Update and add documentation for plugins with deps (ext).

Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered. Fix warnings that gtk-doc points out.

docs/plugins/Makefile.am

@@ -78,6 +78,7 @@ EXTRA_HFILES = \
 	$(top_srcdir)/ext/aalib/gstaasink.h \
 	$(top_srcdir)/ext/annodex/gstcmmldec.h \
 	$(top_srcdir)/ext/annodex/gstcmmlenc.h \
+	$(top_srcdir)/ext/cairo/gsttextoverlay.h \
 	$(top_srcdir)/ext/cairo/gsttimeoverlay.h \
 	$(top_srcdir)/ext/dv/gstdvdec.h \
 	$(top_srcdir)/ext/dv/gstdvdemux.h \
@@ -85,6 +86,10 @@ EXTRA_HFILES = \
 	$(top_srcdir)/ext/flac/gstflacdec.h \
 	$(top_srcdir)/ext/flac/gstflacenc.h \
 	$(top_srcdir)/ext/flac/gstflactag.h \
+	$(top_srcdir)/ext/gconf/gstgconfaudiosrc.h \
+	$(top_srcdir)/ext/gconf/gstgconfaudiosink.h \
+	$(top_srcdir)/ext/gconf/gstgconfvideosrc.h \
+	$(top_srcdir)/ext/gconf/gstgconfvideosink.h \
 	$(top_srcdir)/ext/gdk_pixbuf/gstgdkpixbufsink.h \
 	$(top_srcdir)/ext/hal/gsthalaudiosink.h \
 	$(top_srcdir)/ext/hal/gsthalaudiosrc.h \
@@ -103,6 +108,8 @@ EXTRA_HFILES = \
 	$(top_srcdir)/ext/pulse/pulsesink.h \
 	$(top_srcdir)/ext/pulse/pulsesrc.h \
 	$(top_srcdir)/ext/pulse/pulsemixer.h \
+	$(top_srcdir)/ext/speex/gstspeexenc.h \
+	$(top_srcdir)/ext/speex/gstspeexdec.h \
 	$(top_srcdir)/ext/wavpack/gstwavpackdec.h \
 	$(top_srcdir)/ext/wavpack/gstwavpackenc.h \
 	$(top_srcdir)/ext/wavpack/gstwavpackparse.h \

docs/plugins/gst-plugins-good-plugins-docs.sgml

@@ -40,6 +40,7 @@
     <xi:include href="xml/element-avimux.xml" />
     <xi:include href="xml/element-avisubtitle.xml" />
     <xi:include href="xml/element-cacasink.xml" />
+    <xi:include href="xml/element-cairotextoverlay.xml" />
     <xi:include href="xml/element-cairotimeoverlay.xml" />
     <xi:include href="xml/element-cmmldec.xml" />
     <xi:include href="xml/element-cmmlenc.xml" />
@@ -48,7 +49,6 @@
     <xi:include href="xml/element-directdrawsink.xml" />
     <xi:include href="xml/element-directsoundsink.xml" />
     <xi:include href="xml/element-dv1394src.xml" />
-    <xi:include href="xml/element-hdv1394src.xml" />
     <xi:include href="xml/element-dvdec.xml" />
     <xi:include href="xml/element-dvdemux.xml" />
     <xi:include href="xml/element-equalizer-10bands.xml" />
@@ -59,11 +59,16 @@
     <xi:include href="xml/element-flacenc.xml" />
     <xi:include href="xml/element-flactag.xml" />
     <xi:include href="xml/element-gamma.xml" />
+    <xi:include href="xml/element-gconfaudiosrc.xml" />
+    <xi:include href="xml/element-gconfaudiosink.xml" />
+    <xi:include href="xml/element-gconfvideosrc.xml" />
+    <xi:include href="xml/element-gconfvideosink.xml" />
     <xi:include href="xml/element-gdkpixbufsink.xml" />
     <xi:include href="xml/element-goom.xml" />
     <xi:include href="xml/element-goom2k1.xml" />
     <xi:include href="xml/element-halaudiosink.xml" />
     <xi:include href="xml/element-halaudiosrc.xml" />
+    <xi:include href="xml/element-hdv1394src.xml" />
     <xi:include href="xml/element-icydemux.xml" />
     <xi:include href="xml/element-id3demux.xml" />
     <xi:include href="xml/element-id3v2mux.xml" />
@@ -106,6 +111,8 @@
     <xi:include href="xml/element-smptealpha.xml" />
     <xi:include href="xml/element-souphttpsrc.xml" />
     <xi:include href="xml/element-spectrum.xml" />
+    <xi:include href="xml/element-speexenc.xml" />
+    <xi:include href="xml/element-speexdec.xml" />
     <xi:include href="xml/element-taginject.xml" />
     <xi:include href="xml/element-udpsrc.xml" />
     <xi:include href="xml/element-udpsink.xml" />

docs/plugins/gst-plugins-good-plugins-sections.txt

@@ -103,7 +103,6 @@ gst_audio_amplify_get_type
 GstAudioChebBand
 <SUBSECTION Standard>
 GstAudioChebBandClass
-GstAudioChebBandProcessFunc
 GST_AUDIO_CHEB_BAND
 GST_AUDIO_CHEB_BAND_CLASS
 GST_AUDIO_CHEB_BAND_GET_CLASS
@@ -119,7 +118,6 @@ gst_audio_cheb_band_get_type
 GstAudioChebLimit
 <SUBSECTION Standard>
 GstAudioChebLimitClass
-GstAudioChebLimitProcessFunc
 GST_AUDIO_CHEB_LIMIT
 GST_AUDIO_CHEB_LIMIT_CLASS
 GST_AUDIO_CHEB_LIMIT_GET_CLASS
@@ -243,7 +241,6 @@ gst_audio_panorama_get_type
 GstAudioWSincBand
 <SUBSECTION Standard>
 GstAudioWSincBandClass
-GstAudioWSincBandProcessFunc
 GST_AUDIO_WSINC_BAND
 GST_AUDIO_WSINC_BAND_CLASS
 GST_IS_AUDIO_WSINC_BAND
@@ -258,7 +255,6 @@ gst_audio_wsincband_get_type
 GstAudioWSincLimit
 <SUBSECTION Standard>
 GstAudioWSincLimitClass
-GstAudioWSincLimitProcessFunc
 GST_AUDIO_WSINC_LIMIT
 GST_AUDIO_WSINC_LIMIT_CLASS
 GST_IS_AUDIO_WSINC_LIMIT
@@ -396,6 +392,20 @@ GST_IS_CACASINK_CLASS
 gst_cacasink_get_type
 </SECTION>
 
