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docs: Port all docstring to gtk-doc markdown

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This commit is contained in:
Thibault Saunier 2017-03-08 15:01:13 -03:00
parent 90f766cc51
commit 78022a6e0c
257 changed files with 943 additions and 1141 deletions
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6
ext/assrender/gstassrender.c
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@ -20,15 +20,15 @@
/**
* SECTION:element-assrender
* @title: assrender
*
* Renders timestamped SSA/ASS subtitles on top of a video stream.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v filesrc location=/path/to/mkv ! matroskademux name=d ! queue ! mpegaudioparse ! mpg123audiodec ! audioconvert ! autoaudiosink d. ! queue ! h264parse ! avdec_h264 ! videoconvert ! r. d. ! queue ! "application/x-ass" ! assrender name=r ! videoconvert ! autovideosink
* ]| This pipeline demuxes a Matroska file with h.264 video, MP3 audio and embedded ASS subtitles and renders the subtitles on top of the video.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/bs2b/gstbs2b.c
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@ -20,17 +20,17 @@
/**
* SECTION:element-bs2b
* @title: bs2b
*
* Improve headphone listening of stereo audio records using the bs2b library.
* It does so by mixing the left and right channel in a way that simulates
* a stereo speaker setup while using headphones.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 audiotestsrc ! "audio/x-raw,channel-mask=(bitmask)0x1" ! interleave name=i ! bs2b ! autoaudiosink audiotestsrc freq=330 ! "audio/x-raw,channel-mask=(bitmask)0x2" ! i.
* ]| Play two independent sine test sources and crossfeed them.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/chromaprint/gstchromaprint.c
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@ -23,18 +23,18 @@
/**
* SECTION:element-chromaprint
* @title: chromaprint
*
* The chromaprint element calculates an acoustic fingerprint for an
* audio stream which can be used to identify a song and look up
* further metadata from the <ulink url="http://acoustid.org/">Acoustid</ulink>
* and Musicbrainz databases.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -m uridecodebin uri=file:///path/to/song.ogg ! audioconvert ! chromaprint ! fakesink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/curl/gstcurlbasesink.c
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@ -19,14 +19,14 @@
/**
* SECTION:element-curlsink
* @title: curlsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* a server (e.g. a HTTP/FTP server).
*
* <refsect2>
* <title>Example launch line (upload a JPEG file to an HTTP server)</title>
* ## Example launch line (upload a JPEG file to an HTTP server)
* |[
* gst-launch-1.0 filesrc location=image.jpg ! jpegparse ! curlsink \
* file-name=image.jpg \
@ -35,7 +35,7 @@
* content-type=image/jpeg \
* use-content-length=false
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

7
ext/curl/gstcurlfilesink.c
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@ -19,21 +19,20 @@
/**
* SECTION:element-curlfilesink
* @title: curlfilesink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* a local or network drive.
*
* <refsect2>
* <title>Example launch line (upload a JPEG file to /home/test/images
* directory)</title>
* ## Example launch line (upload a JPEG file to /home/test/images directory)
* |[
* gst-launch-1.0 filesrc location=image.jpg ! jpegparse ! curlfilesink \
* file-name=image.jpg \
* location=file:///home/test/images/
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

10
ext/curl/gstcurlftpsink.c
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@ -19,21 +19,23 @@
/**
* SECTION:element-curlftpsink
* @title: curlftpsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* an FTP server.
*
* <refsect2>
* <title>Example launch line (upload a JPEG file to /home/test/images
* directory)</title>
* ## Example launch line
*
* Upload a JPEG file to /home/test/images * directory)
*
* |[
* gst-launch-1.0 filesrc location=image.jpg ! jpegparse ! curlftpsink \
* file-name=image.jpg \
* location=ftp://192.168.0.1/images/
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

8
ext/curl/gstcurlhttpsink.c
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@ -19,14 +19,17 @@
/**
* SECTION:element-curlhttpsink
* @title: curlhttpsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* an HTTP server.
*
* <refsect2>
* <title>Example launch line (upload a JPEG file to an HTTP server)</title>
* ## Example launch line
*
* Upload a JPEG file to an HTTP server.
*
* |[
* gst-launch-1.0 filesrc location=image.jpg ! jpegparse ! curlhttpsink \
* file-name=image.jpg \
@ -35,7 +38,6 @@
* content-type=image/jpeg \
* use-content-length=false
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H

9
ext/curl/gstcurlsftpsink.c
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@ -19,14 +19,17 @@
/**
* SECTION:element-curlsftpsink
* @title: curlsftpsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* a SFTP (SSH File Transfer Protocol) server.
*
* <refsect2>
* <title>Example launch line (upload a file to /home/john/sftp_tests/)</title>
* ## Example launch line
*
* Upload a file to /home/john/sftp_tests/
*
* |[
* gst-launch-1.0 filesrc location=/home/jdoe/some.file ! curlsftpsink \
* file-name=some.file.backup \
@ -36,8 +39,6 @@
* ssh-priv-keyfile=/home/jdoe/.ssh/id_rsa \
* create-dirs=TRUE
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

9
ext/curl/gstcurlsmtpsink.c
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@ -19,14 +19,17 @@
/**
* SECTION:element-curlsink
* @title: curlsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*
* This is a network sink that uses libcurl as a client to upload data to
* an SMTP server.
*
* <refsect2>
* <title>Example launch line (upload a JPEG file to an SMTP server)</title>
* ## Example launch line
*
* Upload a JPEG file to an SMTP server.
*
* |[
* gst-launch-1.0 filesrc location=image.jpg ! jpegparse ! curlsmtpsink \
* file-name=image.jpg \
@ -38,7 +41,7 @@
* use-ssl=TRUE \
* insecure=TRUE
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
ext/curl/gstcurlsshsink.c
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@ -19,6 +19,7 @@
/**
* SECTION:element-curlsshsink
* @title: curlsshsink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*

1
ext/curl/gstcurltlssink.c
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@ -19,6 +19,7 @@
/**
* SECTION:element-curltlssink
* @title: curltlssink
* @short_description: sink that uploads data to a server using libcurl
* @see_also:
*

6
ext/daala/gstdaaladec.c
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@ -23,6 +23,7 @@
/**
* SECTION:element-daaladec
* @title: daaladec
* @see_also: daalaenc, oggdemux
*
* This element decodes daala streams into raw video
@ -30,13 +31,12 @@
* video codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
* Foundation</ulink>.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 -v filesrc location=videotestsrc.ogg ! oggdemux ! daaladec ! xvimagesink
* ]| This example pipeline will decode an ogg stream and decodes the daala video. Refer to
* the daalaenc example to create the ogg file.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/daala/gstdaalaenc.c
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@ -23,6 +23,7 @@
/**
* SECTION:element-daalaenc
* @title: daalaenc
* @see_also: daaladec, oggmux
*
* This element encodes raw video into a Daala stream.
@ -30,14 +31,13 @@
* video codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
* Foundation</ulink>.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 -v videotestsrc num-buffers=1000 ! daalaenc ! oggmux ! filesink location=videotestsrc.ogg
* ]| This example pipeline will encode a test video source to daala muxed in an
* ogg container. Refer to the daaladec documentation to decode the create
* stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

3
ext/dash/gstdashdemux.c
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@ -29,9 +29,10 @@
*/
/**
* SECTION:element-dashdemux
* @title: dashdemux
*
* DASH demuxer element.
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 playbin uri="http://www-itec.uni-klu.ac.at/ftp/datasets/mmsys12/RedBullPlayStreets/redbull_4s/RedBullPlayStreets_4s_isoffmain_DIS_23009_1_v_2_1c2_2011_08_30.mpd"
* ]|

6
ext/dc1394/gstdc1394src.c
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@ -20,6 +20,7 @@
/**
* SECTION:element-dc1394src
* @title: dc1394src
*
* Source for IIDC (Instrumentation & Industrial Digital Camera) firewire
* cameras. If several cameras are connected to the system, the desired one
@ -31,8 +32,7 @@
* corresponding video formats are exposed in the capabilities.
* The Bayer pattern is left unspecified.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 -v dc1394src ! videoconvert ! autovideosink
* ]| Capture and display frames from the first camera available in the system.
@ -41,7 +41,7 @@
* ! "video/x-bayer,format=gbrg,width=1280,height=960,framerate=15/2" \
* ! bayer2rgb ! videoconvert ! autovideosink
* ]| Capture and display frames from a specific camera in the desired format.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

