mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-26 19:51:11 +00:00
Code Issues 1 Projects Releases Wiki Activity

docs: Port all docstring to gtk-doc markdown

Browse source
This commit is contained in:
Thibault Saunier 2018-10-22 11:39:24 +02:00
parent 135e12565b
commit 0a6a62aa76
205 changed files with 806 additions and 1082 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

6
ext/aalib/gstaasink.c
View file

@ -18,19 +18,19 @@
*/
/**
* SECTION:element-aasink
* @title: aasink
* @see_also: #GstCACASink
*
* Displays video as b/w ascii art.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=test.avi ! decodebin ! videoconvert ! aasink
* ]| This pipeline renders a video to ascii art into a separate window.
* |[
* gst-launch-1.0 filesrc location=test.avi ! decodebin ! videoconvert ! aasink driver=curses
* ]| This pipeline renders a video to ascii art into the current terminal.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

19
ext/cairo/gstcairooverlay.c
View file

@ -19,12 +19,13 @@
/**
* SECTION:element-cairooverlay
* @title: cairooverlay
*
* cairooverlay renders an overlay using a application provided render function.
*
* The full example can be found in tests/examples/cairo/cairo_overlay.c
* <refsect2>
* <title>Example code</title>
*
* ## Example code
* |[
*
* #include &lt;gst/gst.h&gt;
@ -37,7 +38,7 @@
* int width;
* int height;
* } CairoOverlayState;
*
*
* ...
*
* static void
@ -50,7 +51,7 @@
* }
*
* static void
* draw_overlay (GstElement * overlay, cairo_t * cr, guint64 timestamp,
* draw_overlay (GstElement * overlay, cairo_t * cr, guint64 timestamp,
* guint64 duration, gpointer user_data)
* {
* CairoOverlayState *s = (CairoOverlayState *)user_data;
@ -66,7 +67,7 @@
* cairo_move_to (cr, 0, 0);
* cairo_curve_to (cr, 0,-30, -50,-30, -50,0);
* cairo_curve_to (cr, -50,30, 0,35, 0,60 );
* cairo_curve_to (cr, 0,35, 50,30, 50,0 ); *
* cairo_curve_to (cr, 0,35, 50,30, 50,0 ); *
* cairo_curve_to (cr, 50,-30, 0,-30, 0,0 );
* cairo_set_source_rgba (cr, 0.9, 0.0, 0.1, 0.7);
* cairo_fill (cr);
@ -78,12 +79,12 @@
*
* g_signal_connect (cairo_overlay, &quot;draw&quot;, G_CALLBACK (draw_overlay),
* overlay_state);
* g_signal_connect (cairo_overlay, &quot;caps-changed&quot;,
* g_signal_connect (cairo_overlay, &quot;caps-changed&quot;,
* G_CALLBACK (prepare_overlay), overlay_state);
* ...
*
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
@ -538,7 +539,7 @@ gst_cairo_overlay_class_init (GstCairoOverlayClass * klass)
* @cr: Cairo context to draw to.
* @timestamp: Timestamp (see #GstClockTime) of the current buffer.
* @duration: Duration (see #GstClockTime) of the current buffer.
*
*
* This signal is emitted when the overlay should be drawn.
*/
gst_cairo_overlay_signals[SIGNAL_DRAW] =
@ -555,7 +556,7 @@ gst_cairo_overlay_class_init (GstCairoOverlayClass * klass)
* GstCairoOverlay::caps-changed:
* @overlay: Overlay element emitting the signal.
* @caps: The #GstCaps of the element.
*
*
* This signal is emitted when the caps of the element has changed.
*/
gst_cairo_overlay_signals[SIGNAL_CAPS_CHANGED] =

8
ext/dv/gstdvdec.c
View file

@ -20,20 +20,20 @@
/**
* SECTION:element-dvdec
* @title: dvdec
*
* dvdec decodes DV video into raw video. The element expects a full DV frame
* as input, which is 120000 bytes for NTSC and 144000 for PAL video.
*
* This element can perform simple frame dropping with the #GstDVDec:drop-factor
* property. Setting this property to a value N > 1 will only decode every
* property. Setting this property to a value N > 1 will only decode every
* Nth frame.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! dvdec ! xvimagesink
* ]| This pipeline decodes and renders the raw DV stream to a videosink.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/dv/gstdvdemux.c
View file

@ -31,6 +31,7 @@
/**
* SECTION:element-dvdemux
* @title: dvdemux
*
* dvdemux splits raw DV into its audio and video components. The audio will be
* decoded raw samples and the video will be encoded DV video.
@ -38,12 +39,11 @@
* This element can operate in both push and pull mode depending on the
* capabilities of the upstream peer.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! queue ! audioconvert ! alsasink demux. ! queue ! dvdec ! xvimagesink
* ]| This pipeline decodes and renders the raw DV stream to an audio and a videosink.
* </refsect2>
*
*/
/* DV output has two modes, normal and wide. The resolution is the same in both

6
ext/flac/gstflacdec.c
View file

@ -21,21 +21,21 @@
/**
* SECTION:element-flacdec
* @title: flacdec
* @see_also: #GstFlacEnc
*
* flacdec decodes FLAC streams.
* <ulink url="http://flac.sourceforge.net/">FLAC</ulink>
* is a Free Lossless Audio Codec.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
* |[
* gst-launch-1.0 souphttpsrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacparse ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/flac/gstflacenc.c
View file

@ -18,6 +18,7 @@
*/
/**
* SECTION:element-flacenc
* @title: flacenc
* @see_also: #GstFlacDec
*
* flacenc encodes FLAC streams.
@ -25,8 +26,7 @@
* is a Free Lossless Audio Codec. FLAC audio can directly be written into
* a file, or embedded into containers such as oggmux or matroskamux.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc num-buffers=100 ! flacenc ! filesink location=beep.flac
* ]| Encode a short sine wave into FLAC
@ -36,7 +36,7 @@
* |[
* gst-launch-1.0 cdparanoiasrc track=5 ! queue ! audioconvert ! flacenc ! filesink location=track5.flac
* ]| Rip track 5 of an audio CD and encode it losslessly to a FLAC file
* </refsect2>
*
*/
/* TODO: - We currently don't handle discontinuities in the stream in a useful

6
ext/flac/gstflactag.c
View file

@ -23,6 +23,7 @@
/**
* SECTION:element-flactag
* @title: flactag
* @see_also: #flacenc, #flacdec, #GstTagSetter
*
* The flactag element can change the tag contained within a raw
@ -34,14 +35,13 @@
* 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.flac ! flactag ! filesink location=bar.flac
* ]| This element is not useful with gst-launch, 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

6
ext/gdk_pixbuf/gstgdkpixbufoverlay.c
View file

@ -19,6 +19,7 @@
/**
* SECTION:element-gdkpixbufoverlay
* @title: gdkpixbufoverlay
*
* The gdkpixbufoverlay element overlays an image loaded from file onto
* a video stream.
@ -32,14 +33,13 @@
*
* Negative offsets are also not yet supported.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! gdkpixbufoverlay location=image.png ! autovideosink
* ]|
* Overlays the image in image.png onto the test video picture produced by
* videotestsrc.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