+<SECTION>
+<FILE>element-cairotextoverlay</FILE>
+<TITLE>cairotextoverlay</TITLE>
+GstCairoTextOverlay
+<SUBSECTION Standard>
+GstCairoTextOverlayClass
+GST_TYPE_CAIRO_TEXT_OVERLAY
+GST_CAIRO_TEXT_OVERLAY
+GST_CAIRO_TEXT_OVERLAY_CLASS
+GST_IS_CAIRO_TEXT_OVERLAY
+GST_IS_CAIRO_TEXT_OVERLAY_CLASS
+gst_text_overlay_get_type
+</SECTION>
+
 <SECTION>
 <FILE>element-cairotimeoverlay</FILE>
 <TITLE>cairotimeoverlay</TITLE>
@@ -447,7 +457,6 @@ GstCutter
 <SUBSECTION Standard>
 GST_CUTTER
 GST_CUTTER_CLASS
-GST_CUTTER_GET_CLASS
 GST_IS_CUTTER
 GST_IS_CUTTER_CLASS
 GST_TYPE_CUTTER
@@ -697,6 +706,62 @@ GST_IS_GAMMA_CLASS
 gst_gamma_get_type
 </SECTION>
 
+<SECTION>
+<FILE>element-gconfaudiosrc</FILE>
+<TITLE>gconfaudiosrc</TITLE>
+GstGConfAudioSrc
+<SUBSECTION Standard>
+GstGConfAudioSrcClass
+GST_GCONF_AUDIO_SRC
+GST_IS_GCONF_AUDIO_SRC
+GST_TYPE_GCONF_AUDIO_SRC
+GST_GCONF_AUDIO_SRC_CLASS
+GST_IS_GCONF_AUDIO_SRC_CLASS
+gst_gconf_audio_src_get_type
+</SECTION>
+
+<SECTION>
+<FILE>element-gconfaudiosink</FILE>
+<TITLE>gconfaudiosink</TITLE>
+GstGConfAudioSink
+<SUBSECTION Standard>
+GstGConfAudioSinkClass
+GST_GCONF_AUDIO_SINK
+GST_IS_GCONF_AUDIO_SINK
+GST_TYPE_GCONF_AUDIO_SINK
+GST_GCONF_AUDIO_SINK_CLASS
+GST_IS_GCONF_AUDIO_SINK_CLASS
+gst_gconf_audio_sink_get_type
+</SECTION>
+
+<SECTION>
+<FILE>element-gconfvideosrc</FILE>
+<TITLE>gconfvideosrc</TITLE>
+GstGConfVideoSrc
+<SUBSECTION Standard>
+GstGConfVideoSrcClass
+GST_GCONF_VIDEO_SRC
+GST_IS_GCONF_VIDEO_SRC
+GST_TYPE_GCONF_VIDEO_SRC
+GST_GCONF_VIDEO_SRC_CLASS
+GST_IS_GCONF_VIDEO_SRC_CLASS
+gst_gconf_video_src_get_type
+</SECTION>
+
+<SECTION>
+<FILE>element-gconfvideosink</FILE>
+<TITLE>gconfvideosink</TITLE>
+GstGConfVideoSink
+<SUBSECTION Standard>
+GstGConfVideoSinkClass
+GST_GCONF_VIDEO_SINK
+GST_IS_GCONF_VIDEO_SINK
+GST_TYPE_GCONF_VIDEO_SINK
+GST_GCONF_VIDEO_SINK_CLASS
+GST_IS_GCONF_VIDEO_SINK_CLASS
+gst_gconf_video_sink_get_type
+</SECTION>
+
 <SECTION>
 <FILE>element-gdkpixbufsink</FILE>
 <TITLE>gdkpixbufsink</TITLE>
@@ -881,7 +946,6 @@ GST_MATROSKA_MUX
 GST_MATROSKA_MUX_CLASS
 GST_IS_MATROSKA_MUX
 GST_IS_MATROSKA_MUX_CLASS
-gst_matroska_mux_get_type
 gst_matroska_mux_plugin_init
 </SECTION>
 
@@ -897,7 +961,6 @@ GST_MATROSKA_DEMUX
 GST_MATROSKA_DEMUX_CLASS
 GST_IS_MATROSKA_DEMUX
 GST_IS_MATROSKA_DEMUX_CLASS
-gst_matroska_demux_get_type
 gst_matroska_demux_plugin_init
 </SECTION>
 
@@ -1271,7 +1334,7 @@ GST_IS_RTP_JPEG_PAY
 GST_TYPE_RTP_JPEG_PAY
 GST_RTP_JPEG_PAY_CLASS
 GST_IS_RTP_JPEG_PAY_CLASS
-gst_rtp_jpeg_pay_get_type
+gst_rtp_jpeg_pay_plugin_init
 </SECTION>
 
 <SECTION>
@@ -1375,8 +1438,6 @@ GST_IS_SMOKEENC_CLASS
 GstSpectrum
 <SUBSECTION Standard>
 GstSpectrumClass
-GstSpectrumFFTFreeFunc
-GstSpectrumProcessFunc
 GST_SPECTRUM
 GST_SPECTRUM_CLASS
 GST_IS_SPECTRUM
@@ -1385,6 +1446,34 @@ GST_TYPE_SPECTRUM
 gst_spectrum_get_type
 </SECTION>
 
+<SECTION>
+<FILE>element-speexdec</FILE>
+<TITLE>speexdec</TITLE>
+GstSpeexDec
+<SUBSECTION Standard>
+GstSpeexDecClass
+GST_TYPE_SPEEX_DEC
+GST_SPEEX_DEC
+GST_SPEEX_DEC_CLASS
+GST_IS_SPEEX_DEC
+GST_IS_SPEEX_DEC_CLASS
+gst_speex_dec_get_type
+</SECTION>
+
+<SECTION>
+<FILE>element-speexenc</FILE>
+<TITLE>speexenc</TITLE>
+GstSpeexEnc
+<SUBSECTION Standard>
+GstSpeexEncClass
+GST_TYPE_SPEEX_ENC
+GST_SPEEX_ENC
+GST_SPEEX_ENC_CLASS
+GST_IS_SPEEX_ENC
+GST_IS_SPEEX_ENC_CLASS
+gst_speex_enc_get_type
+</SECTION>
+
 <SECTION>
 <FILE>element-taginject</FILE>
 <TITLE>taginject</TITLE>
@@ -1446,12 +1535,6 @@ gst_udpsink_get_type
 <TITLE>videobox</TITLE>
 GstVideoBox
 <SUBSECTION Standard>
-GstVideoBoxClass
-GST_IS_VIDEO_BOX
-GST_IS_VIDEO_BOX_CLASS
-GST_VIDEO_BOX
-GST_VIDEO_BOX_CLASS
-GST_TYPE_VIDEO_BOX
 </SECTION>
 
