mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-28 12:41:05 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/audiofx/audiochebyshevfreqlimit.c
Sebastian Dröge 5f32a4bac6 gst/audiofx/: Add small comparision with the windowed sinc filters in the docs. 
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqlimit.c:
Add small comparision with the windowed sinc filters in the docs.
2007-08-19 19:11:04 +00:00

823 lines
24 KiB
C
Raw Blame History

/*
* GStreamer
* Copyright (C) 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.
*/
/*
* Chebyshev type 1 filter design based on
* "The Scientist and Engineer's Guide to DSP", Chapter 20.
* http://www.dspguide.com/
*
* For type 2 and Chebyshev filters in general read
* http://en.wikipedia.org/wiki/Chebyshev_filter
*
*/
/**
* SECTION:element-audiochebyshevfreqlimit
* @short_description: Chebyshev low pass and high pass filter
*
* <refsect2>
* <para>
* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
* cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
* </para>
* <para>
* This element has the advantage over the windowed sinc lowpass and highpass filter that it is
* much faster and produces almost as good results. It's only disadvantages are the highly
* non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
* </para>
* <para>
* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
* a faster rolloff.
* </para>
* <para>
* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
* be at most this value. A lower ripple value will allow a faster rolloff.
* </para>
* <para>
* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
* </para>
* <para><note>
* Be warned that a too large number of poles can produce noise. The most poles are possible with
* a cutoff frequency at a quarter of the sampling rate.
* </note></para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>
#include <gst/controller/gstcontroller.h>
#include <math.h>
#include "audiochebyshevfreqlimit.h"
#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static const GstElementDetails element_details =
GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
"Filter/Effect/Audio",
"Chebyshev low pass and high pass filter",
"Sebastian Dröge <slomo@circular-chaos.org>");
/* Filter signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
PROP_0,
PROP_MODE,
PROP_TYPE,
PROP_CUTOFF,
PROP_RIPPLE,
PROP_POLES
};
#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) \
GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
DEBUG_INIT);
static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
GstRingBufferSpec * format);
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
static void process_64 (GstAudioChebyshevFreqLimit * filter,
gdouble * data, guint num_samples);
static void process_32 (GstAudioChebyshevFreqLimit * filter,
gfloat * data, guint num_samples);
enum
{
MODE_LOW_PASS = 0,
MODE_HIGH_PASS
};
#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
static GType
gst_audio_chebyshev_freq_limit_mode_get_type (void)
{
static GType gtype = 0;
if (gtype == 0) {
static const GEnumValue values[] = {
{MODE_LOW_PASS, "Low pass (default)",
"low-pass"},
{MODE_HIGH_PASS, "High pass",
"high-pass"},
{0, NULL, NULL}
};
gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values);
}
return gtype;
}
/* GObject vmethod implementations */
static void
gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstCaps *caps;
gst_element_class_set_details (element_class, &element_details);
caps = gst_caps_from_string (ALLOWED_CAPS);
gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
caps);
gst_caps_unref (caps);
}
static void
gst_audio_chebyshev_freq_limit_dispose (GObject * object)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
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 = gst_audio_chebyshev_freq_limit_set_property;
gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Low pass or high pass mode",
GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_TYPE,
g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: Don't use the complete possible range but restrict the upper boundary
* so automatically generated UIs can use a slider without */
g_object_class_install_property (gobject_class, PROP_CUTOFF,
g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
100000.0, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
g_object_class_install_property (gobject_class, PROP_RIPPLE,
g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
200.0, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
/* FIXME: What to do about this upper boundary? With a cutoff frequency of
* rate/4 32 poles are completely possible, with a cutoff frequency very low
* or very high 16 poles already produces only noise */
g_object_class_install_property (gobject_class, PROP_POLES,
g_param_spec_int ("poles", "Poles",
"Number of poles to use, will be rounded up to the next even number",
2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
filter_class->setup =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
trans_class->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
}
static void
gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitClass * klass)
{
filter->cutoff = 0.0;
filter->mode = MODE_LOW_PASS;
filter->type = 1;
filter->poles = 4;
filter->ripple = 0.25;
gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
filter->have_coeffs = FALSE;
filter->num_a = 0;
filter->num_b = 0;
filter->channels = NULL;
}
static void
generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter,
gint p, gdouble * a0, gdouble * a1, gdouble * a2,
gdouble * b1, gdouble * b2)
{
gint np = filter->poles;
gdouble ripple = filter->ripple;
/* pole location in s-plane */
gdouble rp, ip;
/* zero location in s-plane */
gdouble rz = 0.0, iz = 0.0;
/* transfer function coefficients for the z-plane */
gdouble x0, x1, x2, y1, y2;
gint type = filter->type;
/* Calculate pole location for lowpass at frequency 1 */
{
gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;
rp = -sin (angle);
ip = cos (angle);
}
/* If we allow ripple, move the pole from the unit
* circle to an ellipse and keep cutoff at frequency 1 */
if (ripple > 0 && type == 1) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (1.0 / es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
} else if (type == 2) {
gdouble es, vx;
es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
vx = (1.0 / np) * asinh (es);
rp = rp * sinh (vx);
ip = ip * cosh (vx);
}
/* Calculate inverse of the pole location to convert from
* type I to type II */
if (type == 2) {
gdouble mag2 = rp * rp + ip * ip;
rp /= mag2;
ip /= mag2;
}
/* Calculate zero location for frequency 1 on the
* unit circle for type 2 */
if (type == 2) {
gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);
gdouble mag2;
rz = 0.0;
iz = cos (angle);
mag2 = rz * rz + iz * iz;
rz /= mag2;
iz /= mag2;
}
/* Convert from s-domain to z-domain by
* using the bilinear Z-transform, i.e.