34
ext/directfb/dfbvideosink.c
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@ -20,20 +20,17 @@
/**
* SECTION:element-dfbvideosink
* @title: dfbvideosink
*
* DfbVideoSink renders video frames using the
* <ulink url="http://www.directfb.org/">DirectFB</ulink> library.
* Rendering can happen in two different modes :
* <itemizedlist>
* <listitem>
* <para>
* Standalone: this mode will take complete control of the monitor forcing
*
* * Standalone: this mode will take complete control of the monitor forcing
* <ulink url="http://www.directfb.org/">DirectFB</ulink> to fullscreen layout.
* This is convenient to test using the gst-launch-1.0 command line tool or
* other simple applications. It is possible to interrupt playback while
* being in this mode by pressing the Escape key.
* </para>
* <para>
* This mode handles navigation events for every input device supported by
* the <ulink url="http://www.directfb.org/">DirectFB</ulink> library, it will
* look for available video modes in the fb.modes file and try to switch
@ -41,17 +38,12 @@
* hardware acceleration capabilities the element will handle scaling or not.
* If no acceleration is available it will do clipping or centering of the
* video frames respecting the original aspect ratio.
* </para>
* </listitem>
* <listitem>
* <para>
* Embedded: this mode will render video frames in a
*
* * Embedded: this mode will render video frames in a
* #GstDfbVideoSink:surface provided by the
* application developer. This is a more advanced usage of the element and
* it is required to integrate video playback in existing
* <ulink url="http://www.directfb.org/">DirectFB</ulink> applications.
* </para>
* <para>
* When using this mode the element just renders to the
* #GstDfbVideoSink:surface provided by the
* application, that means it won't handle navigation events and won't resize
@ -59,27 +51,21 @@
* frames geometry. Application has to implement the necessary code to grab
* informations about the negotiated geometry and resize there
* #GstDfbVideoSink:surface accordingly.
* </para>
* </listitem>
* </itemizedlist>
*
* For both modes the element implements a buffer pool allocation system to
* optimize memory allocation time and handle reverse negotiation. Indeed if
* you insert an element like videoscale in the pipeline the video sink will
* negotiate with it to try get a scaled video for either the fullscreen layout
* or the application provided external #GstDfbVideoSink:surface.
*
* <refsect2>
* <title>Example application</title>
* <para>
* ## Example application
*
* <include xmlns="http://www.w3.org/2003/XInclude" href="element-dfb-example.xml" />
* </para>
* </refsect2>
* <refsect2>
* <title>Example pipelines</title>
*
* ## Example pipelines
* |[
* gst-launch-1.0 -v videotestsrc ! dfbvideosink hue=20000 saturation=40000 brightness=25000
* ]| test the colorbalance interface implementation in dfbvideosink
* </refsect2>
*/
#ifdef HAVE_CONFIG_H

6
ext/dts/gstdtsdec.c
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@ -20,18 +20,18 @@
/**
* SECTION:element-dtsdec
* @title: dtsdec
*
* Digital Theatre System (DTS) audio decoder
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! dtsdec ! audioresample ! audioconvert ! alsasink
* ]| Play a DTS audio track from a dvd.
* |[
* gst-launch-1.0 filesrc location=abc.dts ! dtsdec ! audioresample ! audioconvert ! alsasink
* ]| Decode a standalone file and play it.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/faac/gstfaac.c
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@ -20,16 +20,16 @@
/**
* SECTION:element-faac
* @title: faac
* @see_also: faad
*
* faac encodes raw audio to AAC (MPEG-4 part 3) streams.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc wave=sine num-buffers=100 ! audioconvert ! faac ! matroskamux ! filesink location=sine.mkv
* ]| Encode a sine beep as aac and write to matroska container.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/faad/gstfaad.c
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@ -20,19 +20,19 @@
/**
* SECTION:element-faad
* @title: faad
* @seealso: faac
*
* faad decodes AAC (MPEG-4 part 3) stream.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 filesrc location=example.mp4 ! qtdemux ! faad ! audioconvert ! audioresample ! autoaudiosink
* ]| Play aac from mp4 file.
* |[
* gst-launch-1.0 filesrc location=example.adts ! faad ! audioconvert ! audioresample ! autoaudiosink
* ]| Play standalone aac bitstream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/fluidsynth/gstfluiddec.c
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@ -21,19 +21,19 @@
/**
* SECTION:element-fluiddec
* @title: fluiddec
* @see_also: timidity, wildmidi
*
* This element renders midi-events as audio streams using
* <ulink url="http://fluidsynth.sourceforge.net//">Fluidsynth</ulink>.
* It offers better sound quality compared to the timidity or wildmidi element.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 filesrc location=song.mid ! midiparse ! fluiddec ! pulsesink
* ]| This example pipeline will parse the midi and render to raw audio which is
* played via pulseaudio.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglbumper.c
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@ -21,16 +21,16 @@
/**
* SECTION:element-glbumper
* @title: glbumper
*
* Bump mapping using the normal method.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 -v videotestsrc ! glupload ! glbumper location=normalmap.bmp ! glimagesink
* ]| A pipeline to test normal mapping.
* FBO (Frame Buffer Object) and GLSL (OpenGL Shading Language) are required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglcolorbalance.c
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@ -22,16 +22,16 @@
/**
* SECTION:element-glcolorbalance
* @title: glcolorbalance
*
* Adjusts brightness, contrast, hue, saturation on a video stream.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 videotestsrc ! glupload ! glcolorbalance saturation=0.0 ! glcolorconvert ! gldownload ! ximagesink
* ]| This pipeline converts the image to black and white by setting the
* saturation to 0.0.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

14
ext/gl/gstglcolorscale.c
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@ -20,17 +20,15 @@
/**
* SECTION:element-glcolorscale
* @title: glcolorscale
*
* video frame scaling and colorspace conversion.
*
* <refsect2>
* <title>Scaling and Color space conversion</title>
* <para>
* ## Scaling and Color space conversion
*
* Equivalent to glupload ! gldownload.
* </para>
* </refsect2>
* <refsect2>
* <title>Examples</title>
*
* ## Examples
* |[
* gst-launch-1.0 -v videotestsrc ! video/x-raw ! glcolorscale ! ximagesink
* ]| A pipeline to test colorspace conversion.
@ -40,7 +38,7 @@
* video/x-raw, width=320, height=240, format=YV12 ! videoconvert ! autovideosink
* ]| A pipeline to test hardware scaling and colorspace conversion.
* FBO and GLSL are required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstgldeinterlace.c
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@ -20,16 +20,16 @@
/**
* SECTION:element-deinterlace
* @title: deinterlace
*
* Deinterlacing using based on fragment shaders.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glupload ! gldeinterlace ! glimagesink
* ]|
* FBO (Frame Buffer Object) and GLSL (OpenGL Shading Language) are required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstgldifferencematte.c
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@ -20,16 +20,16 @@
/**
* SECTION:element-gldifferencematte.
* @title: gldifferencematte.
*
* Saves a background frame and replace it with a pixbuf.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glupload ! gldifferencemate location=backgroundimagefile ! glimagesink
* ]|
* FBO (Frame Buffer Object) and GLSL (OpenGL Shading Language) are required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstgleffects.c
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@ -20,16 +20,16 @@
/**
* SECTION:element-gleffects.
* @title: gleffects.
*
* GL Shading Language effects.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glupload ! gleffects effect=5 ! glimagesink
* ]|
* FBO (Frame Buffer Object) and GLSL (OpenGL Shading Language) are required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

14
ext/gl/gstglfilterapp.c
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@ -20,20 +20,18 @@
/**
* SECTION:element-glfilterapp
* @title: glfilterapp
*
* The resize and redraw callbacks can be set from a client code.
*
* <refsect2>
* <title>CLient callbacks</title>
* <para>
* ## CLient callbacks
*
* The graphic scene can be written from a client code through the
* two glfilterapp properties.
* </para>
* </refsect2>
* <refsect2>
* <title>Examples</title>
*
* ## Examples
* see gst-plugins-gl/tests/examples/generic/recordgraphic
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglfiltercube.c
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@ -20,11 +20,11 @@
/**
* SECTION:element-glfiltercube
* @title: glfiltercube
*
* The resize and redraw callbacks can be set from a client code.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 -v videotestsrc ! glfiltercube ! glimagesink
* ]| A pipeline to mpa textures on the 6 cube faces..
@ -37,7 +37,7 @@
* gst-launch-1.0 -v videotestsrc ! video/x-raw, width=640, height=480 ! glfiltercube ! glimagesink
* ]| Resize scene before drawing the cube.
* The scene size is greater than the input video size.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglfilterglass.c
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@ -21,11 +21,11 @@
/**
* SECTION:element-glfilterglass
* @title: glfilterglass
*
* Map textures on moving glass.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 -v videotestsrc ! glfilterglass ! glimagesink
* ]| A pipeline inspired from http://www.mdk.org.pl/2007/11/17/gl-colorspace-conversions
@ -33,7 +33,7 @@
* |[
* gst-launch-1.0 -v videotestsrc ! glfilterglass ! video/x-raw, width=640, height=480 ! glimagesink
* ]| The scene is greater than the input size.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglfilterreflectedscreen.c
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@ -20,15 +20,15 @@
/**
* SECTION:element-glfilterreflectedscreen
* @title: glfilterreflectedscreen
*
* Map Video Texture upon a screen, on a reflecting surface
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glupload ! glfilterreflectedscreen ! glimagesink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

