docs: Port all docstring to gtk-doc markdown

ext/aalib/gstaasink.c

@@ -18,19 +18,19 @@
  */
 /**
  * SECTION:element-aasink
+ * @title: aasink
  * @see_also: #GstCACASink
  *
  * Displays video as b/w ascii art.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example pipelines
- * <title>Example pipelines</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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 `preroll-pixbuf` or `pixbuf` on the bus, which
- * Watching for element messages named <classname>&quot;preroll-pixbuf&quot;
- * </classname> or <classname>&quot;pixbuf&quot;</classname> 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>
+ * ## Message details
- * <title>Message details</title>
+ *
  * As mentioned above, this element will by default post element messages
- * containing structures named <classname>&quot;preroll-pixbuf&quot;
+ * containing structures named `preroll-pixbuf`
- * </classname> or <classname>&quot;pixbuf&quot;</classname> on the bus (this
+ * ` 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>
+ * * `pixbuf`: the #GdkPixbuf object
- *   <classname>&quot;pixbuf&quot;</classname>: the #GdkPixbuf object
+ * * `pixel-aspect-ratio`: the pixel aspect ratio (PAR) of the input image
- * </listitem>
+ *   (this field contains a #GstFraction); the
- * <listitem>
- *   <classname>&quot;pixel-aspect-ratio&quot;</classname>: 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>
+ * ## Example pipeline
- * <title>Example pipeline</title>
  * |[
  * gst-launch-1.0 -m -v videotestsrc num-buffers=1 ! gdkpixbufsink
  * ]| Process one single test image as pixbuf (note that the output you see will
  * be slightly misleading. The message structure does contain a valid pixbuf
  * object even if the structure string says &apos;(NULL)&apos;).
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H

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 *

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 audiotestsrc ! jackaudiosink
  * ]| Play a sine wave to using jack.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

ext/jack/gstjackaudiosink.h

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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
  * ]| Get audio input into gstreamer from jack.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

ext/jpeg/gstjpegdec.c

@@ -22,15 +22,15 @@
 
 /**
  * SECTION:element-jpegdec
+ * @title: jpegdec
  *
  * Decodes jpeg images.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

ext/jpeg/gstjpegenc.c

@@ -20,16 +20,16 @@
  */
 /**
  * SECTION:element-jpegenc
+ * @title: jpegenc
  *
  * Encodes jpeg images.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

ext/jpeg/gstsmokedec.c

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

ext/jpeg/gstsmokeenc.c

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

ext/libcaca/gstcacasink.c

@@ -18,12 +18,12 @@
  */
 /**
  * SECTION:element-cacasink
+ * @title: cacasink
  * @see_also: #GstAASink
  *
  * Displays video as color ascii art.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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.

ext/libpng/gstpngenc.c

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

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>
+ * ## Example pipelines
- * <title>Example pipelines</title>
  * |[
  * 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

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>
+ * ## Example pipelines
- * <title>Example pipelines</title>
  * |[
  * 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

ext/raw1394/gstdv1394src.c

@@ -21,16 +21,16 @@
  */
 /**
  * SECTION:element-dv1394src
+ * @title: dv1394src
  *
  * Read DV (digital video) data from firewire port.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

ext/raw1394/gsthdv1394src.c

@@ -18,18 +18,18 @@
  */
 /**
  * SECTION:element-hdv1394src
+ * @title: hdv1394src
  *
  * Read MPEG-TS data from firewire port.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

ext/shout2/gstshout2.c

@@ -21,18 +21,18 @@
 
 /**
  * SECTION:element-shout2send
+ * @title: shout2send
  *
  * shout2send pushes a media stream to an Icecast server
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example pipelines
- * <title>Example pipelines</title>
  * |[
  * 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

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>
+ * ## Example pipelines
- * <title>Example pipelines</title>
  * |[
  * gst-launch-1.0 audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
  * ]| Encode an Ogg/Speex file.
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example pipeline
- * <title>Example pipeline</title>
  * |[
  * 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

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>
+ * ## Example pipeline
- * <title>Example pipeline</title>
  * |[
  * 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

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>
+ * ## Example pipeline
- * <title>Example pipeline</title>
  * |[
  * 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

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>
+ * ## Example pipeline
- * <title>Example pipeline</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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>
+ *
  */
 
 /*

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

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

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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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"

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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
- * 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.
- * a cutoff frequency at a quarter of the sampling rate.
- * </note>
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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
- * 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.
- * a cutoff frequency at a quarter of the sampling rate.
- * </para></note>
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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 */

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>
+ * ## Example launch lines
- * <title>Example launch lines</title>
  * |[
  * 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

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>
+ * ## Example application
- * <title>Example application</title>
+ * <programlisting language="C">
- * <informalexample><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

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>
+ * ## Example application
- * <title>Example application</title>
+ * <programlisting language="C">
- * <informalexample><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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=song.ogg ! oggdemux ! vorbisdec ! audiokaraoke ! audioconvert ! alsasink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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",

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example pipelines
- * <title>Example pipelines</title>
+ *
- * <para>
  * |[
  * 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>
  */
 
 /*

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.aac ! aacparse ! faad ! audioresample ! audioconvert ! alsasink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioresample ! audioconvert ! autoaudiosink
  * ]|
- * </refsect2>
+ *
  */
 
 /* TODO:

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrdec ! audioresample ! audioconvert ! alsasink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.dts ! dcaparse ! dtsdec ! audioresample ! audioconvert ! autoaudiosink
  * ]|
- * </refsect2>
+ *
  */
 
 /* TODO:

