gstreamer/gst-libs/gst/audio/gstaudiofiltertemplate.c

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2003> David Schleef <ds@schleef.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* This file was (probably) generated from
* $Id$
* and
* MAKEFILTERVERSION
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audio_filter_template_debug);
#define GST_CAT_DEFAULT audio_filter_template_debug
make GstElementDetails const Original commit message from CVS: * ext/alsa/gstalsamixerelement.c: * ext/alsa/gstalsasrc.c: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: * ext/ogg/gstogmparse.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: * gst-libs/gst/audio/gstaudiofiltertemplate.c: * gst/audioconvert/gstaudioconvert.c: * gst/audiorate/gstaudiorate.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: * gst/subparse/gstsubparse.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpclientsink.c: * gst/tcp/gsttcpclientsrc.c: * gst/tcp/gsttcpserversink.c: * gst/tcp/gsttcpserversrc.c: * gst/typefind/gsttypefindfunctions.c: (plugin_init): * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/v4l/gstv4ljpegsrc.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * sys/v4l/gstv4lsrc.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: * tests/check/libs/cddabasesrc.c: make GstElementDetails const
2006-04-28 19:46:37 +00:00
static const GstElementDetails audio_filter_template_details =
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
2006-03-24 10:42:11 +00:00
GST_ELEMENT_DETAILS ("Audio filter template",
"Filter/Effect/Audio",
"Filters audio",
"David Schleef <ds@schleef.org>");
typedef struct _GstAudioFilterTemplate GstAudioFilterTemplate;
typedef struct _GstAudioFilterTemplateClass GstAudioFilterTemplateClass;
#define GST_TYPE_AUDIO_FILTER_TEMPLATE \
(gst_audio_filter_template_get_type())
#define GST_AUDIO_FILTER_TEMPLATE(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_FILTER_TEMPLATE,GstAudioFilterTemplate))
#define GST_AUDIO_FILTER_TEMPLATE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_FILTER_TEMPLATE,GstAudioFilterTemplateClass))
#define GST_IS_AUDIO_FILTER_TEMPLATE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_FILTER_TEMPLATE))
#define GST_IS_AUDIO_FILTER_TEMPLATE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_FILTER_TEMPLATE))
struct _GstAudioFilterTemplate
{
GstAudioFilter audiofilter;
};
struct _GstAudioFilterTemplateClass
{
GstAudioFilterClass parent_class;
};
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0
/* FILL ME */
};
GST_BOILERPLATE (GstAudioFilterTemplate, gst_audio_filter_template,
GstAudioFilter, GST_TYPE_AUDIO_FILTER);
static void gst_audio_filter_template_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_filter_template_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_filter_template_setup (GstAudioFilter * filter,
GstRingBufferSpec * spec);
static GstFlowReturn gst_audio_filter_template_filter (GstBaseTransform * bt,
GstBuffer * outbuf, GstBuffer * inbuf);
static GstFlowReturn
gst_audio_filter_template_filter_inplace (GstBaseTransform * base_transform,
GstBuffer * buf);
#define ALLOWED_CAPS_STRING \
GST_AUDIO_INT_STANDARD_PAD_TEMPLATE_CAPS
static void
gst_audio_filter_template_base_init (gpointer g_class)
{
GstAudioFilterTemplateClass *klass = (GstAudioFilterTemplateClass *) g_class;
GstAudioFilterClass *audiofilter_class = GST_AUDIO_FILTER_CLASS (g_class);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &audio_filter_template_details);
caps = gst_caps_from_string (ALLOWED_CAPS_STRING);
gst_audio_filter_class_add_pad_templates (audiofilter_class, caps);
gst_caps_unref (caps);
}
static void
gst_audio_filter_template_class_init (GstAudioFilterTemplateClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *btrans_class;
GstAudioFilterClass *audio_filter_class;
gobject_class = (GObjectClass *) klass;
btrans_class = (GstBaseTransformClass *) klass;
audio_filter_class = (GstAudioFilterClass *) klass;
#if 0
g_object_class_install_property (gobject_class, ARG_METHOD,
g_param_spec_enum ("method", "method", "method",
GST_TYPE_AUDIOTEMPLATE_METHOD, GST_AUDIOTEMPLATE_METHOD_1,
G_PARAM_READWRITE));
#endif
gobject_class->set_property = gst_audio_filter_template_set_property;
gobject_class->get_property = gst_audio_filter_template_get_property;
/* this function will be called whenever the format changes */
audio_filter_class->setup = gst_audio_filter_template_setup;
/* here you set up functions to process data (either in place, or from
* one input buffer to another output buffer); only one is required */
btrans_class->transform = gst_audio_filter_template_filter;
btrans_class->transform_ip = gst_audio_filter_template_filter_inplace;
}
static void
gst_audio_filter_template_init (GstAudioFilterTemplate * audio_filter_template,
GstAudioFilterTemplateClass * g_class)
{
GST_DEBUG ("init");
/* do stuff if you need to */
}
static void
gst_audio_filter_template_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioFilterTemplate *filter;
filter = GST_AUDIO_FILTER_TEMPLATE (object);
GST_DEBUG ("set property %u", prop_id);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (filter);
}
static void
gst_audio_filter_template_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioFilterTemplate *filter;
filter = GST_AUDIO_FILTER_TEMPLATE (object);
GST_DEBUG ("get property %u", prop_id);
GST_OBJECT_LOCK (filter);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (filter);
}
static gboolean
gst_audio_filter_template_setup (GstAudioFilter * filter,
GstRingBufferSpec * spec)
{
GstAudioFilterTemplate *audio_filter_template;
audio_filter_template = GST_AUDIO_FILTER_TEMPLATE (filter);
/* if any setup needs to be done, do it here */
return TRUE; /* it's all good */
}
/* You may choose to implement either a copying filter or an
* in-place filter (or both). Implementing only one will give
* full functionality, however, implementing both will cause
* audiofilter to use the optimal function in every situation,
* with a minimum of memory copies. */
static GstFlowReturn
gst_audio_filter_template_filter (GstBaseTransform * base_transform,
GstBuffer * inbuf, GstBuffer * outbuf)
{
GstAudioFilterTemplate *audio_filter_template;
GstAudioFilter *audiofilter;
audiofilter = GST_AUDIO_FILTER (base_transform);
audio_filter_template = GST_AUDIO_FILTER_TEMPLATE (base_transform);
/* do something interesting here. This simply copies the source
* to the destination. */
memcpy (GST_BUFFER_DATA (outbuf), GST_BUFFER_DATA (inbuf),
GST_BUFFER_SIZE (inbuf));
return GST_FLOW_OK;
}
static GstFlowReturn
gst_audio_filter_template_filter_inplace (GstBaseTransform * base_transform,
GstBuffer * buf)
{
GstAudioFilterTemplate *audio_filter_template;
GstAudioFilter *audiofilter;
audiofilter = GST_AUDIO_FILTER (base_transform);
audio_filter_template = GST_AUDIO_FILTER_TEMPLATE (base_transform);
/* do something interesting here. This simply copies the source
* to the destination. */
return GST_FLOW_OK;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (audio_filter_template_debug, "audiofiltertemplate",
0, "audiofiltertemplate");
return gst_element_register (plugin, "audiofiltertemplate", GST_RANK_NONE,
GST_TYPE_AUDIO_FILTER_TEMPLATE);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"gstaudio_filter_template",
"Audio filter template",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);