5
ext/gl/gstglfiltershader.c
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@ -21,11 +21,11 @@
/**
* SECTION:element-glshader
* @title: glshader
*
* OpenGL fragment shader filter
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glupload ! glshader fragment="\"`cat myshader.frag`\"" ! glimagesink
* ]|
@ -51,7 +51,6 @@
* }
* ]|
*
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"

22
ext/gl/gstglimagesink.c
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@ -23,6 +23,7 @@
/**
* SECTION:element-glimagesink
* @title: glimagesink
*
* glimagesink renders video frames to a drawable on a local or remote
* display using OpenGL. This element can receive a Window ID from the
@ -34,28 +35,23 @@
* See the #GstGLDisplay documentation for a list of environment variables that
* can override window/platform detection.
*
* <refsect2>
* <title>Scaling</title>
* <para>
* ## Scaling
*
* Depends on the driver, OpenGL 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 #GstGLImageSink: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
*
* Through the gl thread, glimagesink handle some events coming from the drawable
* to manage its appearance even when the data 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>Examples</title>
*
* ## Examples
* |[
* gst-launch-1.0 -v videotestsrc ! video/x-raw ! glimagesink
* ]| A pipeline to test hardware scaling.
@ -80,7 +76,7 @@
* ]| The graphic FPS scene can be greater than the input video FPS.
* The graphic scene can be written from a client code through the
* two glfilterapp properties.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglmosaic.c
View file

@ -20,14 +20,14 @@
/**
* SECTION:element-glmosaic
* @title: glmosaic
*
* glmixer sub element. N gl sink pads to 1 source pad.
* N + 1 OpenGL contexts shared together.
* N <= 6 because the rendering is more a like a cube than a mosaic
* Each opengl input stream is rendered on a cube face
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! video/x-raw, format=YUY2 ! queue ! glmosaic name=m ! glimagesink \
* videotestsrc pattern=12 ! video/x-raw, format=I420, framerate=5/1, width=100, height=200 ! queue ! m. \
@ -37,7 +37,7 @@
* videotestsrc ! gleffects effect=6 ! queue ! m.
* ]|
* FBO (Frame Buffer Object) is required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstgloverlay.c
View file

@ -20,16 +20,16 @@
/**
* SECTION:element-gloverlay
* @title: gloverlay
*
* Overlay GL video texture with a PNG image
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! gloverlay location=image.jpg ! glimagesink
* ]|
* FBO (Frame Buffer Object) is required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglstereomix.c
View file

@ -23,6 +23,7 @@
/**
* SECTION:element-glstereomix
* @title: glstereomix
*
* Combine 2 input streams to produce a stereoscopic output
* stream. Input views are taken from the left pad and right pad
@ -34,8 +35,7 @@
* The multiview representation on the output is chosen according to
* the downstream caps.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 -v videotestsrc pattern=ball name=left \
* videotestsrc name=right glstereomix name=mix \
@ -55,7 +55,7 @@
* mp4mux ! progressreport ! filesink location=output.mp4
* ]| Mix the input from a camera to the left view, and videotestsrc to the right view,
* and encode as a top-bottom frame packed H.264 video.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"

6
ext/gl/gstglstereosplit.c
View file

@ -20,16 +20,16 @@
/**
* SECTION:element-glstereosplit
* @title: glstereosplit
*
* Receive a stereoscopic video stream and split into left/right
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glstereosplit name=s ! queue ! glimagesink s. ! queue ! glimagesink
* ]|
* FBO (Frame Buffer Object) and GLSL (OpenGL Shading Language) are required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

16
ext/gl/gstgltestsrc.c
View file

@ -23,21 +23,19 @@
/**
* SECTION:element-gltestsrc
* @title: gltestsrc
*
* <refsect2>
* <para>
* The gltestsrc element is used to produce test video texture.
* The video test produced can be controlled with the "pattern"
* property.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
*
* ## Example launch line
*
* |[
* gst-launch-1.0 -v gltestsrc pattern=smpte ! glimagesink
* </programlisting>
* ]|
* Shows original SMPTE color bars in a window.
* </para>
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstgltransformation.c
View file

@ -21,11 +21,11 @@
/**
* SECTION:element-gltransformation
* @title: gltransformation
*
* Transforms video on the GPU.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 gltestsrc ! gltransformation rotation-z=45 ! glimagesink
* ]| A pipeline to rotate by 45 degrees
@ -38,7 +38,7 @@
* |[
* gst-launch-1.0 gltestsrc ! gltransformation rotation-x=-45 ortho=True ! glimagesink
* ]| Rotate the video around the X-Axis by -45° with an orthographic projection
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglvideoflip.c
View file

@ -20,15 +20,15 @@
/**
* SECTION:element-glvideo_flip
* @title: glvideo_flip
*
* Transforms video on the GPU.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glupload ! glvideoflip method=clockwise ! glimagesinkelement
* ]| This pipeline flips the test image 90 degrees clockwise.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglvideomixer.c
View file

@ -20,13 +20,13 @@
/**
* SECTION:element-glvideomixer
* @title: glvideomixer
*
* Composites a number of streams into a single output scene using OpenGL in
* a similar fashion to compositor and videomixer. See the compositor plugin
* for documentation about the #GstGLVideoMixerPad properties.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 glvideomixer name=m ! glimagesink \
* videotestsrc ! video/x-raw, format=YUY2 ! glupload ! glcolorconvert ! m. \
@ -36,7 +36,7 @@
* videotestsrc ! glupload ! glfiltercube ! queue ! m. \
* videotestsrc ! glupload ! gleffects effect=6 ! queue ! m.gst-launch-1.0 glvideomixer name=m ! glimagesink \
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/gl/gstglviewconvert.c
View file

@ -22,14 +22,14 @@
/**
* SECTION:element-glviewconvert
* @title: glviewconvert
*
* Convert stereoscopic video between different representations using fragment shaders.
*
* The element can use either property settings or caps negotiation to choose the
* input and output formats to process.
*
* <refsect2>
* <title>Examples</title>
* ## Examples
* |[
* gst-launch-1.0 videotestsrc ! glupload ! glviewconvert ! glimagesink
* ]| Simple placebo example demonstrating identity passthrough of mono video
@ -39,7 +39,7 @@
* ]| Force re-interpretation of the input checkers pattern as a side-by-side stereoscopic
* image and display in glimagesink.
* FBO (Frame Buffer Object) and GLSL (OpenGL Shading Language) are required.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

12
ext/gl/gstopengl.c
View file

@ -23,20 +23,20 @@
/**
* SECTION:plugin-opengl
* @title: GstOpengl
*
* Cross-platform OpenGL plugin.
* <refsect2>
* <title>Debugging</title>
* </refsect2>
* <refsect2>
* <title>Examples</title>
*
* ## Debugging
*
* ## Examples
* |[
* gst-launch-1.0 --gst-debug=gldisplay:3 videotestsrc ! glimagesink
* ]| A debugging pipeline.
|[
* GST_DEBUG=gl*:6 gst-launch-1.0 videotestsrc ! glimagesink
* ]| A debugging pipelines related to shaders.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
ext/gtk/gstgtkbasesink.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:gtkgstsink
* @title: GstGtkBaseSink
*
*/

4
ext/gtk/gstgtkglsink.c
View file

@ -19,8 +19,8 @@
*/
/**
* SECTION:gstgtkglsink
*
* SECTION:element-gtkglsink
* @title: gtkglsink
*/
#ifdef HAVE_CONFIG_H

3
ext/gtk/gstgtksink.c
View file

@ -19,7 +19,8 @@
*/
/**
* SECTION:gtkgstsink
* SECTION:element-gtkgstsink
* @title: gtkgstsink
*
*/

1
ext/gtk/gtkgstglwidget.c
View file

@ -41,6 +41,7 @@
/**
* SECTION:gtkgstglwidget
* @title: GtkGstGlWidget
* @short_description: a #GtkGLArea that renders GStreamer video #GstBuffers
* @see_also: #GtkGLArea, #GstBuffer
*

1
ext/gtk/gtkgstwidget.c
View file

@ -29,6 +29,7 @@
/**
* SECTION:gtkgstwidget
* @title: GtkGstWidget
* @short_description: a #GtkWidget that renders GStreamer video #GstBuffers
* @see_also: #GtkDrawingArea, #GstBuffer
*

