diff options
| author | Thibault Saunier <tsaunier@igalia.com> | 2018-10-22 11:39:24 +0200 |
|---|---|---|
| committer | Thibault Saunier <tsaunier@igalia.com> | 2019-05-13 10:24:40 -0400 |
| commit | 0a6a62aa7652f0b5a689d4cecbc031b763c2000a (patch) | |
| tree | b997ae369140d0711df167b672149cd1319a3cf0 | |
| parent | 135e12565b5b79ea5d2d2f15b569060ce2c64b76 (diff) | |
| download | gstreamer-plugins-good-0a6a62aa7652f0b5a689d4cecbc031b763c2000a.tar.gz | |
docs: Port all docstring to gtk-doc markdown
205 files changed, 805 insertions, 1081 deletions
diff --git a/ext/aalib/gstaasink.c b/ext/aalib/gstaasink.c index 880e47957..ba21633cd 100644 --- a/ext/aalib/gstaasink.c +++ b/ext/aalib/gstaasink.c @@ -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 diff --git a/ext/cairo/gstcairooverlay.c b/ext/cairo/gstcairooverlay.c index 6cea06f54..e751ad05f 100644 --- a/ext/cairo/gstcairooverlay.c +++ b/ext/cairo/gstcairooverlay.c @@ -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 <gst/gst.h> @@ -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, "draw", G_CALLBACK (draw_overlay), * overlay_state); - * g_signal_connect (cairo_overlay, "caps-changed", + * g_signal_connect (cairo_overlay, "caps-changed", * 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] = diff --git a/ext/dv/gstdvdec.c b/ext/dv/gstdvdec.c index c7c193d75..c9c39bf47 100644 --- a/ext/dv/gstdvdec.c +++ b/ext/dv/gstdvdec.c @@ -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 diff --git a/ext/dv/gstdvdemux.c b/ext/dv/gstdvdemux.c index 65d24d3e4..f0494e269 100644 --- a/ext/dv/gstdvdemux.c +++ b/ext/dv/gstdvdemux.c @@ -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 diff --git a/ext/flac/gstflacdec.c b/ext/flac/gstflacdec.c index 4583ceb4e..e0c54f832 100644 --- a/ext/flac/gstflacdec.c +++ b/ext/flac/gstflacdec.c @@ -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 diff --git a/ext/flac/gstflacenc.c b/ext/flac/gstflacenc.c index 6f3312a58..4bd318fa3 100644 --- a/ext/flac/gstflacenc.c +++ b/ext/flac/gstflacenc.c @@ -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 diff --git a/ext/flac/gstflactag.c b/ext/flac/gstflactag.c index e2e239f4a..12131cbb8 100644 --- a/ext/flac/gstflactag.c +++ b/ext/flac/gstflactag.c @@ -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 diff --git a/ext/gdk_pixbuf/gstgdkpixbufoverlay.c b/ext/gdk_pixbuf/gstgdkpixbufoverlay.c index 38b8ecc97..8981d9603 100644 --- a/ext/gdk_pixbuf/gstgdkpixbufoverlay.c +++ b/ext/gdk_pixbuf/gstgdkpixbufoverlay.c @@ -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 diff --git a/ext/gdk_pixbuf/gstgdkpixbufsink.c b/ext/gdk_pixbuf/gstgdkpixbufsink.c index 6c4cf3dc9..007971f56 100644 --- a/ext/gdk_pixbuf/gstgdkpixbufsink.c +++ b/ext/gdk_pixbuf/gstgdkpixbufsink.c @@ -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>"preroll-pixbuf" - * </classname> or <classname>"pixbuf"</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>"preroll-pixbuf" - * </classname> or <classname>"pixbuf"</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>"pixbuf"</classname>: the #GdkPixbuf object - * </listitem> - * <listitem> - * <classname>"pixel-aspect-ratio"</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 '(NULL)'). - * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index 19022c082..bba2d7de6 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -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 * diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 701d25a84..abf7aa58f 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -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_<name>_<num> where + * + * It will create N Jack ports named out_<name>_<num> where * <name> is the element name and <num> 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 diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index 7fccc8a48..f12504c1d 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -44,7 +44,7 @@ typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass; /** * GstJackAudioSink: - * + * * Opaque #GstJackAudioSink. */ struct _GstJackAudioSink { diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 931f4dddb..1d2cf82ee 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -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_<name>_<num> where + * + * It will create N Jack ports named in_<name>_<num> where * <name> is the element name and <num> 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 diff --git a/ext/jpeg/gstjpegdec.c b/ext/jpeg/gstjpegdec.c index d88b2583d..c38259914 100644 --- a/ext/jpeg/gstjpegdec.c +++ b/ext/jpeg/gstjpegdec.c @@ -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 diff --git a/ext/jpeg/gstjpegenc.c b/ext/jpeg/gstjpegenc.c index 504f1953b..f2a325f05 100644 --- a/ext/jpeg/gstjpegenc.c +++ b/ext/jpeg/gstjpegenc.c @@ -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 diff --git a/ext/jpeg/gstsmokedec.c b/ext/jpeg/gstsmokedec.c index 816d9546d..82b69eefa 100644 --- a/ext/jpeg/gstsmokedec.c +++ b/ext/jpeg/gstsmokedec.c @@ -19,6 +19,7 @@ /** * SECTION:element-smokedec + * @title: smokedec * * Decodes images in smoke format. */ diff --git a/ext/jpeg/gstsmokeenc.c b/ext/jpeg/gstsmokeenc.c index 32b068738..e50dfab03 100644 --- a/ext/jpeg/gstsmokeenc.c +++ b/ext/jpeg/gstsmokeenc.c @@ -18,6 +18,7 @@ */ /** * SECTION:element-smokeenc + * @title: smokeenc * * Encodes images in smoke format. */ diff --git a/ext/libcaca/gstcacasink.c b/ext/libcaca/gstcacasink.c index 40fb52a70..843fbf504 100644 --- a/ext/libcaca/gstcacasink.c +++ b/ext/libcaca/gstcacasink.c @@ -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 diff --git a/ext/libpng/gstpngdec.c b/ext/libpng/gstpngdec.c index 581e3a9f4..4592fef06 100644 --- a/ext/libpng/gstpngdec.c +++ b/ext/libpng/gstpngdec.c @@ -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. diff --git a/ext/libpng/gstpngenc.c b/ext/libpng/gstpngenc.c index e9050b13a..bb0c5cb94 100644 --- a/ext/libpng/gstpngenc.c +++ b/ext/libpng/gstpngenc.c @@ -20,6 +20,7 @@ */ /** * SECTION:element-pngenc + * @title: pngenc * * Encodes png images. */ diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index ca7591506..7ce9cd851 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -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 diff --git a/ext/pulse/pulsesrc.c b/ext/pulse/pulsesrc.c index 30c6e7ff4..8beb1652c 100644 --- a/ext/pulse/pulsesrc.c +++ b/ext/pulse/pulsesrc.c @@ -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 diff --git a/ext/raw1394/gstdv1394src.c b/ext/raw1394/gstdv1394src.c index 30c443ac3..f84259a15 100644 --- a/ext/raw1394/gstdv1394src.c +++ b/ext/raw1394/gstdv1394src.c @@ -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 diff --git a/ext/raw1394/gsthdv1394src.c b/ext/raw1394/gsthdv1394src.c index e9e39805a..9ecb4e8ba 100644 --- a/ext/raw1394/gsthdv1394src.c +++ b/ext/raw1394/gsthdv1394src.c @@ -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 diff --git a/ext/shout2/gstshout2.c b/ext/shout2/gstshout2.c index 9e2d60f4d..3b64642c8 100644 --- a/ext/shout2/gstshout2.c +++ b/ext/shout2/gstshout2.c @@ -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 diff --git a/ext/soup/gstsouphttpclientsink.c b/ext/soup/gstsouphttpclientsink.c index c81248874..17c69f5ca 100644 --- a/ext/soup/gstsouphttpclientsink.c +++ b/ext/soup/gstsouphttpclientsink.c @@ -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 diff --git a/ext/soup/gstsouphttpsrc.c b/ext/soup/gstsouphttpsrc.c index 1bedd4e92..1484a2bdf 100644 --- a/ext/soup/gstsouphttpsrc.c +++ b/ext/soup/gstsouphttpsrc.c @@ -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 diff --git a/ext/speex/gstspeexdec.c b/ext/speex/gstspeexdec.c index cc123ce06..b6106f692 100644 --- a/ext/speex/gstspeexdec.c +++ b/ext/speex/gstspeexdec.c @@ -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 diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c index 4266af63b..5dd205fc8 100644 --- a/ext/speex/gstspeexenc.c +++ b/ext/speex/gstspeexenc.c @@ -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 diff --git a/ext/vpx/gstvp8dec.c b/ext/vpx/gstvp8dec.c index 6a9625cd4..fcfca75e0 100644 --- a/ext/vpx/gstvp8dec.c +++ b/ext/vpx/gstvp8dec.c @@ -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 diff --git a/ext/vpx/gstvp8enc.c b/ext/vpx/gstvp8enc.c index 819b38827..310d4ddeb 100644 --- a/ext/vpx/gstvp8enc.c +++ b/ext/vpx/gstvp8enc.c @@ -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 diff --git a/ext/vpx/gstvp9dec.c b/ext/vpx/gstvp9dec.c index 69bb0fbd5..2cb8fb974 100644 --- a/ext/vpx/gstvp9dec.c +++ b/ext/vpx/gstvp9dec.c @@ -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 diff --git a/ext/vpx/gstvp9enc.c b/ext/vpx/gstvp9enc.c index 0d8f9a2c6..5d77ad7e5 100644 --- a/ext/vpx/gstvp9enc.c +++ b/ext/vpx/gstvp9enc.c @@ -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 diff --git a/ext/wavpack/gstwavpackdec.c b/ext/wavpack/gstwavpackdec.c index 7cce54322..0d0648670 100644 --- a/ext/wavpack/gstwavpackdec.c +++ b/ext/wavpack/gstwavpackdec.c @@ -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 diff --git a/ext/wavpack/gstwavpackenc.c b/ext/wavpack/gstwavpackenc.c index 5d6205cad..95bd8f2b5 100644 --- a/ext/wavpack/gstwavpackenc.c +++ b/ext/wavpack/gstwavpackenc.c @@ -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> + * */ /* diff --git a/gst/alpha/gstalpha.c b/gst/alpha/gstalpha.c index 82ad98167..867e00043 100644 --- a/gst/alpha/gstalpha.c +++ b/gst/alpha/gstalpha.c @@ -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 diff --git a/gst/alpha/gstalpha.h b/gst/alpha/gstalpha.h index 948fb78a5..05b6ab681 100644 --- a/gst/alpha/gstalpha.h +++ b/gst/alpha/gstalpha.h @@ -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 diff --git a/gst/alpha/gstalphacolor.c b/gst/alpha/gstalphacolor.c index e082e74a8..295ee9bc6 100644 --- a/gst/alpha/gstalphacolor.c +++ b/gst/alpha/gstalphacolor.c @@ -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 diff --git a/gst/apetag/gstapedemux.c b/gst/apetag/gstapedemux.c index b3e62894f..ce83835f8 100644 --- a/gst/apetag/gstapedemux.c +++ b/gst/apetag/gstapedemux.c @@ -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" diff --git a/gst/audiofx/audioamplify.c b/gst/audiofx/audioamplify.c index 726ce0e3a..f96093a8f 100644 --- a/gst/audiofx/audioamplify.c +++ b/gst/audiofx/audioamplify.c @@ -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 diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c index 78febe9e3..0751963ee 100644 --- a/gst/audiofx/audiochebband.c +++ b/gst/audiofx/audiochebband.c @@ -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 diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c index e27888666..c02b62e5a 100644 --- a/gst/audiofx/audiocheblimit.c +++ b/gst/audiofx/audiocheblimit.c @@ -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 diff --git a/gst/audiofx/audiodynamic.c b/gst/audiofx/audiodynamic.c index 9b3a62c89..215577cb9 100644 --- a/gst/audiofx/audiodynamic.c +++ b/gst/audiofx/audiodynamic.c @@ -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 */ diff --git a/gst/audiofx/audioecho.c b/gst/audiofx/audioecho.c index c65bef1b7..948abf7ca 100644 --- a/gst/audiofx/audioecho.c +++ b/gst/audiofx/audioecho.c @@ -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 diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c index 0ab32f375..1e3a2abaa 100644 --- a/gst/audiofx/audiofirfilter.c +++ b/gst/audiofx/audiofirfilter.c @@ -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 diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c index eb4f20ce8..81b0bbca8 100644 --- a/gst/audiofx/audioiirfilter.c +++ b/gst/audiofx/audioiirfilter.c @@ -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 diff --git a/gst/audiofx/audioinvert.c b/gst/audiofx/audioinvert.c index a68dd1071..eeb3e784b 100644 --- a/gst/audiofx/audioinvert.c +++ b/gst/audiofx/audioinvert.c @@ -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 diff --git a/gst/audiofx/audiokaraoke.c b/gst/audiofx/audiokaraoke.c index 76697ca17..0d0292603 100644 --- a/gst/audiofx/audiokaraoke.c +++ b/gst/audiofx/audiokaraoke.c @@ -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 diff --git a/gst/audiofx/audiopanorama.c b/gst/audiofx/audiopanorama.c index b901464ae..fe1f2a1cf 100644 --- a/gst/audiofx/audiopanorama.c +++ b/gst/audiofx/audiopanorama.c @@ -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", diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index 3a66d410d..6159bb8cb 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -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 diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index 38eb00717..54bb90332 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.c @@ -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 diff --git a/gst/audiofx/gstscaletempo.c b/gst/audiofx/gstscaletempo.c index 83ee8fe24..f83e8a51a 100644 --- a/gst/audiofx/gstscaletempo.c +++ b/gst/audiofx/gstscaletempo.c @@ -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> + * */ /* diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c index ad81a5468..0d4bf84d7 100644 --- a/gst/audioparsers/gstaacparse.c +++ b/gst/audioparsers/gstaacparse.c @@ -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 diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c index afc8770bc..0c4cbe61a 100644 --- a/gst/audioparsers/gstac3parse.c +++ b/gst/audioparsers/gstac3parse.c @@ -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: diff --git a/gst/audioparsers/gstamrparse.c b/gst/audioparsers/gstamrparse.c index ca0135923..2b6277736 100644 --- a/gst/audioparsers/gstamrparse.c +++ b/gst/audioparsers/gstamrparse.c @@ -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 diff --git a/gst/audioparsers/gstdcaparse.c b/gst/audioparsers/gstdcaparse.c index dbb0b5e27..0d06f539a 100644 --- a/gst/audioparsers/gstdcaparse.c +++ b/gst/audioparsers/gstdcaparse.c @@ -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: diff --git a/gst/audioparsers/gstflacparse.c b/gst/audioparsers/gstflacparse.c index 2758d4cfc..c2a09ffcf 100644 --- a/gst/audioparsers/gstflacparse.c +++ b/gst/audioparsers/gstflacparse.c @@ -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> * */ diff --git a/gst/audioparsers/gstmpegaudioparse.c b/gst/audioparsers/gstmpegaudioparse.c index cfad8833a..b570b1aac 100644 --- a/gst/audioparsers/gstmpegaudioparse.c +++ b/gst/audioparsers/gstmpegaudioparse.c @@ -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 diff --git a/gst/audioparsers/gstsbcparse.c b/gst/audioparsers/gstsbcparse.c index 12c39d95e..fd521f86b 100644 --- a/gst/audioparsers/gstsbcparse.c +++ b/gst/audioparsers/gstsbcparse.c @@ -23,6 +23,7 @@ /** * SECTION:element-sbcparse + * @title: sbcparse * @see_also: sbcdec, sbcenc * * The sbcparse element will parse a bluetooth SBC audio stream into diff --git a/gst/audioparsers/gstwavpackparse.c b/gst/audioparsers/gstwavpackparse.c index 7dc34cf44..df0e2236a 100644 --- a/gst/audioparsers/gstwavpackparse.c +++ b/gst/audioparsers/gstwavpackparse.c @@ -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 diff --git a/gst/auparse/gstauparse.c b/gst/auparse/gstauparse.c index b814ad813..78d100ab4 100644 --- a/gst/auparse/gstauparse.c +++ b/gst/auparse/gstauparse.c @@ -20,6 +20,7 @@ /** * SECTION:element-auparse + * @title: auparse * * Parses .au files mostly originating from sun os based computers. */ diff --git a/gst/autodetect/gstautoaudiosink.c b/gst/autodetect/gstautoaudiosink.c index 1edf6d879..f8f53ac4c 100644 --- a/gst/autodetect/gstautoaudiosink.c +++ b/gst/autodetect/gstautoaudiosink.c @@ -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 diff --git a/gst/autodetect/gstautoaudiosrc.c b/gst/autodetect/gstautoaudiosrc.c index 8adda00d5..828d6ad07 100644 --- a/gst/autodetect/gstautoaudiosrc.c +++ b/gst/autodetect/gstautoaudiosrc.c @@ -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 diff --git a/gst/autodetect/gstautovideosink.c b/gst/autodetect/gstautovideosink.c index e4fd91ebd..8607b46cf 100644 --- a/gst/autodetect/gstautovideosink.c +++ b/gst/autodetect/gstautovideosink.c @@ -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 diff --git a/gst/autodetect/gstautovideosrc.c b/gst/autodetect/gstautovideosrc.c index 2f4343144..f0d12a6b4 100644 --- a/gst/autodetect/gstautovideosrc.c +++ b/gst/autodetect/gstautovideosrc.c @@ -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 diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 834af76cb..ebd120ba0 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -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 diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index 593ed4425..b8d995a58 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -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 diff --git a/gst/avi/gstavisubtitle.c b/gst/avi/gstavisubtitle.c index 25c9cca20..0b36b561d 100644 --- a/gst/avi/gstavisubtitle.c +++ b/gst/avi/gstavisubtitle.c @@ -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 diff --git a/gst/cutter/gstcutter.c b/gst/cutter/gstcutter.c index b13ddf6c3..a8d5d449d 100644 --- a/gst/cutter/gstcutter.c +++ b/gst/cutter/gstcutter.c @@ -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>"cutter"</classname>. + * `cutter`. + * * The message's structure contains two fields: - * <itemizedlist> - * <listitem> - * <para> - * #GstClockTime - * <classname>"timestamp"</classname>: - * the timestamp of the buffer that triggered the message. - * </para> - * </listitem> - * <listitem> - * <para> - * gboolean - * <classname>"above"</classname>: - * %TRUE for begin of silence and %FALSE for end of silence. - * </para> - * </listitem> - * </itemizedlist> * - * <refsect2> - * <title>Example launch line</title> + * * #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 diff --git a/gst/debugutils/breakmydata.c b/gst/debugutils/breakmydata.c index 3722d30f5..4bc9a6423 100644 --- a/gst/debugutils/breakmydata.c +++ b/gst/debugutils/breakmydata.c @@ -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. diff --git a/gst/debugutils/gstcapssetter.c b/gst/debugutils/gstcapssetter.c index 6b3db76d8..24297797a 100644 --- a/gst/debugutils/gstcapssetter.c +++ b/gst/debugutils/gstcapssetter.c @@ -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 diff --git a/gst/debugutils/gstpushfilesrc.c b/gst/debugutils/gstpushfilesrc.c index c60accb09..f51dd0460 100644 --- a/gst/debugutils/gstpushfilesrc.c +++ b/gst/debugutils/gstpushfilesrc.c @@ -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 diff --git a/gst/debugutils/gsttaginject.c b/gst/debugutils/gsttaginject.c index 46b0d9afb..7344de54d 100644 --- a/gst/debugutils/gsttaginject.c +++ b/gst/debugutils/gsttaginject.c @@ -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 diff --git a/gst/debugutils/progressreport.c b/gst/debugutils/progressreport.c index 906bf0f05..a3be8adc8 100644 --- a/gst/debugutils/progressreport.c +++ b/gst/debugutils/progressreport.c @@ -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 diff --git a/gst/debugutils/rndbuffersize.c b/gst/debugutils/rndbuffersize.c index d33a23c7c..c99cbdcb6 100644 --- a/gst/debugutils/rndbuffersize.c +++ b/gst/debugutils/rndbuffersize.c @@ -18,6 +18,7 @@ */ /** * SECTION:element-rndbuffersize + * @title: rndbuffersize * * This element pulls buffers with random sizes from the source. */ diff --git a/gst/deinterlace/gstdeinterlace.c b/gst/deinterlace/gstdeinterlace.c index 595091c9c..8c218b87c 100644 --- a/gst/deinterlace/gstdeinterlace.c +++ b/gst/deinterlace/gstdeinterlace.c @@ -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", diff --git a/gst/dtmf/gstdtmfsrc.c b/gst/dtmf/gstdtmfsrc.c index bd011667d..c72801975 100644 --- a/gst/dtmf/gstdtmfsrc.c +++ b/gst/dtmf/gstdtmfsrc.c @@ -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" diff --git a/gst/dtmf/gstrtpdtmfdepay.c b/gst/dtmf/gstrtpdtmfdepay.c index b7827be7a..67bb54af3 100644 --- a/gst/dtmf/gstrtpdtmfdepay.c +++ b/gst/dtmf/gstrtpdtmfdepay.c @@ -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 diff --git a/gst/dtmf/gstrtpdtmfsrc.c b/gst/dtmf/gstrtpdtmfsrc.c index 9c783d392..dfd3b9d1f 100644 --- a/gst/dtmf/gstrtpdtmfsrc.c +++ b/gst/dtmf/gstrtpdtmfsrc.c @@ -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" diff --git a/gst/effectv/gstaging.c b/gst/effectv/gstaging.c index a91b63b32..d6d28a3a8 100644 --- a/gst/effectv/gstaging.c +++ b/gst/effectv/gstaging.c @@ -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 diff --git a/gst/effectv/gstdice.c b/gst/effectv/gstdice.c index 5c6e11107..e6acd9458 100644 --- a/gst/effectv/gstdice.c +++ b/gst/effectv/gstdice.c @@ -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 diff --git a/gst/effectv/gstedge.c b/gst/effectv/gstedge.c index 07feb990b..79632a69b 100644 --- a/gst/effectv/gstedge.c +++ b/gst/effectv/gstedge.c @@ -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 diff --git a/gst/effectv/gstop.c b/gst/effectv/gstop.c index 283878f8c..ec636b38e 100644 --- a/gst/effectv/gstop.c +++ b/gst/effectv/gstop.c @@ -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 diff --git a/gst/effectv/gstquark.c b/gst/effectv/gstquark.c index 12ade750c..ebbf4b1d7 100644 --- a/gst/effectv/gstquark.c +++ b/gst/effectv/gstquark.c @@ -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 diff --git a/gst/effectv/gstradioac.c b/gst/effectv/gstradioac.c index b663684c4..a36a5b9a7 100644 --- a/gst/effectv/gstradioac.c +++ b/gst/effectv/gstradioac.c @@ -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 diff --git a/gst/effectv/gstrev.c b/gst/effectv/gstrev.c index 2f95950fb..59c2e36f9 100644 --- a/gst/effectv/gstrev.c +++ b/gst/effectv/gstrev.c @@ -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 diff --git a/gst/effectv/gstripple.c b/gst/effectv/gstripple.c index a4ac6f79e..577c7d6fb 100644 --- a/gst/effectv/gstripple.c +++ b/gst/effectv/gstripple.c @@ -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 diff --git a/gst/effectv/gstshagadelic.c b/gst/effectv/gstshagadelic.c index 721a1513a..e220234ef 100644 --- a/gst/effectv/gstshagadelic.c +++ b/gst/effectv/gstshagadelic.c @@ -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 diff --git a/gst/effectv/gststreak.c b/gst/effectv/gststreak.c index 44ed1507d..312dccad1 100644 --- a/gst/effectv/gststreak.c +++ b/gst/effectv/gststreak.c @@ -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 diff --git a/gst/effectv/gstvertigo.c b/gst/effectv/gstvertigo.c index f49a7d2ee..ac047fe2a 100644 --- a/gst/effectv/gstvertigo.c +++ b/gst/effectv/gstvertigo.c @@ -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 diff --git a/gst/effectv/gstwarp.c b/gst/effectv/gstwarp.c index 841ed8ccb..651fcd74d 100644 --- a/gst/effectv/gstwarp.c +++ b/gst/effectv/gstwarp.c @@ -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 diff --git a/gst/equalizer/gstiirequalizer10bands.c b/gst/equalizer/gstiirequalizer10bands.c index 71bbf53ee..d8492f658 100644 --- a/gst/equalizer/gstiirequalizer10bands.c +++ b/gst/equalizer/gstiirequalizer10bands.c @@ -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 diff --git a/gst/equalizer/gstiirequalizer3bands.c b/gst/equalizer/gstiirequalizer3bands.c index 02f14ed4a..67ab2c4cc 100644 --- a/gst/equalizer/gstiirequalizer3bands.c +++ b/gst/equalizer/gstiirequalizer3bands.c @@ -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 diff --git a/gst/equalizer/gstiirequalizernbands.c b/gst/equalizer/gstiirequalizernbands.c index c8325c622..da7ea3341 100644 --- a/gst/equalizer/gstiirequalizernbands.c +++ b/gst/equalizer/gstiirequalizernbands.c @@ -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 <gst/gst.h> - * + * * ... * 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 < 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 diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c index 377d64733..6612dc911 100644 --- a/gst/flv/gstflvdemux.c +++ b/gst/flv/gstflvdemux.c @@ -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 diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index f02c08d1b..cbabec76b 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -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 diff --git a/gst/flv/gstindex.c b/gst/flv/gstindex.c index d64a38a45..8a2edfe28 100644 --- a/gst/flv/gstindex.c +++ b/gst/flv/gstindex.c @@ -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. */ diff --git a/gst/flx/gstflxdec.c b/gst/flx/gstflxdec.c index 37d552aae..63ba67261 100644 --- a/gst/flx/gstflxdec.c +++ b/gst/flx/gstflxdec.c @@ -19,6 +19,7 @@ */ /** * SECTION:element-flxdec + * @title: flxdec * * This element decodes fli/flc/flx-video into raw video */ diff --git a/gst/goom/filters.c b/gst/goom/filters.c index 703d20263..d0521fef0 100644 --- a/gst/goom/filters.c +++ b/gst/goom/filters.c @@ -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); } diff --git a/gst/goom/goom_config.h b/gst/goom/goom_config.h index 7264bfe4e..39b95a0f7 100644 --- a/gst/goom/goom_config.h +++ b/gst/goom/goom_config.h @@ -26,7 +26,7 @@ #if 1 /* ndef COLOR_BGRA */ -/** position des composantes **/ +/* position des composantes */ #define BLEU 0 #define VERT 1 #define ROUGE 2 diff --git a/gst/goom/goom_filters.h b/gst/goom/goom_filters.h index 13096e270..e4cfaeb0c 100644 --- a/gst/goom/goom_filters.h +++ b/gst/goom/goom_filters.h @@ -35,10 +35,10 @@ struct _ZOOM_FILTER_DATA int middleX, middleY; /* milieu de l'effet */ char reverse; /* inverse la vitesse */ char mode; /* type d'effet � appliquer (cf les #define) */ - /** @since June 2001 */ + /* @since June 2001 */ int hPlaneEffect; /* deviation horitontale */ int vPlaneEffect; /* deviation verticale */ - /** @since April 2002 */ + /* @since April 2002 */ int waveEffect; /* applique une "surcouche" de wave effect */ int hypercosEffect; /* applique une "surcouche de hypercos effect */ diff --git a/gst/goom/goom_plugin_info.h b/gst/goom/goom_plugin_info.h index da0e96e58..907d780c2 100644 --- a/gst/goom/goom_plugin_info.h +++ b/gst/goom/goom_plugin_info.h @@ -43,7 +43,7 @@ typedef struct { #define STATES_MAX_NB 128 -/** +/* * Gives informations about the sound. */ struct _SOUND_INFO { @@ -86,7 +86,7 @@ struct _SOUND_INFO { }; -/** +/* * Allows FXs to know the current state of the plugin. */ struct _PLUGIN_INFO { @@ -109,21 +109,21 @@ struct _PLUGIN_INFO { int nbVisuals; VisualFX **visuals; /* pointers on all the visual fx */ - /** The known FX */ + /* The known FX */ VisualFX convolve_fx; VisualFX star_fx; VisualFX zoomFilter_fx; VisualFX tentacles_fx; VisualFX ifs_fx; - /** image buffers */ + /* image buffers */ guint32 *pixel; guint32 *back; Pixel *p1, *p2; Pixel *conv; Pixel *outputBuf; - /** state of goom */ + /* state of goom */ guint32 cycle; GoomState states[STATES_MAX_NB]; int statesNumber; @@ -131,16 +131,16 @@ struct _PLUGIN_INFO { GoomState *curGState; - /** effet de ligne.. */ + /* effet de ligne.. */ GMLine *gmline1; GMLine *gmline2; - /** sinus table */ + /* sinus table */ int sintable[0x10000]; /* INTERNALS */ - /** goom_update internals. + /* goom_update internals. * I took all static variables from goom_update and put them here.. for the moment. */ struct { diff --git a/gst/goom/gstgoom.c b/gst/goom/gstgoom.c index dbbe94184..494234550 100644 --- a/gst/goom/gstgoom.c +++ b/gst/goom/gstgoom.c @@ -22,17 +22,17 @@ /** * SECTION:element-goom + * @title: goom * @see_also: synaesthesia * * Goom is an audio visualisation element. It creates warping structures * based on the incoming audio signal. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v audiotestsrc ! goom ! videoconvert ! xvimagesink * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/goom/ifs.c b/gst/goom/ifs.c index 5438e31b4..5af721fb6 100644 --- a/gst/goom/ifs.c +++ b/gst/goom/ifs.c @@ -725,7 +725,7 @@ ifs_update (PluginInfo * goomInfo, Pixel * data, Pixel * back, int increment, | (col[ROUGE] << (ROUGE * 8)); } -/** VISUAL_FX WRAPPER FOR IFS */ +/* VISUAL_FX WRAPPER FOR IFS */ static void ifs_vfx_apply (VisualFX * _this, Pixel * src, Pixel * dest, diff --git a/gst/goom/sound_tester.h b/gst/goom/sound_tester.h index 2651d5f9d..26418c5e2 100644 --- a/gst/goom/sound_tester.h +++ b/gst/goom/sound_tester.h @@ -22,7 +22,7 @@ #include "goom_plugin_info.h" #include "goom_config.h" -/** change les donnees du SoundInfo */ +/* change les donnees du SoundInfo */ void evaluate_sound(gint16 data[2][512], SoundInfo *sndInfo); #endif diff --git a/gst/goom2k1/filters.h b/gst/goom2k1/filters.h index 102004b9a..0dff5642d 100644 --- a/gst/goom2k1/filters.h +++ b/gst/goom2k1/filters.h @@ -22,7 +22,7 @@ struct ZoomFilterData int middleY; char reverse; char mode; - /** @since June 2001 */ + /* @since June 2001 */ int hPlaneEffect; int vPlaneEffect; char noisify; diff --git a/gst/goom2k1/goom_core.h b/gst/goom2k1/goom_core.h index 1fbc0ee89..3e07af022 100644 --- a/gst/goom2k1/goom_core.h +++ b/gst/goom2k1/goom_core.h @@ -7,9 +7,9 @@ typedef struct ZoomFilterData ZoomFilterData; typedef struct { -/**-----------------------------------------------------** - ** SHARED DATA ** - **-----------------------------------------------------**/ +/*-----------------------------------------------------* + * SHARED DATA * + *-----------------------------------------------------*/ guint32 *pixel; guint32 *back; guint32 *p1, *p2; diff --git a/gst/goom2k1/gstgoom.c b/gst/goom2k1/gstgoom.c index 19eda10bc..c8baf0259 100644 --- a/gst/goom2k1/gstgoom.c +++ b/gst/goom2k1/gstgoom.c @@ -21,18 +21,18 @@ /** * SECTION:element-goom2k1 + * @title: goom2k1 * @see_also: goom, synaesthesia * * Goom2k1 is an audio visualisation element. It creates warping structures * based on the incoming audio signal. Goom2k1 is the older version of the * visualisation. Also available is goom2k4, with a different look. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v audiotestsrc ! goom2k1 ! videoconvert ! xvimagesink * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/icydemux/gsticydemux.c b/gst/icydemux/gsticydemux.c index 594b14712..035e7245f 100644 --- a/gst/icydemux/gsticydemux.c +++ b/gst/icydemux/gsticydemux.c @@ -21,22 +21,22 @@ /** * SECTION:element-icydemux + * @title: icydemux * * icydemux accepts data streams with ICY metadata at known intervals, as * transmitted from an upstream element (usually read as response headers from * an HTTP stream). The mime type of the data between the tag blocks is * detected using typefind functions, and the appropriate output mime type set - * on outgoing buffers. + * on outgoing buffers. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 souphttpsrc location=http://some.server/ iradio-mode=true ! icydemux ! fakesink -t * ]| This pipeline should read any available ICY tag information and output it. * The contents of the stream should be detected, and the appropriate mime * type set on buffers produced from icydemux. (Using gnomevfssrc, neonhttpsrc * or giosrc instead of souphttpsrc should also work.) - * </refsect2> + * */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/id3demux/gstid3demux.c b/gst/id3demux/gstid3demux.c index d491bb87b..b96eeb353 100644 --- a/gst/id3demux/gstid3demux.c +++ b/gst/id3demux/gstid3demux.c @@ -21,11 +21,12 @@ /** * SECTION:element-id3demux + * @title: id3demux * * id3demux accepts data streams with either (or both) ID3v2 regions at the * start, or ID3v1 at the end. The mime type of the data between the tag blocks * is detected using typefind functions, and the appropriate output mime type - * set on outgoing buffers. + * set on outgoing buffers. * * The element is only able to read ID3v1 tags from a seekable stream, because * they are at the end of the stream. That is, when get_range mode is supported @@ -36,14 +37,13 @@ * This id3demux element replaced an older element with the same name which * relied on libid3tag from the MAD project. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 filesrc location=file.mp3 ! id3demux ! fakesink -t * ]| This pipeline should read any available ID3 tag information and output it. * The contents of the file inside the ID3 tag regions should be detected, and * the appropriate mime type set on buffers produced from id3demux. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/imagefreeze/gstimagefreeze.c b/gst/imagefreeze/gstimagefreeze.c index 9ac5d7b67..f2933ceef 100644 --- a/gst/imagefreeze/gstimagefreeze.c +++ b/gst/imagefreeze/gstimagefreeze.c @@ -20,17 +20,17 @@ /** * SECTION:element-imagefreeze + * @title: imagefreeze * * The imagefreeze element generates a still frame video stream from * the input. It duplicates the first frame with the framerate requested * by downstream, allows seeking and answers queries. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v filesrc location=some.png ! decodebin ! imagefreeze ! autovideosink * ]| This pipeline shows a still frame stream of a PNG file. - * </refsect2> + * */ /* This is based on the imagefreeze element from PiTiVi: diff --git a/gst/interleave/deinterleave.c b/gst/interleave/deinterleave.c index 465e31db9..46245cf46 100644 --- a/gst/interleave/deinterleave.c +++ b/gst/interleave/deinterleave.c @@ -32,20 +32,20 @@ /** * SECTION:element-deinterleave + * @title: deinterleave * @see_also: interleave * * Splits one interleaved multichannel audio stream into many mono audio streams. - * + * * This element handles all raw audio formats and supports changing the input caps as long as * all downstream elements can handle the new caps and the number of channels and the channel * positions stay the same. This restriction will be removed in later versions by adding or * removing some source pads as required. - * + * * In most cases a queue and an audioconvert element should be added after each source pad * before further processing of the audio data. - * - * <refsect2> - * <title>Example launch line</title> + * + * ## Example launch line * |[ * gst-launch-1.0 filesrc location=/path/to/file.mp3 ! decodebin ! audioconvert ! "audio/x-raw,channels=2 ! deinterleave name=d d.src_0 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel1.ogg d.src_1 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel2.ogg * ]| Decodes an MP3 file and encodes the left and right channel into separate @@ -55,7 +55,7 @@ * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and * then interleaves the channels again to a WAV file with the channel with the * channels exchanged. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H @@ -186,7 +186,7 @@ gst_deinterleave_class_init (GstDeinterleaveClass * klass) /** * GstDeinterleave:keep-positions - * + * * Keep positions: When enable the caps on the output buffers will * contain the original channel positions. This can be used to correctly * interleave the output again later but can also lead to unwanted effects diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c index 808d0ff31..35cebf31f 100644 --- a/gst/interleave/interleave.c +++ b/gst/interleave/interleave.c @@ -30,22 +30,22 @@ /** * SECTION:element-interleave + * @title: interleave * @see_also: deinterleave * * Merges separate mono inputs into one interleaved stream. - * + * * This element handles all raw floating point sample formats and all signed integer sample formats. The first * caps on one of the sinkpads will set the caps of the output so usually an audioconvert element should be * placed before every sinkpad of interleave. - * + * * It's possible to change the number of channels while the pipeline is running by adding or removing * some of the request pads but this will change the caps of the output buffers. Changing the input * caps is _not_ supported yet. - * + * * The channel number of every sinkpad in the out can be retrieved from the "channel" property of the pad. - * - * <refsect2> - * <title>Example launch line</title> + * + * ## Example launch line * |[ * gst-launch-1.0 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src_0 ! queue ! audioconvert ! i.sink_1 d.src_1 ! queue ! audioconvert ! i.sink_0 * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and @@ -57,7 +57,7 @@ * channel-masks defined in the sink pads ensures a sane mapping of the mono * streams into the stereo stream. NOTE: the proper way to map channels in * code is by using the channel-positions property of the interleave element. - * </refsect2> + * */ /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray @@ -376,7 +376,7 @@ gst_interleave_class_init (GstInterleaveClass * klass) /** * GstInterleave:channel-positions - * + * * Channel positions: This property controls the channel positions * that are used on the src caps. The number of elements should be * the same as the number of sink pads and the array should contain @@ -400,7 +400,7 @@ gst_interleave_class_init (GstInterleaveClass * klass) /** * GstInterleave:channel-positions-from-input - * + * * Channel positions from input: If this property is set to %TRUE the channel * positions will be taken from the input caps if valid channel positions for * the output can be constructed from them. If this is set to %TRUE setting the diff --git a/gst/isomp4/gstqtmoovrecover.c b/gst/isomp4/gstqtmoovrecover.c index f5d2b91dc..185109fc9 100644 --- a/gst/isomp4/gstqtmoovrecover.c +++ b/gst/isomp4/gstqtmoovrecover.c @@ -43,20 +43,18 @@ /** * SECTION:element-qtmoovrecover + * @title: qtmoovrecover * @short_description: Utility element for recovering unfinished quicktime files * - * <refsect2> - * <para> * This element recovers quicktime files created with qtmux using the moov * recovery feature. - * </para> - * <title>Example pipelines</title> - * <para> - * <programlisting> + * + * ## Example pipelines + * + * |[ * TODO - * </programlisting> - * </para> - * </refsect2> + * ]| + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/isomp4/gstqtmux-doc.c b/gst/isomp4/gstqtmux-doc.c index 3b857d84a..375eae5f2 100644 --- a/gst/isomp4/gstqtmux-doc.c +++ b/gst/isomp4/gstqtmux-doc.c @@ -47,6 +47,7 @@ /** * SECTION:element-mp4mux + * @title: mp4mux * @short_description: Muxer for ISO MPEG-4 (.mp4) files * * This element merges streams (audio and video) into ISO MPEG-4 (.mp4) files. @@ -81,20 +82,20 @@ * #GstMp4Mux:streamable allows foregoing to add index metadata (at the end of * file). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 gst-launch-1.0 v4l2src num-buffers=50 ! queue ! x264enc ! mp4mux ! filesink location=video.mp4 * ]| * Records a video stream captured from a v4l2 device, encodes it into H.264 * and muxes it into an mp4 file. - * </refsect2> + * */ /* ============================= 3gppmux ==================================== */ /** * SECTION:element-3gppmux + * @title: 3gppmux * @short_description: Muxer for 3GPP (.3gp) files * * This element merges streams (audio and video) into 3GPP (.3gp) files. @@ -129,14 +130,12 @@ * #Gst3GPPMux:streamable allows foregoing to add index metadata (at the end of * file). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 v4l2src num-buffers=50 ! queue ! ffenc_h263 ! 3gppmux ! filesink location=video.3gp * ]| * Records a video stream captured from a v4l2 device, encodes it into H.263 * and muxes it into an 3gp file. - * </refsect2> * * Documentation last reviewed on 2011-04-21 */ @@ -145,6 +144,7 @@ /** * SECTION:element-mj2mux + * @title: mj2mux * @short_description: Muxer for Motion JPEG-2000 (.mj2) files * * This element merges streams (audio and video) into MJ2 (.mj2) files. @@ -179,14 +179,12 @@ * #GstMJ2Mux:streamable allows foregoing to add index metadata (at the end of * file). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 v4l2src num-buffers=50 ! queue ! jp2kenc ! mj2mux ! filesink location=video.mj2 * ]| * Records a video stream captured from a v4l2 device, encodes it into JPEG-2000 * and muxes it into an mj2 file. - * </refsect2> * * Documentation last reviewed on 2011-04-21 */ @@ -195,6 +193,7 @@ /** * SECTION:element-ismlmux + * @title: ismlmux * @short_description: Muxer for ISML smooth streaming (.isml) files * * This element merges streams (audio and video) into MJ2 (.mj2) files. @@ -229,14 +228,12 @@ * #GstISMLMux:streamable allows foregoing to add index metadata (at the end of * file). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 v4l2src num-buffers=50 ! queue ! jp2kenc ! mj2mux ! filesink location=video.mj2 * ]| * Records a video stream captured from a v4l2 device, encodes it into JPEG-2000 * and muxes it into an mj2 file. - * </refsect2> * * Documentation last reviewed on 2011-04-21 */ diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index dcc28662c..14e451c12 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -47,6 +47,7 @@ /** * SECTION:element-qtmux + * @title: qtmux * @short_description: Muxer for quicktime(.mov) files * * This element merges streams (audio and video) into QuickTime(.mov) files. @@ -108,13 +109,12 @@ * a fixed sample size (such as raw audio and Prores Video) and that don't * have reordered samples. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 v4l2src num-buffers=500 ! video/x-raw,width=320,height=240 ! videoconvert ! qtmux ! filesink location=video.mov * ]| * Records a video stream captured from a v4l2 device and muxes it into a qt file. - * </refsect2> + * */ /* diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c index cf15fad46..c12074ba0 100644 --- a/gst/isomp4/qtdemux.c +++ b/gst/isomp4/qtdemux.c @@ -29,21 +29,21 @@ /** * SECTION:element-qtdemux + * @title: qtdemux * * Demuxes a .mov 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.mov ! qtdemux name=demux demux.audio_0 ! queue ! decodebin ! audioconvert ! audioresample ! autoaudiosink demux.video_0 ! queue ! decodebin ! videoconvert ! videoscale ! autovideosink * ]| Play (parse and decode) a .mov file and try to output it to * an automatically detected soundcard and videosink. If the MOV file contains * compressed audio or video data, this will only work if you have the * right decoder elements/plugins installed. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/law/alaw-decode.c b/gst/law/alaw-decode.c index eea23e5a9..7587fbde7 100644 --- a/gst/law/alaw-decode.c +++ b/gst/law/alaw-decode.c @@ -18,6 +18,7 @@ */ /** * SECTION:element-alawdec + * @title: alawdec * * This element decodes alaw audio. Alaw coding is also known as G.711. */ diff --git a/gst/law/alaw-encode.c b/gst/law/alaw-encode.c index ff17000a7..dae82ac73 100644 --- a/gst/law/alaw-encode.c +++ b/gst/law/alaw-encode.c @@ -18,6 +18,7 @@ */ /** * SECTION:element-alawenc + * @title: alawenc * * This element encode alaw audio. Alaw coding is also known as G.711. */ diff --git a/gst/law/mulaw-conversion.c b/gst/law/mulaw-conversion.c index 15cde174a..b0f39e573 100644 --- a/gst/law/mulaw-conversion.c +++ b/gst/law/mulaw-conversion.c @@ -59,7 +59,7 @@ mulaw_encode (gint16 * in, guint8 * out, gint numsamples) for (i = 0; i < numsamples; i++) { sample = in[i]; - /** get the sample into sign-magnitude **/ + /* get the sample into sign-magnitude */ sign = (sample >> 8) & 0x80; /* set aside the sign */ if (sign != 0) { sample = -sample; /* get magnitude */ @@ -69,7 +69,7 @@ mulaw_encode (gint16 * in, guint8 * out, gint numsamples) if (((guint16) sample) > CLIP) sample = CLIP; /* clip the magnitude */ - /** convert from 16 bit linear to ulaw **/ + /* convert from 16 bit linear to ulaw */ sample = sample + BIAS; exponent = exp_lut[(sample >> 7) & 0xFF]; mantissa = (sample >> (exponent + 3)) & 0x0F; diff --git a/gst/law/mulaw-decode.c b/gst/law/mulaw-decode.c index 1a70d0b7c..044e8fee1 100644 --- a/gst/law/mulaw-decode.c +++ b/gst/law/mulaw-decode.c @@ -18,6 +18,7 @@ */ /** * SECTION:element-mulawdec + * @title: mulawdec * * This element decodes mulaw audio. Mulaw coding is also known as G.711. */ diff --git a/gst/law/mulaw-encode.c b/gst/law/mulaw-encode.c index b22ce0dba..ccf0611ca 100644 --- a/gst/law/mulaw-encode.c +++ b/gst/law/mulaw-encode.c @@ -18,6 +18,7 @@ */ /** * SECTION:element-mulawenc + * @title: mulawenc * * This element encode mulaw audio. Mulaw coding is also known as G.711. */ diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c index bcf290ece..c386f50bd 100644 --- a/gst/level/gstlevel.c +++ b/gst/level/gstlevel.c @@ -21,82 +21,34 @@ /** * SECTION:element-level + * @title: level * * Level analyses incoming audio buffers and, if the #GstLevel:message property * is %TRUE, generates an element message named - * <classname>"level"</classname>: - * after each interval of time given by the #GstLevel:interval property. + * `level`: after each interval of time given by the #GstLevel:interval property. * The message's structure contains these fields: - * <itemizedlist> - * <listitem> - * <para> - * #GstClockTime - * <classname>"timestamp"</classname>: - * the timestamp of the buffer that triggered the message. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"stream-time"</classname>: - * the stream time of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"running-time"</classname>: - * the running_time of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"duration"</classname>: - * the duration of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"endtime"</classname>: - * the end time of the buffer that triggered the message as stream time (this - * is deprecated, as it can be calculated from stream-time + duration) - * </para> - * </listitem> - * <listitem> - * <para> - * #GValueArray of #gdouble - * <classname>"peak"</classname>: - * the peak power level in dB for each channel - * </para> - * </listitem> - * <listitem> - * <para> - * #GValueArray of #gdouble - * <classname>"decay"</classname>: - * the decaying peak power level in dB for each channel + * + * * #GstClockTime `timestamp`: the timestamp of the buffer that triggered the message. + * * #GstClockTime `stream-time`: the stream time of the buffer. + * * #GstClockTime `running-time`: the running_time of the buffer. + * * #GstClockTime `duration`: the duration of the buffer. + * * #GstClockTime `endtime`: the end time of the buffer that triggered the message as + * stream time (this is deprecated, as it can be calculated from stream-time + duration) + * * #GValueArray of #gdouble `peak`: the peak power level in dB for each channel + * * #GValueArray of #gdouble `decay`: the decaying peak power level in dB for each channel * The decaying peak level follows the peak level, but starts dropping if no * new peak is reached after the time given by the #GstLevel:peak-ttl. * When the decaying peak level drops, it does so at the decay rate as * specified by the #GstLevel:peak-falloff. - * </para> - * </listitem> - * <listitem> - * <para> - * #GValueArray of #gdouble - * <classname>"rms"</classname>: - * the Root Mean Square (or average power) level in dB for each channel - * </para> - * </listitem> - * </itemizedlist> + * * #GValueArray of #gdouble `rms`: the Root Mean Square (or average power) level in dB + * for each channel + * + * ## Example application * - * <refsect2> - * <title>Example application</title> * <informalexample><programlisting language="C"> * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" /> * </programlisting></informalexample> - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index acafe1231..588bcb5f6 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -33,15 +33,15 @@ /** * SECTION:element-matroskademux + * @title: matroskademux * * matroskademux demuxes a Matroska file into the different contained streams. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v filesrc location=/path/to/mkv ! matroskademux ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink * ]| This pipeline demuxes a Matroska file and outputs the contained Vorbis audio. - * </refsect2> + * */ diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index a5168a6cf..4665157f4 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -28,18 +28,18 @@ /** * SECTION:element-matroskamux + * @title: matroskamux * * matroskamux muxes different input streams into a Matroska file. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v filesrc location=/path/to/mp3 ! mpegaudioparse ! matroskamux name=mux ! filesink location=test.mkv filesrc location=/path/to/theora.ogg ! oggdemux ! theoraparse ! mux. * ]| This pipeline muxes an MP3 file and a Ogg Theora video into a Matroska file. * |[ * gst-launch-1.0 -v audiotestsrc num-buffers=100 ! audioconvert ! vorbisenc ! matroskamux ! filesink location=test.mka * ]| This pipeline muxes a 440Hz sine wave encoded with the Vorbis codec into a Matroska file. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/matroska/matroska-parse.c b/gst/matroska/matroska-parse.c index 069f581dd..b34817ffb 100644 --- a/gst/matroska/matroska-parse.c +++ b/gst/matroska/matroska-parse.c @@ -33,15 +33,15 @@ /** * SECTION:element-matroskaparse + * @title: matroskaparse * * matroskaparse parsees a Matroska file into the different contained streams. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v filesrc location=/path/to/mkv ! matroskaparse ! vorbisdec ! audioconvert ! audioresample ! autoaudiosink * ]| This pipeline parsees a Matroska file and outputs the contained Vorbis audio. - * </refsect2> + * */ diff --git a/gst/matroska/webm-mux.c b/gst/matroska/webm-mux.c index 188f27304..6d2993135 100644 --- a/gst/matroska/webm-mux.c +++ b/gst/matroska/webm-mux.c @@ -19,11 +19,11 @@ /** * SECTION:element-webmmux + * @title: webmmux * * webmmux muxes VP8 video and Vorbis audio streams into a WebM file. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 webmmux name=mux ! filesink location=newfile.webm \ * uridecodebin uri=file:///path/to/somefile.ogv name=demux \ @@ -35,7 +35,7 @@ * videotestsrc num-buffers=250 ! video/x-raw,framerate=25/1 ! videoconvert ! vp8enc ! queue ! mux.video_0 \ * audiotestsrc samplesperbuffer=44100 num-buffers=10 ! audio/x-raw,rate=44100 ! vorbisenc ! queue ! mux.audio_0 * ]| This pipeline muxes a test video and a sine wave into a WebM file. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/monoscope/gstmonoscope.c b/gst/monoscope/gstmonoscope.c index 1c8f9a8e4..6ecf6bba4 100644 --- a/gst/monoscope/gstmonoscope.c +++ b/gst/monoscope/gstmonoscope.c @@ -21,17 +21,17 @@ /** * SECTION:element-monoscope + * @title: monoscope * @see_also: goom * * Monoscope is an audio visualisation element. It creates a coloured * curve of the audio signal like on an oscilloscope. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! monoscope ! videoconvert ! ximagesink * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/multifile/gstmultifilesink.c b/gst/multifile/gstmultifilesink.c index d2a55ef96..e5ab912ac 100644 --- a/gst/multifile/gstmultifilesink.c +++ b/gst/multifile/gstmultifilesink.c @@ -25,6 +25,7 @@ */ /** * SECTION:element-multifilesink + * @title: multifilesink * @see_also: #GstFileSrc * * Write incoming data to a series of sequentially-named files. @@ -41,77 +42,25 @@ * be substituted with the index for each filename. * * If the #GstMultiFileSink:post-messages property is %TRUE, it sends an application - * message named - * <classname>"GstMultiFileSink"</classname> after writing each - * buffer. + * message named `GstMultiFileSink` after writing each buffer. * * The message's structure contains these fields: - * <itemizedlist> - * <listitem> - * <para> - * #gchar * - * <classname>"filename"</classname>: - * the filename where the buffer was written. - * </para> - * </listitem> - * <listitem> - * <para> - * #gint - * <classname>"index"</classname>: - * the index of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"timestamp"</classname>: - * the timestamp of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"stream-time"</classname>: - * the stream time of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"running-time"</classname>: - * the running_time of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"duration"</classname>: - * the duration of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"offset"</classname>: - * the offset of the buffer that triggered the message. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"offset-end"</classname>: - * the offset-end of the buffer that triggered the message. - * </para> - * </listitem> - * </itemizedlist> * - * <refsect2> - * <title>Example launch line</title> + * * #gchar *`filename`: the filename where the buffer was written. + * * #gint `index`: index of the buffer. + * * #GstClockTime `timestamp`: the timestamp of the buffer. + * * #GstClockTime `stream-time`: the stream time of the buffer. + * * #GstClockTime running-time`: the running_time of the buffer. + * * #GstClockTime `duration`: the duration of the buffer. + * * #guint64 `offset`: the offset of the buffer that triggered the message. + * * #guint64 `offset-end`: the offset-end of the buffer that triggered the message. + * + * ## Example launch line * |[ * gst-launch-1.0 audiotestsrc ! multifilesink * gst-launch-1.0 videotestsrc ! multifilesink post-messages=true location="frame%d" * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/multifile/gstmultifilesrc.c b/gst/multifile/gstmultifilesrc.c index 3ad1a1b88..0f8dd6e78 100644 --- a/gst/multifile/gstmultifilesrc.c +++ b/gst/multifile/gstmultifilesrc.c @@ -20,6 +20,7 @@ */ /** * SECTION:element-multifilesrc + * @title: multifilesrc * @see_also: #GstFileSrc * * Reads buffers from sequentially named files. If used together with an image @@ -30,15 +31,14 @@ * * File names are created by replacing "\%d" with the index using printf(). * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 multifilesrc location="img.%04d.png" index=0 caps="image/png,framerate=\(fraction\)12/1" ! \ * pngdec ! videoconvert ! videorate ! theoraenc ! oggmux ! \ * filesink location="images.ogg" * ]| This pipeline creates a video file "images.ogg" by joining multiple PNG * files named img.0000.png, img.0001.png, etc. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/multifile/gstsplitfilesrc.c b/gst/multifile/gstsplitfilesrc.c index 2b2d0f80f..a30dfefc5 100644 --- a/gst/multifile/gstsplitfilesrc.c +++ b/gst/multifile/gstsplitfilesrc.c @@ -18,6 +18,7 @@ */ /** * SECTION:element-splitfilesrc + * @title: splitfilesrc * @see_also: #GstFileSrc, #GstMultiFileSrc * * Reads data from multiple files, presenting those files as one continuous @@ -29,15 +30,14 @@ * (and expects) shell-style wildcards (but only for the filename, not for * directories). The results will be sorted. * - * <refsect2> - * <title>Example launch lines</title> + * ## Example launch lines * |[ * gst-launch-1.0 splitfilesrc location="/path/to/part-*.mpg" ! decodebin ! ... * ]| Plays the different parts as if they were one single MPEG file. * |[ * gst-launch-1.0 playbin uri="splitfile://path/to/foo.avi.*" * ]| Plays the different parts as if they were one single AVI file. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/multifile/gstsplitmuxsink.c b/gst/multifile/gstsplitmuxsink.c index 0cbe54188..a9cd723c5 100644 --- a/gst/multifile/gstsplitmuxsink.c +++ b/gst/multifile/gstsplitmuxsink.c @@ -19,6 +19,7 @@ /** * SECTION:element-splitmuxsink + * @title: splitmuxsink * @short_description: Muxer wrapper for splitting output stream by size or time * * This element wraps a muxer and a sink, and starts a new file when the mux @@ -45,8 +46,7 @@ * muxer-factory and sink-factory properties are used to construct the new * objects, together with muxer-properties and sink-properties. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 -e v4l2src num-buffers=500 ! video/x-raw,width=320,height=240 ! videoconvert ! queue ! timeoverlay ! x264enc key-int-max=10 ! h264parse ! splitmuxsink location=video%02d.mov max-size-time=10000000000 max-size-bytes=1000000 * ]| @@ -60,7 +60,6 @@ * Records a video stream captured from a v4l2 device and muxer it into * streamable Matroska files, splitting as needed to limit size/duration to 10 * seconds. Each file will finalize asynchronously. - * </refsect2> */ #ifdef HAVE_CONFIG_H diff --git a/gst/multifile/gstsplitmuxsrc.c b/gst/multifile/gstsplitmuxsrc.c index 9a80dde25..7bef5f2a9 100644 --- a/gst/multifile/gstsplitmuxsrc.c +++ b/gst/multifile/gstsplitmuxsrc.c @@ -21,6 +21,7 @@ /** * SECTION:element-splitmuxsrc + * @title: splitmuxsrc * @short_description: Split Demuxer bin that recombines files created by * the splitmuxsink element. * @@ -31,15 +32,14 @@ * streams in each file part at the demuxed elementary level, rather than * as a single larger bytestream. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 splitmuxsrc location=video*.mov ! decodebin ! xvimagesink * ]| Demux each file part and output the video stream as one continuous stream * |[ * gst-launch-1.0 playbin uri="splitmux://path/to/foo.mp4.*" * ]| Play back a set of files created by splitmuxsink - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/multipart/multipartdemux.c b/gst/multipart/multipartdemux.c index 23e67c2e0..8efe2ea77 100644 --- a/gst/multipart/multipartdemux.c +++ b/gst/multipart/multipartdemux.c @@ -22,10 +22,11 @@ /** * SECTION:element-multipartdemux + * @title: multipartdemux * @see_also: #GstMultipartMux * - * MultipartDemux uses the Content-type field of incoming buffers to demux and - * push data to dynamic source pads. Most of the time multipart streams are + * MultipartDemux uses the Content-type field of incoming buffers to demux and + * push data to dynamic source pads. Most of the time multipart streams are * sequential JPEG frames generated from a live source such as a network source * or a camera. * @@ -37,13 +38,12 @@ * be configured specifically with the #GstMultipartDemux:boundary property * otherwise it will be autodetected. * - * <refsect2> - * <title>Sample pipelines</title> + * ## Sample pipelines * |[ * gst-launch-1.0 filesrc location=/tmp/test.multipart ! multipartdemux ! image/jpeg,framerate=\(fraction\)5/1 ! jpegparse ! jpegdec ! videoconvert ! autovideosink * ]| a simple pipeline to demux a multipart file muxed with #GstMultipartMux * containing JPEG frames. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/multipart/multipartmux.c b/gst/multipart/multipartmux.c index e6271ff08..f213cb587 100644 --- a/gst/multipart/multipartmux.c +++ b/gst/multipart/multipartmux.c @@ -19,18 +19,18 @@ /** * SECTION:element-multipartmux + * @title: multipartmux * * MultipartMux uses the #GstCaps of the sink pad as the Content-type field for - * incoming buffers when muxing them to a multipart stream. Most of the time + * incoming buffers when muxing them to a multipart stream. Most of the time * multipart streams are sequential JPEG frames. * - * <refsect2> - * <title>Sample pipelines</title> + * ## Sample pipelines * |[ * gst-launch-1.0 videotestsrc ! video/x-raw, framerate='(fraction)'5/1 ! jpegenc ! multipartmux ! filesink location=/tmp/test.multipart * ]| a pipeline to mux 5 JPEG frames per second into a multipart stream * stored to a file. - * </refsect2> + * */ /* FIXME: drop/merge tag events, or at least send them delayed after stream-start */ diff --git a/gst/replaygain/gstrganalysis.c b/gst/replaygain/gstrganalysis.c index 214901d03..f8c350091 100644 --- a/gst/replaygain/gstrganalysis.c +++ b/gst/replaygain/gstrganalysis.c @@ -22,6 +22,7 @@ /** * SECTION:element-rganalysis + * @title: rganalysis * @see_also: #GstRgVolume * * This element analyzes raw audio sample data in accordance with the proposed @@ -33,7 +34,7 @@ * posted on the message bus with a tag message. The EOS event is forwarded as * normal afterwards. Result tag lists at least contain the tags * #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL. - * + * * Because the generated metadata tags become available at the end of streams, * downstream muxer and encoder elements are normally unable to save them in * their output since they generally save metadata in the file header. @@ -42,9 +43,8 @@ * needed for album processing (see #GstRgAnalysis:num-tracks property) since * the album gain and peak values need to be associated with all tracks of an * album, not just the last one. - * - * <refsect2> - * <title>Example launch lines</title> + * + * ## Example launch lines * |[ * gst-launch-1.0 -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink * ]| Analyze a simple test waveform @@ -56,21 +56,18 @@ * gst-launch-1.0 -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \ * ! wavparse ! rganalysis ! fakesink * ]| Analyze the pink noise reference file - * <para> + * * The above launch line yields a result gain of +6 dB (instead of the expected * +0 dB). This is not in error, refer to the #GstRgAnalysis:reference-level * property documentation for more information. - * </para> - * </refsect2> - * <refsect2> - * <title>Acknowledgements</title> - * <para> + * + * ## Acknowledgements + * * This element is based on code used in the <ulink * url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program and many * others. The relevant parts are copyrighted by David Robinson, Glen Sawyer * and Frank Klemm. - * </para> - * </refsect2> + * */ #ifdef HAVE_CONFIG_H @@ -169,7 +166,7 @@ gst_rg_analysis_class_init (GstRgAnalysisClass * klass) * GstRgAnalysis:num-tracks: * * Number of remaining album tracks. - * + * * Analyzing several streams sequentially and assigning them a common result * gain is known as "album processing". If this gain is used during playback * (by switching to "album mode"), all tracks of an album receive the same diff --git a/gst/replaygain/gstrglimiter.c b/gst/replaygain/gstrglimiter.c index 5e04e7d4f..95c83a120 100644 --- a/gst/replaygain/gstrglimiter.c +++ b/gst/replaygain/gstrglimiter.c @@ -22,21 +22,21 @@ /** * SECTION:element-rglimiter + * @title: rglimiter * @see_also: #GstRgVolume * * This element applies signal compression/limiting to raw audio data. It * performs strict hard limiting with soft-knee characteristics, using a * threshold of -6 dB. This type of filter is mentioned in the proposed <ulink * url="http://replaygain.org">ReplayGain standard</ulink>. - * - * <refsect2> - * <title>Example launch line</title> + * + * ## Example launch line * |[ * gst-launch-1.0 filesrc location=filename.ext ! decodebin ! audioconvert \ * ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \ * ! audioconvert ! audioresample ! alsasink * ]|Playback of a file - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/replaygain/gstrgvolume.c b/gst/replaygain/gstrgvolume.c index 7c4f2811e..fe75b60dd 100644 --- a/gst/replaygain/gstrgvolume.c +++ b/gst/replaygain/gstrgvolume.c @@ -22,37 +22,37 @@ /** * SECTION:element-rgvolume + * @title: rgvolume * @see_also: #GstRgLimiter, #GstRgAnalysis * * This element applies volume changes to streams as lined out in the proposed * <ulink url="http://replaygain.org">ReplayGain standard</ulink>. It * interprets the ReplayGain meta data tags and carries out the adjustment (by * using a volume element internally). The relevant tags are: - * <itemizedlist> - * <listitem>#GST_TAG_TRACK_GAIN</listitem> - * <listitem>#GST_TAG_TRACK_PEAK</listitem> - * <listitem>#GST_TAG_ALBUM_GAIN</listitem> - * <listitem>#GST_TAG_ALBUM_PEAK</listitem> - * <listitem>#GST_TAG_REFERENCE_LEVEL</listitem> - * </itemizedlist> + * + * * #GST_TAG_TRACK_GAIN + * * #GST_TAG_TRACK_PEAK + * * #GST_TAG_ALBUM_GAIN + * * #GST_TAG_ALBUM_PEAK + * * #GST_TAG_REFERENCE_LEVEL + * * The information carried by these tags must have been calculated beforehand by * performing the ReplayGain analysis. This is implemented by the <link * linkend="GstRgAnalysis">rganalysis</link> element. - * + * * The signal compression/limiting recommendations outlined in the proposed * standard are not implemented by this element. This has to be handled by * separate elements because applications might want to have additional filters * between the volume adjustment and the limiting stage. A basic limiter is * included with this plugin: The <link linkend="GstRgLimiter">rglimiter</link> * element applies -6 dB hard limiting as mentioned in the ReplayGain standard. - * - * <refsect2> - * <title>Example launch line</title> + * + * ## Example launch line * |[ * gst-launch-1.0 filesrc location=filename.ext ! decodebin ! audioconvert \ * ! rgvolume ! audioconvert ! audioresample ! alsasink * ]| Playback of a file - * </refsect2> + * */ #ifdef HAVE_CONFIG_H @@ -251,11 +251,11 @@ gst_rg_volume_class_init (GstRgVolumeClass * klass) * presence of ReplayGain tags in the stream, this is set according to one of * these simple formulas: * - * <itemizedlist> - * <listitem>#GstRgVolume:pre-amp + album gain of the stream</listitem> - * <listitem>#GstRgVolume:pre-amp + track gain of the stream</listitem> - * <listitem>#GstRgVolume:pre-amp + #GstRgVolume:fallback-gain</listitem> - * </itemizedlist> + * + * * #GstRgVolume:pre-amp + album gain of the stream + * * #GstRgVolume:pre-amp + track gain of the stream + * * #GstRgVolume:pre-amp + #GstRgVolume:fallback-gain + * */ g_object_class_install_property (gobject_class, PROP_TARGET_GAIN, g_param_spec_double ("target-gain", "Target-gain", diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c index 601f16edc..e4aa71a00 100644 --- a/gst/rtp/gstrtpL16depay.c +++ b/gst/rtp/gstrtpL16depay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpL16depay + * @title: rtpL16depay * @see_also: rtpL16pay * * Extract raw audio from RTP packets according to RFC 3551. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L16, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL16depay ! pulsesink * ]| This example pipeline will depayload an RTP raw audio stream. Refer to * the rtpL16pay example to create the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c index 4783a650f..7e358d3d0 100644 --- a/gst/rtp/gstrtpL16pay.c +++ b/gst/rtp/gstrtpL16pay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpL16pay + * @title: rtpL16pay * @see_also: rtpL16depay * * Payload raw audio into RTP packets according to RFC 3551. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL16pay ! udpsink * ]| This example pipeline will payload raw audio. Refer to * the rtpL16depay example to depayload and play the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpL24depay.c b/gst/rtp/gstrtpL24depay.c index 8b28ee831..b37b06312 100644 --- a/gst/rtp/gstrtpL24depay.c +++ b/gst/rtp/gstrtpL24depay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpL24depay + * @title: rtpL24depay * @see_also: rtpL24pay * * Extract raw audio from RTP packets according to RFC 3190, section 4. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L24, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL24depay ! pulsesink * ]| This example pipeline will depayload an RTP raw audio stream. Refer to * the rtpL24pay example to create the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpL24pay.c b/gst/rtp/gstrtpL24pay.c index 936bd4431..d2ad72535 100644 --- a/gst/rtp/gstrtpL24pay.c +++ b/gst/rtp/gstrtpL24pay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpL24pay + * @title: rtpL24pay * @see_also: rtpL24depay * * Payload raw 24-bit audio into RTP packets according to RFC 3190, section 4. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3190.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! rtpL24pay ! udpsink * ]| This example pipeline will payload raw audio. Refer to * the rtpL24depay example to depayload and play the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpac3depay.c b/gst/rtp/gstrtpac3depay.c index ec2b3baa1..54339f343 100644 --- a/gst/rtp/gstrtpac3depay.c +++ b/gst/rtp/gstrtpac3depay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpac3depay + * @title: rtpac3depay * @see_also: rtpac3pay * * Extract AC3 audio from RTP packets according to RFC 4184. * For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)AC3, payload=(int)96' ! rtpac3depay ! a52dec ! pulsesink * ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to * the rtpac3pay example to create the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c index 7c797b767..8c5c2350d 100644 --- a/gst/rtp/gstrtpac3pay.c +++ b/gst/rtp/gstrtpac3pay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpac3pay + * @title: rtpac3pay * @see_also: rtpac3depay * * Payload AC3 audio into RTP packets according to RFC 4184. * For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink * ]| This example pipeline will encode and payload AC3 stream. Refer to * the rtpac3depay example to depayload and decode the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpamrdepay.c b/gst/rtp/gstrtpamrdepay.c index 7a7c79757..de0275801 100644 --- a/gst/rtp/gstrtpamrdepay.c +++ b/gst/rtp/gstrtpamrdepay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpamrdepay + * @title: rtpamrdepay * @see_also: rtpamrpay * * Extract AMR audio from RTP packets according to RFC 3267. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)AMR, encoding-params=(string)1, octet-align=(string)1, payload=(int)96' ! rtpamrdepay ! amrnbdec ! pulsesink * ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to * the rtpamrpay example to create the RTP stream. - * </refsect2> + * */ /* diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c index 5e70f0a44..7921d9713 100644 --- a/gst/rtp/gstrtpamrpay.c +++ b/gst/rtp/gstrtpamrpay.c @@ -19,18 +19,18 @@ /** * SECTION:element-rtpamrpay + * @title: rtpamrpay * @see_also: rtpamrdepay * * Payload AMR audio into RTP packets according to RFC 3267. * For detailed information see: http://www.rfc-editor.org/rfc/rfc3267.txt * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 -v audiotestsrc ! amrnbenc ! rtpamrpay ! udpsink * ]| This example pipeline will encode and payload an AMR stream. Refer to * the rtpamrdepay example to depayload and decode the RTP stream. - * </refsect2> + * */ /* references: diff --git a/gst/rtp/gstrtpbvdepay.c b/gst/rtp/gstrtpbvdepay.c index a0faf8320..625bb3731 100644 --- a/gst/rtp/gstrtpbvdepay.c +++ b/gst/rtp/gstrtpbvdepay.c @@ -19,6 +19,7 @@ /** * SECTION:element-rtpbvdepay + * @title: rtpbvdepay * @see_also: rtpbvpay * * Extract BroadcomVoice audio from RTP packets according to RFC 4298. diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c index e202015e6..a396d2642 100644 --- a/gst/rtp/gstrtpbvpay.c +++ b/gst/rtp/gstrtpbvpay.c @@ -19,6 +19,7 @@ /** * SECTION:element-rtpbvpay + * @title: rtpbvpay * @see_also: rtpbvdepay * * Payload BroadcomVoice audio into RTP packets according to RFC 4298. diff --git a/gst/rtp/gstrtph261depay.c b/gst/rtp/gstrtph261depay.c index 5d372293c..164d2f04d 100644 --- a/gst/rtp/gstrtph261depay.c +++ b/gst/rtp/gstrtph261depay.c @@ -20,6 +20,7 @@ /** * SECTION:element-rtph261depay + * @title: rtph261depay * @see_also: rtph261pay * * Extract encoded H.261 video frames from RTP packets according to RFC 4587. @@ -29,13 +30,12 @@ * aggregates the extracted stream until a complete frame is received before * it pushes it downstream. * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 udpsrc caps='application/x-rtp, payload=31' ! rtph261depay ! avdec_h261 ! autovideosink * ]| This example pipeline will depayload and decode an RTP H.261 video stream. * Refer to the rtph261pay example to create the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtph261pay.c b/gst/rtp/gstrtph261pay.c index b592d1158..94f75f35a 100644 --- a/gst/rtp/gstrtph261pay.c +++ b/gst/rtp/gstrtph261pay.c @@ -20,6 +20,7 @@ /** * SECTION:element-rtph261pay + * @title: rtph261pay * @see_also: rtph261depay * * Payload encoded H.261 video frames into RTP packets according to RFC 4587. @@ -35,13 +36,12 @@ * encoder does not produce a continuous bit-stream but the decoder requires * it. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 videotestsrc ! avenc_h261 ! rtph261pay ! udpsink * ]| This will encode a test video and payload it. Refer to the rtph261depay * example to depayload and play the RTP stream. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtph264depay.c b/gst/rtp/gstrtph264depay.c index 275a9a099..254d12215 100644 --- a/gst/rtp/gstrtph264depay.c +++ b/gst/rtp/gstrtph264depay.c @@ -59,7 +59,7 @@ GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"video\", " "clock-rate = (int) 90000, " "encoding-name = (string) \"H264\"") - /** optional parameters **/ + /* optional parameters */ /* "profile-level-id = (string) ANY, " */ /* "max-mbps = (string) ANY, " */ /* "max-fs = (string) ANY, " */ diff --git a/gst/rtp/gstrtph265depay.c b/gst/rtp/gstrtph265depay.c index 551e08a5e..f39ff9451 100644 --- a/gst/rtp/gstrtph265depay.c +++ b/gst/rtp/gstrtph265depay.c @@ -62,7 +62,7 @@ GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_CAPS ("application/x-rtp, " "media = (string) \"video\", " "clock-rate = (int) 90000, " "encoding-name = (string) \"H265\"") - /** optional parameters **/ + /* optional parameters */ /* "profile-space = (int) [ 0, 3 ], " */ /* "profile-id = (int) [ 0, 31 ], " */ /* "tier-flag = (int) [ 0, 1 ], " */ diff --git a/gst/rtp/gstrtph265pay.c b/gst/rtp/gstrtph265pay.c index d1a3a4229..55d794dbb 100644 --- a/gst/rtp/gstrtph265pay.c +++ b/gst/rtp/gstrtph265pay.c @@ -69,7 +69,7 @@ GST_STATIC_PAD_TEMPLATE ("src", "media = (string) \"video\", " "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", " "clock-rate = (int) 90000, " "encoding-name = (string) \"H265\"") - /** optional parameters **/ + /* optional parameters */ /* "profile-space = (int) [ 0, 3 ], " */ /* "profile-id = (int) [ 0, 31 ], " */ /* "tier-flag = (int) [ 0, 1 ], " */ diff --git a/gst/rtp/gstrtpj2kdepay.c b/gst/rtp/gstrtpj2kdepay.c index c30dd34a6..132bcf5df 100644 --- a/gst/rtp/gstrtpj2kdepay.c +++ b/gst/rtp/gstrtpj2kdepay.c @@ -20,6 +20,7 @@ /** * SECTION:element-rtpj2kdepay + * @title: rtpj2kdepay * * Depayload an RTP-payloaded JPEG 2000 image into RTP packets according to RFC 5371 * and RFC 5372. diff --git a/gst/rtp/gstrtpj2kpay.c b/gst/rtp/gstrtpj2kpay.c index f1c6c03c2..9ead8a21a 100644 --- a/gst/rtp/gstrtpj2kpay.c +++ b/gst/rtp/gstrtpj2kpay.c @@ -19,6 +19,7 @@ /** * SECTION:element-rtpj2kpay + * @title: rtpj2kpay * * Payload encode JPEG 2000 images into RTP packets according to RFC 5371 * and RFC 5372. @@ -30,7 +31,6 @@ * codestream. A "packetization unit" is defined as either a JPEG 2000 main header, * a JPEG 2000 tile-part header, or a JPEG 2000 packet. * - * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpjpegpay.c b/gst/rtp/gstrtpjpegpay.c index fe9301684..7af04ecb6 100644 --- a/gst/rtp/gstrtpjpegpay.c +++ b/gst/rtp/gstrtpjpegpay.c @@ -21,6 +21,7 @@ /** * SECTION:element-rtpjpegpay + * @title: rtpjpegpay * * Payload encode JPEG pictures into RTP packets according to RFC 2435. * For detailed information see: http://www.rfc-editor.org/rfc/rfc2435.txt diff --git a/gst/rtp/gstrtpklvdepay.c b/gst/rtp/gstrtpklvdepay.c index 424f6aade..a5026712b 100644 --- a/gst/rtp/gstrtpklvdepay.c +++ b/gst/rtp/gstrtpklvdepay.c @@ -20,18 +20,18 @@ /** * SECTION:element-rtpklvdepay + * @title: rtpklvdepay * @see_also: rtpklvpay * * Extract KLV metadata from RTP packets according to RFC 6597. * For detailed information see: http://tools.ietf.org/html/rfc6597 * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 udpsrc caps='application/x-rtp, media=(string)application, clock-rate=(int)90000, encoding-name=(string)SMPTE336M' ! rtpklvdepay ! fakesink dump=true * ]| This example pipeline will depayload an RTP KLV stream and display * a hexdump of the KLV data on stdout. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/rtp/gstrtpklvpay.c b/gst/rtp/gstrtpklvpay.c index 5a07d2cb3..f24d29f3c 100644 --- a/gst/rtp/gstrtpklvpay.c +++ b/gst/rtp/gstrtpklvpay.c @@ -20,18 +20,18 @@ /** * SECTION:element-rtpklvpay + * @title: rtpklvpay * @see_also: rtpklvdepay * * Payloads KLV metadata into RTP packets according to RFC 6597. * For detailed information see: http://tools.ietf.org/html/rfc6597 * - * <refsect2> - * <title>Example pipeline</title> + * ## Example pipeline * |[ * gst-launch-1.0 filesrc location=video-with-klv.ts ! tsdemux ! rtpklvpay ! udpsink * ]| This example pipeline will payload an RTP KLV stream extracted from an * MPEG-TS stream and send it via UDP to an RTP receiver. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/rtp/gstrtpstreamdepay.c b/gst/rtp/gstrtpstreamdepay.c index 34500e9cd..bcaec19ea 100644 --- a/gst/rtp/gstrtpstreamdepay.c +++ b/gst/rtp/gstrtpstreamdepay.c @@ -19,16 +19,17 @@ /** * SECTION:element-rtpstreamdepay + * @title: rtpstreamdepay * * Implements stream depayloading of RTP and RTCP packets for connection-oriented * transport protocols according to RFC4571. - * <refsect2> - * <title>Example launch line</title> + * + * ## Example launch line * |[ * gst-launch-1.0 audiotestsrc ! "audio/x-raw,rate=48000" ! vorbisenc ! rtpvorbispay config-interval=1 ! rtpstreampay ! tcpserversink port=5678 * gst-launch-1.0 tcpclientsrc port=5678 host=127.0.0.1 do-timestamp=true ! "application/x-rtp-stream,media=audio,clock-rate=48000,encoding-name=VORBIS" ! rtpstreamdepay ! rtpvorbisdepay ! decodebin ! audioconvert ! audioresample ! autoaudiosink * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpstreampay.c b/gst/rtp/gstrtpstreampay.c index 87848c4f2..b5758228b 100644 --- a/gst/rtp/gstrtpstreampay.c +++ b/gst/rtp/gstrtpstreampay.c @@ -20,16 +20,17 @@ /** * SECTION:element-rtpstreampay + * @title: rtpstreampay * * Implements stream payloading of RTP and RTCP packets for connection-oriented * transport protocols according to RFC4571. - * <refsect2> - * <title>Example launch line</title> + * + * ## Example launch line * |[ * gst-launch-1.0 audiotestsrc ! "audio/x-raw,rate=48000" ! vorbisenc ! rtpvorbispay config-interval=1 ! rtpstreampay ! tcpserversink port=5678 * gst-launch-1.0 tcpclientsrc port=5678 host=127.0.0.1 do-timestamp=true ! "application/x-rtp-stream,media=audio,clock-rate=48000,encoding-name=VORBIS" ! rtpstreamdepay ! rtpvorbisdepay ! decodebin ! audioconvert ! audioresample ! autoaudiosink * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 8b68266df..d7a6b6c59 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -19,6 +19,7 @@ /** * SECTION:element-rtpbin + * @title: rtpbin * @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux * * RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux, @@ -85,8 +86,7 @@ * and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad * when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \ * rtpbin ! rtptheoradepay ! theoradec ! xvimagesink @@ -131,7 +131,7 @@ * synchronisation. * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1 * on port 5007. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpdtmfmux.c b/gst/rtpmanager/gstrtpdtmfmux.c index cc6d74756..cff68e518 100644 --- a/gst/rtpmanager/gstrtpdtmfmux.c +++ b/gst/rtpmanager/gstrtpdtmfmux.c @@ -27,6 +27,7 @@ /** * SECTION:element-rtpdtmfmux + * @title: rtpdtmfmux * @see_also: rtpdtmfsrc, dtmfsrc, rtpmux * * The RTP "DTMF" Muxer muxes multiple RTP streams into a valid RTP diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 1bd3668c8..e07b4630a 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -30,6 +30,7 @@ /** * SECTION:element-rtpjitterbuffer + * @title: rtpjitterbuffer * * This element reorders and removes duplicate RTP packets as they are received * from a network source. @@ -88,14 +89,13 @@ * * This element will automatically be used inside rtpbin. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is * inserted into the pipeline to smooth out network jitter and to reorder the * out-of-order RTP packets. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H @@ -798,64 +798,14 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass) * Various jitterbuffer statistics. This property returns a GstStructure * with name application/x-rtp-jitterbuffer-stats with the following fields: * - * <itemizedlist> - * <listitem> - * <para> - * #guint64 - * <classname>"num-pushed"</classname>: - * the number of packets pushed out. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"num-lost"</classname>: - * the number of packets considered lost. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"num-late"</classname>: - * the number of packets arriving too late. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"num-duplicates"</classname>: - * the number of duplicate packets. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"rtx-count"</classname>: - * the number of retransmissions requested. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"rtx-success-count"</classname>: - * the number of successful retransmissions. - * </para> - * </listitem> - * <listitem> - * <para> - * #gdouble - * <classname>"rtx-per-packet"</classname>: - * average number of RTX per packet. - * </para> - * </listitem> - * <listitem> - * <para> - * #guint64 - * <classname>"rtx-rtt"</classname>: - * average round trip time per RTX. - * </para> - * </listitem> - * </itemizedlist> + * * #guint64 `num-pushed`: the number of packets pushed out. + * * #guint64 `num-lost`: the number of packets considered lost. + * * #guint64 `num-late`: the number of packets arriving too late. + * * #guint64 `num-duplicates`: the number of duplicate packets. + * * #guint64 `rtx-count`: the number of retransmissions requested. + * * #guint64 `rtx-success-count`: the number of successful retransmissions. + * * #gdouble `rtx-per-packet`: average number of RTX per packet. + * * #guint64 `rtx-rtt`: average round trip time per RTX. * * Since: 1.4 */ diff --git a/gst/rtpmanager/gstrtpmux.c b/gst/rtpmanager/gstrtpmux.c index c6cfc6c58..4d16d5d3c 100644 --- a/gst/rtpmanager/gstrtpmux.c +++ b/gst/rtpmanager/gstrtpmux.c @@ -27,13 +27,13 @@ /** * SECTION:element-rtpmux + * @title: rtpmux * @see_also: rtpdtmfmux * * The rtp muxer takes multiple RTP streams having the same clock-rate and * muxes into a single stream with a single SSRC. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 rtpmux name=mux ! udpsink host=127.0.0.1 port=8888 \ * alsasrc ! alawenc ! rtppcmapay ! \ @@ -45,7 +45,7 @@ * In this example, an audio stream is captured from ALSA and another is * generated, both are encoded into different payload types and muxed together * so they can be sent on the same port. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpptdemux.c b/gst/rtpmanager/gstrtpptdemux.c index 483966bfe..e4da02ea6 100644 --- a/gst/rtpmanager/gstrtpptdemux.c +++ b/gst/rtpmanager/gstrtpptdemux.c @@ -1,4 +1,4 @@ -/* +/* * RTP Demux element * * Copyright (C) 2005 Nokia Corporation. @@ -25,27 +25,27 @@ /** * SECTION:element-rtpptdemux + * @title: rtpptdemux * * rtpptdemux acts as a demuxer for RTP packets based on the payload type of * the packets. Its main purpose is to allow an application to easily receive * and decode an RTP stream with multiple payload types. - * + * * For each payload type that is detected, a new pad will be created and the * #GstRtpPtDemux::new-payload-type signal will be emitted. When the payload for * the RTP stream changes, the #GstRtpPtDemux::payload-type-change signal will be * emitted. - * + * * The element will try to set complete and unique application/x-rtp caps * on the output pads based on the result of the #GstRtpPtDemux::request-pt-map * signal. - * - * <refsect2> - * <title>Example pipelines</title> + * + * ## Example pipelines * |[ * gst-launch-1.0 udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink * ]| Takes an RTP stream and send the RTP packets with the first detected * payload type to fakesink, discarding the other payload types. - * </refsect2> + * */ /* @@ -99,8 +99,8 @@ GST_DEBUG_CATEGORY_STATIC (gst_rtp_pt_demux_debug); */ struct _GstRtpPtDemuxPad { - GstPad *pad; /**< pointer to the actual pad */ - gint pt; /**< RTP payload-type attached to pad */ + GstPad *pad; /*< pointer to the actual pad */ + gint pt; /*< RTP payload-type attached to pad */ gboolean newcaps; }; diff --git a/gst/rtpmanager/gstrtpptdemux.h b/gst/rtpmanager/gstrtpptdemux.h index 578e489fa..95d374e60 100644 --- a/gst/rtpmanager/gstrtpptdemux.h +++ b/gst/rtpmanager/gstrtpptdemux.h @@ -34,12 +34,12 @@ typedef struct _GstRtpPtDemuxPad GstRtpPtDemuxPad; struct _GstRtpPtDemux { - GstElement parent; /**< parent class */ + GstElement parent; /*< parent class */ - GstPad *sink; /**< the sink pad */ - guint16 last_pt; /**< pt of the last packet 0xFFFF if none */ - GSList *srcpads; /**< a linked list of GstRtpPtDemuxPad objects */ - GValue ignored_pts; /**< a GstValueArray of payload types that will not have pads created for */ + GstPad *sink; /*< the sink pad */ + guint16 last_pt; /*< pt of the last packet 0xFFFF if none */ + GSList *srcpads; /*< a linked list of GstRtpPtDemuxPad objects */ + GValue ignored_pts; /*< a GstValueArray of payload types that will not have pads created for */ }; struct _GstRtpPtDemuxClass diff --git a/gst/rtpmanager/gstrtprtxqueue.c b/gst/rtpmanager/gstrtprtxqueue.c index a0d6cbe37..582ff47cb 100644 --- a/gst/rtpmanager/gstrtprtxqueue.c +++ b/gst/rtpmanager/gstrtprtxqueue.c @@ -22,6 +22,7 @@ /** * SECTION:element-rtprtxqueue + * @title: rtprtxqueue * * rtprtxqueue maintains a queue of transmitted RTP packets, up to a * configurable limit (see #GstRTPRtxQueue::max-size-time, @@ -45,13 +46,16 @@ * See also #GstRtpRtxSend, #GstRtpRtxReceive * * # Example pipelines + * * |[ * gst-launch-1.0 rtpbin name=b rtp-profile=avpf \ * audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! rtprtxqueue ! b.send_rtp_sink_0 \ * b.send_rtp_src_0 ! identity drop-probability=0.01 ! udpsink host="127.0.0.1" port=5000 \ * udpsrc port=5001 ! b.recv_rtcp_sink_0 \ * b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5002 sync=false async=false - * ]| Sender pipeline + * ]| + * Sender pipeline + * * |[ * gst-launch-1.0 rtpbin name=b rtp-profile=avpf do-retransmission=true \ * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \ @@ -59,7 +63,8 @@ * b. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \ * udpsrc port=5002 ! b.recv_rtcp_sink_0 \ * b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5001 sync=false async=false - * ]| Receiver pipeline + * ]| + * Receiver pipeline */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtprtxreceive.c b/gst/rtpmanager/gstrtprtxreceive.c index 9a8a66697..4c97691cf 100644 --- a/gst/rtpmanager/gstrtprtxreceive.c +++ b/gst/rtpmanager/gstrtprtxreceive.c @@ -23,6 +23,7 @@ /** * SECTION:element-rtprtxreceive + * @title: rtprtxreceive * @see_also: rtprtxsend, rtpsession, rtpjitterbuffer * * rtprtxreceive listens to the retransmission events from the @@ -45,7 +46,8 @@ * rtpbin instead, with its #GstRtpBin::request-aux-sender and * #GstRtpBin::request-aux-receiver signals. See #GstRtpBin. * - * # Example pipelines + * ## Example pipelines + * * |[ * gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \ * audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! \ @@ -58,6 +60,7 @@ * sync=false async=false * ]| Send audio stream through port 5000 (5001 and 5002 are just the rtcp * link with the receiver) + * * |[ * gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \ * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \ @@ -69,7 +72,8 @@ * rtpsession.send_rtcp_src ! \ * udpsink host="127.0.0.1" port=5001 sync=false async=false \ * udpsrc port=5002 ! rtpsession.recv_rtcp_sink - * ]| Receive audio stream from port 5000 (5001 and 5002 are just the rtcp + * ]| + * Receive audio stream from port 5000 (5001 and 5002 are just the rtcp * link with the sender) * * In this example we can see a simple streaming of an OPUS stream with some @@ -102,7 +106,8 @@ * udpsrc port=5001 ! rtpsession.recv_rtcp_sink \ * rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 \ * sync=false async=false - * ]| Send two audio streams to port 5000. + * ]| + * Send two audio streams to port 5000. * |[ * gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \ * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)97" ! \ @@ -117,7 +122,8 @@ * udpsrc port=5002 ! rtpsession.recv_rtcp_sink \ * rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001 \ * sync=false async=false - * ]| Receive two audio streams from port 5000. + * ]| + * Receive two audio streams from port 5000. * * In this example we are streaming two streams of the same type through the * same port. They, however, are using a different SSRC (ssrc is randomly diff --git a/gst/rtpmanager/gstrtprtxsend.c b/gst/rtpmanager/gstrtprtxsend.c index 8c08f328e..bf35f257e 100644 --- a/gst/rtpmanager/gstrtprtxsend.c +++ b/gst/rtpmanager/gstrtprtxsend.c @@ -23,9 +23,10 @@ /** * SECTION:element-rtprtxsend + * @title: rtprtxsend * * See #GstRtpRtxReceive for examples - * + * * The purpose of the sender RTX object is to keep a history of RTP packets up * to a configurable limit (max-size-time or max-size-packets). It will listen * for upstream custom retransmission events (GstRTPRetransmissionRequest) that diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 1d2e96ec5..a6f70dc2d 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -19,6 +19,7 @@ /** * SECTION:element-rtpsession + * @title: rtpsession * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux * * The RTP session manager models participants with unique SSRC in an RTP @@ -27,23 +28,16 @@ * functionality can be activated. * * The session manager currently implements RFC 3550 including: - * <itemizedlist> - * <listitem> - * <para>RTP packet validation based on consecutive sequence numbers.</para> - * </listitem> - * <listitem> - * <para>Maintainance of the SSRC participant database.</para> - * </listitem> - * <listitem> - * <para>Keeping per participant statistics based on received RTCP packets.</para> - * </listitem> - * <listitem> - * <para>Scheduling of RR/SR RTCP packets.</para> - * </listitem> - * <listitem> - * <para>Support for multiple sender SSRC.</para> - * </listitem> - * </itemizedlist> + * + * * RTP packet validation based on consecutive sequence numbers. + * + * * Maintainance of the SSRC participant database. + * + * * Keeping per participant statistics based on received RTCP packets. + * + * * Scheduling of RR/SR RTCP packets. + * + * * Support for multiple sender SSRC. * * The rtpsession will not demux packets based on SSRC or payload type, nor will * it correct for packet reordering and jitter. Use #GstRtpsSrcDemux, @@ -75,8 +69,7 @@ * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map * signal. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink * ]| Receive theora RTP packets from port 5000 and send them to the depayloader, @@ -105,7 +98,7 @@ * correctly because the second udpsink will not preroll correctly (no RTCP * packets are sent in the PAUSED state). Applications should manually set and * keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c index 1661e790e..830ee1df5 100644 --- a/gst/rtpmanager/gstrtpssrcdemux.c +++ b/gst/rtpmanager/gstrtpssrcdemux.c @@ -21,21 +21,21 @@ /** * SECTION:element-rtpssrcdemux + * @title: rtpssrcdemux * * rtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the * packets. Its main purpose is to allow an application to easily receive and * decode an RTP stream with multiple SSRCs. - * + * * For each SSRC that is detected, a new pad will be created and the - * #GstRtpSsrcDemux::new-ssrc-pad signal will be emitted. - * - * <refsect2> - * <title>Example pipelines</title> + * #GstRtpSsrcDemux::new-ssrc-pad signal will be emitted. + * + * ## Example pipelines * |[ * gst-launch-1.0 udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink * ]| Takes an RTP stream and send the RTP packets with the first detected SSRC * to fakesink, discarding the other SSRCs. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index fefa259d4..19d56bae3 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -516,9 +516,8 @@ rtp_session_class_init (RTPSessionClass * klass) * * Get a GValue Array of all sources in the session. * - * <example> - * <title>Getting the #RTPSources of a session - * <programlisting> + * ## Getting the #RTPSources of a session + * |[ * { * GValueArray *arr; * GValue *val; @@ -534,8 +533,7 @@ rtp_session_class_init (RTPSessionClass * klass) * } * g_value_array_free (arr); * } - * </programlisting> - * </example> + * ]| */ g_object_class_install_property (gobject_class, PROP_SOURCES, g_param_spec_boxed ("sources", "Sources", diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c index 3cf7c7913..3a8ebc861 100644 --- a/gst/rtsp/gstrtpdec.c +++ b/gst/rtsp/gstrtpdec.c @@ -43,6 +43,7 @@ /** * SECTION:element-rtpdec + * @title: rtpdec * * A simple RTP session manager used internally by rtspsrc. */ @@ -241,7 +242,7 @@ gst_rtp_dec_class_init (GstRTPDecClass * g_class) * GstRTPDec::on-new-ssrc: * @rtpbin: the object which received the signal * @session: the session - * @ssrc: the SSRC + * @ssrc: the SSRC * * Notify of a new SSRC that entered @session. */ @@ -254,7 +255,7 @@ gst_rtp_dec_class_init (GstRTPDecClass * g_class) * GstRTPDec::on-ssrc_collision: * @rtpbin: the object which received the signal * @session: the session - * @ssrc: the SSRC + * @ssrc: the SSRC * * Notify when we have an SSRC collision */ @@ -267,7 +268,7 @@ gst_rtp_dec_class_init (GstRTPDecClass * g_class) * GstRTPDec::on-ssrc_validated: * @rtpbin: the object which received the signal * @session: the session - * @ssrc: the SSRC + * @ssrc: the SSRC * * Notify of a new SSRC that became validated. */ @@ -281,7 +282,7 @@ gst_rtp_dec_class_init (GstRTPDecClass * g_class) * GstRTPDec::on-bye-ssrc: * @rtpbin: the object which received the signal * @session: the session - * @ssrc: the SSRC + * @ssrc: the SSRC * * Notify of an SSRC that became inactive because of a BYE packet. */ @@ -294,7 +295,7 @@ gst_rtp_dec_class_init (GstRTPDecClass * g_class) * GstRTPDec::on-bye-timeout: * @rtpbin: the object which received the signal * @session: the session - * @ssrc: the SSRC + * @ssrc: the SSRC * * Notify of an SSRC that has timed out because of BYE */ @@ -307,7 +308,7 @@ gst_rtp_dec_class_init (GstRTPDecClass * g_class) * GstRTPDec::on-timeout: * @rtpbin: the object which received the signal * @session: the session - * @ssrc: the SSRC + * @ssrc: the SSRC * * Notify of an SSRC that has timed out */ diff --git a/gst/shapewipe/gstshapewipe.c b/gst/shapewipe/gstshapewipe.c index 3a0dfdaaf..35d322efc 100644 --- a/gst/shapewipe/gstshapewipe.c +++ b/gst/shapewipe/gstshapewipe.c @@ -19,6 +19,7 @@ /** * SECTION:element-shapewipe + * @title: shapewipe * * The shapewipe element provides custom transitions on video streams * based on a grayscale bitmap. The state of the transition can be @@ -29,12 +30,11 @@ * <ulink url="http://cinelerra.org/transitions.php">Cinelerra transition</ulink> * page. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v videotestsrc ! video/x-raw,format=AYUV,width=640,height=480 ! shapewipe position=0.5 name=shape ! videomixer name=mixer ! videoconvert ! autovideosink filesrc location=mask.png ! typefind ! decodebin ! videoconvert ! videoscale ! queue ! shape.mask_sink videotestsrc pattern=snow ! video/x-raw,format=AYUV,width=640,height=480 ! queue ! mixer. * ]| This pipeline adds the transition from mask.png with position 0.5 to an SMPTE test screen and snow. - * </refsect2> + * */ diff --git a/gst/smpte/gstsmpte.c b/gst/smpte/gstsmpte.c index 7e8864198..5e555cb39 100644 --- a/gst/smpte/gstsmpte.c +++ b/gst/smpte/gstsmpte.c @@ -19,6 +19,7 @@ /** * SECTION:element-smpte + * @title: smpte * * smpte can accept I420 video streams with the same width, height and * framerate. The two incoming buffers are blended together using an effect @@ -28,15 +29,14 @@ * higher presision will create a mask with smoother gradients in order to avoid * banding. * - * <refsect2> - * <title>Sample pipelines</title> + * ## Sample pipelines * |[ * gst-launch-1.0 -v videotestsrc pattern=1 ! smpte name=s border=20000 type=234 duration=2000000000 ! videoconvert ! ximagesink videotestsrc ! s. * ]| A pipeline to demonstrate the smpte transition. * It shows a pinwheel transition a from a snow videotestsrc to an smpte * pattern videotestsrc. The transition will take 2 seconds to complete. The * edges of the transition are smoothed with a 20000 big border. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/smpte/gstsmptealpha.c b/gst/smpte/gstsmptealpha.c index 750748f25..ee093f502 100644 --- a/gst/smpte/gstsmptealpha.c +++ b/gst/smpte/gstsmptealpha.c @@ -19,6 +19,7 @@ /** * SECTION:element-smptealpha + * @title: smptealpha * * smptealpha can accept an I420 or AYUV video stream. An alpha channel is added * using an effect specific SMPTE mask in the I420 input case. In the AYUV case, @@ -33,19 +34,17 @@ * A higher presision will create a mask with smoother gradients in order to * avoid banding. * - * <refsect2> - * <title>Sample pipelines</title> - * <para> + * ## Sample pipelines + * * Here is a pipeline to demonstrate the smpte transition : - * <programlisting> + * |[ * gst-launch-1.0 -v videotestsrc ! smptealpha border=20000 type=44 * position=0.5 ! videomixer ! videoconvert ! ximagesink - * </programlisting> + * ]| * This shows a midway bowtie-h transition a from a videotestsrc to a * transparent image. The edges of the transition are smoothed with a * 20000 big border. - * </para> - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c index 4be896025..1c5efc62d 100644 --- a/gst/spectrum/gstspectrum.c +++ b/gst/spectrum/gstspectrum.c @@ -20,80 +20,39 @@ */ /** * SECTION:element-spectrum + * @title: spectrum * * The Spectrum element analyzes the frequency spectrum of an audio signal. * If the #GstSpectrum:post-messages property is %TRUE, it sends analysis results * as element messages named - * <classname>"spectrum"</classname> after each interval of time given + * `spectrum` after each interval of time given * by the #GstSpectrum:interval property. * * The message's structure contains some combination of these fields: - * <itemizedlist> - * <listitem> - * <para> - * #GstClockTime - * <classname>"timestamp"</classname>: - * the timestamp of the buffer that triggered the message. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"stream-time"</classname>: - * the stream time of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"running-time"</classname>: - * the running_time of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"duration"</classname>: - * the duration of the buffer. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstClockTime - * <classname>"endtime"</classname>: - * the end time of the buffer that triggered the message as stream time (this + * + * * #GstClockTime `timestamp`: the timestamp of the buffer that triggered the message. + * * #GstClockTime `stream-time`: the stream time of the buffer. + * * #GstClockTime `running-time`: the running_time of the buffer. + * * #GstClockTime `duration`: the duration of the buffer. + * * #GstClockTime `endtime`: the end time of the buffer that triggered the message as stream time (this * is deprecated, as it can be calculated from stream-time + duration) - * </para> - * </listitem> - * <listitem> - * <para> - * #GstValueList of #gfloat - * <classname>"magnitude"</classname>: - * the level for each frequency band in dB. All values below the value of the + * * #GstValueList of #gfloat `magnitude`: the level for each frequency band in dB. + * All values below the value of the * #GstSpectrum:threshold property will be set to the threshold. Only present * if the #GstSpectrum:message-magnitude property is %TRUE. - * </para> - * </listitem> - * <listitem> - * <para> - * #GstValueList of #gfloat - * <classname>"phase"</classname>: - * The phase for each frequency band. The value is between -pi and pi. Only + * * #GstValueList of #gfloat `phase`: The phase for each frequency band. The value is between -pi and pi. Only * present if the #GstSpectrum:message-phase property is %TRUE. - * </para> - * </listitem> - * </itemizedlist> * * If #GstSpectrum:multi-channel property is set to true. magnitude and phase * fields will be each a nested #GstValueArray. The first dimension are the * channels and the second dimension are the values. * - * <refsect2> - * <title>Example application</title> + * ## Example application + * * <informalexample><programlisting language="C"> * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" /> * </programlisting></informalexample> - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c index b6859075d..13f1cc203 100644 --- a/gst/udp/gstmultiudpsink.c +++ b/gst/udp/gstmultiudpsink.c @@ -22,6 +22,7 @@ /** * SECTION:element-multiudpsink + * @title: multiudpsink * @see_also: udpsink, multifdsink * * multiudpsink is a network sink that sends UDP packets to multiple diff --git a/gst/udp/gstudpsink.c b/gst/udp/gstudpsink.c index 224d57818..cfe8fdeeb 100644 --- a/gst/udp/gstudpsink.c +++ b/gst/udp/gstudpsink.c @@ -20,17 +20,17 @@ */ /** * SECTION:element-udpsink + * @title: udpsink * @see_also: udpsrc, multifdsink * * udpsink is a network sink that sends UDP packets to the network. * It can be combined with RTP payloaders to implement RTP streaming. * - * <refsect2> - * <title>Examples</title> + * ## Examples * |[ * gst-launch-1.0 -v audiotestsrc ! udpsink * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index c0fa37f79..681e77d27 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -24,6 +24,7 @@ /** * SECTION:element-udpsrc + * @title: udpsrc * @see_also: udpsink, multifdsink * * udpsrc is a network source that reads UDP packets from the network. @@ -65,19 +66,13 @@ * type URIs. * * If the #GstUDPSrc:timeout property is set to a value bigger than 0, udpsrc - * will generate an element message named - * <classname>"GstUDPSrcTimeout"</classname> + * will generate an element message named `GstUDPSrcTimeout` * if no data was received in the given timeout. + * * The message's structure contains one field: - * <itemizedlist> - * <listitem> - * <para> - * #guint64 - * <classname>"timeout"</classname>: the timeout in microseconds that - * expired when waiting for data. - * </para> - * </listitem> - * </itemizedlist> + * + * * #guint64 `timeout`: the timeout in microseconds that expired when waiting for data. + * * The message is typically used to detect that no UDP arrives in the receiver * because it is blocked by a firewall. * @@ -87,8 +82,7 @@ * with the #GstUDPSrc:close-socket property, in which case the * application is responsible for closing the file descriptor. * - * <refsect2> - * <title>Examples</title> + * ## Examples * |[ * gst-launch-1.0 -v udpsrc ! fakesink dump=1 * ]| A pipeline to read from the default port and dump the udp packets. @@ -101,7 +95,7 @@ * |[ * gst-launch-1.0 -v udpsrc port=0 ! fakesink * ]| read udp packets from a free port. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/videobox/gstvideobox.c b/gst/videobox/gstvideobox.c index f8f984c5a..d657e64cd 100644 --- a/gst/videobox/gstvideobox.c +++ b/gst/videobox/gstvideobox.c @@ -20,21 +20,22 @@ */ /** * SECTION:element-videobox + * @title: videobox * @see_also: #GstVideoCrop * * This plugin crops or enlarges the image. It takes 4 values as input, a * top, bottom, left and right offset. Positive values will crop that much * pixels from the respective border of the image, negative values will add - * that much pixels. When pixels are added, you can specify their color. + * that much pixels. When pixels are added, you can specify their color. * Some predefined colors are usable with an enum property. - * + * * The plugin is alpha channel aware and will try to negotiate with a format * that supports alpha channels first. When alpha channel is active two * other properties, alpha and border_alpha can be used to set the alpha * values of the inner picture and the border respectively. an alpha value of * 0.0 means total transparency, 1.0 is opaque. - * - * The videobox plugin has many uses such as doing a mosaic of pictures, + * + * The videobox plugin has many uses such as doing a mosaic of pictures, * letterboxing video, cutting out pieces of video, picture in picture, etc.. * * Setting autocrop to true changes the behavior of the plugin so that @@ -42,11 +43,11 @@ * input and output dimensions, the crop values are selected so that the * smaller frame is effectively centered in the larger frame. This * involves either cropping or padding. - * + * * If you use autocrop there is little point in setting the other * properties manually because they will be overriden if the caps change, * but nothing stops you from doing so. - * + * * Sample pipeline: * |[ * gst-launch-1.0 videotestsrc ! videobox autocrop=true ! \ diff --git a/gst/videocrop/gstaspectratiocrop.c b/gst/videocrop/gstaspectratiocrop.c index 34028647a..9ca93c1ff 100644 --- a/gst/videocrop/gstaspectratiocrop.c +++ b/gst/videocrop/gstaspectratiocrop.c @@ -19,6 +19,7 @@ /** * SECTION:element-aspectratiocrop + * @title: aspectratiocrop * @see_also: #GstVideoCrop * * This element crops video frames to a specified #GstAspectRatioCrop:aspect-ratio. @@ -26,12 +27,11 @@ * If the aspect-ratio is already correct, the element will operate * in pass-through mode. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v videotestsrc ! video/x-raw,height=640,width=480 ! aspectratiocrop aspect-ratio=16/9 ! ximagesink * ]| This pipeline generates a videostream in 4/3 and crops it to 16/9. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/videocrop/gstvideocrop.c b/gst/videocrop/gstvideocrop.c index 31e7d6769..03f6e292f 100644 --- a/gst/videocrop/gstvideocrop.c +++ b/gst/videocrop/gstvideocrop.c @@ -19,6 +19,7 @@ /** * SECTION:element-videocrop + * @title: videocrop * @see_also: #GstVideoBox * * This element crops video frames, meaning it can remove parts of the @@ -36,15 +37,14 @@ * Note that no special efforts are made to handle chroma-subsampled formats * in the case of odd-valued cropping and compensate for sub-unit chroma plane * shifts for such formats in the case where the #GstVideoCrop:left or - * #GstVideoCrop:top property is set to an odd number. This doesn't matter for + * #GstVideoCrop:top property is set to an odd number. This doesn't matter for * most use cases, but it might matter for yours. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 -v videotestsrc ! videocrop top=42 left=1 right=4 bottom=0 ! ximagesink * ]| - * </refsect2> + * */ /* TODO: diff --git a/gst/videofilter/gstgamma.c b/gst/videofilter/gstgamma.c index fd4409f33..769e6f060 100644 --- a/gst/videofilter/gstgamma.c +++ b/gst/videofilter/gstgamma.c @@ -30,18 +30,18 @@ /** * SECTION:element-gamma + * @title: gamma * * Performs gamma correction on a video stream. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 videotestsrc ! gamma gamma=2.0 ! videoconvert ! ximagesink * ]| This pipeline will make the image "brighter". * |[ * gst-launch-1.0 videotestsrc ! gamma gamma=0.5 ! videoconvert ! ximagesink * ]| This pipeline will make the image "darker". - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c index 068bcd7bc..c82787f21 100644 --- a/gst/videofilter/gstvideobalance.c +++ b/gst/videofilter/gstvideobalance.c @@ -26,16 +26,16 @@ /** * SECTION:element-videobalance + * @title: videobalance * * Adjusts brightness, contrast, hue, saturation on a video stream. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 videotestsrc ! videobalance saturation=0.0 ! videoconvert ! ximagesink * ]| This pipeline converts the image to black and white by setting the * saturation to 0.0. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/videofilter/gstvideoflip.c b/gst/videofilter/gstvideoflip.c index c6132f587..f5afffe7a 100644 --- a/gst/videofilter/gstvideoflip.c +++ b/gst/videofilter/gstvideoflip.c @@ -26,15 +26,15 @@ */ /** * SECTION:element-videoflip + * @title: videoflip * * Flips and rotates video. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 videotestsrc ! videoflip method=clockwise ! videoconvert ! ximagesink * ]| This pipeline flips the test image 90 degrees clockwise. - * </refsect2> + * */ diff --git a/gst/videomixer/videomixer2.c b/gst/videomixer/videomixer2.c index c6f2da8a7..a9fc8e68e 100644 --- a/gst/videomixer/videomixer2.c +++ b/gst/videomixer/videomixer2.c @@ -20,6 +20,7 @@ /** * SECTION:element-videomixer + * @title: videomixer * * Videomixer can accept AYUV, ARGB and BGRA video streams. For each of the requested * sink pads it will compare the incoming geometry and framerate to define the @@ -27,12 +28,11 @@ * biggest incoming video stream and the framerate of the fastest incoming one. * * Videomixer will do colorspace conversion. - * + * * Individual parameters for each input stream can be configured on the * #GstVideoMixer2Pad. * - * <refsect2> - * <title>Sample pipelines</title> + * ## Sample pipelines * |[ * gst-launch-1.0 \ * videotestsrc pattern=1 ! \ @@ -55,7 +55,7 @@ * videomixer name=mix ! videoconvert ! ximagesink \ * videotestsrc ! \ * video/x-raw, framerate=\(fraction\)5/1, width=320, height=240 ! mix. - * ]| A pipeline to demostrate bgra mixing. (This does not demonstrate alpha blending). + * ]| A pipeline to demostrate bgra mixing. (This does not demonstrate alpha blending). * |[ * gst-launch-1.0 videotestsrc pattern=1 ! \ * video/x-raw,format =I420, framerate=\(fraction\)10/1, width=100, height=100 ! \ @@ -73,7 +73,7 @@ * "video/x-raw,format=AYUV,width=800,height=600,framerate=(fraction)10/1" ! \ * timeoverlay ! queue2 ! mixer. * ]| A pipeline to demonstrate synchronized mixing (the second stream starts after 3 seconds) - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/gst/wavenc/gstwavenc.c b/gst/wavenc/gstwavenc.c index 94a8bd0d0..e7c95d66c 100644 --- a/gst/wavenc/gstwavenc.c +++ b/gst/wavenc/gstwavenc.c @@ -21,18 +21,17 @@ */ /** * SECTION:element-wavenc + * @title: wavenc * * Format an audio stream into the wav format. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 cdparanoiasrc mode=continuous ! queue ! audioconvert ! wavenc ! filesink location=cd.wav * ]| Rip a whole audio CD into a single wav file, with the track table written into a CUE sheet inside the file * |[ * gst-launch-1.0 cdparanoiasrc track=5 ! queue ! audioconvert ! wavenc ! filesink location=track5.wav * ]| Rip track 5 of an audio CD into a single wav file containing unencoded raw audio samples. - * </refsect2> * */ #ifdef HAVE_CONFIG_H diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 22ed04287..496eeeef3 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -21,14 +21,14 @@ /** * SECTION:element-wavparse + * @title: wavparse * * Parse a .wav file into raw or compressed audio. * * Wavparse supports both push and pull mode operations, making it possible to * stream from a network source. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 filesrc location=sine.wav ! wavparse ! audioconvert ! alsasink * ]| Read a wav file and output to the soundcard using the ALSA element. The @@ -36,7 +36,7 @@ * |[ * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink * ]| Stream data from a network url. - * </refsect2> + * */ /* diff --git a/gst/y4m/gsty4mencode.c b/gst/y4m/gsty4mencode.c index 2a113bc93..e4a1e6027 100644 --- a/gst/y4m/gsty4mencode.c +++ b/gst/y4m/gsty4mencode.c @@ -19,21 +19,18 @@ */ /** * SECTION:element-y4menc + * @title: y4menc * - * <refsect2> - * <para> * Creates a YU4MPEG2 raw video stream as defined by the mjpegtools project. - * </para> - * <title>Example launch line</title> - * <para> + * + * ## Example launch line + * * (write everything in one line, without the backslash characters) - * <programlisting> + * |[ * gst-launch-1.0 videotestsrc num-buffers=250 \ * ! 