42
ext/gdk_pixbuf/gstgdkpixbufsink.c
View file

@ -19,6 +19,7 @@
/**
* SECTION:element-gdkpixbufsink
* @title: gdkpixbufsink
*
* This sink element takes RGB or RGBA images as input and wraps them into
* #GdkPixbuf objects, for easy saving to file via the
@ -27,23 +28,18 @@
*
* There are two ways to use this element and obtain the #GdkPixbuf objects
* created:
* <itemizedlist>
* <listitem>
* Watching for element messages named <classname>&quot;preroll-pixbuf&quot;
* </classname> or <classname>&quot;pixbuf&quot;</classname> on the bus, which
*
* * Watching for element messages named `preroll-pixbuf` or `pixbuf` on the bus, which
* will be posted whenever an image would usually be rendered. See below for
* more details on these messages and how to extract the pixbuf object
* contained in them.
* </listitem>
* <listitem>
* Retrieving the current pixbuf via the #GstGdkPixbufSink:last-pixbuf property
*
* * Retrieving the current pixbuf via the #GstGdkPixbufSink:last-pixbuf property
* when needed. This is the easiest way to get at pixbufs for snapshotting
* purposes - just wait until the pipeline is prerolled (ASYNC_DONE message
* on the bus), then read the property. If you use this method, you may want
* to disable message posting by setting the #GstGdkPixbufSink:post-messages
* property to %FALSE. This avoids unnecessary memory overhead.
* </listitem>
* </itemizedlist>
*
* The primary purpose of this element is to abstract away the #GstBuffer to
* #GdkPixbuf conversion. Other than that it's very similar to the fakesink
@ -54,20 +50,17 @@
* ximagesink, xvimagesink or some other suitable video sink in connection
* with the #GstXOverlay interface instead if you want to do video playback.
*
* <refsect2>
* <title>Message details</title>
* ## Message details
*
* As mentioned above, this element will by default post element messages
* containing structures named <classname>&quot;preroll-pixbuf&quot;
* </classname> or <classname>&quot;pixbuf&quot;</classname> on the bus (this
* containing structures named `preroll-pixbuf`
* ` or `pixbuf` on the bus (this
* can be disabled by setting the #GstGdkPixbufSink:post-messages property
* to %FALSE though). The element message structure has the following fields:
* <itemizedlist>
* <listitem>
* <classname>&quot;pixbuf&quot;</classname>: the #GdkPixbuf object
* </listitem>
* <listitem>
* <classname>&quot;pixel-aspect-ratio&quot;</classname>: the pixel aspect
* ratio (PAR) of the input image (this field contains a #GstFraction); the
*
* * `pixbuf`: the #GdkPixbuf object
* * `pixel-aspect-ratio`: the pixel aspect ratio (PAR) of the input image
* (this field contains a #GstFraction); the
* PAR is usually 1:1 for images, but is often something non-1:1 in the case
* of video input. In this case the image may be distorted and you may need
* to rescale it accordingly before saving it to file or displaying it. This
@ -76,20 +69,15 @@
* according to the size of the output window, in which case it is much more
* efficient to only scale once rather than twice). You can put a videoscale
* element and a capsfilter element with
* <literal>video/x-raw-rgb,pixel-aspect-ratio=(fraction)1/1</literal> caps
* `video/x-raw-rgb,pixel-aspect-ratio=(fraction)1/1` caps
* in front of this element to make sure the pixbufs always have a 1:1 PAR.
* </listitem>
* </itemizedlist>
* </refsect2>
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 -m -v videotestsrc num-buffers=1 ! gdkpixbufsink
* ]| Process one single test image as pixbuf (note that the output you see will
* be slightly misleading. The message structure does contain a valid pixbuf
* object even if the structure string says &apos;(NULL)&apos;).
* </refsect2>
*/
#ifdef HAVE_CONFIG_H

2
ext/jack/gstjackaudioclient.c
View file

@ -494,7 +494,7 @@ gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn,
* Get the jack client connection for @id and @server. Connections to the same
* @id and @server will receive the same physical Jack client connection and
* will therefore be scheduled in the same process callback.
*
*
* Returns: a #GstJackAudioClient.
*/
GstJackAudioClient *

20
ext/jack/gstjackaudiosink.c
View file

@ -21,36 +21,36 @@
/**
* SECTION:element-jackaudiosink
* @title: jackaudiosink
* @see_also: #GstAudioBaseSink, #GstAudioRingBuffer
*
* A Sink that outputs data to Jack ports.
*
* It will create N Jack ports named out_&lt;name&gt;_&lt;num&gt; where
*
* It will create N Jack ports named out_&lt;name&gt;_&lt;num&gt; where
* &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
* Each port corresponds to a gstreamer channel.
*
*
* The samplerate as exposed on the caps is always the same as the samplerate of
* the jack server.
*
*
* When the #GstJackAudioSink:connect property is set to auto, this element
* will try to connect each output port to a random physical jack input pin. In
* this mode, the sink will expose the number of physical channels on its pad
* caps.
*
*
* When the #GstJackAudioSink:connect property is set to none, the element will
* accept any number of input channels and will create (but not connect) an
* output port for each channel.
*
*
* The element will generate an error when the Jack server is shut down when it
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
* size changes at runtime.
*
* <refsect2>
* <title>Example launch line</title>
*
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc ! jackaudiosink
* ]| Play a sine wave to using jack.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

2
ext/jack/gstjackaudiosink.h
View file

@ -44,7 +44,7 @@ typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass;
/**
* GstJackAudioSink:
*
*
* Opaque #GstJackAudioSink.
*/
struct _GstJackAudioSink {

22
ext/jack/gstjackaudiosrc.c
View file

@ -42,34 +42,34 @@
/**
* SECTION:element-jackaudiosrc
* @title: jackaudiosrc
* @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer
*
* A Src that inputs data from Jack ports.
*
* It will create N Jack ports named in_&lt;name&gt;_&lt;num&gt; where
*
* It will create N Jack ports named in_&lt;name&gt;_&lt;num&gt; where
* &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
* Each port corresponds to a gstreamer channel.
*
*
* The samplerate as exposed on the caps is always the same as the samplerate of
* the jack server.
*
*
* When the #GstJackAudioSrc:connect property is set to auto, this element
* will try to connect each input port to a random physical jack output pin.
*
* will try to connect each input port to a random physical jack output pin.
*
* When the #GstJackAudioSrc:connect property is set to none, the element will
* accept any number of output channels and will create (but not connect) an
* input port for each channel.
*
*
* The element will generate an error when the Jack server is shut down when it
* was PAUSED or PLAYING. This element does not support dynamic rate and buffer
* size changes at runtime.
*
* <refsect2>
* <title>Example launch line</title>
*
* ## Example launch line
* |[
* gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
* ]| Get audio input into gstreamer from jack.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/jpeg/gstjpegdec.c
View file

@ -22,15 +22,15 @@
/**
* SECTION:element-jpegdec
* @title: jpegdec
*
* Decodes jpeg images.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v filesrc location=mjpeg.avi ! avidemux ! queue ! jpegdec ! videoconvert ! videoscale ! autovideosink
* ]| The above pipeline decode the mjpeg stream and renders it to the screen.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/jpeg/gstjpegenc.c
View file

@ -20,16 +20,16 @@
*/
/**
* SECTION:element-jpegenc
* @title: jpegenc
*
* Encodes jpeg images.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 videotestsrc num-buffers=50 ! video/x-raw, framerate='(fraction)'5/1 ! jpegenc ! avimux ! filesink location=mjpeg.avi
* ]| a pipeline to mux 5 JPEG frames per second into a 10 sec. long motion jpeg
* avi.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
ext/jpeg/gstsmokedec.c
View file

@ -19,6 +19,7 @@
/**
* SECTION:element-smokedec
* @title: smokedec
*
* Decodes images in smoke format.
*/

1
ext/jpeg/gstsmokeenc.c
View file

@ -18,6 +18,7 @@
*/
/**
* SECTION:element-smokeenc
* @title: smokeenc
*
* Encodes images in smoke format.
*/

6
ext/libcaca/gstcacasink.c
View file

@ -18,12 +18,12 @@
*/
/**
* SECTION:element-cacasink
* @title: cacasink
* @see_also: #GstAASink
*
* Displays video as color ascii art.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* CACA_GEOMETRY=160x60 CACA_FONT=5x7 gst-launch-1.0 filesrc location=test.avi ! decodebin ! videoconvert ! cacasink
* ]| This pipeline renders a video to ascii art into a separate window using a
@ -31,7 +31,7 @@
* |[
* CACA_DRIVER=ncurses gst-launch-1.0 filesrc location=test.avi ! decodebin ! videoconvert ! cacasink
* ]| This pipeline renders a video to ascii art into the current terminal.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
ext/libpng/gstpngdec.c
View file

@ -19,6 +19,7 @@
*/
/**
* SECTION:element-pngdec
* @title: pngdec
*
* Decodes png images. If there is no framerate set on sink caps, it sends EOS
* after the first picture.

