Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top

Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
This commit is contained in:
Stefan Kost 2006-03-24 10:42:11 +00:00
parent 34c9c8dfd6
commit 2d826700fa
32 changed files with 261 additions and 192 deletions

View file

@ -1,3 +1,47 @@
2006-03-24 Stefan Kost <ensonic@users.sf.net>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* ext/cdparanoia/gstcdparanoiasrc.c:
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
(gst_gnome_vfs_sink_class_init):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
* ext/ogg/gstoggmux.c:
* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
* ext/pango/gsttextoverlay.c:
* ext/pango/gsttextrender.c:
* ext/theora/theoradec.c:
* ext/theora/theoraenc.c:
* ext/vorbis/vorbisdec.c:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/audio/gstaudiofilter.c:
(gst_audio_filter_base_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
(gst_audio_filter_template_base_init):
* gst/adder/gstadder.c: (gst_adder_get_type):
* gst/adder/gstadder.h:
* gst/audioconvert/gstaudioconvert.c:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
(gst_audio_test_src_create):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
* gst/playback/gstdecodebin.c:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
(gst_stream_selector_base_init):
* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
* gst/volume/gstvolume.c:
* sys/v4l/gstv4lmjpegsink.c:
* sys/v4l/gstv4lmjpegsrc.c:
* tests/check/libs/cddabasesrc.c:
* tests/old/examples/gob/gst-identity2.gob:
Add docs for adder, use GST_ELEMENT_DETAILS macro,
define GstElementDetails at the top
2006-03-23 Sebastien Moutte <sebastien@moutte.net>
* win32/common/libgstinterfaces.def:

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@ -74,10 +74,10 @@ IGNORE_CFILES = utils.c mem.c imgconvert.c
# the registry won't have the element
EXTRA_HFILES = \
$(top_srcdir)/ext/cdparanoia/gstcdparanoiasrc.h \
$(top_srcdir)/ext/alsa/gstalsamixerelement.h \
$(top_srcdir)/ext/alsa/gstalsasink.h \
$(top_srcdir)/ext/alsa/gstalsasrc.h \
$(top_srcdir)/ext/cdparanoia/gstcdparanoiasrc.h \
$(top_srcdir)/ext/gnomevfs/gstgnomevfssink.h \
$(top_srcdir)/ext/gnomevfs/gstgnomevfssrc.h \
$(top_srcdir)/ext/pango/gstclockoverlay.h \
@ -89,6 +89,7 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/vorbis/vorbisdec.h \
$(top_srcdir)/ext/vorbis/vorbisenc.h \
$(top_srcdir)/ext/vorbis/vorbisparse.h \
$(top_srcdir)/gst/adder/gstadder.h \
$(top_srcdir)/gst/audioconvert/gstaudioconvert.h \
$(top_srcdir)/gst/audioresample/gstaudioresample.h \
$(top_srcdir)/gst/audiotestsrc/gstaudiotestsrc.h \

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@ -12,6 +12,7 @@
<chapter>
<title>gst-plugins-base Elements</title>
<xi:include href="xml/element-adder.xml" />
<xi:include href="xml/element-alsamixer.xml" />
<xi:include href="xml/element-alsasink.xml" />
<xi:include href="xml/element-alsasrc.xml" />

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@ -1,3 +1,19 @@
<SECTION>
<FILE>element-adder</FILE>
<TITLE>adder</TITLE>
GstAdder
<SUBSECTION Standard>
GST_ADDER
GST_IS_ADDER
GST_TYPE_ADDER
gst_adder_factory_init
gst_adder_get_type
GST_ADDER_CLASS
GST_IS_ADDER_CLASS
GstAdderClass
GstAdderFormat
</SECTION>
<SECTION>
<FILE>element-alsamixer</FILE>
GstAlsaMixerElement

