gstreamer/gst/audiofx/audiowsincband.c

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/* -*- c-basic-offset: 2 -*-
*
* GStreamer
* Copyright (C) 1999-2001 Erik Walthinsen <omega@cse.ogi.edu>
* 2006 Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>
* 2007 Sebastian Dröge <slomo@circular-chaos.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 windowed sinc filter is taken from the freely downloadable DSP book,
* "The Scientist and Engineer's Guide to Digital Signal Processing",
* chapter 16
* available at http://www.dspguide.com/
*
* TODO: - Implement the convolution in place, probably only makes sense
* when using FFT convolution as currently the convolution itself
* is probably the bottleneck
* - Maybe allow cascading the filter to get a better stopband attenuation.
* Can be done by convolving a filter kernel with itself
Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: * gst/filter/gstbpwsinc.c: * gst/filter/gstbpwsinc.h: * gst/filter/gstlpwsinc.c: * gst/filter/gstlpwsinc.h: Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other docs via make update in docs/plugins.
2007-08-13 13:50:39 +00:00
* - Drop the first kernel_length/2 samples and append the same number of
* samples on EOS as the first few samples are essentialy zero.
*/
/**
* SECTION:element-bpwsinc
* @short_description: Windowed Sinc band pass and band reject filter
*
* <refsect2>
* <para>
* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
* band. The length parameter controls the rolloff, the window parameter
* controls rolloff and stopband attenuation. The Hamming window provides a faster rolloff but a bit
* worse stopband attenuation, the other way around for the Blackman window.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! bpwsinc mode=band-pass lower-frequency=3000 upper-frequency=10000 length=501 window=blackman ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! bpwsinc mode=band-reject lower-frequency=59 upper-frequency=61 length=10001 window=hamming ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! bpwsinc mode=band-pass lower-frequency=1000 upper-frequency=2000 length=31 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <math.h>
#include <gst/gst.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include "gstbpwsinc.h"
#define GST_CAT_DEFAULT gst_bpwsinc_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails bpwsinc_details =
Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: * gst/filter/gstbpwsinc.c: * gst/filter/gstbpwsinc.h: * gst/filter/gstlpwsinc.c: * gst/filter/gstlpwsinc.h: Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other docs via make update in docs/plugins.
2007-08-13 13:50:39 +00:00
GST_ELEMENT_DETAILS ("Band-pass and Band-reject Windowed sinc filter",
"Filter/Effect/Audio",
"Band-pass Windowed sinc filter",
"Thomas <thomas@apestaart.org>, "
"Steven W. Smith, "
"Dreamlab Technologies Ltd. <mathis.hofer@dreamlab.net>, "
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_LENGTH,
PROP_LOWER_FREQUENCY,
PROP_UPPER_FREQUENCY,
PROP_MODE,
PROP_WINDOW
};
enum
{
MODE_BAND_PASS = 0,
MODE_BAND_REJECT
};
#define GST_TYPE_BPWSINC_MODE (gst_bpwsinc_mode_get_type ())
static GType
gst_bpwsinc_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_BAND_PASS, "Band pass (default)",
"band-pass"},
{MODE_BAND_REJECT, "Band reject",
"band-reject"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstBPWSincMode", values);
}
return gtype;
}
enum
{
WINDOW_HAMMING = 0,
WINDOW_BLACKMAN
};
#define GST_TYPE_BPWSINC_WINDOW (gst_bpwsinc_window_get_type ())
static GType
gst_bpwsinc_window_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{WINDOW_HAMMING, "Hamming window (default)",
"hamming"},
{WINDOW_BLACKMAN, "Blackman window",
"blackman"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstBPWSincWindow", values);
}
return gtype;
}
#define ALLOWED_CAPS \
"audio/x-raw-float, " \
" width = (int) { 32, 64 }, " \
" endianness = (int) BYTE_ORDER, " \
" rate = (int) [ 1, MAX ], " \
" channels = (int) [ 1, MAX ] "
#define DEBUG_INIT(bla) \
Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc... Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: * gst/filter/gstbpwsinc.c: * gst/filter/gstbpwsinc.h: * gst/filter/gstlpwsinc.c: * gst/filter/gstlpwsinc.h: Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other docs via make update in docs/plugins.