* substitute s by (2/t)*((z-1)/(z+1))
* with t = 2 * tan(0.5).
*/
if (type == 1) {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t) / d;
x1 = 2.0 * x0;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
} else {
gdouble t, m, d;
t = 2.0 * tan (0.5);
m = rp * rp + ip * ip;
d = 4.0 - 4.0 * rp * t + m * t * t;
x0 = (t * t * iz * iz + 4.0) / d;
x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
x2 = x0;
y1 = (8.0 - 2.0 * m * t * t) / d;
y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
}
/* Convert from lowpass at frequency 1 to either lowpass
* or highpass.
*
* For lowpass substitute z^(-1) with:
* -1
* z - k
* ------------
* -1
* 1 - k * z
*
* k = sin((1-w)/2) / sin((1+w)/2)
*
* For highpass substitute z^(-1) with:
*
* -1
* -z - k
* ------------
* -1
* 1 + k * z
*
* k = -cos((1+w)/2) / cos((1-w)/2)
*
*/
{
gdouble k, d;
gdouble omega =
2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
if (filter->mode == MODE_LOW_PASS)
k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
else
k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
d = 1.0 + y1 * k - y2 * k * k;
*a0 = (x0 + k * (-x1 + k * x2)) / d;
*a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
*a2 = (x0 * k * k - x1 * k + x2) / d;
*b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
*b2 = (-k * k - y1 * k + y2) / d;
if (filter->mode == MODE_HIGH_PASS) {
*a1 = -*a1;
*b1 = -*b1;
}
}
}
/* Evaluate the transfer function that corresponds to the IIR
* coefficients at zr + zi*I and return the magnitude */
static gdouble
calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
gdouble zi)
{
gdouble sum_ar, sum_ai;
gdouble sum_br, sum_bi;
gdouble gain_r, gain_i;
gdouble sum_r_old;
gdouble sum_i_old;
gint i;
sum_ar = 0.0;
sum_ai = 0.0;
for (i = num_a; i >= 0; i--) {
sum_r_old = sum_ar;
sum_i_old = sum_ai;
sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
}
sum_br = 0.0;
sum_bi = 0.0;
for (i = num_b; i >= 0; i--) {
sum_r_old = sum_br;
sum_i_old = sum_bi;
sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
}
sum_br += 1.0;
sum_bi += 0.0;
gain_r =
(sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
gain_i =
(sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
return (sqrt (gain_r * gain_r + gain_i * gain_i));
}
static void
generate_coefficients (GstAudioChebyshevFreqLimit * filter)
{
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
if (filter->a) {
g_free (filter->a);
filter->a = NULL;
}
if (filter->b) {
g_free (filter->b);
filter->b = NULL;
}
if (filter->channels) {
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
g_free (ctx->x);
g_free (ctx->y);
}
g_free (filter->channels);
filter->channels = NULL;
}
if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "rate was not set yet");
return;
}
filter->have_coeffs = TRUE;
if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
return;
} else if (filter->cutoff <= 0.0) {
filter->num_a = 1;
filter->a = g_new0 (gdouble, 1);
filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
filter->num_b = 0;
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
GST_LOG_OBJECT (filter, "cutoff is lower than zero");
return;
}
/* Calculate coefficients for the chebyshev filter */
{
gint np = filter->poles;
gdouble *a, *b;
gint i, p;
filter->num_a = np + 1;
filter->a = a = g_new0 (gdouble, np + 3);
filter->num_b = np + 1;
filter->b = b = g_new0 (gdouble, np + 3);
filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
for (i = 0; i < channels; i++) {
GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
ctx->x = g_new0 (gdouble, np + 1);
ctx->y = g_new0 (gdouble, np + 1);
}
/* Calculate transfer function coefficients */
a[2] = 1.0;
b[2] = 1.0;
for (p = 1; p <= np / 2; p++) {
gdouble a0, a1, a2, b1, b2;
gdouble *ta = g_new0 (gdouble, np + 3);
gdouble *tb = g_new0 (gdouble, np + 3);
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
memcpy (ta, a, sizeof (gdouble) * (np + 3));
memcpy (tb, b, sizeof (gdouble) * (np + 3));
/* add the new coefficients for the new two poles
* to the cascade by multiplication of the transfer
* functions */
for (i = 2; i < np + 3; i++) {
a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
}
g_free (ta);
g_free (tb);
}
/* Move coefficients to the beginning of the array
* and multiply the b coefficients with -1 to move from
* the transfer function's coefficients to the difference
* equation's coefficients */
b[2] = 0.0;
for (i = 0; i <= np; i++) {
a[i] = a[i + 2];
b[i] = -b[i + 2];
}
/* Normalize to unity gain at frequency 0 for lowpass
* and frequency 0.5 for highpass */
{
gdouble gain;
if (filter->mode == MODE_LOW_PASS)