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@ -2,7 +2,7 @@
/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* <2005> Wim Taymans <wim@fluendo.com>
* <2005> Tim-Philipp Müller <tim centricular net>
* <2005> Tim-Philipp M??ller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@ -68,12 +68,11 @@ static gboolean gst_cd_paranoia_src_open (GstCddaBaseSrc * src,
const gchar * device);
static void gst_cd_paranoia_src_close (GstCddaBaseSrc * src);
static GstElementDetails cdparanoia_details = {
"CD Audio (cdda) Source, Paranoia IV",
"Source/File",
"Read audio from CD in paranoid mode",
"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>"
};
static GstElementDetails cdparanoia_details =
GST_ELEMENT_DETAILS ("CD Audio (cdda) Source, Paranoia IV",
"Source/File",
"Read audio from CD in paranoid mode",
"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
/* We use these to serialize calls to paranoia_read() among several
* cdparanoiasrc instances. We do this because it's the only reasonably

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@ -3,7 +3,7 @@
* 2000 Wim Taymans <wtay@chello.be>
* 2001 Bastien Nocera <hadess@hadess.net>
* 2003 Colin Walters <walters@verbum.org>
* 2005 Tim-Philipp Müller <tim centricular net>
* 2005 Tim-Philipp M??ller <tim centricular net>
*
* gstgnomevfssink.c:
*
@ -73,6 +73,12 @@
#include <string.h>
#include <errno.h>
static GstElementDetails gst_gnome_vfs_sink_details =
GST_ELEMENT_DETAILS ("GnomeVFS Sink",
"Sink/File",
"Write a stream to a GnomeVFS URI",
"Bastien Nocera <hadess@hadess.net>");
enum
{
SIGNAL_ERASE_ASK,
@ -139,11 +145,6 @@ static void
gst_gnome_vfs_sink_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
static GstElementDetails gst_gnome_vfs_sink_details =
GST_ELEMENT_DETAILS ("GnomeVFS Sink",
"Sink/File",
"Write a stream to a GnomeVFS URI",
"Bastien Nocera <hadess@hadess.net>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sinktemplate));

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@ -96,6 +96,12 @@
GST_DEBUG_CATEGORY_STATIC (gnomevfssrc_debug);
#define GST_CAT_DEFAULT gnomevfssrc_debug
static GstElementDetails gst_gnome_vfs_src_details =
GST_ELEMENT_DETAILS ("GnomeVFS Source",
"Source/File",
"Read from any GnomeVFS-supported file",
"Bastien Nocera <hadess@hadess.net>\n"
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>");
static GStaticMutex count_lock = G_STATIC_MUTEX_INIT;
static gint ref_count = 0;
@ -183,12 +189,6 @@ static void
gst_gnome_vfs_src_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
static GstElementDetails gst_gnome_vfs_src_details =
GST_ELEMENT_DETAILS ("GnomeVFS Source",
"Source/File",
"Read from any GnomeVFS-supported file",
"Bastien Nocera <hadess@hadess.net>\n"
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>");
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&srctemplate));

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@ -27,6 +27,12 @@
#include <string.h>
#include <gst/gst-i18n-plugin.h>
static GstElementDetails gst_ogg_demux_details =
GST_ELEMENT_DETAILS ("ogg demuxer",
"Codec/Demuxer",
"demux ogg streams (info about ogg: http://xiph.org)",
"Wim Taymand <wim@fluendo.com>");
#define CHUNKSIZE (8500) /* this is out of vorbisfile */
#define SKELETON_FISHEAD_SIZE 64
#define SKELETON_FISBONE_MIN_SIZE 52
@ -1277,11 +1283,6 @@ static void
gst_ogg_demux_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
static GstElementDetails gst_ogg_demux_details =
GST_ELEMENT_DETAILS ("ogg demuxer",
"Codec/Demuxer",
"demux ogg streams (info about ogg: http://xiph.org)",
"Wim Taymand <wim@fluendo.com>");
gst_element_class_set_details (element_class, &gst_ogg_demux_details);

