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.
$(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 \
$(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 \
$(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 \
<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" />
<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" />
<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" />
<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" />
GstAudioChebBand
<SUBSECTION Standard>
GstAudioChebBandClass
-GstAudioChebBandProcessFunc
GST_AUDIO_CHEB_BAND
GST_AUDIO_CHEB_BAND_CLASS
GST_AUDIO_CHEB_BAND_GET_CLASS
GstAudioChebLimit
<SUBSECTION Standard>
GstAudioChebLimitClass
-GstAudioChebLimitProcessFunc
GST_AUDIO_CHEB_LIMIT
GST_AUDIO_CHEB_LIMIT_CLASS
GST_AUDIO_CHEB_LIMIT_GET_CLASS
GstAudioWSincBand
<SUBSECTION Standard>
GstAudioWSincBandClass
-GstAudioWSincBandProcessFunc
GST_AUDIO_WSINC_BAND
GST_AUDIO_WSINC_BAND_CLASS
GST_IS_AUDIO_WSINC_BAND
GstAudioWSincLimit
<SUBSECTION Standard>
GstAudioWSincLimitClass
-GstAudioWSincLimitProcessFunc
GST_AUDIO_WSINC_LIMIT
GST_AUDIO_WSINC_LIMIT_CLASS
GST_IS_AUDIO_WSINC_LIMIT
</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>
GstCairoTimeOverlay
<SUBSECTION Standard>
GST_CUTTER
GST_CUTTER_CLASS
-GST_CUTTER_GET_CLASS
GST_IS_CUTTER
GST_IS_CUTTER_CLASS
GST_TYPE_CUTTER
</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>
GstGdkPixbufSink
GST_MATROSKA_MUX_CLASS
GST_IS_MATROSKA_MUX
GST_IS_MATROSKA_MUX_CLASS
-gst_matroska_mux_get_type
gst_matroska_mux_plugin_init
</SECTION>
GST_MATROSKA_DEMUX_CLASS
GST_IS_MATROSKA_DEMUX
GST_IS_MATROSKA_DEMUX_CLASS
-gst_matroska_demux_get_type
gst_matroska_demux_plugin_init
</SECTION>
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>
GstSpectrum
<SUBSECTION Standard>
GstSpectrumClass
-GstSpectrumFFTFreeFunc
-GstSpectrumProcessFunc
GST_SPECTRUM
GST_SPECTRUM_CLASS
GST_IS_SPECTRUM
</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>
GstTagInject
<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>
<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
GST_V4L2_MAX_SIZE
GstV4l2BufferPool
GstV4l2Buffer
-GstV4l2Src
GstV4l2SrcClass
GST_V4L2SRC
GST_IS_V4L2SRC
GST_TYPE_WAVENC
GST_WAVENC_CLASS
GST_IS_WAVENC_CLASS
-gst_wavenc_get_type
</SECTION>
<SECTION>
* 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>
*/
* 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>
*/
* 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>
*/
* 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>
/**
* 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>
*/
/**
* 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
/**
* 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
* 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
* ]| Record from audioinput
* </refsect2>
*/
+
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
* 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>
*/
/**
* 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>
*/
* 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
* 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"
* 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"
* 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"
* 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"
/**
* 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>
* contained in them.
* </listitem>
* <listitem>
- * Retrieving the current pixbuf via the <classname>"last-pixbuf"
- * </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>"preroll-pixbuf"
* </classname> or <classname>"pixbuf"</classname> on the bus (this
- * can be disabled by setting the
- * <link linkend="GstGdkPixbufSink--send-messages">"send-messages"
- * property</link> to #FALSE though). The element message'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>"pixbuf"</classname>: the GdkPixbuf object
+ * <classname>"pixbuf"</classname>: the #GdkPixbuf object
* </listitem>
* <listitem>
* <classname>"pixel-aspect-ratio"</classname>: the pixel aspect
* 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>
+ * </refsect2>
+ *
+ * <refsect2>
* <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
+ * ]| Process one single test image as pixbuf (note that the output you see will
* be slightly misleading. The message structure does contain a valid pixbuf
* object even if the structure string says '(NULL)').
- * </para>
* </refsect2>
- *
- * Since: 0.10.8
*/
#ifdef HAVE_CONFIG_H
/**
* 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>
*/
/**
* 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>
*/
#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.
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.
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.
* 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"
* Boston, MA 02111-1307, USA.
*/
+/**
+ * SECTION:element-smokedec
+ *
+ * Decodes images in smoke format.
+ */
#ifdef HAVE_CONFIG_H
#include "config.h"
* 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"
* 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>
*/
#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",
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) \
* Boston, MA 02111-1307, USA.
*
*/
-
/**
* SECTION:element-pngdec
*
* Boston, MA 02111-1307, USA.
*
*/
+/**
+ * SECTION:element-pngenc
+ *
+ * Encodes png images.
+ */
#ifdef HAVE_CONFIG_H
#include "config.h"
/**
* 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.
/**
* SECTION:element-pulsesink
- * @short_description: Output audio to a PulseAudio sound server
* @see_also: pulsesrc, pulsemixer
*
+ * This element outputs audio to a
+ * <ulink href="http://www.pulseaudio.org">PulseAudio sound server</ulink>.
+ *
* <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>
+ * ]| Play an Ogg/Vorbis file.
+ * |[
* gst-launch -v audiotestsrc ! audioconvert ! volume volume=0.4 ! pulsesink
- * </programlisting>
- * Play a 440Hz sine wave.
- * </para>
+ * ]| Play a 440Hz sine wave.
* </refsect2>
- *
*/
#ifdef HAVE_CONFIG_H
/**
* 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>
*/
/**
* 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>
*/
* 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"
{
return GST_URI_SRC;
}
+
static gchar **
gst_hdv1394src_uri_get_protocols (void)
{
return protocols;
}
+
static const gchar *
gst_hdv1394src_uri_get_uri (GstURIHandler * handler)
{
/**
* 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
* 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
/**
* 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)
* 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"
* 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> /dev/null | grep taglist
- * </programlisting>
- * </para>
+ * ]| Verify that tags have been written.
* </refsect2>
*/
* 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> /dev/null | grep taglist
- * </programlisting>
- * </para>
+ * ]| Verify that tags have been written.
* </refsect2>
*/
/**
* 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>
*/
/**
* 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>
*/
/**
* 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>
*/
GstElementClass parent;
} GstMatroskaMuxClass;
-GType gst_matroska_mux_get_type (void);
-
gboolean gst_matroska_mux_plugin_init (GstPlugin *plugin);
G_END_DECLS
* #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
projects / platform / upstream / gstreamer.git / commitdiff