/**
* SECTION:element-a52dec
+ * @title: a52dec
*
* Dolby Digital (AC-3) audio decoder.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 dvdreadsrc title=1 ! mpegpsdemux ! a52dec ! audioconvert ! audioresample ! autoaudiosink
* ]| Play audio part of a dvd title.
* |[
* gst-launch-1.0 filesrc location=abc.ac3 ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
* ]| Decode and play a stand alone AC-3 file.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-amrnbdec
+ * @title: amrnbdec
* @see_also: #GstAmrnbEnc, #GstAmrParse
*
- * AMR narrowband decoder based on the
+ * AMR narrowband decoder based on the
* <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
- *
- * <refsect2>
- * <title>Example launch line</title>
+ *
+ * ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrnbdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-amrnbenc
+ * @title: amrnbenc
* @see_also: #GstAmrnbDec, #GstAmrnbParse
*
- * AMR narrowband encoder based on the
+ * AMR narrowband encoder based on the
* <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
- *
- * <refsect2>
- * <title>Example launch line</title>
+ *
+ * ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
* ]|
* Please note that the above stream misses the header, that is needed to play
* the stream.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-amrwbdec
+ * @title: amrwbdec
* @see_also: #GstAmrwbEnc
*
- * AMR wideband decoder based on the
+ * AMR wideband decoder based on the
* <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 filesrc location=abc.amr ! amrparse ! amrwbdec ! audioconvert ! audioresample ! autoaudiosink
* ]|
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-cdiocddasrc
+ * @title: cdiocddasrc
* @see_also: GstCdParanoiaSrc, GstAudioCdSrc
*
- * <refsect2>
- * <para>
* cdiocddasrc reads and extracts raw audio from Audio CDs. It can operate
* in one of two modes:
- * <itemizedlist>
- * <listitem><para>
- * treat each track as a separate stream, counting time from the start
+ *
+ * * treat each track as a separate stream, counting time from the start
* of the track to the end of the track and posting EOS at the end of
* a track, or
- * </para></listitem>
- * <listitem><para>
- * treat the entire disc as one stream, counting time from the start of
+ * * treat the entire disc as one stream, counting time from the start of
* the first track to the end of the last track, posting EOS only at
* the end of the last track.
- * </para></listitem>
- * </itemizedlist>
- * </para>
- * <para>
+ *
* With a recent-enough version of libcdio, the element will extract
* CD-TEXT if this is supported by the CD-drive and CD-TEXT information
* is available on the CD. The information will be posted on the bus in
* form of a tag message.
- * </para>
- * <para>
+ *
* When opened, the element will also calculate a CDDB disc ID and a
* MusicBrainz disc ID, which applications can use to query online
* databases for artist/title information. These disc IDs will also be
* posted on the bus as part of the tag messages.
- * </para>
- * <para>
+ *
* cdiocddasrc supports the GstUriHandler interface, so applications can use
* playbin with cdda://<track-number> URIs for playback (they will have
* to connect to playbin's notify::source signal and set the device on the
* Applications should use seeks in "track" format to switch between different
* tracks of the same CD (passing a new cdda:// URI to playbin involves opening
* and closing the CD device, which is much slower).
- * </para>
- * <title>Example launch line</title>
- * <para>
- * <programlisting>
+ *
+ * ## Example launch line
+ *
+ * |[
* gst-launch-1.0 cdiocddasrc track=5 device=/dev/cdrom ! audioconvert ! vorbisenc ! oggmux ! filesink location=track5.ogg
- * </programlisting>
+ * ]|
* This pipeline extracts track 5 of the audio CD and encodes it into an
* Ogg/Vorbis file.
- * </para>
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-x264enc
+ * @title: x264enc
* @see_also: faac
*
* This element encodes raw video into H264 compressed data,
* applied, followed by the user-set properties, fast first pass restrictions and
* finally the profile restrictions.
*
- * <note>Some settings, including the default settings, may lead to quite
- * some latency (i.e. frame buffering) in the encoder. This may cause problems
- * with pipeline stalling in non-trivial pipelines, because the encoder latency
- * is often considerably higher than the default size of a simple queue
- * element. Such problems are caused by one of the queues in the other
- * non-x264enc streams/branches filling up and blocking upstream. They can
- * be fixed by relaxing the default time/size/buffer limits on the queue
- * elements in the non-x264 branches, or using a (single) multiqueue element
- * for all branches. Also see the last example below. You can also work around
- * this problem by setting the tune=zerolatency property, but this will affect
- * overall encoding quality so may not be appropriate for your use case.