gain = calculate_gain (a, b, np, np, 1.0, 0.0);
else
gain = calculate_gain (a, b, np, np, -1.0, 0.0);
for (i = 0; i <= np; i++) {
a[i] /= gain;
}
}
GST_LOG_OBJECT (filter,
"Generated IIR coefficients for the Chebyshev filter");
GST_LOG_OBJECT (filter,
"mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB",
(filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
filter->type, filter->poles, filter->cutoff, filter->ripple);
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0)));
{
gdouble wc =
2.0 * M_PI * (filter->cutoff /
GST_AUDIO_FILTER (filter)->format.rate);
gdouble zr = cos (wc), zi = sin (wc);
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
(int) filter->cutoff);
}
GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
GST_AUDIO_FILTER (filter)->format.rate / 2);
}
}
static void
gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->mode = g_value_get_enum (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_TYPE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->type = g_value_get_int (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_CUTOFF:
GST_BASE_TRANSFORM_LOCK (filter);
filter->cutoff = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_RIPPLE:
GST_BASE_TRANSFORM_LOCK (filter);
filter->ripple = g_value_get_float (value);
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
case PROP_POLES:
GST_BASE_TRANSFORM_LOCK (filter);
filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
generate_coefficients (filter);
GST_BASE_TRANSFORM_UNLOCK (filter);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, filter->mode);
break;
case PROP_TYPE:
g_value_set_int (value, filter->type);
break;
case PROP_CUTOFF:
g_value_set_float (value, filter->cutoff);
break;
case PROP_RIPPLE:
g_value_set_float (value, filter->ripple);
break;
case PROP_POLES:
g_value_set_int (value, filter->poles);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstAudioFilter vmethod implementations */
static gboolean
gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base,
GstRingBufferSpec * format)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gboolean ret = TRUE;
if (format->width == 32)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_32;
else if (format->width == 64)
filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
process_64;
else
ret = FALSE;
filter->have_coeffs = FALSE;
return ret;
}
static inline gdouble
process (GstAudioChebyshevFreqLimit * filter,
GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
{
gdouble val = filter->a[0] * x0;
gint i, j;
for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
val += filter->a[i] * ctx->x[j];
j--;
if (j < 0)
j = filter->num_a - 1;
}
for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
val += filter->b[i] * ctx->y[j];
j--;
if (j < 0)
j = filter->num_b - 1;
}
if (ctx->x) {
ctx->x_pos++;
if (ctx->x_pos > filter->num_a - 1)
ctx->x_pos = 0;
ctx->x[ctx->x_pos] = x0;
}
if (ctx->y) {
ctx->y_pos++;
if (ctx->y_pos > filter->num_b - 1)
ctx->y_pos = 0;
ctx->y[ctx->y_pos] = val;
}
return val;
}
#define DEFINE_PROCESS_FUNC(width,ctype) \
static void \
process_##width (GstAudioChebyshevFreqLimit * filter, \
g##ctype * data, guint num_samples) \
{ \
gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; \
gdouble val; \
\
for (i = 0; i < num_samples / channels; i++) { \
for (j = 0; j < channels; j++) { \
val = process (filter, &filter->channels[j], *data); \
*data++ = val; \
} \
} \
}
DEFINE_PROCESS_FUNC (32, float);
DEFINE_PROCESS_FUNC (64, double);
#undef DEFINE_PROCESS_FUNC
/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
GstBuffer * buf)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
guint num_samples =
GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
if (!gst_buffer_is_writable (buf))
return GST_FLOW_OK;
if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
if (!filter->have_coeffs)
generate_coefficients (filter);
filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
return GST_FLOW_OK;
}
static gboolean
gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
{
GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
gint channels = GST_AUDIO_FILTER (filter)->format.channels;
GstAudioChebyshevFreqLimitChannelCtx *ctx;
gint i;
/* Reset the history of input and output values if
* already existing */
if (channels && filter->channels) {
for (i = 0; i < channels; i++) {
ctx = &filter->channels[i];
if (ctx->x)
memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
if (ctx->y)
memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
}
}
return TRUE;
}
Reference in a new issue View git blame Copy permalink