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@ -36,6 +36,12 @@
#include <ogg/ogg.h>
#include <string.h>
static GstElementDetails gst_ogg_parse_details =
GST_ELEMENT_DETAILS ("ogg parser",
"Codec/Parser",
"parse ogg streams into pages (info about ogg: http://xiph.org)",
"Michael Smith <msmith@fluendo.com>");
GST_DEBUG_CATEGORY_STATIC (gst_ogg_parse_debug);
#define GST_CAT_DEFAULT gst_ogg_parse_debug
@ -211,11 +217,6 @@ static void
gst_ogg_parse_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
static GstElementDetails gst_ogg_parse_details =
GST_ELEMENT_DETAILS ("ogg parser",
"Codec/Parser",
"parse ogg streams into pages (info about ogg: http://xiph.org)",
"Michael Smith <msmith@fluendo.com>");
gst_element_class_set_details (element_class, &gst_ogg_parse_details);

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@ -55,6 +55,24 @@ GST_DEBUG_CATEGORY_STATIC (gst_ogm_parse_debug);
#define GST_OGM_PARSE_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_OGM_PARSE, GstOgmParseClass))
static GstElementDetails gst_ogm_audio_parse_details =
GST_ELEMENT_DETAILS ("OGM audio stream parser",
"Codec/Decoder/Audio",
"parse an OGM audio header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
static GstElementDetails gst_ogm_video_parse_details =
GST_ELEMENT_DETAILS ("OGM video stream parser",
"Codec/Decoder/Video",
"parse an OGM video header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
static GstElementDetails gst_ogm_text_parse_details =
GST_ELEMENT_DETAILS ("OGM text stream parser",
"Codec/Decoder/Subtitle",
"parse an OGM text header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
typedef struct _stream_header_video
{
gint32 width;
@ -260,11 +278,6 @@ static void
gst_ogm_audio_parse_base_init (GstOgmParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails gst_ogm_audio_parse_details =
GST_ELEMENT_DETAILS ("OGM audio stream parser",
"Codec/Decoder/Audio",
"parse an OGM audio header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GstCaps *caps = gst_riff_create_audio_template_caps ();
gst_element_class_set_details (element_class, &gst_ogm_audio_parse_details);
@ -280,11 +293,6 @@ static void
gst_ogm_video_parse_base_init (GstOgmParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails gst_ogm_video_parse_details =
GST_ELEMENT_DETAILS ("OGM video stream parser",
"Codec/Decoder/Video",
"parse an OGM video header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GstCaps *caps = gst_riff_create_video_template_caps ();
gst_element_class_set_details (element_class, &gst_ogm_video_parse_details);
@ -300,11 +308,6 @@ static void
gst_ogm_text_parse_base_init (GstOgmParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails gst_ogm_text_parse_details =
GST_ELEMENT_DETAILS ("OGM text stream parser",
"Codec/Decoder/Subtitle",
"parse an OGM text header and stream",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GstCaps *caps = gst_caps_new_simple ("text/plain", NULL, NULL);
gst_element_class_set_details (element_class, &gst_ogm_text_parse_details);

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@ -53,18 +53,18 @@
* If you do not have such a subtitle file, create one looking like this
* in a text editor:
* <programlisting>
* 1
* 00:00:03,000 --> 00:00:05,000
* Hello? (3-5s)
*
* 2
* 00:00:08,000 --> 00:00:13,000
* 1
* 00:00:03,000 --> 00:00:05,000
* Hello? (3-5s)
*
* 2
* 00:00:08,000 --> 00:00:13,000
* Yes, this is a subtitle. Don&apos;t
* you like it? (8-13s)
*
* 3
* 00:00:18,826 --> 00:01:02,886
* Uh? What are you talking about?
* you like it? (8-13s)
*
* 3
* 00:00:18,826 --> 00:01:02,886
* Uh? What are you talking about?
* I don&apos;t understand (18-62s)
* </programlisting>
* </para>
@ -93,12 +93,11 @@
GST_DEBUG_CATEGORY (pango_debug);
#define GST_CAT_DEFAULT pango_debug
static GstElementDetails text_overlay_details = {
"Text Overlay",
"Filter/Editor/Video",
"Adds text strings on top of a video buffer",
"David Schleef <ds@schleef.org>"
};
static GstElementDetails text_overlay_details =
GST_ELEMENT_DETAILS ("Text Overlay",
"Filter/Editor/Video",
"Adds text strings on top of a video buffer",
"David Schleef <ds@schleef.org>");
#define DEFAULT_PROP_TEXT ""