 <SECTION>
@@ -1475,8 +1558,6 @@ GST_TYPE_VIDEO_CROP
 GstAspectRatioCrop
 <SUBSECTION Standard>
 GstAspectRatioCropClass                 
-AspectRatioCropPixelFormat
-GstAspectRatioCropImageDetails
 GST_IS_ASPECT_RATIO_CROP
 GST_IS_ASPECT_RATIO_CROP_CLASS
 GST_ASPECT_RATIO_CROP
@@ -1552,7 +1633,6 @@ GST_V4L2_MIN_BUFFERS
 GST_V4L2_MAX_SIZE
 GstV4l2BufferPool
 GstV4l2Buffer
-GstV4l2Src
 GstV4l2SrcClass
 GST_V4L2SRC
 GST_IS_V4L2SRC
@@ -1594,7 +1674,6 @@ GST_IS_WAVENC
 GST_TYPE_WAVENC
 GST_WAVENC_CLASS
 GST_IS_WAVENC_CLASS
-gst_wavenc_get_type
 </SECTION>
 
 <SECTION>

ext/aalib/gstaasink.c

@@ -20,23 +20,16 @@
  * SECTION:element-aasink
  * @see_also: #GstCACASink
  *
- * <refsect2>
- * <para>
  * Displays video as b/w ascii art.
- * </para>
+ *
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! aasink
- * </programlisting>
+ * ]| This pipeline renders a video to ascii art into a separate window.
- * This pipeline renders a video to ascii art into a separate window.
+ * |[
- * </para>
- * <para>
- * <programlisting>
  * gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! aasink driver=curses
- * </programlisting>
+ * ]| This pipeline renders a video to ascii art into the current terminal.
- * This pipeline renders a video to ascii art into the current terminal.
- * </para>
  * </refsect2>
  */
 

ext/annodex/gstcmmldec.c

@@ -25,17 +25,16 @@
  * SECTION:element-cmmldec
  * @see_also: cmmlenc, oggdemux
  *
- * <refsect2>
- * <para>
  * Cmmldec extracts a CMML document from a CMML bitstream.<ulink
  * url="http://www.annodex.net/TR/draft-pfeiffer-cmml-02.html">CMML</ulink> is
  * an XML markup language for time-continuous data maintained by the <ulink
  * url="http:/www.annodex.org/">Annodex Foundation</ulink>.
- * </para>
+ *
+ * <refsect2>
  * <title>Example pipeline</title>
- * <programlisting>
+ * |[
  * gst-launch -v filesrc location=annotated.ogg ! oggdemux ! cmmldec ! filesink location=annotations.cmml
- * </programlisting>
+ * ]|
  * </refsect2>
  */
 

ext/annodex/gstcmmlenc.c

@@ -25,16 +25,16 @@
  * SECTION:element-cmmlenc
  * @see_also: cmmldec, oggmux
  *
- * <refsect2>
+ * Cmmlenc encodes a CMML document into a CMML stream.  <ulink
- * <para> Cmmlenc encodes a CMML document into a CMML stream.  <ulink
  * url="http://www.annodex.net/TR/draft-pfeiffer-cmml-02.html">CMML</ulink> is
  * an XML markup language for time-continuous data maintained by the <ulink
  * url="http:/www.annodex.org/">Annodex Foundation</ulink>.
- * </para>
+ *
+ * <refsect2>
  * <title>Example pipeline</title>
- * <programlisting>
+ * |[
  * gst-launch -v filesrc location=annotations.cmml ! cmmlenc ! oggmux name=mux ! filesink location=annotated.ogg
- * </programlisting>
+ * ]|
  * </refsect2>
  */
 

ext/cairo/gsttextoverlay.c

@@ -17,6 +17,18 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
+/**
+ * SECTION:element-cairotextoverlay
+ *
+ * cairotextoverlay renders the text on top of the video frames.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch videotestsrc ! cairotextoverlay text="hello" ! autovideosink
+ * ]|
+ * </refsect2>
+ */
 
 #ifdef HAVE_CONFIG_H
 #include <config.h>

ext/cairo/gsttimeoverlay.c

@@ -21,17 +21,14 @@
 /**
  * SECTION:element-cairotimeoverlay
  *
- * <refsect2>
- * <para>
  * cairotimeoverlay renders the buffer timestamp for each frame on top of
  * the frame.
- * </para>
+ *
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
+ * |[
- * <programlisting>
+ * gst-launch videotestsrc ! cairotimeoverlay ! autovideosink
- * gst-launch -v -m videotestsrc ! cairotimeoverlay ! autovideosink
+ * ]|
- * </programlisting>
- * </para>
  * </refsect2>
  */
 

ext/dv/gstdvdec.c

@@ -21,25 +21,21 @@
 /**
  * SECTION:element-dvdec
  *
- * <refsect2>
- * <para>
  * 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.
- * </para>
+ *
- * <para>
+ * This element can perform simple frame dropping with the #GstDVDec:drop-factor
- * This element can perform simple frame dropping with the drop-factor
  * property. Setting this property to a value N > 1 will only decode every 
  * Nth frame.
- * </para>
+ *
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch filesrc location=test.dv ! dvdemux name=demux ! dvdec ! xvimagesink
- * </programlisting>
+ * ]| This pipeline decodes and renders the raw DV stream to a videosink.
- * This pipeline decodes and renders the raw DV stream to a videosink.
- * </para>
- * Last reviewed on 2006-02-28 (0.10.3)
  * </refsect2>
+ *
+ * Last reviewed on 2006-02-28 (0.10.3)
  */
 
 #ifdef HAVE_CONFIG_H

ext/dv/gstdvdemux.c

@@ -30,24 +30,20 @@
 /**
  * SECTION:element-dvdemux
  *
- * <refsect2>
- * <para>
  * 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.
- * </para>
+ *
- * <para>
+ * This element can operate in both push and pull mode depending on the
- * This element can operate in both push and pull mode depending on the capabilities
+ * capabilities of the upstream peer.
- * of the upstream peer.
+ *
- * </para>
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch filesrc location=test.dv ! dvdemux name=demux ! queue ! audioconvert ! alsasink demux. ! queue ! dvdec ! xvimagesink
- * </programlisting>
+ * ]| This pipeline decodes and renders the raw DV stream to an audio and a videosink.
- * This pipeline decodes and renders the raw DV stream to an audio and a videosink.
- * </para>
- * Last reviewed on 2006-02-27 (0.10.3)
  * </refsect2>
+ *
+ * Last reviewed on 2006-02-27 (0.10.3)
  */
 