2007-08-13 13:50:39 +00:00
GST_DEBUG_CATEGORY_INIT (gst_bpwsinc_debug, "bpwsinc", 0, "Band-pass and Band-reject Windowed sinc filter plugin");
GST_BOILERPLATE_FULL (GstBPWSinc, gst_bpwsinc, GstAudioFilter,
GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
static void bpwsinc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void bpwsinc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstFlowReturn bpwsinc_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean bpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps,
guint * size);
static gboolean bpwsinc_start (GstBaseTransform * base);
static gboolean bpwsinc_setup (GstAudioFilter * base,
GstRingBufferSpec * format);
/* Element class */
static void
gst_bpwsinc_dispose (GObject * object)
{
GstBPWSinc *self = GST_BPWSINC (object);
if (self->residue) {
g_free (self->residue);
self->residue = NULL;
}
if (self->kernel) {
g_free (self->kernel);
self->kernel = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_bpwsinc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GstCaps *caps;
gst_element_class_set_details (element_class, &bpwsinc_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
caps);
gst_caps_unref (caps);
}
static void
gst_bpwsinc_class_init (GstBPWSincClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
GstAudioFilterClass *filter_class;
gobject_class = (GObjectClass *) klass;
trans_class = (GstBaseTransformClass *) klass;
filter_class = (GstAudioFilterClass *) klass;
gobject_class->set_property = bpwsinc_set_property;
gobject_class->get_property = bpwsinc_get_property;
gobject_class->dispose = gst_bpwsinc_dispose;
g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,
g_param_spec_double ("lower-frequency", "Lower Frequency",
"Cut-off lower frequency (Hz)",
0.0, G_MAXDOUBLE, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,
g_param_spec_double ("upper-frequency", "Upper Frequency",
"Cut-off upper frequency (Hz)",
0.0, G_MAXDOUBLE, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_LENGTH,
g_param_spec_int ("length", "Length",
"Filter kernel length, will be rounded to the next odd number",
3, G_MAXINT, 101, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Band pass or band reject mode", GST_TYPE_BPWSINC_MODE,
MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_WINDOW,
g_param_spec_enum ("window", "Window",
"Window function to use", GST_TYPE_BPWSINC_WINDOW,
WINDOW_HAMMING, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
trans_class->transform = GST_DEBUG_FUNCPTR (bpwsinc_transform);
trans_class->get_unit_size = GST_DEBUG_FUNCPTR (bpwsinc_get_unit_size);
trans_class->start = GST_DEBUG_FUNCPTR (bpwsinc_start);
filter_class->setup = GST_DEBUG_FUNCPTR (bpwsinc_setup);
}
static void
gst_bpwsinc_init (GstBPWSinc * self, GstBPWSincClass * g_class)
{
self->kernel_length = 101;
self->lower_frequency = 0.0;
self->upper_frequency = 0.0;
self->mode = MODE_BAND_PASS;
self->window = WINDOW_HAMMING;
self->kernel = NULL;
self->have_kernel = FALSE;
self->residue = NULL;
}
static void
process_32 (GstBPWSinc * self, gfloat * src, gfloat * dst, guint input_samples)
{
gint kernel_length = self->kernel_length;
gint i, j, k, l;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint res_start;
/* convolution */
for (i = 0; i < input_samples; i++) {
dst[i] = 0.0;
k = i % channels;
l = i / channels;
for (j = 0; j < kernel_length; j++)
if (l < j)
dst[i] +=
self->residue[(kernel_length + l - j) * channels +
k] * self->kernel[j];
else
dst[i] += src[(l - j) * channels + k] * self->kernel[j];
}
/* copy the tail of the current input buffer to the residue, while
* keeping parts of the residue if the input buffer is smaller than
* the kernel length */
if (input_samples < kernel_length * channels)
res_start = kernel_length * channels - input_samples;
