm-format=s16le sample-rate=48000 num-channels=2 \
* audioconvert ! audioresample ! autoaudiosink
- * ]| Read raw data from a local file and parse it as PCM data with 48000 Hz sample
+ * ]|
+ * Read raw data from a local file and parse it as PCM data with 48000 Hz sample
* rate, signed 16 bit integer samples, and 2 channels. use-sink-caps is set to
* false to ensure the property information is used and the parser does not expect
* audio/x-raw or audio/x-unaligned-raw caps.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-rawvideoparse
+ * @title: rawvideoparse
*
* This element parses incoming data as raw video frames and timestamps these.
* It also handles seek queries in said raw video data, and ensures that output
* plane-array properties.
*
* The frame stride property is useful in cases where there is extra data between
- * the frames (for example, trailing metadata, or headers). The parser calculates
+ * the frames (for example, trailing metadata, or headers). The parser calculates
* the actual frame size out of the other properties and compares it with this
* frame-stride value. If the frame stride is larger than the calculated size,
* then the extra bytes after the end of the frame are skipped. For example, with
* no duration set. The first output buffer will have a PTS 0, all subsequent ones
* an unset PTS.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 filesrc location=video.raw ! rawvideoparse use-sink-caps=false \
* width=500 height=400 format=y444 ! autovideosink
- * ]| Read raw data from a local file and parse it as video data with 500x400 pixels
+ * ]|
+ * Read raw data from a local file and parse it as video data with 500x400 pixels
* and Y444 video format.
* |[
* gst-launch-1.0 filesrc location=video.raw ! queue ! "video/x-raw, width=320, \
* height=240, format=I420, framerate=1/1" ! rawvideoparse \
* use-sink-caps=true ! autovideosink
- * ]| Read raw data from a local file and parse it as video data with 320x240 pixels
+ * ]|
+ * Read raw data from a local file and parse it as video data with 320x240 pixels
* and I420 video format. The queue element here is to force push based scheduling.
* See the documentation in #GstRawBaseParse for the reason why.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-multifdsink
+ * @title: multifdsink
* @see_also: tcpserversink
*
* This plugin writes incoming data to a set of file descriptors. The
- * file descriptors can be added to multifdsink by emitting the #GstMultiFdSink::add signal.
+ * file descriptors can be added to multifdsink by emitting the #GstMultiFdSink::add signal.
* For each descriptor added, the #GstMultiFdSink::client-added signal will be called.
*
* The multifdsink element needs to be set into READY, PAUSED or PLAYING state
* before operations such as adding clients are possible.
*
* A client can also be added with the #GstMultiFdSink::add-full signal
- * that allows for more control over what and how much data a client
+ * that allows for more control over what and how much data a client
* initially receives.
*
* Clients can be removed from multifdsink by emitting the #GstMultiFdSink::remove signal. For
* Note that multifdsink still has a reference to the file descriptor when the
* #GstMultiFdSink::client-removed signal is emitted, so that "get-stats" can be performed on
* the descriptor; it is therefore not safe to close the file descriptor in
- * the #GstMultiFdSink::client-removed signal handler, and you should use the
+ * the #GstMultiFdSink::client-removed signal handler, and you should use the
* #GstMultiFdSink::client-fd-removed signal to safely close the fd.
*
* Multifdsink internally keeps a queue of the incoming buffers and uses a
* speeds.
*
* When adding a client to multifdsink, the #GstMultiFdSink:sync-method property will define
- * which buffer in the queued buffers will be sent first to the client. Clients
- * can be sent the most recent buffer (which might not be decodable by the
- * client if it is not a keyframe), the next keyframe received in
+ * which buffer in the queued buffers will be sent first to the client. Clients
+ * can be sent the most recent buffer (which might not be decodable by the
+ * client if it is not a keyframe), the next keyframe received in
* multifdsink (which can take some time depending on the keyframe rate), or the
- * last received keyframe (which will cause a simple burst-on-connect).
+ * last received keyframe (which will cause a simple burst-on-connect).
* Multifdsink will always keep at least one keyframe in its internal buffers
* when the sync-mode is set to latest-keyframe.