1
ext/libpng/gstpngenc.c
View file

@ -20,6 +20,7 @@
*/
/**
* SECTION:element-pngenc
* @title: pngenc
*
* Encodes png images.
*/

6
ext/pulse/pulsesink.c
View file

@ -23,13 +23,13 @@
/**
* SECTION:element-pulsesink
* @title: pulsesink
* @see_also: pulsesrc
*
* This element outputs audio to a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
* ]| Play an Ogg/Vorbis file.
@ -40,7 +40,7 @@
* gst-launch-1.0 -v audiotestsrc ! pulsesink stream-properties="props,media.title=test"
* ]| Play a sine wave and set a stream property. The property can be checked
* with "pactl list".
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/pulse/pulsesrc.c
View file

@ -21,17 +21,17 @@
/**
* SECTION:element-pulsesrc
* @title: pulsesrc
* @see_also: pulsesink
*
* This element captures audio from a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
* ]| Record from a sound card using pulseaudio and encode to Ogg/Vorbis.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/raw1394/gstdv1394src.c
View file

@ -21,16 +21,16 @@
*/
/**
* SECTION:element-dv1394src
* @title: dv1394src
*
* Read DV (digital video) data from firewire port.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 dv1394src ! queue ! dvdemux name=d ! queue ! dvdec ! xvimagesink d. ! queue ! alsasink
* ]| This pipeline captures from the firewire port and displays it (might need
* format converters for audio/video).
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/raw1394/gsthdv1394src.c
View file

@ -18,18 +18,18 @@
*/
/**
* SECTION:element-hdv1394src
* @title: hdv1394src
*
* Read MPEG-TS data from firewire port.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 hdv1394src ! queue ! decodebin name=d ! queue ! xvimagesink d. ! queue ! alsasink
* ]| captures from the firewire port and plays the streams.
* |[
* gst-launch-1.0 hdv1394src ! queue ! filesink location=mydump.ts
* ]| capture to a disk file
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/shout2/gstshout2.c
View file

@ -21,18 +21,18 @@
/**
* SECTION:element-shout2send
* @title: shout2send
*
* shout2send pushes a media stream to an Icecast server
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 uridecodebin uri=file:///path/to/audiofile ! audioconvert ! vorbisenc ! oggmux ! shout2send mount=/stream.ogg port=8000 username=source password=somepassword ip=server_IP_address_or_hostname
* ]| This pipeline demuxes, decodes, re-encodes and re-muxes an audio
* media file into oggvorbis and sends the resulting stream to an Icecast
* server. Properties mount, port, username and password are all server-config
* dependent.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

8
ext/soup/gstsouphttpclientsink.c
View file

@ -18,20 +18,20 @@
*/
/**
* SECTION:element-gstsouphttpclientsink
* @title: gstsouphttpclientsink
*
* The souphttpclientsink element sends pipeline data to an HTTP server
* using HTTP PUT commands.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc num-buffers=300 ! theoraenc ! oggmux !
* souphttpclientsink location=http://server/filename.ogv
* ]|
*
*
* This example encodes 10 seconds of video and sends it to the HTTP
* server "server" using HTTP PUT commands.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/soup/gstsouphttpsrc.c
View file

@ -14,6 +14,7 @@
/**
* SECTION:element-souphttpsrc
* @title: souphttpsrc
*
* This plugin reads data from a remote location specified by a URI.
* Supported protocols are 'http', 'https'.
@ -33,8 +34,7 @@
* need to use the #ICYDemux element as follow-up element to extract the Icecast
* metadata and to determine the underlying media type.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v souphttpsrc location=https://some.server.org/index.html
* ! filesink location=/home/joe/server.html
@ -64,7 +64,7 @@
* These are used by the mime/multipart demultiplexer to emit timestamps
* on the JPEG-encoded video frame buffers. This allows the Matroska
* multiplexer to timestamp the frames in the resulting file.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/speex/gstspeexdec.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-speexdec
* @title: speexdec
* @see_also: speexenc, oggdemux
*
* This element decodes a Speex stream to raw integer audio.
@ -27,13 +28,12 @@
* 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 filesrc location=speex.ogg ! oggdemux ! speexdec ! audioconvert ! audioresample ! alsasink
* ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the
* documentation of speexenc.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/speex/gstspeexenc.c
View file

@ -19,6 +19,7 @@
/**
* SECTION:element-speexenc
* @title: speexenc
* @see_also: speexdec, oggmux
*
* This element encodes audio as a Speex stream.
@ -26,12 +27,11 @@
* 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 audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
* ]| Encode an Ogg/Speex file.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/vpx/gstvp8dec.c
View file

@ -21,6 +21,7 @@
*/
/**
* SECTION:element-vp8dec
* @title: vp8dec
* @see_also: vp8enc, matroskademux
*
* This element decodes VP8 streams into raw video.
@ -29,12 +30,11 @@
* </ulink>. It's the successor of On2 VP3, which was the base of the
* Theora video codec.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 -v filesrc location=videotestsrc.webm ! matroskademux ! vp8dec ! videoconvert ! videoscale ! autovideosink
* ]| This example pipeline will decode a WebM stream and decodes the VP8 video.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/vpx/gstvp8enc.c
View file

@ -21,6 +21,7 @@
*/
/**
* SECTION:element-vp8enc
* @title: vp8enc
* @see_also: vp8dec, webmmux, oggmux
*
* This element encodes raw video into a VP8 stream.
@ -37,13 +38,12 @@
* for explanation, examples for useful encoding parameters and more details
* on the encoding parameters.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 -v videotestsrc num-buffers=1000 ! vp8enc ! webmmux ! filesink location=videotestsrc.webm
* ]| This example pipeline will encode a test video source to VP8 muxed in an
* WebM container.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/vpx/gstvp9dec.c
View file

@ -21,6 +21,7 @@
*/
/**
* SECTION:element-vp9dec
* @title: vp9dec
* @see_also: vp9enc, matroskademux
*
* This element decodes VP9 streams into raw video.
@ -29,12 +30,11 @@
* </ulink>. It's the successor of On2 VP3, which was the base of the
* Theora video codec.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 -v filesrc location=videotestsrc.webm ! matroskademux ! vp9dec ! videoconvert ! videoscale ! autovideosink
* ]| This example pipeline will decode a WebM stream and decodes the VP9 video.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/vpx/gstvp9enc.c
View file

@ -21,6 +21,7 @@
*/
/**
* SECTION:element-vp9enc
* @title: vp9enc
* @see_also: vp9dec, webmmux, oggmux
*
* This element encodes raw video into a VP9 stream.
@ -37,13 +38,12 @@
* for explanation, examples for useful encoding parameters and more details
* on the encoding parameters.
*
* <refsect2>
* <title>Example pipeline</title>
* ## Example pipeline
* |[
* gst-launch-1.0 -v videotestsrc num-buffers=1000 ! vp9enc ! webmmux ! filesink location=videotestsrc.webm
* ]| This example pipeline will encode a test video source to VP9 muxed in an
* WebM container.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/wavpack/gstwavpackdec.c
View file

@ -23,19 +23,19 @@
/**
* SECTION:element-wavpackdec
* @title: wavpackdec
*
* WavpackDec decodes framed (for example by the WavpackParse element)
* Wavpack streams and decodes them to raw audio.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
* ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and
* tries to play it back using an automatically found audio sink.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
ext/wavpack/gstwavpackenc.c
View file

