3 * Copyright (C) 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
18 * Boston, MA 02111-1307, USA.
22 * Chebyshev type 1 filter design based on
23 * "The Scientist and Engineer's Guide to DSP", Chapter 20.
24 * http://www.dspguide.com/
26 * For type 2 and Chebyshev filters in general read
27 * http://en.wikipedia.org/wiki/Chebyshev_filter
32 * SECTION:element-audiocheblimit
34 * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
35 * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
37 * This element has the advantage over the windowed sinc lowpass and highpass filter that it is
38 * much faster and produces almost as good results. It's only disadvantages are the highly
39 * non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
41 * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
42 * some frequencies in the passband will be amplified by that value. A higher ripple value will allow
45 * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
46 * be at most this value. A lower ripple value will allow a faster rolloff.
48 * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
51 * Be warned that a too large number of poles can produce noise. The most poles are possible with
52 * a cutoff frequency at a quarter of the sampling rate.
56 * <title>Example launch line</title>
58 * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
59 * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
60 * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
70 #include <gst/base/gstbasetransform.h>
71 #include <gst/audio/audio.h>
72 #include <gst/audio/gstaudiofilter.h>
73 #include <gst/controller/gstcontroller.h>
77 #include "math_compat.h"
79 #include "audiocheblimit.h"
81 #include "gst/glib-compat-private.h"
83 #define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
84 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
96 #define DEBUG_INIT(bla) \
97 GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element");
99 GST_BOILERPLATE_FULL (GstAudioChebLimit,
100 gst_audio_cheb_limit, GstAudioFXBaseIIRFilter,
101 GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
103 static void gst_audio_cheb_limit_set_property (GObject * object,
104 guint prop_id, const GValue * value, GParamSpec * pspec);
105 static void gst_audio_cheb_limit_get_property (GObject * object,
106 guint prop_id, GValue * value, GParamSpec * pspec);
107 static void gst_audio_cheb_limit_finalize (GObject * object);
109 static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter,
110 GstRingBufferSpec * format);
118 #define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_cheb_limit_mode_get_type ())
120 gst_audio_cheb_limit_mode_get_type (void)
122 static GType gtype = 0;
125 static const GEnumValue values[] = {
126 {MODE_LOW_PASS, "Low pass (default)",
128 {MODE_HIGH_PASS, "High pass",
133 gtype = g_enum_register_static ("GstAudioChebLimitMode", values);
138 /* GObject vmethod implementations */
141 gst_audio_cheb_limit_base_init (gpointer klass)
143 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
145 gst_element_class_set_details_simple (element_class,
146 "Low pass & high pass filter",
147 "Filter/Effect/Audio",
148 "Chebyshev low pass and high pass filter",
149 "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
153 gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
155 GObjectClass *gobject_class = (GObjectClass *) klass;
156 GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
158 gobject_class->set_property = gst_audio_cheb_limit_set_property;
159 gobject_class->get_property = gst_audio_cheb_limit_get_property;
160 gobject_class->finalize = gst_audio_cheb_limit_finalize;
162 g_object_class_install_property (gobject_class, PROP_MODE,
163 g_param_spec_enum ("mode", "Mode",
164 "Low pass or high pass mode",
165 GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
166 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
167 g_object_class_install_property (gobject_class, PROP_TYPE,
168 g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
169 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
171 /* FIXME: Don't use the complete possible range but restrict the upper boundary
172 * so automatically generated UIs can use a slider without */
173 g_object_class_install_property (gobject_class, PROP_CUTOFF,
174 g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