*
* There are additional values for the #GstMultiFdSink:sync-method
* property to allow finer control over burst-on-connect behaviour. By selecting
* the 'burst' method a minimum burst size can be chosen, 'burst-keyframe'
- * additionally requires that the burst begin with a keyframe, and
+ * additionally requires that the burst begin with a keyframe, and
* 'burst-with-keyframe' attempts to burst beginning with a keyframe, but will
* prefer a minimum burst size even if it requires not starting with a keyframe.
*
* Multifdsink can be instructed to keep at least a minimum amount of data
- * expressed in time or byte units in its internal queues with the
+ * expressed in time or byte units in its internal queues with the
* #GstMultiFdSink:time-min and #GstMultiFdSink:bytes-min properties respectively.
- * These properties are useful if the application adds clients with the
+ * These properties are useful if the application adds clients with the
* #GstMultiFdSink::add-full signal to make sure that a burst connect can
- * actually be honored.
+ * actually be honored.
*
* When streaming data, clients are allowed to read at a different rate than
* the rate at which multifdsink receives data. If the client is reading too
* fast, no data will be send to the client until multifdsink receives more
- * data. If the client, however, reads too slowly, data for that client will be
- * queued up in multifdsink. Two properties control the amount of data
- * (buffers) that is queued in multifdsink: #GstMultiFdSink:buffers-max and
+ * data. If the client, however, reads too slowly, data for that client will be
+ * queued up in multifdsink. Two properties control the amount of data
+ * (buffers) that is queued in multifdsink: #GstMultiFdSink:buffers-max and
* #GstMultiFdSink:buffers-soft-max. A client that falls behind by
* #GstMultiFdSink:buffers-max is removed from multifdsink forcibly.
*
* RESYNC_KEYFRAME positions the client at the most recent keyframe in the
* buffer queue.
*
- * multifdsink will by default synchronize on the clock before serving the
- * buffers to the clients. This behaviour can be disabled by setting the sync
+ * multifdsink will by default synchronize on the clock before serving the
+ * buffers to the clients. This behaviour can be disabled by setting the sync
* property to FALSE. Multifdsink will by default not do QoS and will never
* drop late buffers.
*/
/**
* SECTION:element-multihandlesink
+ * @title: multihandlesink
* @see_also: tcpserversink
*
* This plugin writes incoming data to a set of file descriptors. The
- * file descriptors can be added to multihandlesink by emitting the #GstMultiHandleSink::add signal.
+ * file descriptors can be added to multihandlesink by emitting the #GstMultiHandleSink::add signal.
* For each descriptor added, the #GstMultiHandleSink::client-added signal will be called.
*
* A client can also be added with the #GstMultiHandleSink::add-full signal
- * that allows for more control over what and how much data a client
+ * that allows for more control over what and how much data a client
* initially receives.
*
* Clients can be removed from multihandlesink by emitting the #GstMultiHandleSink::remove signal. For
* Note that multihandlesink still has a reference to the file descriptor when the
* #GstMultiHandleSink::client-removed signal is emitted, so that "get-stats" can be performed on
* the descriptor; it is therefore not safe to close the file descriptor in
- * the #GstMultiHandleSink::client-removed signal handler, and you should use the
+ * the #GstMultiHandleSink::client-removed signal handler, and you should use the
* #GstMultiHandleSink::client-fd-removed signal to safely close the fd.
*
* Multisocketsink internally keeps a queue of the incoming buffers and uses a
* speeds.
*
* When adding a client to multihandlesink, the #GstMultiHandleSink:sync-method property will define
- * which buffer in the queued buffers will be sent first to the client. Clients
- * can be sent the most recent buffer (which might not be decodable by the
- * client if it is not a keyframe), the next keyframe received in
+ * which buffer in the queued buffers will be sent first to the client. Clients
+ * can be sent the most recent buffer (which might not be decodable by the
+ * client if it is not a keyframe), the next keyframe received in
* multihandlesink (which can take some time depending on the keyframe rate), or the
- * last received keyframe (which will cause a simple burst-on-connect).
+ * last received keyframe (which will cause a simple burst-on-connect).
* Multisocketsink will always keep at least one keyframe in its internal buffers
* when the sync-mode is set to latest-keyframe.