@ -21,13 +21,13 @@
/**
* SECTION:element-wavpackenc
* @title: wavpackenc
*
* WavpackEnc encodes raw audio into a framed Wavpack stream.
* <ulink url="http://www.wavpack.com/">Wavpack</ulink> is an open-source
* audio codec that features both lossless and lossy encoding.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
* ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
@ -40,7 +40,7 @@
* gst-launch-1.0 cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
* ]| This pipeline encodes audio from an audio CD into a Wavpack file using
* lossy encoding at a certain bitrate (the file will be fairly small).
* </refsect2>
*
*/
/*

3
gst/alpha/gstalpha.c
View file

@ -22,7 +22,8 @@
/**
* SECTION:element-alpha
*
* @title: alpha
*
* The alpha element adds an alpha channel to a video stream. The values
* of the alpha channel can be either be set to a constant or can be
* dynamically calculated via chroma keying, e.g. blue can be set as

3
gst/alpha/gstalpha.h
View file

@ -42,8 +42,7 @@ G_BEGIN_DECLS
typedef struct _GstAlpha GstAlpha;
typedef struct _GstAlphaClass GstAlphaClass;
/**
/**
* GstAlphaMethod:
* @ALPHA_METHOD_SET: Set/adjust alpha channel
* @ALPHA_METHOD_GREEN: Chroma Key green

1
gst/alpha/gstalphacolor.c
View file

@ -19,6 +19,7 @@
/**
* SECTION:element-alphacolor
* @title: alphacolor
*
* The alphacolor element does memory-efficient (in-place) colourspace
* conversion from RGBA to AYUV or AYUV to RGBA while preserving the

6
gst/apetag/gstapedemux.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-apedemux
* @title: apedemux
*
* apedemux accepts data streams with APE tags at the start or at the end
* (or both). The mime type of the data between the tag blocks is detected
@ -33,14 +34,13 @@
* wavparse or musepackdec, can operate on files containing APE tag
* information.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -t filesrc location=file.mpc ! apedemux ! fakesink
* ]| This pipeline should read any available APE tag information and output it.
* The contents of the file inside the APE tag regions should be detected, and
* the appropriate mime type set on buffers produced from apedemux.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"

6
gst/audiofx/audioamplify.c
View file

@ -21,18 +21,18 @@
/**
* SECTION:element-audioamplify
* @title: audioamplify
*
* Amplifies an audio stream by a given factor and allows the selection of different clipping modes.
* The difference between the clipping modes is best evaluated by testing.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc wave=saw ! audioamplify amplification=1.5 ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioamplify amplification=1.5 clipping-method=wrap-negative ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioamplify amplification=1.5 clipping-method=wrap-positive ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

12
gst/audiofx/audiochebband.c
View file

@ -34,6 +34,7 @@
/**
* SECTION:element-audiochebband
* @title: audiochebband
*
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The number of poles and the ripple parameter control the rolloff.
@ -51,19 +52,16 @@
*
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
*
* <note>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
* </note>
* > Be warned that a too large number of poles can produce noise. The most poles are possible with
* > a cutoff frequency at a quarter of the sampling rate.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc freq=1500 ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=6000 poles=4 ! audioconvert ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
* gst-launch-1.0 audiotestsrc wave=white-noise ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

12
gst/audiofx/audiocheblimit.c
View file

@ -30,6 +30,7 @@
/**
* SECTION:element-audiocheblimit
* @title: audiocheblimit
*
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
@ -47,19 +48,16 @@
*
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
*
* <note><para>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
* </para></note>
* > Be warned that a too large number of poles can produce noise. The most poles are possible with
* > a cutoff frequency at a quarter of the sampling rate.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
* gst-launch-1.0 audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/audiofx/audiodynamic.c
View file

@ -20,20 +20,20 @@
/**
* SECTION:element-audiodynamic
* @title: audiodynamic
*
* This element can act as a compressor or expander. A compressor changes the
* amplitude of all samples above a specific threshold with a specific ratio,
* a expander does the same for all samples below a specific threshold. If
* soft-knee mode is selected the ratio is applied smoothly.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 ratio=0.5 ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 ratio=4.0 ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
/* TODO: Implement attack and release parameters */

6
gst/audiofx/audioecho.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-audioecho
* @title: audioecho
*
* audioecho adds an echo or (simple) reverb effect to an audio stream. The echo
* delay, intensity and the percentage of feedback can be configured.
@ -38,14 +39,13 @@
* channels that are configured surround channels for the delay are
* selected using the surround-channels mask property.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 autoaudiosrc ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
* gst-launch-1.0 filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink
* gst-launch-1.0 audiotestsrc ! audioconvert ! audio/x-raw,channels=4 ! audioecho surround-delay=true delay=500000000 ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

10
gst/audiofx/audiofirfilter.c
View file

@ -21,6 +21,7 @@
/**
* SECTION:element-audiofirfilter
* @title: audiofirfilter
*
* audiofirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Finite_impulse_response">FIR filter</ulink>. Before usage the
* "kernel" property has to be set to the filter kernel that should be
@ -37,12 +38,11 @@
* "rate-changed" signal can be used. This should be done for most
* FIR filters as they're depending on the sampling rate.
*
* <refsect2>
* <title>Example application</title>
* <informalexample><programlisting language="C">
* ## Example application
* <programlisting language="C">
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/audiofx/firfilter-example.c" />
* </programlisting></informalexample>
* </refsect2>
* ]|
*
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray

10
gst/audiofx/audioiirfilter.c
View file

@ -21,6 +21,7 @@
/**
* SECTION:element-audioiirfilter
* @title: audioiirfilter
*
* audioiirfilter implements a generic audio <ulink url="http://en.wikipedia.org/wiki/Infinite_impulse_response">IIR filter</ulink>. Before usage the
* "a" and "b" properties have to be set to the filter coefficients that
@ -33,12 +34,11 @@
* "rate-changed" signal can be used. This should be done for most
* IIR filters as they're depending on the sampling rate.
*
* <refsect2>
* <title>Example application</title>
* <informalexample><programlisting language="C">
* ## Example application
* <programlisting language="C">
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/audiofx/iirfilter-example.c" />
* </programlisting></informalexample>
* </refsect2>
* ]|
*
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray

6
gst/audiofx/audioinvert.c
View file

@ -21,19 +21,19 @@
/**
* SECTION:element-audioinvert
* @title: audioinvert
*
* Swaps upper and lower half of audio samples. Mixing an inverted sample on top of
* the original with a slight delay can produce effects that sound like resonance.
* Creating a stereo sample from a mono source, with one channel inverted produces wide-stereo sounds.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc wave=saw ! audioinvert degree=0.4 ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert degree=0.4 ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioinvert degree=0.4 ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/audiofx/audiokaraoke.c
View file

@ -20,16 +20,16 @@
/**
* SECTION:element-audiokaraoke
* @title: audiokaraoke
*
* Remove the voice from audio by filtering the center channel.
* This plugin is useful for karaoke applications.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

8
gst/audiofx/audiopanorama.c
View file

@ -21,19 +21,19 @@
/**
* SECTION:element-audiopanorama
* @title: audiopanorama
*
* Stereo panorama effect with controllable pan position. One can choose between the default psychoacoustic panning method,
* which keeps the same perceived loudness, and a simple panning method that just controls the volume on one channel.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc wave=saw ! audiopanorama panorama=-1.00 ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiopanorama panorama=-1.00 ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiopanorama panorama=-1.00 ! audioconvert ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiopanorama method=simple panorama=-0.50 ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
@ -185,7 +185,7 @@ gst_audio_panorama_class_init (GstAudioPanoramaClass * klass)
*
* Panning method: psychoacoustic mode keeps the same perceived loudness,
* while simple mode just controls the volume of one channel. It's merely
* a matter of taste which method should be chosen.
* a matter of taste which method should be chosen.
*/
g_object_class_install_property (gobject_class, PROP_METHOD,
g_param_spec_enum ("method", "Panning method",