else
res_start = 0;
for (i = 0; i < res_start; i++)
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
}
static void
process_64 (GstBPWSinc * self, gdouble * src, gdouble * dst,
guint input_samples)
{
gint kernel_length = self->kernel_length;
gint i, j, k, l;
gint channels = GST_AUDIO_FILTER (self)->format.channels;
gint res_start;
/* convolution */
for (i = 0; i < input_samples; i++) {
dst[i] = 0.0;
k = i % channels;
l = i / channels;
for (j = 0; j < kernel_length; j++)
if (l < j)
dst[i] +=
self->residue[(kernel_length + l - j) * channels +
k] * self->kernel[j];
else
dst[i] += src[(l - j) * channels + k] * self->kernel[j];
}
/* copy the tail of the current input buffer to the residue, while
* keeping parts of the residue if the input buffer is smaller than
* the kernel length */
if (input_samples < kernel_length * channels)
res_start = kernel_length * channels - input_samples;
else
res_start = 0;
for (i = 0; i < res_start; i++)
self->residue[i] = self->residue[i + input_samples];
for (i = res_start; i < kernel_length * channels; i++)
self->residue[i] = src[input_samples - kernel_length * channels + i];
}
static void
bpwsinc_build_kernel (GstBPWSinc * self)
{
gint i = 0;
gdouble sum = 0.0;
gint len = 0;
gdouble *kernel_lp, *kernel_hp;
gdouble w;
len = self->kernel_length;
if (GST_AUDIO_FILTER (self)->format.rate == 0) {
GST_DEBUG ("rate not set yet");
return;
}
if (GST_AUDIO_FILTER (self)->format.channels == 0) {
GST_DEBUG ("channels not set yet");
return;
}
/* Clamp frequencies */
self->lower_frequency =
CLAMP (self->lower_frequency, 0.0,
GST_AUDIO_FILTER (self)->format.rate / 2);
self->upper_frequency =
CLAMP (self->upper_frequency, 0.0,
GST_AUDIO_FILTER (self)->format.rate / 2);
if (self->lower_frequency > self->upper_frequency) {
gint tmp = self->lower_frequency;
self->lower_frequency = self->upper_frequency;
self->upper_frequency = tmp;
}
GST_DEBUG ("bpwsinc: initializing filter kernel of length %d "
"with lower frequency %.2lf Hz "
", upper frequency %.2lf Hz for mode %s",
len, self->lower_frequency, self->upper_frequency,
(self->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject");
/* fill the lp kernel */
w = 2 * M_PI * (self->lower_frequency / GST_AUDIO_FILTER (self)->format.rate);
kernel_lp = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
kernel_lp[i] = w;
else
kernel_lp[i] = sin (w * (i - len / 2))
/ (i - len / 2);
/* Windowing */
if (self->window == WINDOW_HAMMING)
kernel_lp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
kernel_lp[i] *=
(0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
sum = 0.0;
for (i = 0; i < len; ++i)
sum += kernel_lp[i];
for (i = 0; i < len; ++i)
kernel_lp[i] /= sum;
/* fill the hp kernel */
w = 2 * M_PI * (self->upper_frequency / GST_AUDIO_FILTER (self)->format.rate);
kernel_hp = g_new (gdouble, len);
for (i = 0; i < len; ++i) {
if (i == len / 2)
kernel_hp[i] = w;
else
kernel_hp[i] = sin (w * (i - len / 2))
/ (i - len / 2);
/* Windowing */
if (self->window == WINDOW_HAMMING)
kernel_hp[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / len));
else
kernel_hp[i] *=
(0.42 - 0.5 * cos (2 * M_PI * i / len) +
0.08 * cos (4 * M_PI * i / len));
}
/* normalize for unity gain at DC */
sum = 0.0;
for (i = 0; i < len; ++i)
sum += kernel_hp[i];
for (i = 0; i < len; ++i)
kernel_hp[i] /= sum;
/* do spectral inversion to go from lowpass to highpass */
for (i = 0; i < len; ++i)
kernel_hp[i] = -kernel_hp[i];
kernel_hp[len / 2] += 1;
/* combine the two kernels */
if (self->kernel)
g_free (self->kernel);
self->kernel = g_new (gdouble, len);
for (i = 0; i < len; ++i)
self->kernel[i] = kernel_lp[i] + kernel_hp[i];
/* free the helper kernels */
g_free (kernel_lp);
g_free (kernel_hp);
/* do spectral inversion to go from bandreject to bandpass