*
* There are additional values for the #GstMultiHandleSink:sync-method
* property to allow finer control over burst-on-connect behaviour. By selecting
* the 'burst' method a minimum burst size can be chosen, 'burst-keyframe'
- * additionally requires that the burst begin with a keyframe, and
+ * additionally requires that the burst begin with a keyframe, and
* 'burst-with-keyframe' attempts to burst beginning with a keyframe, but will
* prefer a minimum burst size even if it requires not starting with a keyframe.
*
* Multisocketsink can be instructed to keep at least a minimum amount of data
- * expressed in time or byte units in its internal queues with the
+ * expressed in time or byte units in its internal queues with the
* #GstMultiHandleSink:time-min and #GstMultiHandleSink:bytes-min properties respectively.
- * These properties are useful if the application adds clients with the
+ * These properties are useful if the application adds clients with the
* #GstMultiHandleSink::add-full signal to make sure that a burst connect can
- * actually be honored.
+ * actually be honored.
*
* When streaming data, clients are allowed to read at a different rate than
* the rate at which multihandlesink receives data. If the client is reading too
* fast, no data will be send to the client until multihandlesink receives more
- * data. If the client, however, reads too slowly, data for that client will be
- * queued up in multihandlesink. Two properties control the amount of data
- * (buffers) that is queued in multihandlesink: #GstMultiHandleSink:buffers-max and
+ * data. If the client, however, reads too slowly, data for that client will be
+ * queued up in multihandlesink. Two properties control the amount of data
+ * (buffers) that is queued in multihandlesink: #GstMultiHandleSink:buffers-max and
* #GstMultiHandleSink:buffers-soft-max. A client that falls behind by
* #GstMultiHandleSink:buffers-max is removed from multihandlesink forcibly.
*
* RESYNC_KEYFRAME positions the client at the most recent keyframe in the
* buffer queue.
*
- * multihandlesink will by default synchronize on the clock before serving the
- * buffers to the clients. This behaviour can be disabled by setting the sync
+ * multihandlesink will by default synchronize on the clock before serving the
+ * buffers to the clients. This behaviour can be disabled by setting the sync
* property to FALSE. Multisocketsink will by default not do QoS and will never
* drop late buffers.
*/
* @GST_SYNC_METHOD_NEXT_KEYFRAME : client receives next keyframe
* @GST_SYNC_METHOD_LATEST_KEYFRAME : client receives latest keyframe (burst)
* @GST_SYNC_METHOD_BURST : client receives specific amount of data
- * @GST_SYNC_METHOD_BURST_KEYFRAME : client receives specific amount of data
+ * @GST_SYNC_METHOD_BURST_KEYFRAME : client receives specific amount of data
* starting from latest keyframe
* @GST_SYNC_METHOD_BURST_WITH_KEYFRAME : client receives specific amount of data from
* a keyframe, or if there is not enough data after
/**
* SECTION:element-multisocketsink
+ * @title: multisocketsink
* @see_also: tcpserversink
*
* This plugin writes incoming data to a set of file descriptors. The
- * file descriptors can be added to multisocketsink by emitting the #GstMultiSocketSink::add signal.
+ * file descriptors can be added to multisocketsink by emitting the #GstMultiSocketSink::add signal.
* For each descriptor added, the #GstMultiSocketSink::client-added signal will be called.
*
* A client can also be added with the #GstMultiSocketSink::add-full signal
- * that allows for more control over what and how much data a client
+ * that allows for more control over what and how much data a client
* initially receives.
*
* Clients can be removed from multisocketsink by emitting the #GstMultiSocketSink::remove signal. For
* Note that multisocketsink still has a reference to the file descriptor when the
* #GstMultiSocketSink::client-removed signal is emitted, so that "get-stats" can be performed on
* the descriptor; it is therefore not safe to close the file descriptor in
- * the #GstMultiSocketSink::client-removed signal handler, and you should use the
+ * the #GstMultiSocketSink::client-removed signal handler, and you should use the
* #GstMultiSocketSink::client-fd-removed signal to safely close the fd.
*
* Multisocketsink internally keeps a queue of the incoming buffers and uses a
* speeds.