6
gst/audiofx/audiowsincband.c
View file

@ -32,6 +32,7 @@
/**
* SECTION:element-audiowsincband
* @title: audiowsincband
*
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The length parameter controls the rolloff, the window parameter
@ -42,14 +43,13 @@
* a much better rolloff when using a larger kernel size and almost linear phase. The only
* disadvantage is the much slower execution time with larger kernels.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc freq=1500 ! audioconvert ! audiowsincband mode=band-pass lower-frequency=3000 upper-frequency=10000 length=501 window=blackman ! audioconvert ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsincband mode=band-reject lower-frequency=59 upper-frequency=61 length=10001 window=hamming ! audioconvert ! alsasink
* gst-launch-1.0 audiotestsrc wave=white-noise ! audioconvert ! audiowsincband mode=band-pass lower-frequency=1000 upper-frequency=2000 length=31 ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/audiofx/audiowsinclimit.c
View file

@ -32,6 +32,7 @@
/**
* SECTION:element-audiowsinclimit
* @title: audiowsinclimit
*
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The length parameter controls the rolloff, the window parameter
@ -42,14 +43,13 @@
* a much better rolloff when using a larger kernel size and almost linear phase. The only
* disadvantage is the much slower execution time with larger kernels.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc freq=1500 ! audioconvert ! audiowsinclimit mode=low-pass cutoff=1000 length=501 ! audioconvert ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiowsinclimit mode=high-pass cutoff=15000 length=501 ! audioconvert ! alsasink
* gst-launch-1.0 audiotestsrc wave=white-noise ! audioconvert ! audiowsinclimit mode=low-pass cutoff=1000 length=10001 window=blackman ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

9
gst/audiofx/gstscaletempo.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-scaletempo
* @title: scaletempo
*
* Scale tempo while maintaining pitch
* (WSOLA-like technique with cross correlation)
@ -27,9 +28,8 @@
*
* Use Sceletempo to apply playback rates without the chipmunk effect.
*
* <refsect2>
* <title>Example pipelines</title>
* <para>
* ## Example pipelines
*
* |[
* filesrc location=media.ext ! decodebin name=d \
* d. ! queue ! audioconvert ! audioresample ! scaletempo ! audioconvert ! audioresample ! autoaudiosink \
@ -54,8 +54,7 @@
* correlation (roughly a dot-product). Scaletempo consumes most of its CPU
* cycles here. One can use the #GstScaletempo:search propery to tune how far
* the algoritm looks.
* </para>
* </refsect2>
*
*/
/*

6
gst/audioparsers/gstaacparse.c
View file

@ -21,6 +21,7 @@
/**
* SECTION:element-aacparse
* @title: aacparse
* @short_description: AAC parser
* @see_also: #GstAmrParse
*
@ -30,12 +31,11 @@
* be determined either. However, ADTS format AAC clips can be seeked, and parser
* can also estimate playback position and clip duration.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/audioparsers/gstac3parse.c
View file

@ -21,17 +21,17 @@
*/
/**
* SECTION:element-ac3parse
* @title: ac3parse
* @short_description: AC3 parser
* @see_also: #GstAmrParse, #GstAACParse
*
* This is an AC3 parser.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioresample ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*
*/
/* TODO:

6
gst/audioparsers/gstamrparse.c
View file

@ -22,18 +22,18 @@
/**
* SECTION:element-amrparse
* @title: amrparse
* @short_description: AMR parser
* @see_also: #GstAmrnbDec, #GstAmrnbEnc
*
* This is an AMR parser capable of handling both narrow-band and wideband
* formats.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/audioparsers/gstdcaparse.c
View file

@ -19,17 +19,17 @@
/**
* SECTION:element-dcaparse
* @title: dcaparse
* @short_description: DCA (DTS Coherent Acoustics) parser
* @see_also: #GstAmrParse, #GstAACParse, #GstAc3Parse
*
* This is a DCA (DTS Coherent Acoustics) parser.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*
*/
/* TODO:

5
gst/audioparsers/gstflacparse.c
View file

@ -23,6 +23,7 @@
/**
* SECTION:element-flacparse
* @title: flacparse
* @see_also: flacdec, oggdemux, vorbisparse
*
* The flacparse element will parse the header packets of the FLAC
@ -37,15 +38,13 @@
* which allows you to (for example) remux an ogg/flac or convert a native FLAC
* format file to an ogg bitstream.
*
* <refsect2>
* <title>Example pipelines</title>
* ## Example pipelines
* |[
* gst-launch-1.0 -v filesrc location=sine.flac ! flacparse ! identity \
* ! oggmux ! filesink location=sine-remuxed.ogg
* ]| This pipeline converts a native FLAC format file to an ogg bitstream.
* It also illustrates that the streamheader is set in the caps, and that each
* buffer has the timestamp, duration, offset, and offset_end set.
* </refsect2>
*
*/

6
gst/audioparsers/gstmpegaudioparse.c
View file

@ -21,18 +21,18 @@
*/
/**
* SECTION:element-mpegaudioparse
* @title: mpegaudioparse
* @short_description: MPEG audio parser
* @see_also: #GstAmrParse, #GstAACParse
*
* Parses and frames mpeg1 audio streams. Provides seeking.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=test.mp3 ! mpegaudioparse ! mpg123audiodec
* ! audioconvert ! audioresample ! autoaudiosink
* ]|
* </refsect2>
*
*/
/* FIXME: we should make the base class (GstBaseParse) aware of the

1
gst/audioparsers/gstsbcparse.c
View file

@ -23,6 +23,7 @@
/**
* SECTION:element-sbcparse
* @title: sbcparse
* @see_also: sbcdec, sbcenc
*
* The sbcparse element will parse a bluetooth SBC audio stream into

6
gst/audioparsers/gstwavpackparse.c
View file

@ -20,17 +20,17 @@
*/
/**
* SECTION:element-wavpackparse
* @title: wavpackparse
* @short_description: Wavpack parser
* @see_also: #GstAmrParse, #GstAACParse
*
* This is an Wavpack parser.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
gst/auparse/gstauparse.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-auparse
* @title: auparse
*
* Parses .au files mostly originating from sun os based computers.
*/

6
gst/autodetect/gstautoaudiosink.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-autoaudiosink
* @title: autoaudiosink
* @see_also: autovideosink, alsasink, osssink
*
* autoaudiosink is an audio sink that automatically detects an appropriate
@ -27,12 +28,11 @@
* that have <quote>Sink</quote> and <quote>Audio</quote> in the class field
* of their element information, and also have a non-zero autoplugging rank.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v -m audiotestsrc ! audioconvert ! audioresample ! autoaudiosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/autodetect/gstautoaudiosrc.c
View file

@ -21,6 +21,7 @@
/**
* SECTION:element-autoaudiosrc
* @title: autoaudiosrc
* @see_also: autovideosrc, alsasrc, osssrc
*
* autoaudiosrc is an audio source that automatically detects an appropriate
@ -28,12 +29,11 @@
* that have <quote>Source</quote> and <quote>Audio</quote> in the class field
* of their element information, and also have a non-zero autoplugging rank.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v -m autoaudiosrc ! audioconvert ! audioresample ! autoaudiosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/autodetect/gstautovideosink.c
View file

@ -20,6 +20,7 @@
/**
* SECTION:element-autovideosink
* @title: autovideosink
* @see_also: autoaudiosink, ximagesink, xvimagesink, sdlvideosink
*
* autovideosink is a video sink that automatically detects an appropriate
@ -27,12 +28,11 @@
* that have <quote>Sink</quote> and <quote>Video</quote> in the class field
* of their element information, and also have a non-zero autoplugging rank.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v -m videotestsrc ! autovideosink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/autodetect/gstautovideosrc.c
View file

@ -21,6 +21,7 @@
/**
* SECTION:element-autovideosrc
* @title: autovideosrc
* @see_also: autoaudiosrc, v4l2src, v4lsrc
*
* autovideosrc is a video src that automatically detects an appropriate
@ -28,12 +29,11 @@
* that have <quote>Source</quote> and <quote>Video</quote> in the class field
* of their element information, and also have a non-zero autoplugging rank.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v -m autovideosrc ! xvimagesink
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/avi/gstavidemux.c
View file