 /* DV output has two modes, normal and wide. The resolution is the same in both

ext/esd/esdmon.c

@@ -20,9 +20,10 @@
  * Boston, MA 02111-1307, USA.
  */
 /**
- * SECTION:element-esdmod
+ * SECTION:element-esdmon
+ * @see_also: #GstAlsaSrc, #GstAutoAudioSrc
  *
- * This element outputs sound to an already-running Enlightened Sound Daemon
+ * This element records sound from an already-running Enlightened Sound Daemon
  * (ESound Daemon, esd). Note that a sound daemon will never be auto-spawned
  * through this element (regardless of the system configuration), since this
  * is actively prevented by the element. If you must use esd, you need to
@@ -38,6 +39,7 @@
  * ]| Record from audioinput
  * </refsect2>
  */
+
 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

ext/esd/esdsink.c

@@ -26,24 +26,20 @@
  * SECTION:element-esdsink
  * @see_also: #GstAlsaSink, #GstAutoAudioSink
  *
- * <refsect2>
- * <para>
  * This element outputs sound to an already-running Enlightened Sound Daemon
  * (ESound Daemon, esd). Note that a sound daemon will never be auto-spawned
  * through this element (regardless of the system configuration), since this
  * is actively prevented by the element. If you must use esd, you need to
  * make sure it is started automatically with your session or otherwise.
- * </para>
+ *
- * <para>
  * TODO: insert some comments about how sucky esd is and that all the cool
  * kids use pulseaudio or whatever these days.
- * </para>
+ *
- * <para>
+ * <refsect2>
- * Simple example pipeline that plays an Ogg/Vorbis file via esd:
+ * <title>Example launch line</title>
- * <programlisting>
+ * |[
  * gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! esdsink
- * </programlisting>
+ * ]| play an Ogg/Vorbis audio file via esd
- * </para>
  * </refsect2>
  */
 

ext/flac/gstflacdec.c

@@ -21,26 +21,20 @@
 
 /**
  * SECTION:element-flacdec
- * @seealso: flacenc
+ * @see_also: #GstFlacEnc
  *
- * <refsect2>
- * <para>
  * flacdec decodes FLAC streams.
  * <ulink url="http://flac.sourceforge.net/">FLAC</ulink>
  * is a Free Lossless Audio Codec.
- * </para>
+ *
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch filesrc location=media/small/dark.441-16-s.flac ! flacdec ! audioconvert ! audioresample ! autoaudiosink
- * </programlisting>
+ * ]|
- * </para>
+ * |[
- * <title>Another example launch line</title>
- * <para>
- * <programlisting>
  * gst-launch gnomevfssrc location=http://gstreamer.freedesktop.org/media/small/dark.441-16-s.flac ! flacdec ! audioconvert ! audioresample ! queue min-threshold-buffers=10 ! autoaudiosink
- * </programlisting>
+ * ]|
- * </para>
  * </refsect2>
  */
 

ext/flac/gstflacenc.c

@@ -16,6 +16,21 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
+/**
+ * SECTION:element-flacenc
+ * @see_also: #GstFlacDec
+ *
+ * flacenc encodes FLAC streams.
+ * <ulink url="http://flac.sourceforge.net/">FLAC</ulink>
+ * is a Free Lossless Audio Codec.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch audiotestsrc num-buffers=100 ! flacenc ! filesink location=beep.flac
+ * ]|
+ * </refsect2>
+ */
 
 /* TODO: - We currently don't handle discontinuities in the stream in a useful
  *         way and instead rely on the developer plugging in audiorate if

ext/gconf/gstgconfaudiosink.c

@@ -17,6 +17,19 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
+/**
+ * SECTION:element-gconfaudiosink
+ *
+ * This element outputs sound to the audiosink that has been configured in
+ * GConf by the user.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! gconfaudiosink
+ * ]| Play on configured audiosink
+ * </refsect2>
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/gconf/gstgconfaudiosrc.c

@@ -17,6 +17,20 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
+/**
+ * SECTION:element-gconfaudiosrc
+ * @see_also: #GstAlsaSrc, #GstAutoAudioSrc
+ *
+ * This element records sound from the audiosink that has been configured in
+ * GConf by the user.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch gconfaudiosrc ! audioconvert ! wavenc ! filesink location=record.wav
+ * ]| Record from configured audioinput
+ * </refsect2>
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/gconf/gstgconfvideosink.c

@@ -16,6 +16,19 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
+/**
+ * SECTION:element-gconfvideosink
+ *
+ * This element outputs video to the videosink that has been configured in
+ * GConf by the user.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch filesrc location=foo.ogg ! decodebin ! ffmpegcolorspace ! gconfvideosink
+ * ]| Play on configured videosink
+ * </refsect2>
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/gconf/gstgconfvideosrc.c

@@ -17,6 +17,20 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
+/**
+ * SECTION:element-gconfvideosrc
+ * @see_also: #GstAlsaSrc, #GstAutoVideoSrc
+ *
+ * This element records video from the videosink that has been configured in
+ * GConf by the user.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch gconfvideosrc ! theoraenc ! oggmux ! filesink location=record.ogg
+ * ]| Record from configured videoinput
+ * </refsect2>
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/gdk_pixbuf/gstgdkpixbufsink.c