*
* When adding a client to multisocketsink, the #GstMultiSocketSink:sync-method property will define
- * which buffer in the queued buffers will be sent first to the client. Clients
- * can be sent the most recent buffer (which might not be decodable by the
- * client if it is not a keyframe), the next keyframe received in
+ * which buffer in the queued buffers will be sent first to the client. Clients
+ * can be sent the most recent buffer (which might not be decodable by the
+ * client if it is not a keyframe), the next keyframe received in
* multisocketsink (which can take some time depending on the keyframe rate), or the
- * last received keyframe (which will cause a simple burst-on-connect).
+ * last received keyframe (which will cause a simple burst-on-connect).
* Multisocketsink will always keep at least one keyframe in its internal buffers
* when the sync-mode is set to latest-keyframe.
*
* There are additional values for the #GstMultiSocketSink:sync-method
* property to allow finer control over burst-on-connect behaviour. By selecting
* the 'burst' method a minimum burst size can be chosen, 'burst-keyframe'
- * additionally requires that the burst begin with a keyframe, and
+ * additionally requires that the burst begin with a keyframe, and
* 'burst-with-keyframe' attempts to burst beginning with a keyframe, but will
* prefer a minimum burst size even if it requires not starting with a keyframe.
*
* Multisocketsink can be instructed to keep at least a minimum amount of data
- * expressed in time or byte units in its internal queues with the
+ * expressed in time or byte units in its internal queues with the
* #GstMultiSocketSink:time-min and #GstMultiSocketSink:bytes-min properties respectively.
- * These properties are useful if the application adds clients with the
+ * These properties are useful if the application adds clients with the
* #GstMultiSocketSink::add-full signal to make sure that a burst connect can
- * actually be honored.
+ * actually be honored.
*
* When streaming data, clients are allowed to read at a different rate than
* the rate at which multisocketsink receives data. If the client is reading too
* fast, no data will be send to the client until multisocketsink receives more
- * data. If the client, however, reads too slowly, data for that client will be
- * queued up in multisocketsink. Two properties control the amount of data
- * (buffers) that is queued in multisocketsink: #GstMultiSocketSink:buffers-max and
+ * data. If the client, however, reads too slowly, data for that client will be
+ * queued up in multisocketsink. Two properties control the amount of data
+ * (buffers) that is queued in multisocketsink: #GstMultiSocketSink:buffers-max and
* #GstMultiSocketSink:buffers-soft-max. A client that falls behind by
* #GstMultiSocketSink:buffers-max is removed from multisocketsink forcibly.
*
* RESYNC_KEYFRAME positions the client at the most recent keyframe in the
* buffer queue.
*
- * multisocketsink will by default synchronize on the clock before serving the
- * buffers to the clients. This behaviour can be disabled by setting the sync
+ * multisocketsink will by default synchronize on the clock before serving the
+ * buffers to the clients. This behaviour can be disabled by setting the sync
* property to FALSE. Multisocketsink will by default not do QoS and will never
* drop late buffers.
*/
/**
* SECTION:element-socketsrc
+ * @title: socketsrc
*
* Receive data from a socket.
*
/**
* SECTION:element-tcpclientsink
+ * @title: tcpclientsink
* @see_also: #tcpclientsink
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* # server:
* nc -l -p 3000
* # client:
* gst-launch-1.0 fdsink fd=1 ! tcpclientsink port=3000
- * ]| everything you type in the client is shown on the server (fd=1 means
+ * ]|
+ * everything you type in the client is shown on the server (fd=1 means
* standard input which is the command line input file descriptor)
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-tcpclientsrc
+ * @title: tcpclientsrc
* @see_also: #tcpclientsink
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* # server:
* nc -l -p 3000
* # client:
* gst-launch-1.0 tcpclientsrc port=3000 ! fdsink fd=2
- * ]| everything you type in the server is shown on the client.
+ * ]|
+ * everything you type in the server is shown on the client.
