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

183 lines
5.6 KiB
C
Raw Normal View History

/* GStreamer audio filter base class
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2003> David Schleef <ds@schleef.org>
* Copyright (C) <2007> Tim-Philipp Müller <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
* 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.
*/
/**
* SECTION:gstaudiofilter
* @short_description: Base class for simple audio filters
*
* #GstAudioFilter is a #GstBaseTransform-derived base class for simple audio
* filters, ie. those that output the same format that they get as input.
*
* #GstAudioFilter will parse the input format for you (with error checking)
* before calling your setup function. Also, elements deriving from
* #GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from
* their base_init function to easily configure the set of caps/formats that
* the element is able to handle.
*
* Derived classes should override the GstAudioFilter::setup() and
* GstBaseTransform::transform_ip() and/or GstBaseTransform::transform()
* virtual functions in their class_init function.
*
* Since: 0.10.12
*
* Last reviewed on 2007-02-03 (0.10.11.1)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstaudiofilter.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (audiofilter_dbg);
#define GST_CAT_DEFAULT audiofilter_dbg
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_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 base class",
"Filter/Effect/Audio",
"Filters audio",
"David Schleef <ds@schleef.org>");
static void gst_audio_filter_base_init (gpointer g_class);
static void gst_audio_filter_class_init (gpointer g_class, gpointer class_data);
static void gst_audio_filter_init (GTypeInstance * instance, gpointer g_class);
static gboolean gst_audio_filter_set_caps (GstBaseTransform * btrans,
GstCaps * incaps, GstCaps * outcaps);
static GstElementClass *parent_class = NULL;
GType
gst_audio_filter_get_type (void)
{
static GType audio_filter_type = 0;
if (!audio_filter_type) {
const GTypeInfo audio_filter_info = {
sizeof (GstAudioFilterClass),
gst_audio_filter_base_init,
NULL,
gst_audio_filter_class_init,
NULL,
NULL,
sizeof (GstAudioFilter),
0,
gst_audio_filter_init,
};
GST_DEBUG_CATEGORY_INIT (audiofilter_dbg, "audiofilter", 0, "audiofilter");
audio_filter_type = g_type_register_static (GST_TYPE_BASE_TRANSFORM,
"GstAudioFilter", &audio_filter_info, G_TYPE_FLAG_ABSTRACT);
}
return audio_filter_type;
}
static void
gst_audio_filter_base_init (gpointer g_class)
{
GstAudioFilterClass *klass = (GstAudioFilterClass *) g_class;
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_set_details (element_class, &audio_filter_details);
}
static void
gst_audio_filter_class_init (gpointer klass, gpointer class_data)
{
GstBaseTransformClass *basetrans_class;
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_class_init): * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init): * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init): * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_class_init): * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init): * gst-libs/gst/interfaces/colorbalancechannel.c: (gst_color_balance_channel_class_init): * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/interfaces/tunerchannel.c: (gst_tuner_channel_class_init): * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init): * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_stream_selector_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * sys/v4l/gstv4lcolorbalance.c: (gst_v4l_color_balance_channel_class_init): * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init), (gst_v4l_tuner_norm_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:02:53 +00:00
parent_class = g_type_class_peek_parent (klass);
basetrans_class = (GstBaseTransformClass *) klass;
basetrans_class->set_caps = GST_DEBUG_FUNCPTR (gst_audio_filter_set_caps);
}
static void
gst_audio_filter_init (GTypeInstance * instance, gpointer g_class)
{
GstAudioFilter *filter = GST_AUDIO_FILTER (instance);
/* to make gst_buffer_spec_parse_caps() happy, not used in our case */
filter->format.latency_time = GST_SECOND;
}
static gboolean
gst_audio_filter_set_caps (GstBaseTransform * btrans, GstCaps * incaps,
GstCaps * outcaps)
{
GstAudioFilterClass *klass;
GstAudioFilter *filter;
gboolean ret = TRUE;
g_assert (gst_caps_is_equal (incaps, outcaps));
filter = GST_AUDIO_FILTER (btrans);
GST_LOG_OBJECT (filter, "caps: %" GST_PTR_FORMAT, incaps);
if (!gst_ring_buffer_parse_caps (&filter->format, incaps)) {
GST_WARNING_OBJECT (filter, "couldn't parse %" GST_PTR_FORMAT, incaps);
return FALSE;
}
klass = GST_AUDIO_FILTER_CLASS (G_OBJECT_GET_CLASS (filter));
if (klass->setup)
ret = klass->setup (filter, &filter->format);
return ret;
}
/**
* gst_audio_filter_class_add_pad_templates:
* @klass: an #GstAudioFilterClass
* @allowed_caps: what formats the filter can handle, as #GstCaps
*
* Convenience function to add pad templates to this element class, with
* @allowed_caps as the caps that can be handled.
*
* This function is usually used from within a GObject base_init function.
*
* Since: 0.10.12
*/
void
gst_audio_filter_class_add_pad_templates (GstAudioFilterClass * klass,
const GstCaps * allowed_caps)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
g_return_if_fail (GST_IS_AUDIO_FILTER_CLASS (klass));
g_return_if_fail (allowed_caps != NULL);
g_return_if_fail (GST_IS_CAPS (allowed_caps));
gst_element_class_add_pad_template (element_class,
gst_pad_template_new ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
gst_caps_copy (allowed_caps)));
gst_element_class_add_pad_template (element_class,
gst_pad_template_new ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
gst_caps_copy (allowed_caps)));
}