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

Browse source 
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.
This commit is contained in:
Stefan Kost 2009-01-28 17:46:06 +02:00
parent 1f32369451
commit 9cf73bdd8f
44 changed files with 453 additions and 355 deletions
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7
docs/plugins/Makefile.am
View file

@ -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 \

9
docs/plugins/gst-plugins-good-plugins-docs.sgml
View file

@ -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" />

121
docs/plugins/gst-plugins-good-plugins-sections.txt
View file

@ -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>
@ -1474,9 +1557,7 @@ GST_TYPE_VIDEO_CROP
<TITLE>aspectratiocrop</TITLE>
GstAspectRatioCrop
<SUBSECTION Standard>
GstAspectRatioCropClass
AspectRatioCropPixelFormat
GstAspectRatioCropImageDetails
GstAspectRatioCropClass
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>

19
ext/aalib/gstaasink.c
View file

@ -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.
* </para>
* <para>
* <programlisting>
* ]| This pipeline renders a video to ascii art into a separate window.
* |[
* gst-launch filesrc location=test.avi ! decodebin ! ffmpegcolorspace ! aasink driver=curses
* </programlisting>
* This pipeline renders a video to ascii art into the current terminal.
* </para>
* ]| This pipeline renders a video to ascii art into the current terminal.
* </refsect2>
*/

9
ext/annodex/gstcmmldec.c
View file

@ -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>
*/

10
ext/annodex/gstcmmlenc.c
View file

@ -25,16 +25,16 @@
* SECTION:element-cmmlenc
* @see_also: cmmldec, oggmux
*
* <refsect2>
* <para> Cmmlenc encodes a CMML document into a CMML stream. <ulink
* 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>
*/

12
ext/cairo/gsttextoverlay.c
View file

@ -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>

13
ext/cairo/gsttimeoverlay.c
View file

@ -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 -v -m videotestsrc ! cairotimeoverlay ! autovideosink
* </programlisting>
* </para>
* |[
* gst-launch videotestsrc ! cairotimeoverlay ! autovideosink
* ]|
* </refsect2>
*/

20
ext/dv/gstdvdec.c
View file

@ -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 drop-factor
*
* 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
* 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.
* </para>
* Last reviewed on 2006-02-28 (0.10.3)
* ]| This pipeline decodes and renders the raw DV stream to a videosink.
* </refsect2>
*
* Last reviewed on 2006-02-28 (0.10.3)
*/
#ifdef HAVE_CONFIG_H

22
ext/dv/gstdvdemux.c
View file

@ -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 capabilities
* of the upstream peer.
* </para>
*
* This element can operate in both push and pull mode depending on the
* capabilities of the upstream peer.
*
* <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.
* </para>
* Last reviewed on 2006-02-27 (0.10.3)
* ]| This pipeline decodes and renders the raw DV stream to an audio and a videosink.
* </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

6
ext/esd/esdmon.c
View file

@ -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

16
ext/esd/esdsink.c
View file

@ -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>
* Simple example pipeline that plays an Ogg/Vorbis file via esd:
* <programlisting>
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! audioresample ! esdsink
* </programlisting>
* </para>
* ]| play an Ogg/Vorbis audio file via esd
* </refsect2>
*/

20
ext/flac/gstflacdec.c
View file

@ -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>
*/

15
ext/flac/gstflacenc.c
View file

@ -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

13
ext/gconf/gstgconfaudiosink.c
View file

@ -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"

14
ext/gconf/gstgconfaudiosrc.c
View file

@ -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"

13
ext/gconf/gstgconfvideosink.c
View file

@ -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"

14
ext/gconf/gstgconfvideosrc.c
View file

@ -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"

77
ext/gdk_pixbuf/gstgdkpixbufsink.c
View file

@ -19,20 +19,14 @@
/**
* SECTION:element-gdkpixbufsink
* @short_description: video sink that converts RGB images to GdkPixbufs.
* @see_also:
* @Since: 0.10.8
*
* <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
* <ulink url="http://library.gnome.org/devel/gdk-pixbuf/unstable/index.html">
* GdkPixbuf library API</ulink> or displaying in Gtk+ applications (e.g. using
* the <ulink url="http://library.gnome.org/devel/gtk/unstable/GtkImage.html">
* GtkImage widget</ulink>).
* </para>
* <para>
* There are two ways to use this element and obtain the GdkPixbuf objects
* #GdkPixbuf objects, for easy saving to file via the
* GdkPixbuf library API or displaying in Gtk+ applications (e.g. using
* the #GtkImage widget).
*
* 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;
* </classname> property when needed.
* Retrieving the current pixbuf via the #GstGdkPixbufSink:last-pixbuf property
* when needed.
* </listitem>
* </itemizedlist>
* </para>
* <para>
* 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
*
* 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
* 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.
* </para>
* with the #GstXOverlay interface instead if you want to do video playback.
*
* <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
* <link linkend="GstGdkPixbufSink--send-messages">&quot;send-messages&quot;
* property</link> to #FALSE though). The element message&apos;s structure
* will have the following fields:
* can be disabled by setting the #GstGdkPixbufSink:send-messages property
* to #FALSE though). The element message structure has 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
* <ulink url="http://library.gnome.org/devel/gdk-pixbuf/unstable/index.html">
* GdkPixbuf library API</ulink> (the reason this is not done automatically
* is that the application will often scale the image anyway 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
* can easily be done using gdk_pixbuf_scale() (the reason this is not done
* automatically is that the application will often scale the image anyway
* 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
* 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

20
ext/hal/gsthalaudiosink.c
View file

@ -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>
* Here is a pipeline to test your sound output :
* <programlisting>
* ]| list the UDIs of all your ALSA output devices
* |[
* gst-launch -v audiotestsrc ! halaudiosink udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_playback_0
* </programlisting>
* This pipeline produces a test signal on the specified sound device.
* </para>
* ]| test your soundcard by playing a test signal on the specified sound device.
* </refsect2>
*/

20
ext/hal/gsthalaudiosrc.c
View file

@ -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>
* Here is a pipeline to test your sound input :
* <programlisting>
* ]| list the UDIs of all your ALSA input devices
* |[
* 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>
*/

6
ext/hal/hal.c
View file

@ -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.