@@ -19,20 +19,14 @@
 
 /**
  * SECTION:element-gdkpixbufsink
- * @short_description: video sink that converts RGB images to GdkPixbufs.
+ * @Since: 0.10.8
- * @see_also:
  *
- * <refsect2>
- * <para>
  * This sink element takes RGB or RGBA images as input and wraps them into
- * GdkPixbuf objects, for easy saving to file via the
+ * #GdkPixbuf objects, for easy saving to file via the
- * <ulink url="http://library.gnome.org/devel/gdk-pixbuf/unstable/index.html">
+ * GdkPixbuf library API or displaying in Gtk+ applications (e.g. using
- * GdkPixbuf library API</ulink> or displaying in Gtk+ applications (e.g. using
+ * the #GtkImage widget).
- * the <ulink url="http://library.gnome.org/devel/gtk/unstable/GtkImage.html">
+ *
- * GtkImage widget</ulink>).
+ * There are two ways to use this element and obtain the #GdkPixbuf objects
- * </para>
- * <para>
- * There are two ways to use this element and obtain the GdkPixbuf objects
  * created:
  * <itemizedlist>
  * <listitem>
@@ -43,34 +37,30 @@
  * contained in them.
  * </listitem>
  * <listitem>
- * Retrieving the current pixbuf via the <classname>&quot;last-pixbuf&quot;
+ * Retrieving the current pixbuf via the #GstGdkPixbufSink:last-pixbuf property
- * </classname> property when needed.
+ * when needed.
  * </listitem>
  * </itemizedlist>
- * </para>
+ *
- * <para>
+ * The primary purpose of this element is to abstract away the #GstBuffer to
- * 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
- * GdkPixbuf conversion. Other than that it's very similar to the fakesink
  * element.
- * </para>
+ *
- * <para>
  * This element is meant for easy no-hassle video snapshotting. It is not
  * suitable for video playback or video display at high framerates. Use
  * ximagesink, xvimagesink or some other suitable video sink in connection
- * with the GstXOverlay interface instead if you want to do video playback.
+ * with the #GstXOverlay interface instead if you want to do video playback.
- * </para>
+ *
+ * <refsect2>
  * <title>Message details</title>
- * <para>
  * As mentioned above, this element will by default post element messages
  * containing structures named <classname>&quot;preroll-pixbuf&quot;
  * </classname> or <classname>&quot;pixbuf&quot;</classname> on the bus (this
- * can be disabled by setting the
+ * can be disabled by setting the #GstGdkPixbufSink:send-messages property
- * <link linkend="GstGdkPixbufSink--send-messages">&quot;send-messages&quot;
+ * to #FALSE though). The element message structure has the following fields:
- * property</link> to #FALSE though). The element message&apos;s structure
- * will have the following fields:
  * <itemizedlist>
  * <listitem>
- *   <classname>&quot;pixbuf&quot;</classname>: the GdkPixbuf object
+ *   <classname>&quot;pixbuf&quot;</classname>: the #GdkPixbuf object
  * </listitem>
  * <listitem>
  *   <classname>&quot;pixel-aspect-ratio&quot;</classname>: the pixel aspect
@@ -78,30 +68,25 @@
  *   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
- *   can easily be done using the
+ *   can easily be done using gdk_pixbuf_scale() (the reason this is not done
- *   <ulink url="http://library.gnome.org/devel/gdk-pixbuf/unstable/index.html">
+ *   automatically is that the application will often scale the image anyway
- *   GdkPixbuf library API</ulink> (the reason this is not done automatically
+ *   according to the size of the output window, in which case it is much more
- *   is that the application will often scale the image anyway according to the
+ *   efficient to only scale once rather than twice). You can put a videoscale
- *   size of the output window, in which case it is much more efficient to only
+ *   element and a capsfilter element with
- *   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
  *   in front of this element to make sure the pixbufs always have a 1:1 PAR.
  * </listitem>
  * </itemizedlist>
- * </para>
- * <title>Example pipeline</title>
- * <para>
- * <programlisting>
- * gst-launch -m -v videotestsrc num-buffers=1 ! gdkpixbufsink
- * </programlisting>
- * 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;).
- * </para>
  * </refsect2>
  *
- * Since: 0.10.8
+ * <refsect2>
+ * <title>Example pipeline</title>
+ * |[
+ * gst-launch -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/hal/gsthalaudiosink.c

@@ -21,27 +21,21 @@
 /**
  * SECTION:element-halaudiosink
  *
- * <refsect2>
- * <para>
  * HalAudioSink allows access to output of sound devices by specifying the
  * corresponding persistent Unique Device Id (UDI) from the Hardware Abstraction
- * Layer (HAL) in the <link linkend="GstHalAudioSrc--udi">udi</link> property.
+ * Layer (HAL) in the #GstHalAudioSink:udi property.
  * It currently always embeds alsasink or osssink as HAL doesn't support other
  * sound systems yet. You can also specify the UDI of a device that has ALSA or
  * OSS subdevices. If both are present ALSA is preferred.
- * </para>
+ *
+ * <refsect2>
  * <title>Examples</title>
- * <para>
+ * |[
- * To list the UDIs of all your ALSA output devices :
- * <programlisting>
  * hal-find-by-property --key alsa.type --string playback
- * </programlisting>
+ * ]| list the UDIs of all your ALSA output devices
- * Here is a pipeline to test your sound output :
+ * |[
- * <programlisting>
  * gst-launch -v audiotestsrc ! halaudiosink udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_playback_0
- * </programlisting>
+ * ]| test your soundcard by playing a test signal on the specified sound device.
- * This pipeline produces a test signal on the specified sound device.
- * </para>
  * </refsect2>
  */
 

ext/hal/gsthalaudiosrc.c

@@ -22,28 +22,22 @@
 /**
  * SECTION:element-halaudiosrc
  *
- * <refsect2>
- * <para>
  * HalAudioSrc allows access to input of sound devices by specifying the
  * corresponding persistent Unique Device Id (UDI) from the Hardware Abstraction
- * Layer (HAL) in the <link linkend="GstHalAudioSrc--udi">udi</link> property.
+ * Layer (HAL) in the #GstHalAudioSrc:udi property.
  * It currently always embeds alsasrc or osssrc as HAL doesn't support other
  * sound systems yet. You can also specify the UDI of a device that has ALSA or
  * OSS subdevices. If both are present ALSA is preferred.
- * </para>
+ *
+ * <refsect2>
  * <title>Examples</title>
- * <para>
+ * |[
- * To list the UDIs of all your ALSA input devices :
- * <programlisting>
  * hal-find-by-property --key alsa.type --string capture
- * </programlisting>
+ * ]| list the UDIs of all your ALSA input devices
- * Here is a pipeline to test your sound input :
+ * |[
- * <programlisting>
  * gst-launch -v halaudiosrc udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_capture_0 ! autoaudiosink
- * </programlisting>
+ * ]| You should now hear yourself with a small delay if you have a microphone
- * You should now hear yourself with a small delay if you have a microphone
  * connected to the specified sound device.
- * </para>
  * </refsect2>
  */
 

ext/hal/hal.c

@@ -40,7 +40,7 @@ GST_DEBUG_CATEGORY_EXTERN (hal_debug);
 #define LIBHAL_FREE_DBUS_ERROR(e) dbus_error_free (e)
 #endif
 
-/**
+/*
  * gst_hal_get_alsa_element:
  * @ctx: a #LibHalContext which should be used for querying HAL.
  * @udi: a #gchar corresponding to the UDI you want to get.
@@ -130,7 +130,7 @@ gst_hal_get_alsa_element (LibHalContext * ctx, const gchar * udi,
   return string;
 }
 
-/**
+/*
  * gst_hal_get_oss_element:
  * @ctx: a #LibHalContext which should be used for querying HAL.
  * @udi: a #gchar corresponding to the UDI you want to get.
@@ -203,7 +203,7 @@ gst_hal_get_oss_element (LibHalContext * ctx, const gchar * udi,
   return string;
 }
 
-/**
+/*
  * gst_hal_get_string:
  * @udi: a #gchar corresponding to the UDI you want to get.
  * @device_type: a #GstHalDeviceType specifying the wanted device type.