6
ext/hls/gsthlsdemux.c
View file

@ -26,15 +26,15 @@
*/
/**
* SECTION:element-hlsdemux
* @title: hlsdemux
*
* HTTP Live Streaming demuxer element.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 souphttpsrc location=http://devimages.apple.com/iphone/samples/bipbop/gear4/prog_index.m3u8 ! hlsdemux ! decodebin ! videoconvert ! videoscale ! autovideosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/hls/gsthlssink.c
View file

@ -19,15 +19,15 @@
/**
* SECTION:element-hlssink
* @title: hlssink
*
* HTTP Live Streaming sink/server
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 videotestsrc is-live=true ! x264enc ! mpegtsmux ! hlssink max-files=5
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"

6
ext/iqa/iqa.c
View file

@ -19,6 +19,7 @@
/**
* SECTION:element-iqa
* @title: iqa
* @short_description: Image Quality Assessment plugin.
*
* IQA will perform full reference image quality assessment, with the
@ -48,13 +49,12 @@
* sink_2\=\(double\)0.0082939683976297474\;",
* time=(guint64)0;
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -m uridecodebin uri=file:///test/file/1 ! iqa name=iqa do-dssim=true \
* ! videoconvert ! autovideosink uridecodebin uri=file:///test/file/2 ! iqa.
* ]| This pipeline will output messages to the console for each set of compared frames.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

26
ext/kate/gstkatedec.c
View file

@ -45,33 +45,29 @@
/**
* SECTION:element-katedec
* @title: katedec
* @see_also: oggdemux
*
* <refsect2>
* <para>
* This element decodes Kate streams
* <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
* for text based data, such as subtitles. Any number of kate streams can be
* embedded in an Ogg stream.
* </para>
* <para>
*
* libkate (see above url) is needed to build this plugin.
* </para>
* <title>Example pipeline</title>
* <para>
*
* ## Example pipeline
*
* This explicitely decodes a Kate stream:
* <programlisting>
* |[
* gst-launch-1.0 filesrc location=test.ogg ! oggdemux ! katedec ! fakesink silent=TRUE
* </programlisting>
* </para>
* <para>
* ]|
*
* This will automatically detect and use any Kate streams multiplexed
* in an Ogg stream:
* <programlisting>
* |[
* gst-launch-1.0 playbin uri=file:///tmp/test.ogg
* </programlisting>
* </para>
* </refsect2>
* ]|
*
*/
#ifdef HAVE_CONFIG_H

19
ext/kate/gstkateenc.c
View file

@ -46,26 +46,23 @@
/**
* SECTION:element-kateenc
* @title: kateenc
* @see_also: oggmux
*
* <refsect2>
* <para>
* This element encodes Kate streams
* <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
* for text based data, such as subtitles. Any number of kate streams can be
* embedded in an Ogg stream.
* </para>
* <para>
*
* libkate (see above url) is needed to build this plugin.
* </para>
* <title>Example pipeline</title>
* <para>
*
* ## Example pipeline
*
* This encodes a DVD SPU track to a Kate stream:
* <programlisting>
* |[
* gst-launch-1.0 dvdreadsrc ! dvddemux ! dvdsubparse ! kateenc category=spu-subtitles ! oggmux ! filesink location=test.ogg
* </programlisting>
* </para>
* </refsect2>
* ]|
*
*/
/* FIXME:

25
ext/kate/gstkateparse.c
View file

@ -21,40 +21,35 @@
/**
* SECTION:element-kateparse
* @title: kateparse
* @short_description: parses kate streams
* @see_also: katedec, vorbisparse, oggdemux, theoraparse
*
* <refsect2>
* <para>
* The kateparse element will parse the header packets of the Kate
* stream and put them as the streamheader in the caps. This is used in the
* multifdsink case where you want to stream live kate streams to multiple
* clients, each client has to receive the streamheaders first before they can
* consume the kate packets.
* </para>
* <para>
*
* This element also makes sure that the buffers that it pushes out are properly
* timestamped and that their offset and offset_end are set. The buffers that
* kateparse outputs have all of the metadata that oggmux expects to receive,
* which allows you to (for example) remux an ogg/kate file.
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
*
* ## Example pipelines
*
* |[
* gst-launch-1.0 -v filesrc location=kate.ogg ! oggdemux ! kateparse ! fakesink
* </programlisting>
* ]|
* This pipeline shows that the streamheader is set in the caps, and that each
* buffer has the timestamp, duration, offset, and offset_end set.
* </para>
* <para>
* <programlisting>
*
* |[
* gst-launch-1.0 filesrc location=kate.ogg ! oggdemux ! kateparse \
* ! oggmux ! filesink location=kate-remuxed.ogg
* </programlisting>
* ]|
* This pipeline shows remuxing. kate-remuxed.ogg might not be exactly the same
* as kate.ogg, but they should produce exactly the same decoded data.
* </para>
* </refsect2>
*
*/

29
ext/kate/gstkatetag.c
View file

@ -21,46 +21,41 @@
/**
* SECTION:element-katetag
* @title: katetag
* @see_also: #oggdemux, #oggmux, #kateparse, #GstTagSetter
* @short_description: retags kate streams
*
* <refsect2>
* <para>
* The katetag element can change the tag contained within a raw
* kate stream. Specifically, it modifies the comments header packet
* of the kate stream, as well as the language and category of the
* kate stream.
* </para>
* <para>
*
* The element will also process the stream as the #kateparse element does
* so it can be used when remuxing an Ogg Kate stream, without additional
* elements.
* </para>
* <para>
*
* Applications can set the tags to write using the #GstTagSetter interface.
* Tags contained within the kate stream will be picked up
* automatically (and merged according to the merge mode set via the tag
* setter interface).
* </para>
* <title>Example pipelines</title>
* <para>
*
* ## Example pipelines
*
* This element is only useful with gst-launch-1.0 for modifying the language
* and/or category (which are properties of the stream located in the kate
* beginning of stream header), because it does not support setting the tags
* on a #GstTagSetter interface. Conceptually, the element will usually be
* used like:
* <programlisting>
* |[
* gst-launch-1.0 -v filesrc location=foo.ogg ! oggdemux ! katetag ! oggmux ! filesink location=bar.ogg
* </programlisting>
* </para>
* <para>
* ]|
*
* This pipeline will set the language and category of the stream to the
* given values:
* <programlisting>
* |[
* gst-launch-1.0 -v filesrc location=foo.ogg ! oggdemux ! katetag language=pt_BR category=subtitles ! oggmux ! filesink location=bar.ogg
* </programlisting>
* </para>
* </refsect2>
* ]|
*
*/
#ifdef HAVE_CONFIG_H

19
ext/kate/gstkatetiger.c
View file

@ -45,32 +45,29 @@
/**
* SECTION:element-tiger
* @title: tiger
* @see_also: katedec
*
* <refsect2>
* <para>
* This element decodes and renders Kate streams
* <ulink url="http://libkate.googlecode.com/">Kate</ulink> is a free codec
* for text based data, such as subtitles. Any number of kate streams can be
* embedded in an Ogg stream.
* </para>
* <para>
*
* libkate (see above url) and <ulink url="http://libtiger.googlecode.com/">libtiger</ulink>
* are needed to build this element.
* </para>
* <title>Example pipeline</title>
* <para>
*
* ## Example pipeline
*
* This pipeline renders a Kate stream on top of a Theora video multiplexed
* in the same stream:
* <programlisting>
* |[
* gst-launch-1.0 \
* filesrc location=video.ogg ! oggdemux name=demux \
* demux. ! queue ! theoradec ! videoconvert ! tiger name=tiger \
* demux. ! queue ! kateparse ! tiger. \
* tiger. ! videoconvert ! autovideosink
* </programlisting>
* </para>
* </refsect2>
* ]|
*
*/
#ifdef HAVE_CONFIG_H