@ -22,21 +22,21 @@
/**
* SECTION:element-avidemux
* @title: avidemux
*
* Demuxes an .avi file into raw or compressed audio and/or video streams.
*
* This element supports both push and pull-based scheduling, depending on the
* capabilities of the upstream elements.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=test.avi ! avidemux name=demux demux.audio_00 ! decodebin ! audioconvert ! audioresample ! autoaudiosink demux.video_00 ! queue ! decodebin ! videoconvert ! videoscale ! autovideosink
* ]| Play (parse and decode) an .avi file and try to output it to
* an automatically detected soundcard and videosink. If the AVI file contains
* compressed audio or video data, this will only work if you have the
* right decoder elements/plugins installed.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

8
gst/avi/gstavimux.c
View file

@ -27,12 +27,12 @@
/**
* SECTION:element-avimux
* @title: avimux
*
* Muxes raw or compressed audio and/or video streams into an AVI file.
*
* <refsect2>
* <title>Example launch lines</title>
* <para>(write everything in one line, without the backslash characters)</para>
* ## Example launch lines
* (write everything in one line, without the backslash characters)
* |[
* gst-launch-1.0 videotestsrc num-buffers=250 \
* ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' \
@ -53,7 +53,7 @@
* ]| This will create an .AVI file containing the same test video and sound
* as above, only that both streams will be compressed this time. This will
* only work if you have the necessary encoder elements installed of course.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

16
gst/avi/gstavisubtitle.c
View file

@ -20,19 +20,17 @@
/**
* SECTION:element-avisubtitle
* @title: avisubtitle
*
* <refsect2>
* <para>
* Parses the subtitle stream from an avi file.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
*
* ## Example launch line
*
* |[
* gst-launch-1.0 filesrc location=subtitle.avi ! avidemux name=demux ! queue ! avisubtitle ! subparse ! textoverlay name=overlay ! videoconvert ! autovideosink demux. ! queue ! decodebin ! overlay.
* </programlisting>
* ]|
* This plays an avi file with a video and subtitle stream.
* </para>
* </refsect2>
*
*/
/* example of a subtitle chunk in an avi file

30
gst/cutter/gstcutter.c
View file

@ -20,34 +20,22 @@
*/
/**
* SECTION:element-cutter
* @title: cutter
*
* Analyses the audio signal for periods of silence. The start and end of
* silence is signalled by bus messages named
* <classname>&quot;cutter&quot;</classname>.
* The message's structure contains two fields:
* <itemizedlist>
* <listitem>
* <para>
* #GstClockTime
* <classname>&quot;timestamp&quot;</classname>:
* the timestamp of the buffer that triggered the message.
* </para>
* </listitem>
* <listitem>
* <para>
* gboolean
* <classname>&quot;above&quot;</classname>:
* %TRUE for begin of silence and %FALSE for end of silence.
* </para>
* </listitem>
* </itemizedlist>
* `cutter`.
*
* <refsect2>
* <title>Example launch line</title>
* The message's structure contains two fields:
*
* * #GstClockTime `timestamp`: the timestamp of the buffer that triggered the message.
* * gboolean `above`: %TRUE for begin of silence and %FALSE for end of silence.
*
* ## Example launch line
* |[
* gst-launch-1.0 -m filesrc location=foo.ogg ! decodebin ! audioconvert ! cutter ! autoaudiosink
* ]| Show cut messages.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
gst/debugutils/breakmydata.c
View file

@ -18,6 +18,7 @@
*/
/**
* SECTION:element-breakmydata
* @title: breakmydata
*
* This element modifies the contents of the buffer it is passed randomly
* according to the parameters set.

7
gst/debugutils/gstcapssetter.c
View file

@ -19,21 +19,22 @@
/**
* SECTION:element-capssetter
* @title: capssetter
*
* Sets or merges caps on a stream's buffers. That is, a buffer's caps are
* updated using (fields of) #GstCapsSetter:caps. Note that this may contain
* multiple structures (though not likely recommended), but each of these must
* be fixed (or will otherwise be rejected).
*
*
* If #GstCapsSetter:join is %TRUE, then the incoming caps' mime-type is
* compared to the mime-type(s) of provided caps and only matching structure(s)
* are considered for updating.
*
*
* If #GstCapsSetter:replace is %TRUE, then any caps update is preceded by
* clearing existing fields, making provided fields (as a whole) replace
* incoming ones. Otherwise, no clearing is performed, in which case provided
* fields are added/merged onto incoming caps
*
*
* Although this element might mainly serve as debug helper,
* it can also practically be used to correct a faulty pixel-aspect-ratio,
* or to modify a yuv fourcc value to effectively swap chroma components or such

6
gst/debugutils/gstpushfilesrc.c
View file

@ -19,6 +19,7 @@
/**
* SECTION:element-pushfilesrc
* @title: pushfilesrc
* @see_also: filesrc
*
* This element is only useful for debugging purposes. It implements an URI
@ -28,13 +29,12 @@
* connection with the playbin element (which creates a source based on the
* URI passed).
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -m playbin uri=pushfile:///home/you/some/file.ogg
* ]| This plays back the given file using playbin, with the demuxer operating
* push-based.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/debugutils/gsttaginject.c
View file

@ -20,19 +20,19 @@
*/
/**
* SECTION:element-taginject
* @title: taginject
*
* Element that injects new metadata tags, but passes incoming data through
* unmodified.
*
* <refsect2>
* <title>Example launch lines</title>
* ## Example launch lines
* |[
* gst-launch-1.0 audiotestsrc num-buffers=100 ! taginject tags="title=testsrc,artist=gstreamer" ! vorbisenc ! oggmux ! filesink location=test.ogg
* ]| set title and artist
* |[
* gst-launch-1.0 audiotestsrc num-buffers=100 ! taginject tags="keywords=\{\"testone\",\"audio\"\},title=\"audio\ testtone\"" ! vorbisenc ! oggmux ! filesink location=test.ogg
* ]| set keywords and title demonstrating quoting of special chars and handling lists
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/debugutils/progressreport.c
View file

@ -22,6 +22,7 @@
/**
* SECTION:element-progressreport
* @title: progressreport
*
* The progressreport element can be put into a pipeline to report progress,
* which is done by doing upstream duration and position queries in regular
@ -53,15 +54,14 @@
* is in reference to an internal point of a pipeline and not the pipeline as
* a whole).
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -m filesrc location=foo.ogg ! decodebin ! progressreport update-freq=1 ! audioconvert ! audioresample ! autoaudiosink
* ]| This shows a progress query where a duration is available.
* |[
* gst-launch-1.0 -m audiotestsrc ! progressreport update-freq=1 ! audioconvert ! autoaudiosink
* ]| This shows a progress query where no duration is available.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

1
gst/debugutils/rndbuffersize.c
View file

@ -18,6 +18,7 @@
*/
/**
* SECTION:element-rndbuffersize
* @title: rndbuffersize
*
* This element pulls buffers with random sizes from the source.
*/