ext/jpeg/gstjpegenc.c

@@ -16,7 +16,11 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
-
+/**
+ * SECTION:element-jpegenc
+ *
+ * Encodes jpeg images.
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/jpeg/gstsmokedec.c

@@ -17,6 +17,11 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/**
+ * SECTION:element-smokedec
+ *
+ * Decodes images in smoke format.
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/jpeg/gstsmokeenc.c

@@ -16,7 +16,11 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
-
+/**
+ * SECTION:element-smokeenc
+ *
+ * Encodes images in smoke format.
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/libcaca/gstcacasink.c

@@ -20,24 +20,17 @@
  * SECTION:element-cacasink
  * @see_also: #GstAASink
  *
- * <refsect2>
- * <para>
  * Displays video as color ascii art.
- * </para>
+ *
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
+ * |[
- * <programlisting>
  * CACA_GEOMETRY=160x60 CACA_FONT=5x7 gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! cacasink
- * </programlisting>
+ * ]| This pipeline renders a video to ascii art into a separate window using a
- * This pipeline renders a video to ascii art into a separate window using a
  * small font and specifying the ascii resolution.
- * </para>
+ * |[
- * <para>
- * <programlisting>
  * CACA_DRIVER=ncurses gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! cacasink
- * </programlisting>
+ * ]| This pipeline renders a video to ascii art into the current terminal.
- * This pipeline renders a video to ascii art into the current terminal.
- * </para>
  * </refsect2>
  */
 
@@ -50,6 +43,17 @@
 
 #include "gstcacasink.h"
 
+#define GST_CACA_DEFAULT_SCREEN_WIDTH 80
+#define GST_CACA_DEFAULT_SCREEN_HEIGHT 25
+#define GST_CACA_DEFAULT_BPP 24
+#define GST_CACA_DEFAULT_RED_MASK GST_VIDEO_BYTE1_MASK_32_INT
+#define GST_CACA_DEFAULT_GREEN_MASK GST_VIDEO_BYTE2_MASK_32_INT
+#define GST_CACA_DEFAULT_BLUE_MASK GST_VIDEO_BYTE3_MASK_32_INT
+
+//#define GST_CACA_DEFAULT_RED_MASK R_MASK_32_REVERSE_INT
+//#define GST_CACA_DEFAULT_GREEN_MASK G_MASK_32_REVERSE_INT
+//#define GST_CACA_DEFAULT_BLUE_MASK B_MASK_32_REVERSE_INT
+
 /* elementfactory information */
 static const GstElementDetails gst_cacasink_details =
 GST_ELEMENT_DETAILS ("A colored ASCII art video sink",

ext/libcaca/gstcacasink.h

@@ -34,17 +34,6 @@
 extern "C" {
 #endif /* __cplusplus */
 
-#define GST_CACA_DEFAULT_SCREEN_WIDTH 80
-#define GST_CACA_DEFAULT_SCREEN_HEIGHT 25
-#define GST_CACA_DEFAULT_BPP 24
-#define GST_CACA_DEFAULT_RED_MASK GST_VIDEO_BYTE1_MASK_32_INT
-#define GST_CACA_DEFAULT_GREEN_MASK GST_VIDEO_BYTE2_MASK_32_INT
-#define GST_CACA_DEFAULT_BLUE_MASK GST_VIDEO_BYTE3_MASK_32_INT
-
-//#define GST_CACA_DEFAULT_RED_MASK R_MASK_32_REVERSE_INT
-//#define GST_CACA_DEFAULT_GREEN_MASK G_MASK_32_REVERSE_INT
-//#define GST_CACA_DEFAULT_BLUE_MASK B_MASK_32_REVERSE_INT
-
 #define GST_TYPE_CACASINK \
   (gst_cacasink_get_type())
 #define GST_CACASINK(obj) \

ext/libpng/gstpngdec.c

@@ -12,7 +12,6 @@
  * Boston, MA 02111-1307, USA.
  *
  */
-
 /**
  * SECTION:element-pngdec
  *

ext/libpng/gstpngenc.c

@@ -15,6 +15,11 @@
  * Boston, MA 02111-1307, USA.
  *
  */
+/**
+ * SECTION:element-pngenc
+ *
+ * Encodes png images.
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/pulse/pulsemixer.c

@@ -21,16 +21,14 @@
 
 /**
  * SECTION:element-pulsemixer
- * @short_description: Element to control sound input and output levels for the PulseAudio sound server
  * @see_also: pulsesrc, pulsesink
  *
- * <refsect2>
- * <para>
  * This element lets you adjust sound input and output levels for the
  * PulseAudio sound server. It supports the GstMixer interface, which can be
  * used to obtain a list of available mixer tracks. Set the mixer element to
  * READY state before using the GstMixer interface on it.
- * </para>
+ *
+ * <refsect2>
  * <title>Example pipelines</title>
  * <para>
  * pulsemixer can't be used in a sensible way in gst-launch.

ext/pulse/pulsesink.c

@@ -21,28 +21,20 @@
 
 /**
  * SECTION:element-pulsesink
- * @short_description: Output audio to a PulseAudio sound server
  * @see_also: pulsesrc, pulsemixer
  *
- * <refsect2>
+ * This element outputs audio to a
- * <para>
+ * <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
- * This element outputs audio to a PulseAudio sound server.
- * </para>
- * <title>Example pipelines</title>
- * <para>
- * <programlisting>
- * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
- * </programlisting>
- * Play an Ogg/Vorbis file.
- * </para>
- * <para>
- * <programlisting>
- * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
- * </programlisting>
- * Play a 440Hz sine wave.
- * </para>
- * </refsect2>
  *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! pulsesink
+ * ]| Play an Ogg/Vorbis file.
+ * |[
+ * gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
+ * ]| Play a 440Hz sine wave.
+ * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H

ext/pulse/pulsesrc.c

@@ -21,20 +21,16 @@
 
 /**
  * SECTION:element-pulsesrc
- * @short_description: Capture audio from a PulseAudio sound server
  * @see_also: pulsesink, pulsemixer
  *
+ * This element captures audio from a
+ * <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
+ *
  * <refsect2>
- * <para>
- * This element captures audio from a PulseAudio sound server.
- * </para>
  * <title>Example pipelines</title>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch -v pulsesrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
- * </programlisting>
+ * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
- * Record from a sound card using ALSA and encode to Ogg/Vorbis.
- * </para>
  * </refsect2>
  */
 