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@ -54,13 +54,12 @@
GST_DEBUG_CATEGORY_EXTERN (pango_debug);
#define GST_CAT_DEFAULT pango_debug
static GstElementDetails text_render_details = {
"Text Render",
"Filter/Editor/Video",
"Renders a text string to an image bitmap",
"David Schleef <ds@schleef.org>, "
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>"
};
static GstElementDetails text_render_details =
GST_ELEMENT_DETAILS ("Text Render",
"Filter/Editor/Video",
"Renders a text string to an image bitmap",
"David Schleef <ds@schleef.org>, "
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>");
enum
{

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@ -58,13 +58,11 @@ enum
ARG_CROP
};
static GstElementDetails theora_dec_details = {
"TheoraDec",
"Codec/Decoder/Video",
"decode raw theora streams to raw YUV video",
"Benjamin Otte <in7y118@public.uni-hamburg.de>, "
"Wim Taymans <wim@fluendo.com>",
};
static GstElementDetails theora_dec_details = GST_ELEMENT_DETAILS ("TheoraDec",
"Codec/Decoder/Video",
"decode raw theora streams to raw YUV video",
"Benjamin Otte <in7y118@public.uni-hamburg.de>, "
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate theora_dec_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",

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@ -149,12 +149,10 @@ granulepos_to_timestamp (GstTheoraEnc * theoraenc, ogg_int64_t granulepos)
theoraenc->info.fps_numerator);
}
static GstElementDetails theora_enc_details = {
"TheoraEnc",
"Codec/Encoder/Video",
"encode raw YUV video to a theora stream",
"Wim Taymans <wim@fluendo.com>",
};
static GstElementDetails theora_enc_details = GST_ELEMENT_DETAILS ("TheoraEnc",
"Codec/Encoder/Video",
"encode raw YUV video to a theora stream",
"Wim Taymans <wim@fluendo.com>");
static GstStaticPadTemplate theora_enc_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink",

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@ -54,12 +54,10 @@
GST_DEBUG_CATEGORY_EXTERN (vorbisdec_debug);
#define GST_CAT_DEFAULT vorbisdec_debug
static GstElementDetails vorbis_dec_details = {
"VorbisDec",
"Codec/Decoder/Audio",
"decode raw vorbis streams to float audio",
"Benjamin Otte <in7y118@public.uni-hamburg.de>",
};
static GstElementDetails vorbis_dec_details = GST_ELEMENT_DETAILS ("VorbisDec",
"Codec/Decoder/Audio",
"decode raw vorbis streams to float audio",
"Benjamin Otte <in7y118@public.uni-hamburg.de>");
/* Filter signals and args */
enum

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@ -66,12 +66,10 @@ GST_DEBUG_CATEGORY_EXTERN (vorbisenc_debug);
static GstPadTemplate *gst_vorbisenc_src_template, *gst_vorbisenc_sink_template;
/* elementfactory information */
GstElementDetails vorbisenc_details = {
"Vorbis encoder",
"Codec/Encoder/Audio",
"Encodes audio in Vorbis format",
"Monty <monty@xiph.org>, " "Wim Taymans <wim@fluendo.com>",
};
GstElementDetails vorbisenc_details = GST_ELEMENT_DETAILS ("Vorbis encoder",
"Codec/Encoder/Audio",
"Encodes audio in Vorbis format",
"Monty <monty@xiph.org>, " "Wim Taymans <wim@fluendo.com>");
/* GstVorbisEnc signals and args */
enum