* if specified */
if (self->mode == MODE_BAND_PASS) {
for (i = 0; i < len; ++i)
self->kernel[i] = -self->kernel[i];
self->kernel[len / 2] += 1;
}
/* set up the residue memory space */
if (self->residue)
g_free (self->residue);
self->residue =
g_new0 (gdouble, len * GST_AUDIO_FILTER (self)->format.channels);
self->have_kernel = TRUE;
}
/* GstAudioFilter vmethod implementations */
/* get notified of caps and plug in the correct process function */
static gboolean
bpwsinc_setup (GstAudioFilter * base, GstRingBufferSpec * format)
{
GstBPWSinc *self = GST_BPWSINC (base);
gboolean ret = TRUE;
if (format->width == 32)
self->process = (GstBPWSincProcessFunc) process_32;
else if (format->width == 64)
self->process = (GstBPWSincProcessFunc) process_64;
else
ret = FALSE;
self->have_kernel = FALSE;
return TRUE;
}
/* GstBaseTransform vmethod implementations */
static gboolean
bpwsinc_get_unit_size (GstBaseTransform * base, GstCaps * caps, guint * size)
{
gint width, channels;
GstStructure *structure;
gboolean ret;
g_assert (size);
structure = gst_caps_get_structure (caps, 0);
ret = gst_structure_get_int (structure, "width", &width);
ret &= gst_structure_get_int (structure, "channels", &channels);
*size = width * channels / 8;
return ret;
}
static GstFlowReturn
bpwsinc_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstBPWSinc *self = GST_BPWSINC (base);
GstClockTime timestamp;
gint input_samples =
GST_BUFFER_SIZE (outbuf) / (GST_AUDIO_FILTER (self)->format.width / 8);
/* don't process data in passthrough-mode */
if (gst_base_transform_is_passthrough (base))
return GST_FLOW_OK;
/* FIXME: subdivide GST_BUFFER_SIZE into small chunks for smooth fades */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (G_OBJECT (self), timestamp);
if (!self->have_kernel)
bpwsinc_build_kernel (self);
self->process (self, GST_BUFFER_DATA (inbuf), GST_BUFFER_DATA (outbuf),
input_samples);
return GST_FLOW_OK;
}
static gboolean
bpwsinc_start (GstBaseTransform * base)
{
GstBPWSinc *self = GST_BPWSINC (base);
gint channels = GST_AUDIO_FILTER (self)->format.channels;
/* Reset the residue if already existing */
if (channels && self->residue)
memset (self->residue, 0, channels *
self->kernel_length * sizeof (gdouble));
return TRUE;
}
static void
bpwsinc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstBPWSinc *self = GST_BPWSINC (object);
g_return_if_fail (GST_IS_BPWSINC (self));
switch (prop_id) {
case PROP_LENGTH:{
gint val;
GST_BASE_TRANSFORM_LOCK (self);
val = g_value_get_int (value);
if (val % 2 == 0)
val++;
self->kernel_length = val;
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
}
case PROP_LOWER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (self);
self->lower_frequency = g_value_get_double (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_UPPER_FREQUENCY:
GST_BASE_TRANSFORM_LOCK (self);
self->upper_frequency = g_value_get_double (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (self);
self->mode = g_value_get_enum (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
case PROP_WINDOW:
GST_BASE_TRANSFORM_LOCK (self);
self->window = g_value_get_enum (value);
bpwsinc_build_kernel (self);
GST_BASE_TRANSFORM_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
bpwsinc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstBPWSinc *self = GST_BPWSINC (object);
switch (prop_id) {
case PROP_LENGTH:
g_value_set_int (value, self->kernel_length);
break;
case PROP_LOWER_FREQUENCY:
g_value_set_double (value, self->lower_frequency);
break;
case PROP_UPPER_FREQUENCY:
g_value_set_double (value, self->upper_frequency);
break;
case PROP_MODE:
g_value_set_enum (value, self->mode);
break;
case PROP_WINDOW:
g_value_set_enum (value, self->window);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}