59
ext/ladspa/gstladspa.c
View file

@ -22,6 +22,7 @@
/**
* SECTION:element-ladspa
* @title: ladspa
* @short_description: bridge for LADSPA (Linux Audio Developer's Simple Plugin API)
* @see_also: #GstAudioConvert #GstAudioResample, #GstAudioTestSrc, #GstAutoAudioSink
*
@ -32,8 +33,7 @@
* element classification. The functionality you get depends on the LADSPA plugins
* you have installed.
*
* <refsect2>
* <title>Example LADSPA line without this plugins</title>
* ## Example LADSPA line without this plugins
* |[
* (padsp) listplugins
* (padsp) analyseplugin cmt.so amp_mono
@ -41,16 +41,13 @@
* (padsp) applyplugin testin.wav testout.wav cmt.so amp_mono 2
* gst-launch-1.0 playbin uri=file://"$PWD"/testout.wav
* ]| Decode any audio file into wav with the format expected for the specific ladspa plugin to be applied, apply the ladspa filter and play it.
* </refsect2>
*
* Now with this plugin:
*
* <refsect2>
* <title>Example LADSPA line with this plugins</title>
* ## Example LADSPA line with this plugins
* |[
* gst-launch-1.0 autoaudiosrc ! ladspa-cmt-so-amp-mono gain=2 ! ladspa-caps-so-plate ! ladspa-tap-echo-so-tap-stereo-echo l-delay=500 r-haas-delay=500 ! tee name=myT myT. ! queue ! autoaudiosink myT. ! queue ! audioconvert ! goom ! videoconvert ! xvimagesink pixel-aspect-ratio=3/4
* ]| Get audio input, filter it through CAPS Plate and TAP Stereo Echo, play it and show a visualization (recommended hearphones).
* </refsect2>
*
* In case you wonder the plugin naming scheme, quoting ladspa.h:
* "Plugin types should be identified by file and label rather than by
@ -61,60 +58,52 @@
* on top of the audio in and out one, so some parameters are readable too.
*
* You can see the listing of plugins available with:
* <refsect2>
* <title>Inspecting the plugins list</title>
*
* ## Inspecting the plugins list
* |[
* gst-inspect ladspa
* ]| List available LADSPA plugins on gstreamer.
* </refsect2>
*
* You can see the parameters of any plugin with:
* <refsect2>
* <title>Inspecting the plugins</title>
*
* ## Inspecting the plugins
* |[
* gst-inspect ladspa-retro-flange-1208-so-retroflange
* ]| List details of the plugin, parameters, range and defaults included.
* </refsect2>
*
* The elements categorize in:
* <itemizedlist>
* <listitem><para>Filter/Effect/Audio/LADSPA:</para>
* <refsect2>
* <title>Example Filter/Effect/Audio/LADSPA line with this plugins</title>
*
* * Filter/Effect/Audio/LADSPA:
*
* ## Example Filter/Effect/Audio/LADSPA line with this plugins
* |[
* gst-launch-1.0 filesrc location="$myfile" ! decodebin ! audioconvert ! audioresample ! ladspa-calf-so-reverb decay-time=15 high-frq-damp=20000 room-size=5 diffusion=1 wet-amount=2 dry-amount=2 pre-delay=50 bass-cut=20000 treble-cut=20000 ! ladspa-tap-echo-so-tap-stereo-echo l-delay=500 r-haas-delay=500 ! autoaudiosink
* ]| Decode any audio file, filter it through Calf Reverb LADSPA then TAP Stereo Echo, and play it.
* </refsect2>
* </listitem>
* <listitem><para>Source/Audio/LADSPA:</para>
* <refsect2>
* <title>Example Source/Audio/LADSPA line with this plugins</title>
*
* * Source/Audio/LADSPA:
*
* ## Example Source/Audio/LADSPA line with this plugins
* |[
* gst-launch-1.0 ladspasrc-sine-so-sine-fcac frequency=220 amplitude=100 ! audioconvert ! autoaudiosink
* ]| Generate a sine wave with Sine Oscillator (Freq:control, Amp:control) and play it.
* </refsect2>
* <refsect2>
* <title>Example Source/Audio/LADSPA line with this plugins</title>
*
* ## Example Source/Audio/LADSPA line with this plugins
* |[
* gst-launch-1.0 ladspasrc-caps-so-click bpm=240 volume=1 ! autoaudiosink
* ]| Generate clicks with CAPS Click - Metronome at 240 beats per minute and play it.
* </refsect2>
* <refsect2>
* <title>Example Source/Audio/LADSPA line with this plugins</title>
*
* ## Example Source/Audio/LADSPA line with this plugins
* |[
* gst-launch-1.0 ladspasrc-random-1661-so-random-fcsc-oa ! ladspa-cmt-so-amp-mono gain=1.5 ! ladspa-caps-so-plate ! tee name=myT myT. ! queue ! autoaudiosink myT. ! queue ! audioconvert ! wavescope ! videoconvert ! autovideosink
* ]| Generate random wave, filter it trhough Mono Amplifier and Versatile Plate Reverb, and play, while showing, it.
* </refsect2>
* </listitem>
* <listitem><para>Sink/Audio/LADSPA:</para>
* <refsect2>
* <title>Example Sink/Audio/LADSPA line with this plugins</title>
*
* * Sink/Audio/LADSPA:
*
* ## Example Sink/Audio/LADSPA line with this plugins
* |[
* gst-launch-1.0 autoaudiosrc ! ladspa-cmt-so-amp-mono gain=2 ! ladspa-caps-so-plate ! ladspa-tap-echo-so-tap-stereo-echo l-delay=500 r-haas-delay=500 ! tee name=myT myT. ! audioconvert ! audioresample ! queue ! ladspasink-cmt-so-null-ai myT. ! audioconvert ! audioresample ! queue ! goom ! videoconvert ! xvimagesink pixel-aspect-ratio=3/4
* ]| Get audio input, filter it trhough Mono Amplifier, CAPS Plate LADSPA and TAP Stereo Echo, explicitily anulate audio with Null (Audio Output), and play a visualization (recommended hearphones).
* </refsect2>
* </listitem>
* </itemizedlist>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/libde265/libde265-dec.c
View file

@ -21,15 +21,15 @@
/**
* SECTION:element-libde265dec
* @title: libde265dec
*
* Decodes HEVC/H.265 video.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=bitstream.hevc ! 'video/x-hevc,stream-format=byte-stream,framerate=25/1' ! libde265dec ! autovideosink
* ]| The above pipeline decodes the HEVC/H.265 bitstream and renders it to the screen.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
ext/lv2/gstlv2.c
View file

@ -22,6 +22,7 @@
/**
* SECTION:element-lv2
* @title: lv2
* @short_description: bridge for LV2.
*
* LV2 is a standard for plugins and matching host applications,

6
ext/openal/gstopenalsink.c
View file

@ -24,6 +24,7 @@
/**
* SECTION:element-openalsink
* @title: openalsink
* @see_also: openalsrc
* @short_description: capture raw audio samples through OpenAL
*
@ -31,8 +32,7 @@
*
* Unfortunately the capture API doesn't have a format enumeration/check. all you can do is try opening it and see if it works.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 audiotestsrc ! audioconvert ! volume volume=0.5 ! openalsink
* ]| will play a sine wave (continuous beep sound) through OpenAL.
@ -42,7 +42,7 @@
* |[
* gst-launch-1.0 openalsrc ! "audio/x-raw,format=S16LE,rate=44100" ! audioconvert ! volume volume=0.25 ! openalsink
* ]| will capture and play audio through OpenAL.
* </refsect2>
*
*/
/*

6
ext/openal/gstopenalsrc.c
View file

@ -49,20 +49,20 @@
/**
* SECTION:element-openalsrc
* @title: openalsrc
* @see_also: openalsink
* @short_description: capture raw audio samples through OpenAL
*
* This element captures raw audio samples through OpenAL.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v openalsrc ! audioconvert ! wavenc ! filesink location=stream.wav
* ]| * will capture sound through OpenAL and encode it to a wav file.
* |[
* gst-launch-1.0 openalsrc ! "audio/x-raw,format=S16LE,rate=44100" ! audioconvert ! volume volume=0.25 ! openalsink
* ]| will capture and play audio through OpenAL.
* </refsect2>
*
*/
/*

6
ext/opus/gstopusparse.c
View file

@ -22,16 +22,16 @@
/**
* SECTION:element-opusparse
* @title: opusparse
* @see_also: opusenc, opusdec
*
* This element parses OPUS packets.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=opusdata ! opusparse ! opusdec ! audioconvert ! audioresample ! alsasink
* ]| Decode and plays an unmuxed Opus file.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

4
ext/resindvd/rsndec.c
View file

@ -369,7 +369,7 @@ rsn_dec_get_type (void)
return type;
}
/** Audio decoder subclass */
/* Audio decoder subclass */
static GstStaticPadTemplate audio_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
@ -422,7 +422,7 @@ rsn_audiodec_init (RsnAudioDec * self)
{
}
/** Video decoder subclass */
/* Video decoder subclass */
static GstStaticPadTemplate video_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,