- * </note>
+ * > Some settings, including the default settings, may lead to quite
+ * > some latency (i.e. frame buffering) in the encoder. This may cause problems
+ * > with pipeline stalling in non-trivial pipelines, because the encoder latency
+ * > is often considerably higher than the default size of a simple queue
+ * > element. Such problems are caused by one of the queues in the other
+ * > non-x264enc streams/branches filling up and blocking upstream. They can
+ * > be fixed by relaxing the default time/size/buffer limits on the queue
+ * > elements in the non-x264 branches, or using a (single) multiqueue element
+ * > for all branches. Also see the last example below. You can also work around
+ * > this problem by setting the tune=zerolatency property, but this will affect
+ * > overall encoding quality so may not be appropriate for your use case.
*
- * <refsect2>
- * <title>Example pipeline</title>
+ * ## Example pipeline
* |[
* gst-launch-1.0 -v videotestsrc num-buffers=1000 ! x264enc qp-min=18 ! \
* avimux ! filesink location=videotestsrc.avi
* specific settings are needed in this case to avoid pipeline stalling.
* Depending on goals and context, other approaches are possible, e.g.
* tune=zerolatency might be configured, or queue sizes increased.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
#include "asfheaders.h"
G_BEGIN_DECLS
-
+
#define GST_TYPE_ASF_DEMUX \
(gst_asf_demux_get_type())
#define GST_ASF_DEMUX(obj) \
GST_ASF_3D_SIDE_BY_SIDE_HALF_RL = 0x02,
GST_ASF_3D_TOP_AND_BOTTOM_HALF_LR = 0x03,
GST_ASF_3D_TOP_AND_BOTTOM_HALF_RL = 0x04,
- GST_ASF_3D_DUAL_STREAM = 0x0D, /**< Full format*/
+ GST_ASF_3D_DUAL_STREAM = 0x0D, /*< Full format*/
};
typedef struct
/* extended stream properties (optional) */
AsfStreamExtProps ext_props;
-
+
gboolean inspect_payload;
} AsfStream;
GstClockTime sidx_interval; /* interval between entries in ns */
guint sidx_num_entries; /* number of index entries */
AsfSimpleIndexEntry *sidx_entries; /* packet number for each entry */
-
+
GSList *other_streams; /* remember streams that are in header but have unknown type */
/* For reverse playback */
/**
* SECTION:element-rtspwms
+ * @title: rtspwms
*
* A WMS RTSP extension
*/
#include <gst/gst.h>
G_BEGIN_DECLS
-
-/**
+/**
* GstRDTType:
* @GST_RDT_TYPE_INVALID:
* @GST_RDT_TYPE_ASMACTION:
/**
* GST_RDT_IS_DATA_TYPE:
* @t: the #GstRDTType to check
- *
+ *
* Check if @t is a data packet type.
*/
#define GST_RDT_IS_DATA_TYPE(t) ((t) < 0xff00)
* @buffer: pointer to RDT buffer
* @offset: offset of packet in buffer data
*
- * Data structure that points to a packet at @offset in @buffer.
+ * Data structure that points to a packet at @offset in @buffer.
* The size of the structure is made public to allow stack allocations.
*/
struct _GstRDTPacket
-{
+{
GstBuffer *buffer;
guint offset;
-
+
/*< private >*/
GstRDTType type; /* type of current packet */
guint16 length; /* length of current packet in bytes */
/**
* SECTION:element-rademux
+ * @title: rademux
*
* Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
- *
- * <refsect2>
- * <title>Example launch line</title>
+ *
+ * ## Example launch line
* |[
* gst-launch-1.0 filesrc location=interview.ra ! rademux ! avdec_real_288 ! audioconvert ! audioresample ! autoaudiosink
* ]| Read a RealAudio file and decode it and output it to the soundcard using
* gst-launch-1.0 souphttpsrc location=http://www.example.org/interview.ra ! rademux ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
* ]| Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
* and output it to the soundcard.
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
/**
* SECTION:element-rdtmanager
+ * @title: rdtmanager
* @see_also: GstRtspSrc
*
* A simple RTP session manager used internally by rtspsrc.
/**
* SECTION:element-rtspreal
+ * @title: rtspreal
*
* A RealMedia RTSP extension
*/
/**
* SECTION:element-xingmux
+ * @title: xingmux
*
* xingmux adds a Xing header to MP3 files. This contains information about the duration and size
* of the file and a seek table and is very useful for getting an almost correct duration and better
* seeking on VBR MP3 files.
- *
+ *
* This element will remove any existing Xing, LAME or VBRI headers from the beginning of the file.
*
- * <refsect2>
- * <title>Example launch line</title>
+ * ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc num-buffers=1000 ! audioconvert ! lamemp3enc ! xingmux ! filesink location=test.mp3
* gst-launch-1.0 filesrc location=test.mp3 ! xingmux ! filesink location=test2.mp3
* gst-launch-1.0 filesrc location=test.mp3 ! mp3parse ! xingmux ! filesink location=test2.mp3
* ]|
- * </refsect2>
+ *
*/
#ifdef HAVE_CONFIG_H
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