6
ext/jpeg/gstjpegenc.c
View file

@ -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"

5
ext/jpeg/gstsmokedec.c
View file

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

6
ext/jpeg/gstsmokeenc.c
View file

@ -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"

30
ext/libcaca/gstcacasink.c
View file

@ -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.
* </para>
* ]| This pipeline renders a video to ascii art into the current terminal.
* </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",

11
ext/libcaca/gstcacasink.h
View file

@ -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) \

1
ext/libpng/gstpngdec.c
View file

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

5
ext/libpng/gstpngenc.c
View file

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

6
ext/pulse/pulsemixer.c
View file

@ -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.

30
ext/pulse/pulsesink.c
View file

@ -21,28 +21,20 @@
/**
* SECTION:element-pulsesink
* @short_description: Output audio to a PulseAudio sound server
* @see_also: pulsesrc, pulsemixer
*
* <refsect2>
* <para>
* 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>
* This element outputs audio to a
* <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
*
* <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

14
ext/pulse/pulsesrc.c
View file

@ -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.
* </para>
* ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
* </refsect2>
*/

14
ext/raw1394/gstdv1394src.c
View file

@ -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 ! dvdemux name=d ! queue ! dvdec ! xvimagesink d. ! queue ! alsasink
* </programlisting>
* This pipeline captures from the firewire port and displays it (might need
* |[
* gst-launch 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).
* </para>
* </refsect2>
*/

17
ext/raw1394/gsthdv1394src.c
View file

@ -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)
{

59
ext/soup/gstsouphttpsrc.c
View file

@ -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
* default.
* </para>
* <para>
* In case the element-souphttpsrc::iradio-mode property is set and the
* location is an HTTP resource, souphttpsrc will send special Icecast HTTP
* headers to the server to request additional Icecast meta-information. If
* the server is not an Icecast server, it will behave as if the
* element-souphttpsrc::iradio-mode property were not set. If it is,
* souphttpsrc will output data with a media type of application/x-icy,
* 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
* media type.
* </para>
* <para>
* Example pipeline:
* <programlisting>
* The #GstSoupHTTPSrc:proxy property can be used to override the default.
*
* In case the #GstSoupHTTPSrc:iradio-mode property is set and the location is
* an HTTP resource, souphttpsrc will send special Icecast HTTP headers to the
* server to request additional Icecast meta-information.
* If the server is not an Icecast server, it will behave as if the
* #GstSoupHTTPSrc:iradio-mode property were not set. If it is, souphttpsrc will
* output data with a media type of application/x-icy, 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 media type.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* 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

14
ext/speex/gstspeexdec.c
View file

@ -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 documentation of speexenc.
* </para>
* ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the
* documentation of speexenc.
* </refsect2>
*
* Last reviewed on 2006-04-05 (0.10.2)

16
ext/speex/gstspeexenc.c
View file

@ -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"

25
ext/taglib/gstapev2mux.cc
View file

@ -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>
* Here is a simple pipeline that transcodes a file from Ogg/Vorbis to mp3
* format with an APEv2 that contains the same as the the Ogg/Vorbis file:
* <programlisting>
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! apev2mux ! filesink location=foo.mp3
* </programlisting>
* Make sure the Ogg/Vorbis file actually has comments to preserve.
* You can verify the tags were written using:
* <programlisting>
* ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
* APEv2 that contains the same as the the Ogg/Vorbis file. Make sure the
* Ogg/Vorbis file actually has comments to preserve.
* |[
* gst-launch -m filesrc location=foo.mp3 ! apedemux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist
* </programlisting>
* </para>
* ]| Verify that tags have been written.
* </refsect2>
*/

25
ext/taglib/gstid3v2mux.cc
View file

@ -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>
* Here is a simple pipeline that transcodes a file from Ogg/Vorbis to mp3
* format with an ID3v2 that contains the same as the the Ogg/Vorbis file:
* <programlisting>
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! id3v2mux ! filesink location=foo.mp3
* </programlisting>
* Make sure the Ogg/Vorbis file actually has comments to preserve.
* You can verify the tags were written using:
* <programlisting>
* ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
* ID3v2 that contains the same as the the Ogg/Vorbis file. Make sure the
* Ogg/Vorbis file actually has comments to preserve.
* |[
* gst-launch -m filesrc location=foo.mp3 ! id3demux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist
* </programlisting>
* </para>
* ]| Verify that tags have been written.
* </refsect2>
*/

10
ext/wavpack/gstwavpackdec.c
View file

@ -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>
*/

24
ext/wavpack/gstwavpackenc.c
View file

@ -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>
*/

10
ext/wavpack/gstwavpackparse.c
View file

@ -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.
* </para>
* ]| This pipeline decodes the Wavpack file test.wv into raw audio buffers.
* </refsect2>
*/

2
gst/matroska/matroska-mux.h
View file

@ -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

2
gst/udp/gstudpsrc.c
View file

@ -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