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>
+ * ## Example pipelines
- * <title>Example pipelines</title>
  * |[
  * 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>
  *
  */
 

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.wavpack ! wavpackparse ! wavpackdec ! audioresample ! audioconvert ! autoaudiosink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

gst/auparse/gstauparse.c

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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 -v -m audiotestsrc ! audioconvert ! audioresample ! autoaudiosink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 -v -m autoaudiosrc ! audioconvert ! audioresample ! autoaudiosink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 -v -m videotestsrc ! autovideosink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 -v -m autovideosrc ! xvimagesink
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch lines
- * <title>Example launch lines</title>
+ * (write everything in one line, without the backslash characters)
- * <para>(write everything in one line, without the backslash characters)</para>
  * |[
  * 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

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>
+ * ## Example launch line
- * <para>
+ *
- * <programlisting>
+ * |[
  * 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

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>&quot;cutter&quot;</classname>.
+ * `cutter`.
- * The message's structure contains two fields:
- * <itemizedlist>
- * <listitem>
- *   <para>
- *   #GstClockTime
- *   <classname>&quot;timestamp&quot;</classname>:
- *   the timestamp of the buffer that triggered the message.
- *   </para>
- * </listitem>
- * <listitem>
- *   <para>
- *   gboolean
- *   <classname>&quot;above&quot;</classname>:
- *   %TRUE for begin of silence and %FALSE for end of silence.
- *   </para>
- * </listitem>
- * </itemizedlist>
  *
- * <refsect2>
+ * The message's structure contains two fields:
- * <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

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.

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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch lines
- * <title>Example launch lines</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

gst/debugutils/rndbuffersize.c

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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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>
+   * * tomsmocomp  Motion Adaptive: Motion Search
-   * <listitem>
+   * * greedyh Motion Adaptive: Advanced Detection
-   * <para>
+   * * greedyl Motion Adaptive: Simple Detection
-   * tomsmocomp
+   * * vfir Blur vertical
-   * Motion Adaptive: Motion Search
+   * * linear Linear interpolation
-   * </para>
+   * * linearblend Linear interpolation in time domain.
-   * </listitem>
+   *   Any motion causes significant ghosting, so this
-   * <listitem>
+   *   method should not be used.
-   * <para>
+   * * scalerbob Double lines
-   * greedyh
+   * * weave Weave. Bad quality, do not use.
-   * Motion Adaptive: Advanced Detection
+   * * weavetff Progressive: Top Field First.  Bad quality, do not use.
-   * </para>
+   * * weavebff Progressive: Bottom Field First.  Bad quality, do not use.
-   * </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>
    */
   g_object_class_install_property (gobject_class, PROP_METHOD,
       g_param_spec_enum ("method",

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"

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

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"

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

gst/effectv/gstshagadelic.c

@@ -24,15 +24,15 @@
 
 /**
  * SECTION:element-shagadelictv
+ * @title: shagadelictv
  *
  * Oh behave, ShagedelicTV makes images shagadelic!
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

gst/effectv/gststreak.c

@@ -30,15 +30,15 @@
 
 /**
  * SECTION:element-streaktv
+ * @title: streaktv
  *
  * StreakTV makes after images of moving objects.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

gst/effectv/gstwarp.c

@@ -35,15 +35,15 @@
 
 /**
  * SECTION:element-warptv
+ * @title: warptv
  *
  * WarpTV does realtime goo'ing of the video input.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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>
+ * ## Example code
- * <title>Example code</title>
  * |[
  * #include &lt;gst/gst.h&gt;
- * 
+ *
  * ...
  * typedef struct {
  *   gfloat freq;
  *   gfloat width;
  *   gfloat gain;
  * } GstEqualizerBandState;
- * 
+ *
  * ...
- * 
+ *
  *   GstElement *equalizer;
  *   GObject *band;
  *   gint i;
@@ -55,14 +54,14 @@
  *     {6000.0, 1000.0,   6.0},
  *     {3000.0,  120.0,   2.0}
  *   };
- * 
+ *
  * ...
- * 
+ *
  *   equalizer = gst_element_factory_make ("equalizer-nbands", "equalizer");
  *   g_object_set (G_OBJECT (equalizer), "num-bands", 5, NULL);
- * 
+ *
  * ...
- * 
+ *
  *   for (i = 0; i &lt; 5; i++) {
  *     band = gst_child_proxy_get_child_by_index (GST_CHILD_PROXY (equalizer), i);
  *     g_object_set (G_OBJECT (band), "freq", state[i].freq,
@@ -70,10 +69,10 @@
  * 	"gain", state[i].gain);
  *     g_object_unref (G_OBJECT (band));
  *   }
- * 
+ *
  * ...
  * ]|
- * </refsect2>
+ *
  */
 
 #ifdef HAVE_CONFIG_H

gst/flv/gstflvdemux.c

@@ -19,15 +19,15 @@
 
 /**
  * SECTION:element-flvdemux
+ * @title: flvdemux
  *
  * flvdemux demuxes an FLV file into the different contained streams.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

gst/flv/gstflvmux.c

@@ -23,15 +23,15 @@
 
 /**
  * SECTION:element-flvmux
+ * @title: flvmux
  *
  * flvmux muxes different streams into an FLV file.
  *
- * <refsect2>
+ * ## Example launch line
- * <title>Example launch line</title>
  * |[
  * 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

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
- * 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
- * resolver may call functions that take the object lock as well, and
+ * > the lock is not recursive.
- * the lock is not recursive.
- * </note>
  *
  * Returns: TRUE if the writer would be mapped to an id.
  */

gst/flx/gstflxdec.c

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

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);
 }
 

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