* If you want to detect network failures and/or limit the time your tcp client
* keeps waiting for data from server setting a timeout value can be useful.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-tcpserversink
+ * @title: tcpserversink
* @see_also: #multifdsink
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* # server:
* gst-launch-1.0 fdsrc fd=1 ! tcpserversink port=3000
* # client:
* gst-launch-1.0 tcpclientsrc port=3000 ! fdsink fd=2
- * ]|
- * </refsect2>
+ * ]|
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-tcpserversrc
+ * @title: tcpserversrc
* @see_also: #tcpserversink
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* # server:
* gst-launch-1.0 tcpserversrc port=3000 ! fdsink fd=2
* # client:
* gst-launch-1.0 fdsrc fd=1 ! tcpclientsink port=3000
- * ]|
- * </refsect2>
+ * ]|
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-videoconvert
+ * @title: videoconvert
*
* Convert video frames between a great variety of video formats.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! video/x-raw,format=YUY2 ! videoconvert ! autovideosink
- * ]| This will output a test video (generated in YUY2 format) in a video
+ * ]|
+ * This will output a test video (generated in YUY2 format) in a video
* window. If the video sink selected does not support YUY2 videoconvert will
* automatically convert the video to a format understood by the video sink.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-videorate
+ * @title: videorate
*
* This element takes an incoming stream of timestamped video frames.
* It will produce a perfect stream that matches the source pad's framerate.
* certain factor. It must not be confused with framerate. Think of rate as
* speed and framerate as flow.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/video.ogg ! videoconvert ! videoscale ! videorate ! video/x-raw,framerate=15/1 ! autovideosink
- * ]| Decode a video file and adjust the framerate to 15 fps before playing.
+ * ]|
+ * Decode a video file and adjust the framerate to 15 fps before playing.
* To create a test Ogg/Theora file refer to the documentation of theoraenc.
* |[
* gst-launch-1.0 -v v4l2src ! videorate ! video/x-raw,framerate=25/2 ! theoraenc ! oggmux ! filesink location=recording.ogg
- * ]| Capture video from a V4L device, and adjust the stream to 12.5 fps before
+ * ]|
+ * Capture video from a V4L device, and adjust the stream to 12.5 fps before
* encoding to Ogg/Theora.
* |[
* gst-launch-1.0 -v uridecodebin uri=file:///path/to/video.ogg ! videoconvert ! videoscale ! videorate ! video/x-raw,framerate=1/5 ! jpegenc ! multifilesink location=snapshot-%05d.jpg
- * ]| Decode a video file and save a snapshot every 5 seconds as consecutively numbered jpeg file.
- * </refsect2>
+ * ]|
+ * Decode a video file and save a snapshot every 5 seconds as consecutively numbered jpeg file.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-videoscale
+ * @title: videoscale
* @see_also: videorate, videoconvert
*
* This element resizes video frames. By default the element will try to
* RGB formats and is therefore generally able to operate anywhere in a
* pipeline.
*
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=videotestsrc.ogg ! oggdemux ! theoradec ! videoconvert ! videoscale ! autovideosink
- * ]| Decode an Ogg/Theora and display the video. If the video sink chosen
+ * ]|
+ * Decode an Ogg/Theora and display the video. If the video sink chosen
* cannot perform scaling, the video scaling will be performed by videoscale
* when you resize the video window.
* To create the test Ogg/Theora file refer to the documentation of theoraenc.
* |[
* gst-launch-1.0 -v filesrc location=videotestsrc.ogg ! oggdemux ! theoradec ! videoconvert ! videoscale ! video/x-raw,width=100 ! autovideosink
- * ]| Decode an Ogg/Theora and display the video with a width of 100.
- * </refsect2>
+ * ]|
+ * Decode an Ogg/Theora and display the video with a width of 100.
+ *
*/
/*
/**
* SECTION:element-videotestsrc
+ * @title: videotestsrc
*
* The videotestsrc element is used to produce test video data in a wide variety
* of formats. The video test data produced can be controlled with the "pattern"
* property.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc pattern=snow ! video/x-raw,width=1280,height=720 ! autovideosink
- * ]| Shows random noise in a video window.
- * </refsect2>
+ * ]|
+ * Shows random noise in a video window.
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-volume
+ * @title: volume
*
* The volume element changes the volume of the audio data.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 -v -m audiotestsrc ! volume volume=0.5 ! level ! fakesink silent=TRUE
- * ]| This pipeline shows that the level of audiotestsrc has been halved
+ * ]|
+ * This pipeline shows that the level of audiotestsrc has been halved
* (peak values are around -6 dB and RMS around -9 dB) compared to
* the same pipeline without the volume element.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-ximagesink
+ * @title: ximagesink
*
* XImageSink renders video frames to a drawable (XWindow) on a local or remote
* display. This element can receive a Window ID from the application through
* drawable. If no Window ID was provided by the application, the element will
* create its own internal window and render into it.