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@ -27,6 +27,11 @@
#include <string.h>
static GstElementDetails audio_filter_details =
GST_ELEMENT_DETAILS ("Audio filter base class",
"Filter/Effect/Audio",
"Filters audio",
"David Schleef <ds@schleef.org>");
/* GstAudioFilter signals and args */
enum
@ -83,12 +88,6 @@ gst_audio_filter_get_type (void)
static void
gst_audio_filter_base_init (gpointer g_class)
{
static GstElementDetails audio_filter_details = {
"Audio filter base class",
"Filter/Effect/Audio",
"Filters audio",
"David Schleef <ds@schleef.org>"
};
GstAudioFilterClass *klass = (GstAudioFilterClass *) g_class;
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

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@ -34,6 +34,12 @@
#include <gst/audio/gstaudiofilter.h>
#include <string.h>
static GstElementDetails audio_filter_template_details =
GST_ELEMENT_DETAILS ("Audio filter template",
"Filter/Effect/Audio",
"Filters audio",
"David Schleef <ds@schleef.org>");
typedef struct _GstAudioFilterTemplate GstAudioFilterTemplate;
typedef struct _GstAudioFilterTemplateClass GstAudioFilterTemplateClass;
@ -117,12 +123,6 @@ gst_audio_filter_template_get_type (void)
static void
gst_audio_filter_template_base_init (gpointer g_class)
{
static GstElementDetails audio_filter_template_details = {
"Audio filter template",
"Filter/Effect/Audio",
"Filters audio",
"David Schleef <ds@schleef.org>"
};
GstAudioFilterTemplateClass *klass = (GstAudioFilterTemplateClass *) g_class;
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

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@ -20,6 +20,21 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-adder
*
* <refsect2>
* The Adder allows to mix several streams into one by adding the data.
* Mixed data is clamped to the min/max values of the data format.
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=100 ! adder name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
* </programlisting>
* This pipeline produces two sine waves mixed together.
* </para>
* </refsect2>
*/
/* Element-Checklist-Version: 5 */
#ifdef HAVE_CONFIG_H
@ -90,7 +105,7 @@ gst_adder_get_type (void)
{
static GType adder_type = 0;
if (!adder_type) {
if (G_UNLIKELY (adder_type == 0)) {
static const GTypeInfo adder_info = {
sizeof (GstAdderClass), NULL, NULL,
(GClassInitFunc) gst_adder_class_init, NULL, NULL,

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@ -28,8 +28,6 @@
G_BEGIN_DECLS
extern GstElementDetails gst_adder_details;
#define GST_TYPE_ADDER \
(gst_adder_get_type())
#define GST_ADDER(obj) \

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@ -74,12 +74,11 @@ GST_DEBUG_CATEGORY (audio_convert_debug);
/*** DEFINITIONS **************************************************************/
static GstElementDetails audio_convert_details = {
"Audio Conversion",
"Filter/Converter/Audio",
"Convert audio to different formats",
"Benjamin Otte <in7y118@public.uni-hamburg.de>",
};
static GstElementDetails audio_convert_details =
GST_ELEMENT_DETAILS ("Audio Conversion",
"Filter/Converter/Audio",
"Convert audio to different formats",
"Benjamin Otte <in7y118@public.uni-hamburg.de>");
/* type functions */
static void gst_audio_convert_dispose (GObject * obj);