21
ext/resindvd/rsninputselector.c
View file

@ -25,6 +25,7 @@
/**
* SECTION:element-input-selector
* @title: input-selector
* @see_also: #GstOutputSelector
*
* Direct one out of N input streams to the output pad.
@ -32,21 +33,11 @@
* The input pads are from a GstPad subclass and have additional
* properties, which users may find useful, namely:
*
* <itemizedlist>
* <listitem>
* "running-time": Running time of stream on pad (#gint64)
* </listitem>
* <listitem>
* "tags": The currently active tags on the pad (#GstTagList, boxed type)
* </listitem>
* <listitem>
* "active": If the pad is currently active (#gboolean)
* </listitem>
* <listitem>
* "always-ok" : Make an inactive pads return #GST_FLOW_OK instead of
* #GST_FLOW_NOT_LINKED
* </listitem>
* </itemizedlist>
* * "running-time": Running time of stream on pad (#gint64)
* * "tags": The currently active tags on the pad (#GstTagList, boxed type)
* * "active": If the pad is currently active (#gboolean)
* * "always-ok" : Make an inactive pads return #GST_FLOW_OK instead of #GST_FLOW_NOT_LINKED
*
*/
#ifdef HAVE_CONFIG_H

6
ext/rsvg/gstrsvgdec.c
View file

@ -19,15 +19,15 @@
*/
/**
* SECTION:element-rsvgdec
* @title: rsvgdec
*
* This elements renders SVG graphics.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 filesrc location=image.svg ! rsvgdec ! imagefreeze ! videoconvert ! autovideosink
* ]| render and show a svg image.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/rsvg/gstrsvgoverlay.c
View file

@ -21,6 +21,7 @@
/**
* SECTION:element-rsvgoverlay
* @title: rsvgoverlay
*
* This elements overlays SVG graphics over the video. SVG data can
* either be specified through properties, or fed through the
@ -44,8 +45,7 @@
* the values of the x/y/width/height attributes, by setting
* height-/width-relative to 1.0. and all other attributes to 0.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 -v videotestsrc ! videoconvert ! rsvgoverlay location=foo.svg ! videoconvert ! autovideosink
* ]| specifies the SVG location through the filename property.
@ -55,7 +55,7 @@
* |[
* gst-launch-1.0 -v videotestsrc ! videoconvert ! rsvgoverlay data='&lt;svg viewBox="0 0 800 600"&gt;&lt;image x="80%" y="80%" width="10%" height="10%" xlink:href="foo.jpg" /&gt;&lt;/svg&gt;' ! videoconvert ! autovideosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/rtmp/gstrtmpsink.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-rtmpsink
* @title: rtmpsink
*
* This element delivers data to a streaming server via RTMP. It uses
* librtmp, and supports any protocols/urls that librtmp supports.
@ -27,12 +28,11 @@
* for librtmp, such as 'flashver=version'. See the librtmp documentation
* for more detail
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! ffenc_flv ! flvmux ! rtmpsink location='rtmp://localhost/path/to/stream live=1'
* ]| Encode a test video stream to FLV video format and stream it via RTMP.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/rtmp/gstrtmpsrc.c
View file

@ -26,17 +26,17 @@
/**
* SECTION:element-rtmpsrc
* @title: rtmpsrc
*
* This plugin reads data from a local or remote location specified
* by an URI. This location can be specified using any protocol supported by
* the RTMP library, i.e. rtmp, rtmpt, rtmps, rtmpe, rtmfp, rtmpte and rtmpts.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 -v rtmpsrc location=rtmp://somehost/someurl ! fakesink
* ]| Open an RTMP location and pass its content to fakesink.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/sbc/gstsbcdec.c
View file

@ -21,15 +21,15 @@
/**
* SECTION:element-sbdec
* @title: sbdec
*
* This element decodes a Bluetooth SBC audio streams to raw integer PCM audio
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=audio.sbc ! sbcparse ! sbcdec ! audioconvert ! audioresample ! autoaudiosink
* ]| Decode a raw SBC file.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/sbc/gstsbcenc.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-sbenc
* @title: sbenc
*
* This element encodes raw integer PCM audio into a Bluetooth SBC audio.
*
@ -27,12 +28,11 @@
* allocation-mode can be set by adding a capsfilter element with appropriate
* filtercaps after the sbcenc encoder element.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v audiotestsrc ! sbcenc ! rtpsbcpay ! udpsink
* ]| Encode a sine wave into SBC, RTP payload it and send over the network using UDP
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

2
ext/smoothstreaming/gstmssdemux.c
View file

@ -22,10 +22,10 @@
/**
* SECTION:element-mssdemux
* @title: mssdemux
*
* Demuxes a Microsoft's Smooth Streaming manifest into its audio and/or video streams.
*
*
*/
/*

29
ext/spandsp/gstdtmfdetect.c
View file

@ -24,36 +24,21 @@
/**
* SECTION:element-dtmfdetect
* @title: dtmfdetect
* @short_description: Detects DTMF tones
*
* This element will detect DTMF tones and emit messages.
*
* The message is called <classname>&quot;dtmf-event&quot;</classname> and has
* the following fields:
* <itemizedlist>
* <listitem>
* <para>
* gint <classname>type</classname> (0-1):
* The application uses this field to specify which of the two methods
* The message is called `dtmf-event` and has the following fields:
*
* * gint `type` (0-1): The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are
* specfied by their number. This element can only take events as input.
* Do not confuse with "method" which specified the output.
* </para>
* </listitem>
* <listitem>
* <para>
* gint <classname>number</classname> (0-16):
* The event number.
* </para>
* </listitem>
* <listitem>
* <para>
* gint <classname>method</classname> (2):
* This field will always been 2 (ie sound) from this element.
* </para>
* </listitem>
* </itemizedlist>
* * gint `number` (0-16): The event number.
* * gint `method` (2): This field will always been 2 (ie sound) from this element.
*
*/
#ifdef HAVE_CONFIG_H

1
ext/spandsp/gstspanplc.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-spanplc
* @title: spanplc
*
* The spanplc (Packet Loss Concealment) element provides a synthetic
* fill-in signal, to minimise the audible effect of lost packets in

6
ext/srtp/gstsrtpdec.c
View file

@ -46,6 +46,7 @@
/**
* SECTION:element-srtpdec
* @title: srtpdec
* @see_also: srtpenc
*
* gstrtpdec acts as a decoder that removes security from SRTP and SRTCP
@ -95,8 +96,7 @@
* other means. If no rollover counter is provided by the user, 0 is
* used by default.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 udpsrc port=5004 caps='application/x-srtp, payload=(int)8, ssrc=(uint)1356955624, srtp-key=(buffer)012345678901234567890123456789012345678901234567890123456789, srtp-cipher=(string)aes-128-icm, srtp-auth=(string)hmac-sha1-80, srtcp-cipher=(string)aes-128-icm, srtcp-auth=(string)hmac-sha1-80' ! srtpdec ! rtppcmadepay ! alawdec ! pulsesink
* ]| Receive PCMA SRTP packets through UDP using caps to specify
@ -105,7 +105,7 @@
* gst-launch-1.0 audiotestsrc ! alawenc ! rtppcmapay ! 'application/x-rtp, payload=(int)8, ssrc=(uint)1356955624' ! srtpenc key="012345678901234567890123456789012345678901234567890123456789" ! udpsink port=5004
* ]| Send PCMA SRTP packets through UDP, nothing how the SSRC is forced so
* that the receiver will recognize it.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

3
ext/srtp/gstsrtpenc.c
View file

@ -45,7 +45,8 @@
*/
/**
* SECTION:gst-plugin-bad-plugins-srtpenc
* SECTION:element-srtpenc
* @title: srtpenc
* @see_also: srtpdec
*
* gstrtpenc acts as an encoder that adds security to RTP and RTCP

6
ext/teletextdec/gstteletextdec.c
View file

@ -21,16 +21,16 @@
/**
* SECTION:element-teletextdec
* @title: teletextdec
*
* Decode a stream of raw VBI packets containing teletext information to a RGBA
* stream.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v -m filesrc location=recording.mpeg ! tsdemux ! teletextdec ! videoconvert ! ximagesink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/ttml/gstttmlparse.c
View file

@ -22,6 +22,7 @@
/**
* SECTION:element-ttmlparse
* @title: ttmlparse
*
* Parses timed text subtitle files described using Timed Text Markup Language
* (TTML). Currently, only the EBU-TT-D profile of TTML, designed for
@ -35,13 +36,12 @@
* elements. A downstream renderer element uses this information to correctly
* render the text on top of video frames.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 filesrc location=<media file location> ! video/quicktime ! qtdemux name=q ttmlrender name=r q. ! queue ! h264parse ! avdec_h264 ! autovideoconvert ! r.video_sink filesrc location=<subtitle file location> blocksize=16777216 ! queue ! ttmlparse ! r.text_sink r. ! ximagesink q. ! queue ! aacparse ! avdec_aac ! audioconvert ! alsasink
* ]| Parse and render TTML subtitles contained in a single XML file over an
* MP4 stream containing H.264 video and AAC audio.
* </refsect2>
*
*/
#include <stdio.h>