176 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
177 g_object_class_install_property (gobject_class, PROP_RIPPLE,
178 g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
180 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
182 /* FIXME: What to do about this upper boundary? With a cutoff frequency of
183 * rate/4 32 poles are completely possible, with a cutoff frequency very low
184 * or very high 16 poles already produces only noise */
185 g_object_class_install_property (gobject_class, PROP_POLES,
186 g_param_spec_int ("poles", "Poles",
187 "Number of poles to use, will be rounded up to the next even number",
189 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
191 filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup);
195 gst_audio_cheb_limit_init (GstAudioChebLimit * filter,
196 GstAudioChebLimitClass * klass)
198 filter->cutoff = 0.0;
199 filter->mode = MODE_LOW_PASS;
202 filter->ripple = 0.25;
204 filter->lock = g_mutex_new ();
208 generate_biquad_coefficients (GstAudioChebLimit * filter,
209 gint p, gdouble * a0, gdouble * a1, gdouble * a2,
210 gdouble * b1, gdouble * b2)
212 gint np = filter->poles;
213 gdouble ripple = filter->ripple;
215 /* pole location in s-plane */
218 /* zero location in s-plane */
221 /* transfer function coefficients for the z-plane */
222 gdouble x0, x1, x2, y1, y2;
223 gint type = filter->type;
225 /* Calculate pole location for lowpass at frequency 1 */
227 gdouble angle = (G_PI / 2.0) * (2.0 * p - 1) / np;
233 /* If we allow ripple, move the pole from the unit
234 * circle to an ellipse and keep cutoff at frequency 1 */
235 if (ripple > 0 && type == 1) {
238 es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
240 vx = (1.0 / np) * asinh (1.0 / es);
243 } else if (type == 2) {
246 es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
247 vx = (1.0 / np) * asinh (es);
252 /* Calculate inverse of the pole location to convert from
253 * type I to type II */
255 gdouble mag2 = rp * rp + ip * ip;
261 /* Calculate zero location for frequency 1 on the
262 * unit circle for type 2 */
264 gdouble angle = G_PI / (np * 2.0) + ((p - 1) * G_PI) / (np);
272 /* Convert from s-domain to z-domain by
273 * using the bilinear Z-transform, i.e.
274 * substitute s by (2/t)*((z-1)/(z+1))
275 * with t = 2 * tan(0.5).
281 m = rp * rp + ip * ip;
282 d = 4.0 - 4.0 * rp * t + m * t * t;
287 y1 = (8.0 - 2.0 * m * t * t) / d;
288 y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
293 m = rp * rp + ip * ip;
294 d = 4.0 - 4.0 * rp * t + m * t * t;
296 x0 = (t * t * iz * iz + 4.0) / d;
297 x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
299 y1 = (8.0 - 2.0 * m * t * t) / d;
300 y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
303 /* Convert from lowpass at frequency 1 to either lowpass
306 * For lowpass substitute z^(-1) with:
313 * k = sin((1-w)/2) / sin((1+w)/2)
315 * For highpass substitute z^(-1) with:
323 * k = -cos((1+w)/2) / cos((1-w)/2)
329 2.0 * G_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
331 if (filter->mode == MODE_LOW_PASS)
332 k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
334 k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
336 d = 1.0 + y1 * k - y2 * k * k;
337 *a0 = (x0 + k * (-x1 + k * x2)) / d;
338 *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
339 *a2 = (x0 * k * k - x1 * k + x2) / d;
340 *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
341 *b2 = (-k * k - y1 * k + y2) / d;
343 if (filter->mode == MODE_HIGH_PASS) {
351 generate_coefficients (GstAudioChebLimit * filter)
353 if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
354 gdouble *a = g_new0 (gdouble, 1);
357 gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
358 (filter), a, 1, NULL, 0);
360 GST_LOG_OBJECT (filter, "rate was not set yet");
364 if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
365 gdouble *a = g_new0 (gdouble, 1);
367 a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
368 gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
369 (filter), a, 1, NULL, 0);
370 GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
372 } else if (filter->cutoff <= 0.0) {
373 gdouble *a = g_new0 (gdouble, 1);
375 a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
376 gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
377 (filter), a, 1, NULL, 0);