View file

@ -64,12 +64,11 @@
#define M_PI_M2 ( M_PI + M_PI )
GstElementDetails gst_audio_test_src_details = {
"Audio test source",
"Source/Audio",
"Creates audio test signals of given frequency and volume",
"Stefan Kost <ensonic@users.sf.net>"
};
static GstElementDetails gst_audio_test_src_details =
GST_ELEMENT_DETAILS ("Audio test source",
"Source/Audio",
"Creates audio test signals of given frequency and volume",
"Stefan Kost <ensonic@users.sf.net>");
enum
@ -116,7 +115,7 @@ gst_audiostestsrc_wave_get_type (void)
{0, NULL, NULL},
};
if (!audiostestsrc_wave_type) {
if (G_UNLIKELY (audiostestsrc_wave_type == 0)) {
audiostestsrc_wave_type = g_enum_register_static ("GstAudioTestSrcWave",
audiostestsrc_waves);
}
@ -140,8 +139,10 @@ static gboolean gst_audio_test_src_query (GstBaseSrc * basesrc,
static void gst_audio_test_src_change_wave (GstAudioTestSrc * src);
/*
static void gst_audio_test_src_get_times (GstBaseSrc * basesrc,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
*/
static GstFlowReturn gst_audio_test_src_create (GstBaseSrc * basesrc,
guint64 offset, guint length, GstBuffer ** buffer);
@ -195,8 +196,10 @@ gst_audio_test_src_class_init (GstAudioTestSrcClass * klass)
GST_DEBUG_FUNCPTR (gst_audio_test_src_is_seekable);
gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_audio_test_src_do_seek);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_audio_test_src_query);
gstbasesrc_class->get_times =
GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times);
/*
gstbasesrc_class->get_times =
GST_DEBUG_FUNCPTR (gst_audio_test_src_get_times);
*/
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_audio_test_src_create);
}
@ -576,6 +579,7 @@ gst_audio_test_src_change_volume (GstAudioTestSrc * src)
}
}
#ifdef __DISABLE_NO_LIVE__
static void
gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
@ -598,6 +602,7 @@ gst_audio_test_src_get_times (GstBaseSrc * basesrc, GstBuffer * buffer,
*end = -1;
}
}
#endif
static gboolean
gst_audio_test_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
@ -636,6 +641,7 @@ static GstFlowReturn
gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** buffer)
{
GstFlowReturn res;
GstAudioTestSrc *src;
GstBuffer *buf;
GstClockTime next_time;
@ -646,6 +652,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
if (src->eos_reached)
return GST_FLOW_UNEXPECTED;
/* example for tagging generated data */
if (!src->tags_pushed) {
GstTagList *taglist;
GstEvent *event;
@ -660,6 +667,7 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
src->tags_pushed = TRUE;
}
/* check for eos */
if (src->check_seek_stop &&
(src->n_samples_stop > src->n_samples) &&
(src->n_samples_stop < src->n_samples + src->samples_per_buffer)
@ -673,15 +681,17 @@ gst_audio_test_src_create (GstBaseSrc * basesrc, guint64 offset,
src->generate_samples_per_buffer = src->samples_per_buffer;
n_samples = src->n_samples + src->samples_per_buffer;
}
next_time = n_samples * GST_SECOND / src->samplerate;
next_time = gst_util_uint64_scale (n_samples, GST_SECOND,
(guint64) src->samplerate);
buf =
gst_buffer_new_and_alloc (src->generate_samples_per_buffer *
sizeof (gint16));
gst_buffer_set_caps (buf, GST_PAD_CAPS (basesrc->srcpad));
/* allocate a new buffer suitable for this pad */
if ((res = gst_pad_alloc_buffer (basesrc->srcpad, src->n_samples,
src->generate_samples_per_buffer * sizeof (gint16),
GST_PAD_CAPS (basesrc->srcpad), &buf)) != GST_FLOW_OK) {
return res;
}
GST_BUFFER_TIMESTAMP (buf) = src->timestamp_offset + src->running_time;
GST_BUFFER_OFFSET (buf) = src->n_samples;
GST_BUFFER_OFFSET_END (buf) = n_samples;
GST_BUFFER_DURATION (buf) = next_time - src->running_time;