6
ext/ttml/gstttmlrender.c
View file

@ -25,19 +25,19 @@
/**
* SECTION:element-ttmlrender
* @title: ttmlrender
*
* Renders timed text on top of a video stream. It receives text in buffers
* from a ttmlparse element; each text string is in its own #GstMemory within
* the GstBuffer, and the styling and layout associated with each text string
* is in metadata attached to the #GstBuffer.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 filesrc location=<media file location> ! video/quicktime ! qtdemux name=q ttmlrender name=r q. ! queue ! h264parse ! avdec_h264 ! autovideoconvert ! r.video_sink filesrc location=<subtitle file location> blocksize=16777216 ! queue ! ttmlparse ! r.text_sink r. ! ximagesink q. ! queue ! aacparse ! avdec_aac ! audioconvert ! alsasink
* ]| Parse and render TTML subtitles contained in a single XML file over an
* MP4 stream containing H.264 video and AAC audio:
* </refsect2>
*
*/
#include <gst/video/video.h>

1
ext/ttml/subtitle.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:gstsubtitle
* @title: GstSubtitle
* @short_description: Library for describing sets of static subtitles.
*
* This library enables the description of static text scenes made up of a

1
ext/ttml/subtitlemeta.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:gstsubtitlemeta
* @title: GstSubtitleMeta
* @short_description: Metadata class for timed-text subtitles.
*
* The GstSubtitleMeta class enables the layout and styling information needed

6
ext/voaacenc/gstvoaacenc.c
View file

@ -19,16 +19,16 @@
/**
* SECTION:element-voaacenc
* @title: voaacenc
*
* AAC audio encoder based on vo-aacenc library
* <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voaacenc ! filesink location=abc.aac
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/voamrwbenc/gstvoamrwbenc.c
View file

@ -19,19 +19,19 @@
/**
* SECTION:element-voamrwbenc
* @title: voamrwbenc
* @see_also: #GstAmrWbDec, #GstAmrWbParse
*
* AMR wideband encoder based on the
* <ulink url="http://www.penguin.cz/~utx/amr">reference codec implementation</ulink>.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voamrwbenc ! filesink location=abc.amr
* ]|
* Please note that the above stream misses the header, that is needed to play
* the stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
ext/vulkan/gstvulkan.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:plugin-vulkan
* @title: vulkan
*
* Cross-platform Vulkan plugin.
*/

1
ext/vulkan/vkbuffermemory.c
View file

@ -26,6 +26,7 @@
/**
* SECTION:vkbuffermemory
* @title: vkbuffermemory
* @short_description: memory subclass for Vulkan buffer memory
* @see_also: #GstMemory, #GstAllocator
*

1
ext/vulkan/vkbufferpool.c
View file

@ -26,6 +26,7 @@
/**
* SECTION:vkbufferpool
* @title: GstVulkanBufferPool
* @short_description: buffer pool for #GstVulkanBufferMemory objects
* @see_also: #GstBufferPool, #GstVulkanBufferMemory
*

1
ext/vulkan/vkimagememory.c
View file

@ -26,6 +26,7 @@
/**
* SECTION:vkimagememory
* @title: GstVkImageMemory
* @short_description: memory subclass for Vulkan image memory
* @see_also: #GstMemory, #GstAllocator
*

1
ext/vulkan/vkmemory.c
View file

@ -28,6 +28,7 @@
/**
* SECTION:vkmemory
* @title: GstVkMemory
* @short_description: memory subclass for Vulkan device memory
* @see_also: #GstMemory, #GstAllocator
*

1
ext/vulkan/vksink.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-vulkansink
* @title: vulkansink
*
* vulkansink renders video frames to a drawable on a local or remote
* display using Vulkan.

1
ext/vulkan/vkupload.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-vulkanupload
* @title: vulkanupload
*
* vulkanupload uploads data into Vulkan memory objects.
*/

2
ext/vulkan/vkwindow.c
View file

@ -19,7 +19,7 @@
*/
/**
* SECTION:gstglwindow
* SECTION:vkwindow
* @short_description: window/surface abstraction
* @title: GstVulkanWindow
* @see_also: #GstGLContext, #GstGLDisplay

6
ext/wayland/gstwaylandsink.c
View file

@ -23,18 +23,18 @@
/**
* SECTION:element-waylandsink
* @title: waylandsink
*
* The waylandsink is creating its own window and render the decoded video frames to that.
* Setup the Wayland environment as described in
* <ulink url="http://wayland.freedesktop.org/building.html">Wayland</ulink> home page.
* The current implementaion is based on weston compositor.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v videotestsrc ! waylandsink
* ]| test the video rendering in wayland
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/wildmidi/gstwildmidi.c
View file

@ -21,6 +21,7 @@
/**
* SECTION:element-wildmidi
* @title: wildmidi
* @see_also: timidity
*
* This element renders midi-files as audio streams using
@ -29,13 +30,12 @@
* uses the same sound-patches as timidity (it tries the path in $WILDMIDI_CFG,
* $HOME/.wildmidirc and /etc/wildmidi.cfg)
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 filesrc location=song.mid ! wildmidi ! alsasink
* ]| This example pipeline will parse the midi and render to raw audio which is
* played via alsa.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
ext/x265/gstx265enc.c
View file

@ -22,6 +22,7 @@
/**
* SECTION:element-x265enc
* @title: x265enc
*
* This element encodes raw video into H265 compressed data.
*

57
ext/zbar/gstzbar.c
View file

@ -19,64 +19,31 @@
/**
* SECTION:element-zbar
* @title: zbar
*
* Detect bar codes in the video streams and send them as element messages to
* the #GstBus if .#GstZBar:message property is %TRUE.
* If the .#GstZBar:attach-frame property is %TRUE, the posted barcode message
* includes a sample of the frame where the barcode was detected (Since 1.6).
*
* The element generate messages named
* <classname>&quot;barcode&quot;</classname>. The structure containes these
* fields:
* <itemizedlist>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;timestamp&quot;</classname>:
* the timestamp of the buffer that triggered the message.
* </para>
* </listitem>
* <listitem>
* <para>
* gchar*
* <classname>&quot;type&quot;</classname>:
* the symbol type.
* </para>
* </listitem>
* <listitem>
* <para>
* gchar*
* <classname>&quot;symbol&quot;</classname>:
* the deteted bar code data.
* </para>
* </listitem>
* <listitem>
* <para>
* gint
* <classname>&quot;quality&quot;</classname>:
* an unscaled, relative quantity: larger values are better than smaller
* values.
* </para>
* </listitem>
* <listitem>
* <para>
* GstSample
* <classname>&quot;frame&quot;</classname>:
* the frame in which the barcode message was detected, if
* the .#GstZBar:attach-frame property was set to %TRUE (Since 1.6)
* </para>
* </listitem>
* </itemizedlist>
* The element generate messages named`barcode`. The structure containes these fields:
*
* <refsect2>
* <title>Example launch lines</title>
* * #GstClockTime `timestamp`: the timestamp of the buffer that triggered the message.
* * gchar * `type`: the symbol type.
* * gchar * `symbol`: the deteted bar code data.
* * gint `quality`: an unscaled, relative quantity: larger values are better than smaller
* values.
* * GstSample `frame`: the frame in which the barcode message was detected, if
* the .#GstZBar:attach-frame property was set to %TRUE (Since 1.6)
*
* ## Example launch lines
* |[
* gst-launch-1.0 -m v4l2src ! videoconvert ! zbar ! videoconvert ! xvimagesink
* ]| This pipeline will detect barcodes and send them as messages.
* |[
* gst-launch-1.0 -m v4l2src ! tee name=t ! queue ! videoconvert ! zbar ! fakesink t. ! queue ! xvimagesink
* ]| Same as above, but running the filter on a branch to keep the display in color
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

34
gst-libs/gst/base/gstaggregator.c
View file

@ -21,46 +21,40 @@
*/
/**
* SECTION: gstaggregator
* @title: GstAggregator
* @short_description: manages a set of pads with the purpose of
* aggregating their buffers.
* @see_also: gstcollectpads for historical reasons.
*
* Manages a set of pads with the purpose of aggregating their buffers.
* Control is given to the subclass when all pads have data.
* <itemizedlist>
* <listitem><para>
* Base class for mixers and muxers. Subclasses should at least implement
*
* * Base class for mixers and muxers. Subclasses should at least implement
* the #GstAggregatorClass.aggregate() virtual method.
* </para></listitem>
* <listitem><para>
* When data is queued on all pads, tha aggregate vmethod is called.
* </para></listitem>
* <listitem><para>
* One can peek at the data on any given GstAggregatorPad with the
*
* * When data is queued on all pads, tha aggregate vmethod is called.
*
* * One can peek at the data on any given GstAggregatorPad with the
* gst_aggregator_pad_get_buffer () method, and take ownership of it
* with the gst_aggregator_pad_steal_buffer () method. When a buffer
* has been taken with steal_buffer (), a new buffer can be queued
* on that pad.
* </para></listitem>
* <listitem><para>
* If the subclass wishes to push a buffer downstream in its aggregate
*
* * If the subclass wishes to push a buffer downstream in its aggregate
* implementation, it should do so through the
* gst_aggregator_finish_buffer () method. This method will take care
* of sending and ordering mandatory events such as stream start, caps
* and segment.
* </para></listitem>
* <listitem><para>
* Same goes for EOS events, which should not be pushed directly by the
*
* * Same goes for EOS events, which should not be pushed directly by the
* subclass, it should instead return GST_FLOW_EOS in its aggregate
* implementation.
* </para></listitem>
* <listitem><para>
* Note that the aggregator logic regarding gap event handling is to turn
*
* * Note that the aggregator logic regarding gap event handling is to turn
* these into gap buffers with matching PTS and duration. It will also
* flag these buffers with GST_BUFFER_FLAG_GAP and GST_BUFFER_FLAG_DROPPABLE
* to ease their identification and subsequent processing.
* </para></listitem>
* </itemizedlist>
*
*/
#ifdef HAVE_CONFIG_H