81
gst/deinterlace/gstdeinterlace.c
View file

@ -22,16 +22,16 @@
/**
* SECTION:element-deinterlace
* @title: deinterlace
*
* deinterlace deinterlaces interlaced video frames to progressive video frames.
* For this different algorithms can be selected which will be described later.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v filesrc location=/path/to/file ! decodebin ! videoconvert ! deinterlace ! videoconvert ! autovideosink
* ]| This pipeline deinterlaces a video file with the default deinterlacing options.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
@ -526,69 +526,18 @@ gst_deinterlace_class_init (GstDeinterlaceClass * klass)
* the "method" child via the #GstChildProxy interface and
* setting the appropiate properties on it.
*
* <itemizedlist>
* <listitem>
* <para>
* tomsmocomp
* Motion Adaptive: Motion Search
* </para>
* </listitem>
* <listitem>
* <para>
* greedyh
* Motion Adaptive: Advanced Detection
* </para>
* </listitem>
* <listitem>
* <para>
* greedyl
* Motion Adaptive: Simple Detection
* </para>
* </listitem>
* <listitem>
* <para>
* vfir
* Blur vertical
* </para>
* </listitem>
* <listitem>
* <para>
* linear
* Linear interpolation
* </para>
* </listitem>
* <listitem>
* <para>
* linearblend
* Linear interpolation in time domain. Any motion causes significant
* ghosting, so this method should not be used.
* </para>
* </listitem>
* <listitem>
* <para>
* scalerbob
* Double lines
* </para>
* </listitem>
* <listitem>
* <para>
* weave
* Weave. Bad quality, do not use.
* </para>
* </listitem>
* <listitem>
* <para>
* weavetff
* Progressive: Top Field First. Bad quality, do not use.
* </para>
* </listitem>
* <listitem>
* <para>
* weavebff
* Progressive: Bottom Field First. Bad quality, do not use.
* </para>
* </listitem>
* </itemizedlist>
* * tomsmocomp Motion Adaptive: Motion Search
* * greedyh Motion Adaptive: Advanced Detection
* * greedyl Motion Adaptive: Simple Detection
* * vfir Blur vertical
* * linear Linear interpolation
* * linearblend Linear interpolation in time domain.
* Any motion causes significant ghosting, so this
* method should not be used.
* * scalerbob Double lines
* * weave Weave. Bad quality, do not use.
* * weavetff Progressive: Top Field First. Bad quality, do not use.
* * weavebff Progressive: Bottom Field First. Bad quality, do not use.
*/
g_object_class_install_property (gobject_class, PROP_METHOD,
g_param_spec_enum ("method",

5
gst/dtmf/gstdtmfsrc.c
View file

@ -27,6 +27,7 @@
/**
* SECTION:element-dtmfsrc
* @title: dtmfsrc
* @see_also: rtpdtmsrc, rtpdtmfmuxx
*
* The DTMFSrc element generates DTMF (ITU-T Q.23 Specification) tone packets on request
@ -99,7 +100,7 @@
* DTMFSrc element inside the pipeline) about the start of a DTMF named
* event '1' of volume -25 dBm0:
*
* <programlisting>
* |[
* structure = gst_structure_new ("dtmf-event",
* "type", G_TYPE_INT, 1,
* "number", G_TYPE_INT, 1,
@ -108,7 +109,7 @@
*
* event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
* gst_element_send_event (pipeline, event);
* </programlisting>
* ]|
*
* When a DTMF tone actually starts or stop, a "dtmf-event-processed"
* element #GstMessage with the same fields as the "dtmf-event"

1
gst/dtmf/gstrtpdtmfdepay.c
View file

@ -21,6 +21,7 @@
*/
/**
* SECTION:element-rtpdtmfdepay
* @title: rtpdtmfdepay
* @see_also: rtpdtmfsrc, rtpdtmfmux
*
* This element takes RTP DTMF packets and produces sound. It also emits a

5
gst/dtmf/gstrtpdtmfsrc.c
View file

@ -25,6 +25,7 @@
/**
* SECTION:element-rtpdtmfsrc
* @title: rtpdtmfsrc
* @see_also: dtmfsrc, rtpdtmfdepay, rtpdtmfmux
*
* The RTPDTMFSrc element generates RTP DTMF (RFC 2833) event packets on request
@ -97,7 +98,7 @@
* RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
* event '1' of volume -25 dBm0:
*
* <programlisting>
* |[
* structure = gst_structure_new ("dtmf-event",
* "type", G_TYPE_INT, 1,
* "number", G_TYPE_INT, 1,
@ -106,7 +107,7 @@
*
* event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
* gst_element_send_event (pipeline, event);
* </programlisting>
* ]|
*
* When a DTMF tone actually starts or stop, a "dtmf-event-processed"
* element #GstMessage with the same fields as the "dtmf-event"

6
gst/effectv/gstaging.c
View file

@ -26,16 +26,16 @@
/**
* SECTION:element-agingtv
* @title: agingtv
*
* AgingTV ages a video stream in realtime, changes the colors and adds
* scratches and dust.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! agingtv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of agingtv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstdice.c
View file

@ -28,6 +28,7 @@
/**
* SECTION:element-dicetv
* @title: dicetv
*
* DiceTV 'dices' the screen up into many small squares, each defaulting
* to a size of 16 pixels by 16 pixels.. Each square is rotated randomly
@ -36,12 +37,11 @@
* counterclockwise). The direction of each square normally remains
* consistent between each frame.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! dicetv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of dicetv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstedge.c
View file

@ -26,16 +26,16 @@
/**
* SECTION:element-edgetv
* @title: edgetv
*
* EdgeTV detects edges and display it in good old low resolution
* computer way.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! edgetv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of edgetv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstop.c
View file

@ -26,17 +26,17 @@
/**
* SECTION:element-optv
* @title: optv
*
* Traditional black-white optical animation is now resurrected as a
* real-time video effect. Input images are binarized and combined with
* various optical pattern.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! optv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of optv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstquark.c
View file

@ -26,16 +26,16 @@
/**
* SECTION:element-quarktv
* @title: quarktv
*
* QuarkTV disolves moving objects. It picks up pixels from
* the last frames randomly.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! quarktv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of quarktv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

8
gst/effectv/gstradioac.c
View file

@ -26,10 +26,11 @@
/**
* SECTION:element-radioactv
* @title: radioactv
*
* RadioacTV does *NOT* detect a radioactivity. It detects a difference
* from previous frame and blurs it.
*
*
* RadioacTV has 4 mode, normal, strobe1, strobe2 and trigger.
* In trigger mode, effect appears only when the trigger property is %TRUE.
*
@ -37,12 +38,11 @@
* current frame and previous frame dropped, while strobe2 mode uses the difference from
* previous frame displayed. The effect of strobe2 is stronger than strobe1.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! radioactv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of radioactv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstrev.c
View file

@ -42,17 +42,17 @@
/**
* SECTION:element-revtv
* @title: revtv
*
* RevTV acts like a video waveform monitor for each line of video
* processed. This creates a pseudo 3D effect based on the brightness
* of the video along each line.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! revtv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of revtv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstripple.c
View file

@ -30,16 +30,16 @@
/**
* SECTION:element-rippletv
* @title: rippletv
*
* RippleTV does ripple mark effect on the video input. The ripple is caused
* by motion or random rain drops.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! rippletv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of rippletv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstshagadelic.c
View file

@ -24,15 +24,15 @@
/**
* SECTION:element-shagadelictv
* @title: shagadelictv
*
* Oh behave, ShagedelicTV makes images shagadelic!
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! shagadelictv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of shagadelictv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gststreak.c
View file

@ -30,15 +30,15 @@
/**
* SECTION:element-streaktv
* @title: streaktv
*
* StreakTV makes after images of moving objects.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! streaktv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of streaktv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstvertigo.c
View file

@ -23,15 +23,15 @@
/**
* SECTION:element-vertigotv
* @title: vertigotv
*
* VertigoTV is a loopback alpha blending effector with rotating and scaling.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! vertigotv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of vertigotv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/effectv/gstwarp.c
View file

@ -35,15 +35,15 @@
/**
* SECTION:element-warptv
* @title: warptv
*
* WarpTV does realtime goo'ing of the video input.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v videotestsrc ! warptv ! videoconvert ! autovideosink
* ]| This pipeline shows the effect of warptv on a test stream.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/equalizer/gstiirequalizer10bands.c
View file

@ -19,16 +19,16 @@
/**
* SECTION:element-equalizer-10bands
* @title: equalizer-10bands
*
* The 10 band equalizer element allows to change the gain of 10 equally distributed
* frequency bands between 30 Hz and 15 kHz.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=song.ogg ! oggdemux ! vorbisdec ! audioconvert ! equalizer-10bands band2=3.0 ! alsasink
* ]| This raises the volume of the 3rd band which is at 119 Hz by 3 db.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/equalizer/gstiirequalizer3bands.c
View file