ext/raw1394/gstdv1394src.c

@@ -22,18 +22,14 @@
 /**
  * SECTION:element-dv1394src
  *
- * <refsect2>
- * <para>
  * Read DV (digital video) data from firewire port.
- * </para>
+ *
+ * <refsect2>
  * <title>Example launch line</title>
- * <para>
+ * |[
- * <programlisting>
+ * gst-launch dv1394src ! queue ! dvdemux name=d ! queue ! dvdec ! xvimagesink d. ! queue ! alsasink
- * gst-launch dv1394src ! dvdemux name=d ! queue ! dvdec ! xvimagesink d. ! queue ! alsasink
+ * ]| This pipeline captures from the firewire port and displays it (might need
- * </programlisting>
- * This pipeline captures from the firewire port and displays it (might need
  * format converters for audio/video).
- * </para>
  * </refsect2>
  */
 

ext/raw1394/gsthdv1394src.c

@@ -16,6 +16,21 @@
  * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
  * Boston, MA 02111-1307, USA.
  */
+/**
+ * SECTION:element-hdv1394src
+ *
+ * Read MPEG-TS data from firewire port.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch hdv1394src ! queue ! decodebin name=d ! queue ! xvimagesink d. ! queue ! alsasink
+ * ]| captures from the firewire port and plays the streams.
+ * |[
+ * gst-launch hdv1394src ! queue ! filesink location=mydump.ts
+ * ]| capture to a disk file
+ * </refsect2>
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"
@@ -776,6 +791,7 @@ gst_hdv1394src_uri_get_type (void)
 {
   return GST_URI_SRC;
 }
+
 static gchar **
 gst_hdv1394src_uri_get_protocols (void)
 {
@@ -783,6 +799,7 @@ gst_hdv1394src_uri_get_protocols (void)
 
   return protocols;
 }
+
 static const gchar *
 gst_hdv1394src_uri_get_uri (GstURIHandler * handler)
 {

ext/soup/gstsouphttpsrc.c

@@ -14,65 +14,48 @@
 
 /**
  * SECTION:element-souphttpsrc
- * @short_description: Read from an HTTP/HTTPS/WebDAV/Icecast/Shoutcast
- * location.
  *
- * <refsect2>
- * <para>
  * This plugin reads data from a remote location specified by a URI.
  * Supported protocols are 'http', 'https'.
- * </para>
+ * 
- * <para>
  * An HTTP proxy must be specified by its URL.
  * If the "http_proxy" environment variable is set, its value is used.
  * If built with libsoup's GNOME integration features, the GNOME proxy
  * configuration will be used, or failing that, proxy autodetection.
- * The element-souphttpsrc::proxy property can be used to override the
+ * The #GstSoupHTTPSrc:proxy property can be used to override the default.
- * default.
+ *
- * </para>
+ * In case the #GstSoupHTTPSrc:iradio-mode property is set and the location is
- * <para>
+ * an HTTP resource, souphttpsrc will send special Icecast HTTP headers to the
- * In case the element-souphttpsrc::iradio-mode property is set and the
+ * server to request additional Icecast meta-information.
- * location is an HTTP resource, souphttpsrc will send special Icecast HTTP
+ * If the server is not an Icecast server, it will behave as if the
- * headers to the server to request additional Icecast meta-information. If
+ * #GstSoupHTTPSrc:iradio-mode property were not set. If it is, souphttpsrc will
- * the server is not an Icecast server, it will behave as if the
+ * output data with a media type of application/x-icy, in which case you will
- * element-souphttpsrc::iradio-mode property were not set. If it is,
+ * need to use the #ICYDemux element as follow-up element to extract the Icecast
- * souphttpsrc will output data with a media type of application/x-icy,
+ * metadata and to determine the underlying media type.
- * in which case you will need to use the #ICYDemux element as follow-up
+ *
- * element to extract the Icecast metadata and to determine the underlying
+ * <refsect2>
- * media type.
+ * <title>Example launch line</title>
- * </para>
+ * |[
- * <para>
- * Example pipeline:
- * <programlisting>
  * gst-launch -v souphttpsrc location=https://some.server.org/index.html
  *     ! filesink location=/home/joe/server.html
- * </programlisting>
+ * ]| The above pipeline reads a web page from a server using the HTTPS protocol
- * The above pipeline reads a web page from a server using the HTTPS protocol
  * and writes it to a local file.
- * </para>
+ * |[
- * <para>
- * Another example pipeline:
- * <programlisting>
  * gst-launch -v souphttpsrc user-agent="FooPlayer 0.99 beta"
  *     automatic-redirect=false proxy=http://proxy.intranet.local:8080
  *     location=http://music.foobar.com/demo.mp3 ! mad ! audioconvert
  *     ! audioresample ! alsasink
- * </programlisting>
+ * ]| The above pipeline will read and decode and play an mp3 file from a
- * The above pipeline will read and decode and play an mp3 file from a
  * web server using the HTTP protocol. If the server sends redirects,
  * the request fails instead of following the redirect. The specified
  * HTTP proxy server is used. The User-Agent HTTP request header
  * is set to a custom string instead of "GStreamer souphttpsrc."
- * </para>
+ * |[
- * <para>
- * Yet another example pipeline:
- * <programlisting>
  * gst-launch -v souphttpsrc location=http://10.11.12.13/mjpeg
  *     do-timestamp=true ! multipartdemux
  *     ! image/jpeg,width=640,height=480 ! matroskamux
  *     ! filesink location=mjpeg.mkv
- * </programlisting>
+ * ]| The above pipeline reads a motion JPEG stream from an IP camera
- * The above pipeline reads a motion JPEG stream from an IP camera
  * using the HTTP protocol, encoded as mime/multipart image/jpeg
  * parts, and writes a Matroska motion JPEG file. The width and
  * height properties are set in the caps to provide the Matroska
@@ -81,9 +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.
- * </para>
  * </refsect2>
- *
  */
 
 #ifdef HAVE_CONFIG_H

ext/speex/gstspeexdec.c

@@ -20,23 +20,19 @@
 
 /**
  * SECTION:element-speexdec
- * @short_description: a decoder that decodes Speex to raw audio
  * @see_also: speexenc, oggdemux
  *
- * <refsect2>
- * <para>
  * This element decodes a Speex stream to raw integer audio.
  * <ulink url="http://www.speex.org/">Speex</ulink> is a royalty-free
  * audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
  * Foundation</ulink>.
- * </para>
+ *
+ * <refsect2>
  * <title>Example pipelines</title>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch -v filesrc location=speex.ogg ! oggdemux ! speexdec ! audioconvert ! audioresample ! alsasink
- * </programlisting>
+ * ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the
- * Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the documentation of speexenc.
+ * documentation of speexenc.
- * </para>
  * </refsect2>
  *
  * Last reviewed on 2006-04-05 (0.10.2)

ext/speex/gstspeexenc.c

@@ -17,6 +17,22 @@
  * Boston, MA 02111-1307, USA.
  */
 
+/**
+ * SECTION:element-speexenc
+ * @see_also: speexdec, oggmux
+ *
+ * This element encodes audio as a Speex stream.
+ * <ulink url="http://www.speex.org/">Speex</ulink> is a royalty-free
+ * audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
+ * Foundation</ulink>.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
+ * ]| Encode an Ogg/Speex file.
+ * </refsect2>
+ */
 