54
gst-libs/gst/codecparsers/gsth264parser.c
View file

@ -29,40 +29,30 @@
/**
* SECTION:gsth264parser
* @title: GstH264Parser
* @short_description: Convenience library for h264 video
* bitstream parsing.
*
* It offers bitstream parsing in both AVC (length-prefixed) and Annex B
* (0x000001 start code prefix) format. To identify a NAL unit in a bitstream
* and parse its headers, first call:
* <itemizedlist>
* <listitem>
* #gst_h264_parser_identify_nalu to identify a NAL unit in an Annex B type bitstream
* </listitem>
* <listitem>
* #gst_h264_parser_identify_nalu_avc to identify a NAL unit in an AVC type bitstream
* </listitem>
* </itemizedlist>
*
* * #gst_h264_parser_identify_nalu to identify a NAL unit in an Annex B type bitstream
*
* * #gst_h264_parser_identify_nalu_avc to identify a NAL unit in an AVC type bitstream
*
* The following functions are then available for parsing the structure of the
* #GstH264NalUnit, depending on the #GstH264NalUnitType:
* <itemizedlist>
* <listitem>
* From #GST_H264_NAL_SLICE to #GST_H264_NAL_SLICE_IDR: #gst_h264_parser_parse_slice_hdr
* </listitem>
* <listitem>
* #GST_H264_NAL_SEI: #gst_h264_parser_parse_sei
* </listitem>
* <listitem>
* #GST_H264_NAL_SPS: #gst_h264_parser_parse_sps
* </listitem>
* <listitem>
* #GST_H264_NAL_PPS: #gst_h264_parser_parse_pps
* </listitem>
* <listitem>
* Any other: #gst_h264_parser_parse_nal
* </listitem>
* </itemizedlist>
*
* * From #GST_H264_NAL_SLICE to #GST_H264_NAL_SLICE_IDR: #gst_h264_parser_parse_slice_hdr
*
* * #GST_H264_NAL_SEI: #gst_h264_parser_parse_sei
*
* * #GST_H264_NAL_SPS: #gst_h264_parser_parse_sps
*
* * #GST_H264_NAL_PPS: #gst_h264_parser_parse_pps
*
* * Any other: #gst_h264_parser_parse_nal
*
* One of these functions *must* be called on every NAL unit in the bitstream,
* in order to keep the internal structures of the #GstH264NalParser up to
@ -70,17 +60,13 @@
* type, if no special parsing of the current NAL unit is required by the
* application.
*
* For more details about the structures, look at the ITU-T H.264 and ISO/IEC 14496-10 – MPEG-4
* For more details about the structures, look at the ITU-T H.264 and ISO/IEC 14496-10 – MPEG-4
* Part 10 specifications, available at:
*
* <itemizedlist>
* <listitem>
* ITU-T H.264: http://www.itu.int/rec/T-REC-H.264
* </listitem>
* <listitem>
* ISO/IEC 14496-10: http://www.iso.org/iso/iso_catalogue/catalogue_tc/catalogue_detail.htm?csnumber=56538
* </listitem>
* </itemizedlist>
* * ITU-T H.264: http://www.itu.int/rec/T-REC-H.264
*
* * ISO/IEC 14496-10: http://www.iso.org/iso/iso_catalogue/catalogue_tc/catalogue_detail.htm?csnumber=56538
*
*/
#ifdef HAVE_CONFIG_H

52
gst-libs/gst/codecparsers/gsth265parser.c
View file

@ -22,43 +22,32 @@
/**
* SECTION:gsth265parser
* @title: GstH265Parser
* @short_description: Convenience library for h265 video bitstream parsing.
*
* It offers you bitstream parsing in HEVC mode and non-HEVC mode. To identify
* Nals in a bitstream and parse its headers, you should call:
* <itemizedlist>
* <listitem>
* gst_h265_parser_identify_nalu() to identify the following nalu in
*
* * gst_h265_parser_identify_nalu() to identify the following nalu in
* non-HEVC bitstreams
* </listitem>
* <listitem>
* gst_h265_parser_identify_nalu_hevc() to identify the nalu in
*
* * gst_h265_parser_identify_nalu_hevc() to identify the nalu in
* HEVC bitstreams
* </listitem>
* </itemizedlist>
*
* Then, depending on the #GstH265NalUnitType of the newly parsed #GstH265NalUnit,
* you should call the differents functions to parse the structure:
* <itemizedlist>
* <listitem>
* From #GST_H265_NAL_SLICE_TRAIL_N to #GST_H265_NAL_SLICE_CRA_NUT: gst_h265_parser_parse_slice_hdr()
* </listitem>
* <listitem>
* #GST_H265_NAL_SEI: gst_h265_parser_parse_sei()
* </listitem>
* <listitem>
* #GST_H265_NAL_VPS: gst_h265_parser_parse_vps()
* </listitem>
* <listitem>
* #GST_H265_NAL_SPS: gst_h265_parser_parse_sps()
* </listitem>
* <listitem>
* #GST_H265_NAL_PPS: #gst_h265_parser_parse_pps()
* </listitem>
* <listitem>
* Any other: gst_h265_parser_parse_nal()
* </listitem>
* </itemizedlist>
*
* * From #GST_H265_NAL_SLICE_TRAIL_N to #GST_H265_NAL_SLICE_CRA_NUT: gst_h265_parser_parse_slice_hdr()
*
* * #GST_H265_NAL_SEI: gst_h265_parser_parse_sei()
*
* * #GST_H265_NAL_VPS: gst_h265_parser_parse_vps()
*
* * #GST_H265_NAL_SPS: gst_h265_parser_parse_sps()
*
* * #GST_H265_NAL_PPS: #gst_h265_parser_parse_pps()
*
* * Any other: gst_h265_parser_parse_nal()
*
* Note: You should always call gst_h265_parser_parse_nal() if you don't
* actually need #GstH265NalUnitType to be parsed for your personal use, in
@ -67,11 +56,8 @@
* For more details about the structures, look at the ITU-T H.265
* specifications, you can download them from:
*
* <itemizedlist>
* <listitem>
* ITU-T H.265: http://www.itu.int/rec/T-REC-H.265
* </listitem>
* </itemizedlist>
* * ITU-T H.265: http://www.itu.int/rec/T-REC-H.265
*
*/
#ifdef HAVE_CONFIG_H

1
gst-libs/gst/codecparsers/gstjpeg2000sampling.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:gstjpeg2000sampling
* @title: GstJpeg2000Sampling
* @short_description: Manage JPEG 2000 sampling and colorspace fields
*
*/

6
gst-libs/gst/codecparsers/gstjpegparser.c
View file

@ -20,13 +20,11 @@
/**
* SECTION:gstjpegparser
* @title: GstJpegParser
* @short_description: Convenience library for JPEG bitstream parsing.
*
* <refsect2>
* <para>
* Provides useful functions for parsing JPEG images
* </para>
* </refsect2>
*
*/
#include <string.h>

1
gst-libs/gst/codecparsers/gstmpeg4parser.c
View file

@ -20,6 +20,7 @@
*/
/**
* SECTION:gstmpeg4parser
* @title: GstMpeg4Parser
* @short_description: Convenience library for parsing mpeg4 part 2 video
* bitstream.
*

6
gst-libs/gst/codecparsers/gstmpegvideoparser.c
View file

@ -25,14 +25,12 @@
/**
* SECTION:gstmpegvideoparser
* @title: GstMpegvideoParser
* @short_description: Convenience library for mpeg1 and 2 video
* bitstream parsing.
*
* <refsect2>
* <para>
* Provides useful functions for mpeg videos bitstream parsing.
* </para>
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

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