View file

@ -43,12 +43,11 @@ GST_DEBUG_CATEGORY (ffmpegcolorspace_debug);
#define GST_CAT_DEFAULT ffmpegcolorspace_debug
/* elementfactory information */
static GstElementDetails ffmpegcsp_details = {
"FFMPEG Colorspace converter",
"Filter/Converter/Video",
"Converts video from one colorspace to another",
"Ronald Bultje <rbultje@ronald.bitfreak.net>",
};
static GstElementDetails ffmpegcsp_details =
GST_ELEMENT_DETAILS ("FFMPEG Colorspace converter",
"Filter/Converter/Video",
"Converts video from one colorspace to another",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
/* Stereo signals and args */

View file

@ -142,12 +142,11 @@ static void queue_filled_cb (GstElement * queue, GstDecodeBin * decode_bin);
static GstElementClass *parent_class;
static guint gst_decode_bin_signals[LAST_SIGNAL] = { 0 };
static GstElementDetails gst_decode_bin_details = {
"Decoder Bin",
"Generic/Bin/Decoder",
"Autoplug and decode to raw media",
"Wim Taymans <wim@fluendo.com>"
};
static GstElementDetails gst_decode_bin_details =
GST_ELEMENT_DETAILS ("Decoder Bin",
"Generic/Bin/Decoder",
"Autoplug and decode to raw media",
"Wim Taymans <wim@fluendo.com>");
static GType

View file

@ -128,12 +128,11 @@ static GstElementClass *parent_class;
//static guint gst_play_bin_signals[LAST_SIGNAL] = { 0 };
static GstElementDetails gst_play_bin_details = {
"Player Bin",
"Generic/Bin/Player",
"Autoplug and play media from an uri",
"Wim Taymans <wim@fluendo.com>"
};
static GstElementDetails gst_play_bin_details =
GST_ELEMENT_DETAILS ("Player Bin",
"Generic/Bin/Player",
"Autoplug and play media from an uri",
"Wim Taymans <wim@fluendo.com>");
static GType
gst_play_bin_get_type (void)

View file

@ -37,6 +37,14 @@
GST_DEBUG_CATEGORY_STATIC (stream_selector_debug);
#define GST_CAT_DEFAULT stream_selector_debug
static GstElementDetails gst_stream_selector_details =
GST_ELEMENT_DETAILS ("StreamSelector",
"Generic",
"N-to-1 input stream_selectoring",
"Julien Moutte <julien@moutte.net>\n"
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>\n"
"Jan Schmidt <thaytan@mad.scientist.com>");
static GstStaticPadTemplate gst_stream_selector_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink%d",
GST_PAD_SINK,
@ -106,13 +114,6 @@ static void
gst_stream_selector_base_init (GstStreamSelectorClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails gst_stream_selector_details =
GST_ELEMENT_DETAILS ("StreamSelector",
"Generic",
"N-to-1 input stream_selectoring",
"Julien Moutte <julien@moutte.net>\n"
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>\n"
"Jan Schmidt <thaytan@mad.scientist.com>");
gst_element_class_set_details (element_class, &gst_stream_selector_details);

View file

@ -33,6 +33,13 @@
GST_DEBUG_CATEGORY_STATIC (sub_parse_debug);
#define GST_CAT_DEFAULT sub_parse_debug
static GstElementDetails sub_parse_details =
GST_ELEMENT_DETAILS ("Subtitle parsers",
"Codec/Parser/Subtitle",
"Parses subtitle (.sub) files into text streams",
"Gustavo J. A. M. Carneiro <gjc@inescporto.pt>\n"
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>");
static GstStaticPadTemplate sink_templ = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
@ -88,13 +95,6 @@ static void
gst_sub_parse_base_init (GstSubParseClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
static GstElementDetails sub_parse_details = {
"Subtitle parsers",
"Codec/Parser/Subtitle",
"Parses subtitle (.sub) files into text streams",
"Gustavo J. A. M. Carneiro <gjc@inescporto.pt>\n"
"Ronald S. Bultje <rbultje@ronald.bitfreak.net>"
};
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_templ));