*
- * <refsect2>
- * <title>Scaling</title>
- * <para>
+ * ## Scaling
+ *
* As standard XImage rendering to a drawable is not scaled, XImageSink will use
* reverse caps negotiation to try to get scaled video frames for the drawable.
* This is accomplished by asking the peer pad if it accepts some different caps
* which in most cases implies that there is a scaling element in the pipeline,
- * or that an element generating the video frames can generate them with a
+ * or that an element generating the video frames can generate them with a
* different geometry. This mechanism is handled during buffer allocations, for
* each allocation request the video sink will check the drawable geometry, look
* at the #GstXImageSink:force-aspect-ratio property, calculate the geometry of
* desired video frames and then check that the peer pad accept those new caps.
* If it does it will then allocate a buffer in video memory with this new
* geometry and return it with the new caps.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Events</title>
- * <para>
+ *
+ * ## Events
+ *
* XImageSink creates a thread to handle events coming from the drawable. There
- * are several kind of events that can be grouped in 2 big categories: input
+ * are several kind of events that can be grouped in 2 big categories: input
* events and window state related events. Input events will be translated to
* navigation events and pushed upstream for other elements to react on them.
* This includes events such as pointer moves, key press/release, clicks etc...
* is not flowing (GST_STATE_PAUSED). That means that even when the element is
* paused, it will receive expose events from the drawable and draw the latest
* frame with correct borders/aspect-ratio.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Pixel aspect ratio</title>
- * <para>
+ *
+ * ## Pixel aspect ratio
+ *
* When changing state to GST_STATE_READY, XImageSink will open a connection to
* the display specified in the #GstXImageSink:display property or the default
- * display if nothing specified. Once this connection is open it will inspect
- * the display configuration including the physical display geometry and
+ * display if nothing specified. Once this connection is open it will inspect
+ * the display configuration including the physical display geometry and
* then calculate the pixel aspect ratio. When caps negotiation will occur, the
- * video sink will set the calculated pixel aspect ratio on the caps to make
+ * video sink will set the calculated pixel aspect ratio on the caps to make
* sure that incoming video frames will have the correct pixel aspect ratio for
* this display. Sometimes the calculated pixel aspect ratio can be wrong, it is
* then possible to enforce a specific pixel aspect ratio using the
* #GstXImageSink:pixel-aspect-ratio property.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Examples</title>
+ *
+ * ## Examples
* |[
* gst-launch-1.0 -v videotestsrc ! queue ! ximagesink
- * ]| A pipeline to test reverse negotiation. When the test video signal appears
+ * ]|
+ * A pipeline to test reverse negotiation. When the test video signal appears
* you can resize the window and see that scaled buffers of the desired size are
* going to arrive with a short delay. This illustrates how buffers of desired
* size are allocated along the way. If you take away the queue, scaling will
* happen almost immediately.
* |[
* gst-launch-1.0 -v videotestsrc ! navigationtest ! videoconvert ! ximagesink
- * ]| A pipeline to test navigation events.
+ * ]|
+ * A pipeline to test navigation events.
* While moving the mouse pointer over the test signal you will see a black box
- * following the mouse pointer. If you press the mouse button somewhere on the
+ * following the mouse pointer. If you press the mouse button somewhere on the
* video and release it somewhere else a green box will appear where you pressed
* the button and a red one where you released it. (The navigationtest element
* is part of gst-plugins-good.)
* |[
* gst-launch-1.0 -v videotestsrc ! video/x-raw, pixel-aspect-ratio=(fraction)4/3 ! videoscale ! ximagesink
- * ]| This is faking a 4/3 pixel aspect ratio caps on video frames produced by
+ * ]|
+ * This is faking a 4/3 pixel aspect ratio caps on video frames produced by
* videotestsrc, in most cases the pixel aspect ratio of the display will be
- * 1/1. This means that videoscale will have to do the scaling to convert
+ * 1/1. This means that videoscale will have to do the scaling to convert
* incoming frames to a size that will match the display pixel aspect ratio
- * (from 320x240 to 320x180 in this case). Note that you might have to escape
+ * (from 320x240 to 320x180 in this case). Note that you might have to escape
* some characters for your shell like '\(fraction\)'.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
* @pool_lock: used to protect the buffer pool
* @buffer_pool: a list of #GstXImageBuffer that could be reused at next buffer
* allocation call
- * @synchronous: used to store if XSynchronous should be used or not (for
+ * @synchronous: used to store if XSynchronous should be used or not (for
* debugging purpose only)
* @keep_aspect: used to remember if reverse negotiation scaling should respect
* aspect ratio
/**
* SECTION:element-xvimagesink
+ * @title: xvimagesink
*
* XvImageSink renders video frames to a drawable (XWindow) on a local display
* using the XVideo extension. Rendering to a remote display is theoretically
* application, the element will create its own internal window and render
* into it.