 #ifdef HAVE_CONFIG_H
 #include "config.h"

ext/taglib/gstapev2mux.cc

@@ -23,28 +23,23 @@
  * SECTION:element-apev2mux
  * @see_also: #GstTagSetter
  *
- * <refsect2>
- * <para>
  * This element adds APEv2 tags to the beginning of a stream using the taglib
  * library.
- * </para>
+ *
- * <para>
  * Applications can set the tags to write using the #GstTagSetter interface.
  * Tags sent by upstream elements will be picked up automatically (and merged
  * according to the merge mode set via the tag setter interface).
- * </para>
+ *
- * <para>
+ * <refsect2>
- * Here is a simple pipeline that transcodes a file from Ogg/Vorbis to mp3
+ * <title>Example pipelines</title>
- * format with an APEv2 that contains the same as the the Ogg/Vorbis file:
+ * |[
- * <programlisting>
  * gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! apev2mux ! filesink location=foo.mp3
- * </programlisting>
+ * ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
- * Make sure the Ogg/Vorbis file actually has comments to preserve.
+ * APEv2 that contains the same as the the Ogg/Vorbis file. Make sure the
- * You can verify the tags were written using:
+ * Ogg/Vorbis file actually has comments to preserve.
- * <programlisting>
+ * |[
  * gst-launch -m filesrc location=foo.mp3 ! apedemux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist
- * </programlisting>
+ * ]| Verify that tags have been written.
- * </para>
  * </refsect2>
  */
 

ext/taglib/gstid3v2mux.cc

@@ -22,30 +22,25 @@
  * SECTION:element-id3v2mux
  * @see_also: #GstID3Demux, #GstTagSetter
  *
- * <refsect2>
- * <para>
  * This element adds ID3v2 tags to the beginning of a stream using the taglib
  * library. More precisely, the tags written are ID3 version 2.4.0 tags (which
  * means in practice that some hardware players or outdated programs might not
  * be able to read them properly).
- * </para>
+ *
- * <para>
  * Applications can set the tags to write using the #GstTagSetter interface.
  * Tags sent by upstream elements will be picked up automatically (and merged
  * according to the merge mode set via the tag setter interface).
- * </para>
+ *
- * <para>
+ * <refsect2>
- * Here is a simple pipeline that transcodes a file from Ogg/Vorbis to mp3
+ * <title>Example pipelines</title>
- * format with an ID3v2 that contains the same as the the Ogg/Vorbis file:
+ * |[
- * <programlisting>
  * gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! id3v2mux ! filesink location=foo.mp3
- * </programlisting>
+ * ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
- * Make sure the Ogg/Vorbis file actually has comments to preserve.
+ * ID3v2 that contains the same as the the Ogg/Vorbis file. Make sure the
- * You can verify the tags were written using:
+ * Ogg/Vorbis file actually has comments to preserve.
- * <programlisting>
+ * |[
  * gst-launch -m filesrc location=foo.mp3 ! id3demux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist
- * </programlisting>
+ * ]| Verify that tags have been written.
- * </para>
  * </refsect2>
  */
 

ext/wavpack/gstwavpackdec.c

@@ -24,19 +24,17 @@
 /**
  * SECTION:element-wavpackdec
  *
- * <refsect2>
  * 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>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! audioconvert ! audioresample ! autoaudiosink
- * </programlisting>
+ * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers and
- * This pipeline decodes the Wavpack file test.wv into raw audio buffers and
  * tries to play it back using an automatically found audio sink.
- * </para>
  * </refsect2>
  */
 

ext/wavpack/gstwavpackenc.c

@@ -22,32 +22,24 @@
 /**
  * SECTION:element-wavpackenc
  *
- * <refsect2>
  * 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>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch audiotestsrc num-buffers=500 ! audioconvert ! wavpackenc ! filesink location=sinewave.wv
- * </programlisting>
+ * ]| This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
- * This pipeline encodes audio from audiotestsrc into a Wavpack file. The audioconvert element is needed
  * as the Wavpack encoder only accepts input with 32 bit width (and every depth between 1 and 32 bits).
- * </para>
+ * |[
- * <para>
- * <programlisting>
  * gst-launch cdda://1 ! audioconvert ! wavpackenc ! filesink location=track1.wv
- * </programlisting>
+ * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
- * This pipeline encodes audio from an audio CD into a Wavpack file using
  * lossless encoding (the file output will be fairly large).
- * </para>
+ * |[
- * <para>
- * <programlisting>
  * gst-launch cdda://1 ! audioconvert ! wavpackenc bitrate=128000 ! filesink location=track1.wv
- * </programlisting>
+ * ]| This pipeline encodes audio from an audio CD into a Wavpack file using
- * 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).
- * </para>
  * </refsect2>
  */
 

ext/wavpack/gstwavpackparse.c

@@ -24,19 +24,17 @@
 /**
  * SECTION:element-wavpackparse
  *
- * <refsect2>
  * WavpackParse takes raw, unframed Wavpack streams and splits them into
  * single Wavpack chunks with information like bit depth and the position
  * in the 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>
- * <para>
+ * |[
- * <programlisting>
  * gst-launch filesrc location=test.wv ! wavpackparse ! wavpackdec ! fakesink
- * </programlisting>
+ * ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers.
- * This pipeline decodes the Wavpack file test.wv into raw audio buffers.
- * </para>
  * </refsect2>
  */
 

gst/matroska/matroska-mux.h

@@ -137,8 +137,6 @@ typedef struct _GstMatroskaMuxClass {
   GstElementClass parent;
 } GstMatroskaMuxClass;
 
-GType    gst_matroska_mux_get_type    (void);
-
 gboolean gst_matroska_mux_plugin_init (GstPlugin *plugin);
 
 G_END_DECLS

gst/udp/gstudpsrc.c

@@ -29,7 +29,7 @@
  * #GstUDPSrc:port property to 0. After setting the udpsrc to PAUSED, the
  * allocated port can be obtained by reading the port property.
  *
- * udpsrc can read from multicast groups by setting the #GstUDPSrc:multicast_group
+ * udpsrc can read from multicast groups by setting the #GstUDPSrc:multicast-group
  * property to the IP address of the multicast group.
  *
  * Alternatively one can provide a custom socket to udpsrc with the #GstUDPSrc:sockfd