View file

@ -66,12 +66,10 @@
/* number of steps we use for the mixer interface to go from 0.0 to 1.0 */
# define VOLUME_STEPS 100
static GstElementDetails volume_details = {
"Volume",
"Filter/Effect/Audio",
"Set volume on audio/raw streams",
"Andy Wingo <wingo@pobox.com>",
};
static GstElementDetails volume_details = GST_ELEMENT_DETAILS ("Volume",
"Filter/Effect/Audio",
"Set volume on audio/raw streams",
"Andy Wingo <wingo@pobox.com>");
/* Filter signals and args */
enum

View file

@ -31,12 +31,11 @@ GST_DEBUG_CATEGORY (v4lmjpegsink_debug);
#define GST_CAT_DEFAULT v4lmjpegsink_debug
/* elementfactory information */
static GstElementDetails gst_v4lmjpegsink_details = {
"Video (video4linux/MJPEG) sink",
"Sink/Video",
"Writes MJPEG-encoded frames to a zoran MJPEG/video4linux device",
"Ronald Bultje <rbultje@ronald.bitfreak.net>"
};
static GstElementDetails gst_v4lmjpegsink_details =
GST_ELEMENT_DETAILS ("Video (video4linux/MJPEG) sink",
"Sink/Video",
"Writes MJPEG-encoded frames to a zoran MJPEG/video4linux device",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
/* v4lmjpegsink signals and args */
enum

View file

@ -31,12 +31,11 @@ GST_DEBUG_CATEGORY (v4lmjpegsrc_debug);
#define GST_CAT_DEFAULT v4lmjpegsrc_debug
/* elementfactory information */
static GstElementDetails gst_v4lmjpegsrc_details = {
"Video (video4linux/MJPEG) Source",
"Source/Video",
"Reads MJPEG-encoded frames from a zoran MJPEG/video4linux device",
"Ronald Bultje <rbultje@ronald.bitfreak.net>"
};
static GstElementDetails gst_v4lmjpegsrc_details =
GST_ELEMENT_DETAILS ("Video (video4linux/MJPEG) Source",
"Source/Video",
"Reads MJPEG-encoded frames from a zoran MJPEG/video4linux device",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
/* V4lMjpegSrc signals and args */
enum

View file

@ -2,7 +2,7 @@
*
* unit test for cddabasesrc
*
* Copyright (C) <2005> Tim-Philipp Müller <tim centricular net>
* Copyright (C) <2005> Tim-Philipp M??ller <tim centricular net>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@ -184,12 +184,11 @@ static gboolean gst_cd_foo_src_open (GstCddaBaseSrc * src,
const gchar * device);
static void gst_cd_foo_src_close (GstCddaBaseSrc * src);
static GstElementDetails cdfoo_details = {
"CD Audio (cdda) Source, FooBar",
"Source/File",
"Read audio from CD",
"Foo Bar <foo@bar.com>"
};
static GstElementDetails cdfoo_details =
GST_ELEMENT_DETAILS ("CD Audio (cdda) Source, FooBar",
"Source/File",
"Read audio from CD",
"Foo Bar <foo@bar.com>");
static void
gst_cd_foo_src_base_init (gpointer g_class)

View file

@ -11,14 +11,12 @@ class Gst:Identity2 from Gst:Element {
private gboolean
plugin_init (GModule *module, GstPlugin *plugin)
{
static GstElementDetails identity2_details = {
static GstElementDetails identity2_details =
GST_ELEMENT_DETAILS (
"GOB Identity",
"Filter/Effect",
"Does nothing",
"1.0",
"Wim Taymans <wim.taymans@chello.be>",
"(C) 2001",
};
"Wim Taymans <wim.taymans@chello.be>");
GstElementFactory *factory;
factory = gst_elementfactory_new ("identity2", TYPE_SELF,