@ -19,16 +19,16 @@
/**
* SECTION:element-equalizer-3bands
* @title: equalizer-3bands
*
* The 3-band equalizer element allows to change the gain of a low frequency,
* medium frequency and high frequency band.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=song.ogg ! oggdemux ! vorbisdec ! audioconvert ! equalizer-3bands band1=6.0 ! alsasink
* ]| This raises the volume of the 2nd band, which is at 1110 Hz, by 6 db.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

27
gst/equalizer/gstiirequalizernbands.c
View file

@ -20,31 +20,30 @@
/**
* SECTION:element-equalizer-nbands
* @title: equalizer-nbands
*
* The n-band equalizer element is a fully parametric equalizer. It allows to
* select between 1 and 64 bands and has properties on each band to change
* the center frequency, band width and gain.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 filesrc location=song.ogg ! oggdemux ! vorbisdec ! audioconvert ! equalizer-nbands num-bands=15 band5::gain=6.0 ! alsasink
* ]| This make the equalizer use 15 bands and raises the volume of the 5th band by 6 db.
* </refsect2>
* <refsect2>
* <title>Example code</title>
*
* ## Example code
* |[
* #include &lt;gst/gst.h&gt;
*
*
* ...
* typedef struct {
* gfloat freq;
* gfloat width;
* gfloat gain;
* } GstEqualizerBandState;
*
*
* ...
*
*
* GstElement *equalizer;
* GObject *band;
* gint i;
@ -55,14 +54,14 @@
* {6000.0, 1000.0, 6.0},
* {3000.0, 120.0, 2.0}
* };
*
*
* ...
*
*
* equalizer = gst_element_factory_make ("equalizer-nbands", "equalizer");
* g_object_set (G_OBJECT (equalizer), "num-bands", 5, NULL);
*
*
* ...
*
*
* for (i = 0; i &lt; 5; i++) {
* band = gst_child_proxy_get_child_by_index (GST_CHILD_PROXY (equalizer), i);
* g_object_set (G_OBJECT (band), "freq", state[i].freq,
@ -70,10 +69,10 @@
* "gain", state[i].gain);
* g_object_unref (G_OBJECT (band));
* }
*
*
* ...
* ]|
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/flv/gstflvdemux.c
View file

@ -19,15 +19,15 @@
/**
* SECTION:element-flvdemux
* @title: flvdemux
*
* flvdemux demuxes an FLV file into the different contained streams.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v filesrc location=/path/to/flv ! flvdemux ! audioconvert ! autoaudiosink
* ]| This pipeline demuxes an FLV file and outputs the contained raw audio streams.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

6
gst/flv/gstflvmux.c
View file

@ -23,15 +23,15 @@
/**
* SECTION:element-flvmux
* @title: flvmux
*
* flvmux muxes different streams into an FLV file.
*
* <refsect2>
* <title>Example launch line</title>
* ## Example launch line
* |[
* gst-launch-1.0 -v flvmux name=mux ! filesink location=test.flv audiotestsrc samplesperbuffer=44100 num-buffers=10 ! faac ! mux. videotestsrc num-buffers=250 ! video/x-raw,framerate=25/1 ! x264enc ! mux.
* ]| This pipeline encodes a test audio and video stream and muxes both into an FLV file.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H

9
gst/flv/gstindex.c
View file

@ -22,6 +22,7 @@
/**
* SECTION:gstindex
* @title: GstIndex
* @short_description: Generate indexes on objects
* @see_also: #GstIndexFactory
*
@ -702,11 +703,9 @@ gst_index_gtype_resolver (GstIndex * index, GstObject * writer,
* to a string. That string will be used to register or look up an id
* in the index.
*
* <note>
* The caller must not hold @writer's #GST_OBJECT_LOCK, as the default
* resolver may call functions that take the object lock as well, and
* the lock is not recursive.
* </note>
* > The caller must not hold @writer's #GST_OBJECT_LOCK, as the default
* > resolver may call functions that take the object lock as well, and
* > the lock is not recursive.
*
* Returns: TRUE if the writer would be mapped to an id.
*/

1
gst/flx/gstflxdec.c
View file

@ -19,6 +19,7 @@
*/
/**
* SECTION:element-flxdec
* @title: flxdec
*
* This element decodes fli/flc/flx-video into raw video
*/

21
gst/goom/filters.c
View file

@ -150,14 +150,14 @@ typedef struct _ZOOM_FILTER_FX_WRAPPER_DATA
int mustInitBuffers;
int interlace_start;
/** modif by jeko : fixedpoint : buffration = (16:16) (donc 0<=buffration<=2^16) */
/* modif by jeko : fixedpoint : buffration = (16:16) (donc 0<=buffration<=2^16) */
int buffratio;
int *firedec;
/** modif d'optim by Jeko : precalcul des 4 coefs resultant des 2 pos */
/* modif d'optim by Jeko : precalcul des 4 coefs resultant des 2 pos */
int precalCoef[BUFFPOINTNB][BUFFPOINTNB];
/** calculatePXandPY statics */
/* calculatePXandPY statics */
int wave;
int wavesp;
@ -509,7 +509,7 @@ c_zoom (Pixel * expix1, Pixel * expix2, unsigned int prevX, unsigned int prevY,
}
}
/** generate the water fx horizontal direction buffer */
/* generate the water fx horizontal direction buffer */
static void
generateTheWaterFXHorizontalDirectionBuffer (PluginInfo * goomInfo,
ZoomFilterFXWrapperData * data)
@ -558,13 +558,12 @@ generateTheWaterFXHorizontalDirectionBuffer (PluginInfo * goomInfo,
/**
* Main work for the dynamic displacement map.
/*
* Main work for the dynamic displacement map.
*
* Reads data from pix1, write to pix2.
*
* Useful datas for this FX are stored in ZoomFilterData.
*
* If you think that this is a strange function name, let me say that a long time ago,
* there has been a slow version and a gray-level only one. Then came these function,
* fast and workin in RGB colorspace ! nice but it only was applying a zoom to the image.
@ -583,7 +582,7 @@ zoomFilterFastRGB (PluginInfo * goomInfo, Pixel * pix1, Pixel * pix2,
if (!BVAL (data->enabled_bp))
return;
/** changement de taille **/
/* changement de taille */
if ((data->prevX != resx) || (data->prevY != resy)) {
data->prevX = resx;
data->prevY = resy;
@ -609,7 +608,7 @@ zoomFilterFastRGB (PluginInfo * goomInfo, Pixel * pix1, Pixel * pix2,
if (data->interlace_start != -2)
zf = NULL;
/** changement de config **/
/* changement de config */
if (zf) {
data->reverse = zf->reverse;
data->general_speed = (float) (zf->vitesse - 128) / 128.0f;
@ -786,7 +785,7 @@ zoomFilterVisualFXWrapper_init (struct _VISUAL_FX *_this, PluginInfo * info)
data->hPlaneEffect = 0;
data->noisify = 2;
/** modif by jeko : fixedpoint : buffration = (16:16) (donc 0<=buffration<=2^16) */
/* modif by jeko : fixedpoint : buffration = (16:16) (donc 0<=buffration<=2^16) */
data->buffratio = 0;
data->firedec = 0;
@ -800,7 +799,7 @@ zoomFilterVisualFXWrapper_init (struct _VISUAL_FX *_this, PluginInfo * info)
_this->params = &data->params;
_this->fx_data = (void *) data;
/** modif d'optim by Jeko : precalcul des 4 coefs resultant des 2 pos */
/* modif d'optim by Jeko : precalcul des 4 coefs resultant des 2 pos */
generatePrecalCoef (data->precalCoef);
}

2
gst/goom/goom_config.h
View file

@ -26,7 +26,7 @@
#if 1
/* ndef COLOR_BGRA */
/** position des composantes **/
/* position des composantes */
#define BLEU 0
#define VERT 1
#define ROUGE 2

Some files were not shown because too many files have changed in this diff Show more