*
- * <refsect2>
- * <title>Scaling</title>
- * <para>
+ * ## Scaling
+ *
* The XVideo extension, when it's available, handles hardware accelerated
* scaling of video frames. This means that the element will just accept
* incoming video frames no matter their geometry and will then put them to the
* drawable scaling them on the fly. Using the #GstXvImageSink:force-aspect-ratio
* property it is possible to enforce scaling with a constant aspect ratio,
* which means drawing black borders around the video frame.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Events</title>
- * <para>
+ *
+ * ## Events
+ *
* XvImageSink creates a thread to handle events coming from the drawable. There
* are several kind of events that can be grouped in 2 big categories: input
* events and window state related events. Input events will be translated to
* is not flowing (GST_STATE_PAUSED). That means that even when the element is
* paused, it will receive expose events from the drawable and draw the latest
* frame with correct borders/aspect-ratio.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Pixel aspect ratio</title>
- * <para>
+ *
+ * ## Pixel aspect ratio
+ *
* When changing state to GST_STATE_READY, XvImageSink will open a connection to
* the display specified in the #GstXvImageSink:display property or the
* default display if nothing specified. Once this connection is open it will
* Sometimes the calculated pixel aspect ratio can be wrong, it is
* then possible to enforce a specific pixel aspect ratio using the
* #GstXvImageSink:pixel-aspect-ratio property.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Examples</title>
+ *
+ * ## Examples
* |[
* gst-launch-1.0 -v videotestsrc ! xvimagesink
- * ]| A pipeline to test hardware scaling.
+ * ]|
+ * A pipeline to test hardware scaling.
* When the test video signal appears you can resize the window and see that
* video frames are scaled through hardware (no extra CPU cost). By default
* the image will never be distorted when scaled, instead black borders will
* be added if needed.
* |[
* gst-launch-1.0 -v videotestsrc ! xvimagesink force-aspect-ratio=false
- * ]| Same pipeline with #GstXvImageSink:force-aspect-ratio property set to
+ * ]|
+ * Same pipeline with #GstXvImageSink:force-aspect-ratio property set to
* false. You can observe that no borders are drawn around the scaled image
* now and it will be distorted to fill the entire frame instead of respecting
* the aspect ratio.
* |[
* gst-launch-1.0 -v videotestsrc ! navigationtest ! xvimagesink
- * ]| A pipeline to test navigation events.
+ * ]|
+ * A pipeline to test navigation events.
* While moving the mouse pointer over the test signal you will see a black box
* following the mouse pointer. If you press the mouse button somewhere on the
* video and release it somewhere else a green box will appear where you pressed
* image area
* |[
* gst-launch-1.0 -v videotestsrc ! video/x-raw, pixel-aspect-ratio=4/3 ! xvimagesink
- * ]| This is faking a 4/3 pixel aspect ratio caps on video frames produced by
+ * ]|
+ * This is faking a 4/3 pixel aspect ratio caps on video frames produced by
* videotestsrc, in most cases the pixel aspect ratio of the display will be
* 1/1. This means that XvImageSink will have to do the scaling to convert
* incoming frames to a size that will match the display pixel aspect ratio
* (from 320x240 to 320x180 in this case).
* |[
* gst-launch-1.0 -v videotestsrc ! xvimagesink hue=100 saturation=-100 brightness=100
- * ]| Demonstrates how to use the colorbalance interface.
- * </refsect2>
+ * ]|
+ * Demonstrates how to use the colorbalance interface.
+ *
*/
/* for developers: there are two useful tools : xvinfo and xvattr */
projects / platform / upstream / gstreamer.git / commitdiff