'video/x-raw,format=(string)I420,width=320,height=240,framerate=(fraction)25/1' \ * ! y4menc ! filesink location=test.yuv - * </programlisting> - * </para> - * </refsect2> + * ]| * */ diff --git a/sys/directsound/gstdirectsoundsink.c b/sys/directsound/gstdirectsoundsink.c index c0da18685..66e2dfd88 100644 --- a/sys/directsound/gstdirectsoundsink.c +++ b/sys/directsound/gstdirectsoundsink.c @@ -28,6 +28,7 @@ /** * SECTION:element-directsoundsink + * @title: directsoundsink * * This element lets you output sound using the DirectSound API. * @@ -36,8 +37,7 @@ * your pipeline works under all circumstances (those conversion elements will * act in passthrough-mode if no conversion is necessary). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! directsoundsink * ]| will output a sine wave (continuous beep sound) to your sound card (with @@ -45,7 +45,7 @@ * |[ * gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! directsoundsink * ]| will play an Ogg/Vorbis audio file and output it. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/oss/gstosssink.c b/sys/oss/gstosssink.c index 2e6b8c5c7..75b2cf788 100644 --- a/sys/oss/gstosssink.c +++ b/sys/oss/gstosssink.c @@ -22,6 +22,7 @@ /** * SECTION:element-osssink + * @title: osssink * * This element lets you output sound using the Open Sound System (OSS). * @@ -30,8 +31,7 @@ * your pipeline works under all circumstances (those conversion elements will * act in passthrough-mode if no conversion is necessary). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! osssink * ]| will output a sine wave (continuous beep sound) to your sound card (with @@ -39,7 +39,7 @@ * |[ * gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! osssink * ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/oss/gstosssrc.c b/sys/oss/gstosssrc.c index 4923ad26d..c4c55f904 100644 --- a/sys/oss/gstosssrc.c +++ b/sys/oss/gstosssrc.c @@ -22,17 +22,17 @@ /** * SECTION:element-osssrc + * @title: osssrc * * This element lets you record sound using the Open Sound System (OSS). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 -v osssrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg * ]| will record sound from your sound card using OSS and encode it to an * Ogg/Vorbis file (this will only work if your mixer settings are right * and the right inputs enabled etc.) - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/oss4/oss4-sink.c b/sys/oss4/oss4-sink.c index 84028e7a5..177d4dc22 100644 --- a/sys/oss4/oss4-sink.c +++ b/sys/oss4/oss4-sink.c @@ -18,17 +18,17 @@ */ /** * SECTION:element-oss4sink + * @title: oss4sink * * This element lets you output sound using the Open Sound System (OSS) * version 4. - * + * * Note that you should almost always use generic audio conversion elements * like audioconvert and audioresample in front of an audiosink to make sure * your pipeline works under all circumstances (those conversion elements will * act in passthrough-mode if no conversion is necessary). - * - * <refsect2> - * <title>Example pipelines</title> + * + * ## Example pipelines * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! oss4sink * ]| will output a sine wave (continuous beep sound) to your sound card (with @@ -37,7 +37,7 @@ * gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! oss4sink * ]| will play an Ogg/Vorbis audio file and output it using the Open Sound System * version 4. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/oss4/oss4-source.c b/sys/oss4/oss4-source.c index 84b315d35..92a620ad4 100644 --- a/sys/oss4/oss4-source.c +++ b/sys/oss4/oss4-source.c @@ -19,18 +19,18 @@ /** * SECTION:element-oss4src + * @title: oss4src * * This element lets you record sound using the Open Sound System (OSS) * version 4. - * - * <refsect2> - * <title>Example pipelines</title> + * + * ## Example pipelines * |[ * gst-launch-1.0 -v oss4src ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=mymusic.ogg * ]| will record sound from your sound card using OSS4 and encode it to an * Ogg/Vorbis file (this will only work if your mixer settings are right * and the right inputs areenabled etc.) - * </refsect2> + * */ /* FIXME: make sure we're not doing ioctls from the app thread (e.g. via the diff --git a/sys/osxaudio/gstosxaudiosink.c b/sys/osxaudio/gstosxaudiosink.c index a1fd15875..0e9608b69 100644 --- a/sys/osxaudio/gstosxaudiosink.c +++ b/sys/osxaudio/gstosxaudiosink.c @@ -49,15 +49,15 @@ /** * SECTION:element-osxaudiosink + * @title: osxaudiosink * * This element renders raw audio samples using the CoreAudio api. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink * ]| Play an Ogg/Vorbis file. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/osxaudio/gstosxaudiosrc.c b/sys/osxaudio/gstosxaudiosrc.c index 4e56778bc..c3c445b47 100644 --- a/sys/osxaudio/gstosxaudiosrc.c +++ b/sys/osxaudio/gstosxaudiosrc.c @@ -44,15 +44,15 @@ /** * SECTION:element-osxaudiosrc + * @title: osxaudiosrc * * This element captures raw audio samples using the CoreAudio api. * - * <refsect2> - * <title>Example launch line</title> + * ## Example launch line * |[ * gst-launch-1.0 osxaudiosrc ! wavenc ! filesink location=audio.wav * ]| - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/v4l2/gstv4l2radio.c b/sys/v4l2/gstv4l2radio.c index 6bfbefb6f..5a18e5aa2 100644 --- a/sys/v4l2/gstv4l2radio.c +++ b/sys/v4l2/gstv4l2radio.c @@ -21,19 +21,19 @@ /** * SECTION:element-v4l2radio + * @title: v4l2radio * * v4l2radio can be used to control radio device * and to tune it to different radiostations. * - * <refsect2> - * <title>Example launch lines</title> + * ## Example launch lines * |[ * gst-launch-1.0 v4l2radio device=/dev/radio0 frequency=101200000 * gst-launch-1.0 alsasrc device=hw:1 ! audioconvert ! audioresample ! alsasink * ]| * First pipeline tunes the radio device /dev/radio0 to station 101.2 MHz, * second pipeline connects digital audio out (hw:1) to default sound card. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/v4l2/gstv4l2sink.c b/sys/v4l2/gstv4l2sink.c index 48ed8f4b0..162835eac 100644 --- a/sys/v4l2/gstv4l2sink.c +++ b/sys/v4l2/gstv4l2sink.c @@ -24,12 +24,12 @@ /** * SECTION:element-v4l2sink + * @title: v4l2sink * * v4l2sink can be used to display video to v4l2 devices (screen overlays * provided by the graphics hardware, tv-out, etc) * - * <refsect2> - * <title>Example launch lines</title> + * ## Example launch lines * |[ * gst-launch-1.0 videotestsrc ! v4l2sink device=/dev/video1 * ]| This pipeline displays a test pattern on /dev/video1 @@ -45,7 +45,7 @@ * original video frame geometry so that the box can be drawn to the correct * position. This also handles borders correctly, limiting coordinates to the * image area - * </refsect2> + * */ diff --git a/sys/v4l2/gstv4l2src.c b/sys/v4l2/gstv4l2src.c index 3997114d1..615554169 100644 --- a/sys/v4l2/gstv4l2src.c +++ b/sys/v4l2/gstv4l2src.c @@ -23,12 +23,12 @@ /** * SECTION:element-v4l2src + * @title: v4l2src * * v4l2src can be used to capture video from v4l2 devices, like webcams and tv * cards. * - * <refsect2> - * <title>Example launch lines</title> + * ## Example launch lines * |[ * gst-launch-1.0 v4l2src ! xvimagesink * ]| This pipeline shows the video captured from /dev/video0 tv card and for @@ -37,7 +37,6 @@ * gst-launch-1.0 v4l2src ! jpegdec ! xvimagesink * ]| This pipeline shows the video captured from a webcam that delivers jpeg * images. - * </refsect2> * * Since 1.14, the use of libv4l2 has been disabled due to major bugs in the * emulation layer. To enable usage of this library, set the environment diff --git a/sys/v4l2/tuner.c b/sys/v4l2/tuner.c index b6b3fe917..e0435644f 100644 --- a/sys/v4l2/tuner.c +++ b/sys/v4l2/tuner.c @@ -29,43 +29,36 @@ /** * SECTION:gsttuner + * @title: TunEr.h * @short_description: Interface for elements providing tuner operations - * - * <refsect2> - * <para> + * * The GstTuner interface is provided by elements that have the ability to * tune into multiple input signals, for example TV or radio capture cards. - * </para><para> + * * The interpretation of 'tuning into' an input stream depends on the element * implementing the interface. For v4lsrc, it might imply selection of an - * input source and/or frequency to be configured on a TV card. Another + * input source and/or frequency to be configured on a TV card. Another * GstTuner implementation might be to allow selection of an active input pad * from multiple input pads. - * </para><para> + * * That said, the GstTuner interface functions are biased toward the * TV capture scenario. - * </para><para> + * * The general parameters provided are for configuration are: - * <itemizedlist> - * <listitem>Selection of a current #GstTunerChannel. The current channel - * represents the input source (e.g. Composite, S-Video etc for TV capture). - * </listitem> - * <listitem>The #GstTunerNorm for the channel. The norm chooses the - * interpretation of the incoming signal for the current channel. For example, - * PAL or NTSC, or more specific variants there-of. - * </listitem> - * <listitem>Channel frequency. If the current channel has the ability to tune - * between multiple frequencies (if it has the GST_TUNER_CHANNEL_FREQUENCY flag) - * then the frequency can be changed/retrieved via the - * gst_tuner_set_frequency() and gst_tuner_get_frequency() methods. - * </listitem> - * </itemizedlist> - * </para> - * <para> + * + * * Selection of a current #GstTunerChannel. The current channel + * represents the input source (e.g. Composite, S-Video etc for TV capture). + * * The #GstTunerNorm for the channel. The norm chooses the + * interpretation of the incoming signal for the current channel. For example, + * PAL or NTSC, or more specific variants there-of. + * * Channel frequency. If the current channel has the ability to tune + * between multiple frequencies (if it has the GST_TUNER_CHANNEL_FREQUENCY flag) + * then the frequency can be changed/retrieved via the + * gst_tuner_set_frequency() and gst_tuner_get_frequency() methods. + * * Where applicable, the signal strength can be retrieved and/or monitored * via a signal. - * </para> - * </refsect2> + * */ /* FIXME 0.11: check if we need to add API for sometimes-supportedness @@ -324,7 +317,7 @@ gst_tuner_get_norm (GstTuner * tuner) * checked using GST_TUNER_CHANNEL_HAS_FLAG (), with the proper flag * being GST_TUNER_CHANNEL_FREQUENCY. * - * The frequency is in Hz, with minimum steps indicated by the + * The frequency is in Hz, with minimum steps indicated by the * frequency_multiplicator provided in the #GstTunerChannel. The * valid range is provided in the min_frequency and max_frequency properties * of the #GstTunerChannel. @@ -389,7 +382,7 @@ gst_tuner_get_frequency (GstTuner * tuner, GstTunerChannel * channel) * GST_TUNER_CHANNEL_HAS_FLAG (), and the appropriate flag to check * for is GST_TUNER_CHANNEL_FREQUENCY. * - * The valid range of the signal strength is indicated in the + * The valid range of the signal strength is indicated in the * min_signal and max_signal properties of the #GstTunerChannel. * * Returns: Signal strength, or 0 on error. @@ -416,7 +409,7 @@ gst_tuner_signal_strength (GstTuner * tuner, GstTunerChannel * channel) * gst_tuner_find_norm_by_name: * @tuner: A #GstTuner instance * @norm: A string containing the name of a #GstTunerNorm - * + * * Look up a #GstTunerNorm by name. * * Returns: A #GstTunerNorm, or NULL if no norm with the provided name @@ -443,7 +436,7 @@ gst_tuner_find_norm_by_name (GstTuner * tuner, gchar * norm) * gst_tuner_find_channel_by_name: * @tuner: A #GstTuner instance * @channel: A string containing the name of a #GstTunerChannel - * + * * Look up a #GstTunerChannel by name. * * Returns: A #GstTunerChannel, or NULL if no channel with the provided name @@ -491,7 +484,7 @@ gst_tuner_channel_changed (GstTuner * tuner, GstTunerChannel * channel) * * Called by elements implementing the #GstTuner interface when the * current norm changes. Fires the #GstTuner::norm-changed signal. - * + * */ void gst_tuner_norm_changed (GstTuner * tuner, GstTunerNorm * norm) @@ -534,7 +527,7 @@ gst_tuner_frequency_changed (GstTuner * tuner, * * Called by elements implementing the #GstTuner interface when the * incoming signal strength changes. Fires the #GstTuner::signal-changed - * signal on the tuner and the #GstTunerChannel::signal-changed signal on + * signal on the tuner and the #GstTunerChannel::signal-changed signal on * the channel. */ void diff --git a/sys/v4l2/tunerchannel.c b/sys/v4l2/tunerchannel.c index 6e456967b..a6902a2d3 100644 --- a/sys/v4l2/tunerchannel.c +++ b/sys/v4l2/tunerchannel.c @@ -27,20 +27,17 @@ /** * SECTION:gsttunerchannel + * @title: GstTunerChannel * @short_description: A channel from an element implementing the #GstTuner * interface. * - * <refsect2> - * <para>The #GstTunerChannel object is provided by an element implementing + * The #GstTunerChannel object is provided by an element implementing * the #GstTuner interface. - * </para> - * <para> + * * GstTunerChannel provides a name and flags to determine the type and * capabilities of the channel. If the GST_TUNER_CHANNEL_FREQUENCY flag is * set, then the channel also information about the minimum and maximum * frequency, and range of the reported signal strength. - * </para> - * </refsect2> */ enum diff --git a/sys/v4l2/tunernorm.c b/sys/v4l2/tunernorm.c index 5c57ffb7d..931d290ab 100644 --- a/sys/v4l2/tunernorm.c +++ b/sys/v4l2/tunernorm.c @@ -27,15 +27,14 @@ /** * SECTION:gsttunernorm + * @title: TunErnorm.h * @short_description: Encapsulates information about the data format(s) * for a #GstTunerChannel. * - * <refsect2> - * <para>The #GstTunerNorm object is created by an element implementing the + * The #GstTunerNorm object is created by an element implementing the * #GstTuner interface and encapsulates the selection of a capture/output format * for a selected #GstTunerChannel. - * </para> - * </refsect2> + * */ enum diff --git a/sys/waveform/gstwaveformsink.c b/sys/waveform/gstwaveformsink.c index b670a5faa..7fc6eb121 100644 --- a/sys/waveform/gstwaveformsink.c +++ b/sys/waveform/gstwaveformsink.c @@ -21,6 +21,7 @@ /** * SECTION:element-waveformsink + * @title: waveformsink * * This element lets you output sound using the Windows WaveForm API. * @@ -29,8 +30,7 @@ * your pipeline works under all circumstances (those conversion elements will * act in passthrough-mode if no conversion is necessary). * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 -v audiotestsrc ! audioconvert ! volume volume=0.1 ! waveformsink * ]| will output a sine wave (continuous beep sound) to your sound card (with @@ -38,7 +38,7 @@ * |[ * gst-launch-1.0 -v filesrc location=music.ogg ! decodebin ! audioconvert ! audioresample ! waveformsink * ]| will play an Ogg/Vorbis audio file and output it. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c index 8aad4e2fd..22bbb9de2 100644 --- a/sys/ximage/gstximagesrc.c +++ b/sys/ximage/gstximagesrc.c @@ -20,6 +20,7 @@ /** * SECTION:element-ximagesrc + * @title: ximagesrc * * This element captures your X Display and creates raw RGB video. It uses * the XDamage extension if available to only capture areas of the screen that @@ -27,12 +28,11 @@ * available to also capture your mouse pointer. By default it will fixate to * 25 frames per second. * - * <refsect2> - * <title>Example pipelines</title> + * ## Example pipelines * |[ * gst-launch-1.0 ximagesrc ! video/x-raw,framerate=5/1 ! videoconvert ! theoraenc ! oggmux ! filesink location=desktop.ogg * ]| Encodes your X display to an Ogg theora video at 5 frames per second. - * </refsect2> + * */ #ifdef HAVE_CONFIG_H |