378 GST_LOG_OBJECT (filter, "cutoff is lower than zero");
382 /* Calculate coefficients for the chebyshev filter */
384 gint np = filter->poles;
388 a = g_new0 (gdouble, np + 3);
389 b = g_new0 (gdouble, np + 3);
391 /* Calculate transfer function coefficients */
395 for (p = 1; p <= np / 2; p++) {
396 gdouble a0, a1, a2, b1, b2;
397 gdouble *ta = g_new0 (gdouble, np + 3);
398 gdouble *tb = g_new0 (gdouble, np + 3);
400 generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
402 memcpy (ta, a, sizeof (gdouble) * (np + 3));
403 memcpy (tb, b, sizeof (gdouble) * (np + 3));
405 /* add the new coefficients for the new two poles
406 * to the cascade by multiplication of the transfer
408 for (i = 2; i < np + 3; i++) {
409 a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
410 b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
416 /* Move coefficients to the beginning of the array
417 * and multiply the b coefficients with -1 to move from
418 * the transfer function's coefficients to the difference
419 * equation's coefficients */
421 for (i = 0; i <= np; i++) {
426 /* Normalize to unity gain at frequency 0 for lowpass
427 * and frequency 0.5 for highpass */
431 if (filter->mode == MODE_LOW_PASS)
433 gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
437 gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
440 for (i = 0; i <= np; i++) {
445 gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
446 (filter), a, np + 1, b, np + 1);
448 GST_LOG_OBJECT (filter,
449 "Generated IIR coefficients for the Chebyshev filter");
450 GST_LOG_OBJECT (filter,
451 "mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB",
452 (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
453 filter->type, filter->poles, filter->cutoff, filter->ripple);
454 GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
455 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
458 #ifndef GST_DISABLE_GST_DEBUG
461 2.0 * G_PI * (filter->cutoff /
462 GST_AUDIO_FILTER (filter)->format.rate);
463 gdouble zr = cos (wc), zi = sin (wc);
465 GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
466 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1,
467 b, np + 1, zr, zi)), (int) filter->cutoff);
471 GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
472 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
474 GST_AUDIO_FILTER (filter)->format.rate / 2);
479 gst_audio_cheb_limit_finalize (GObject * object)
481 GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
483 g_mutex_free (filter->lock);
486 G_OBJECT_CLASS (parent_class)->finalize (object);
490 gst_audio_cheb_limit_set_property (GObject * object, guint prop_id,
491 const GValue * value, GParamSpec * pspec)
493 GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
497 g_mutex_lock (filter->lock);
498 filter->mode = g_value_get_enum (value);
499 generate_coefficients (filter);
500 g_mutex_unlock (filter->lock);
503 g_mutex_lock (filter->lock);
504 filter->type = g_value_get_int (value);
505 generate_coefficients (filter);
506 g_mutex_unlock (filter->lock);
509 g_mutex_lock (filter->lock);
510 filter->cutoff = g_value_get_float (value);
511 generate_coefficients (filter);
512 g_mutex_unlock (filter->lock);
515 g_mutex_lock (filter->lock);
516 filter->ripple = g_value_get_float (value);
517 generate_coefficients (filter);
518 g_mutex_unlock (filter->lock);
521 g_mutex_lock (filter->lock);
522 filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
523 generate_coefficients (filter);
524 g_mutex_unlock (filter->lock);
527 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
533 gst_audio_cheb_limit_get_property (GObject * object, guint prop_id,
534 GValue * value, GParamSpec * pspec)
536 GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
540 g_value_set_enum (value, filter->mode);
543 g_value_set_int (value, filter->type);
546 g_value_set_float (value, filter->cutoff);
549 g_value_set_float (value, filter->ripple);
552 g_value_set_int (value, filter->poles);
555 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
560 /* GstAudioFilter vmethod implementations */
563 gst_audio_cheb_limit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
565 GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
567 generate_coefficients (filter);
569 return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);