diff -r 29ecd5cb86b3 -r d43ce56a1534 gst_plugins_good/gst/audiofx/audiochebband.c --- a/gst_plugins_good/gst/audiofx/audiochebband.c Tue Jul 06 14:35:10 2010 +0300 +++ /dev/null Thu Jan 01 00:00:00 1970 +0000 @@ -1,666 +0,0 @@ -/* - * GStreamer - * Copyright (C) 2007-2009 Sebastian Dröge - * - * 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 - * - * Transformation from lowpass to bandpass/bandreject: - * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm - * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm - * - */ - -/** - * SECTION:element-audiochebband - * - * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency - * band. The number of poles and the ripple parameter control the rolloff. - * - * This element has the advantage over the windowed sinc bandpass and bandreject 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. - * - * 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. - * - * 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. - * - * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. - * - * - * 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. - * - * - * - * Example launch line - * |[ - * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink - * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink - * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink - * ]| - * - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include -#include -#include -#include -#include - -#include - -#include "math_compat.h" - -#include "audiochebband.h" - -#define GST_CAT_DEFAULT gst_audio_cheb_band_debug -GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); - -enum -{ - PROP_0, - PROP_MODE, - PROP_TYPE, - PROP_LOWER_FREQUENCY, - PROP_UPPER_FREQUENCY, - PROP_RIPPLE, - PROP_POLES -}; - -#define DEBUG_INIT(bla) \ - GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0, "audiochebband element"); - -GST_BOILERPLATE_FULL (GstAudioChebBand, gst_audio_cheb_band, - GstAudioFXBaseIIRFilter, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT); - -static void gst_audio_cheb_band_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_audio_cheb_band_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec); -static void gst_audio_cheb_band_finalize (GObject * object); - -static gboolean gst_audio_cheb_band_setup (GstAudioFilter * filter, - GstRingBufferSpec * format); - -enum -{ - MODE_BAND_PASS = 0, - MODE_BAND_REJECT -}; - -#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_cheb_band_mode_get_type ()) -static GType -gst_audio_cheb_band_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 ("GstAudioChebBandMode", values); - } - return gtype; -} - -/* GObject vmethod implementations */ - -static void -gst_audio_cheb_band_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_set_details_simple (element_class, - "Band pass & band reject filter", "Filter/Effect/Audio", - "Chebyshev band pass and band reject filter", - "Sebastian Dröge "); -} - -static void -gst_audio_cheb_band_class_init (GstAudioChebBandClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass; - - gobject_class->set_property = gst_audio_cheb_band_set_property; - gobject_class->get_property = gst_audio_cheb_band_get_property; - gobject_class->finalize = gst_audio_cheb_band_finalize; - - 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_BAND_MODE, - MODE_BAND_PASS, - G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); - 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 | G_PARAM_STATIC_STRINGS)); - - /* 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_LOWER_FREQUENCY, - g_param_spec_float ("lower-frequency", "Lower frequency", - "Start frequency of the band (Hz)", 0.0, 100000.0, - 0.0, - G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); - g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY, - g_param_spec_float ("upper-frequency", "Upper frequency", - "Stop frequency of the band (Hz)", 0.0, 100000.0, 0.0, - G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); - 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 | G_PARAM_STATIC_STRINGS)); - /* FIXME: What to do about this upper boundary? With a frequencies near - * rate/4 32 poles are completely possible, with frequencies 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 multiply of four", - 4, 32, 4, - G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); - - filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_setup); -} - -static void -gst_audio_cheb_band_init (GstAudioChebBand * filter, - GstAudioChebBandClass * klass) -{ - filter->lower_frequency = filter->upper_frequency = 0.0; - filter->mode = MODE_BAND_PASS; - filter->type = 1; - filter->poles = 4; - filter->ripple = 0.25; - - filter->lock = g_mutex_new (); -} - -static void -generate_biquad_coefficients (GstAudioChebBand * filter, - gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3, - gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4) -{ - gint np = filter->poles / 2; - gdouble ripple = filter->ripple; - - /* pole location in s-plane */ - gdouble rp, ip; - - /* zero location in s-plane */ - gdouble 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 move 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; - - iz = cos (angle); - mag2 = iz * iz; - 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 bandpass - * or band reject. - * - * For bandpass substitute z^(-1) with: - * - * -2 -1 - * -z + alpha * z - beta - * ---------------------------- - * -2 -1 - * beta * z - alpha * z + 1 - * - * alpha = (2*a*b)/(1+b) - * beta = (b-1)/(b+1) - * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) - * b = tan(1/2) * cot((w1 - w0)/2) - * - * For bandreject substitute z^(-1) with: - * - * -2 -1 - * z - alpha * z + beta - * ---------------------------- - * -2 -1 - * beta * z - alpha * z + 1 - * - * alpha = (2*a)/(1+b) - * beta = (1-b)/(1+b) - * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) - * b = tan(1/2) * tan((w1 - w0)/2) - * - */ - { - gdouble a, b, d; - gdouble alpha, beta; - gdouble w0 = - 2.0 * M_PI * (filter->lower_frequency / - GST_AUDIO_FILTER (filter)->format.rate); - gdouble w1 = - 2.0 * M_PI * (filter->upper_frequency / - GST_AUDIO_FILTER (filter)->format.rate); - - if (filter->mode == MODE_BAND_PASS) { - a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); - b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0); - - alpha = (2.0 * a * b) / (1.0 + b); - beta = (b - 1.0) / (b + 1.0); - - d = 1.0 + beta * (y1 - beta * y2); - - *a0 = (x0 + beta * (-x1 + beta * x2)) / d; - *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d; - *a2 = - (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) + - alpha * alpha * (x0 - x1 + x2)) / d; - *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d; - *a4 = (beta * (beta * x0 - x1) + x2) / d; - *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d; - *b2 = - (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) + - 2.0 * beta * (-1.0 + y2)) / d; - *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d; - *b4 = (-beta * beta - beta * y1 + y2) / d; - } else { - a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); - b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0); - - alpha = (2.0 * a) / (1.0 + b); - beta = (1.0 - b) / (1.0 + b); - - d = -1.0 + beta * (beta * y2 + y1); - - *a0 = (-x0 - beta * x1 - beta * beta * x2) / d; - *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d; - *a2 = - (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) - - alpha * alpha * (x0 + x1 + x2)) / d; - *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d; - *a4 = (-beta * beta * x0 - beta * x1 - x2) / d; - *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d; - *b2 = - -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) + - alpha * alpha * (-1.0 + y1 + y2)) / d; - *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d; - *b4 = -(-beta * beta + beta * y1 + y2) / d; - } - } -} - -static void -generate_coefficients (GstAudioChebBand * filter) -{ - if (GST_AUDIO_FILTER (filter)->format.rate == 0) { - gdouble *a = g_new0 (gdouble, 1); - - a[0] = 1.0; - gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER - (filter), a, 1, NULL, 0); - GST_LOG_OBJECT (filter, "rate was not set yet"); - return; - } - - if (filter->upper_frequency <= filter->lower_frequency) { - gdouble *a = g_new0 (gdouble, 1); - - a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0; - gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER - (filter), a, 1, NULL, 0); - - GST_LOG_OBJECT (filter, "frequency band had no or negative dimension"); - return; - } - - if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) { - filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2; - GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency"); - } - - if (filter->lower_frequency < 0.0) { - filter->lower_frequency = 0.0; - GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0"); - } - - /* Calculate coefficients for the chebyshev filter */ - { - gint np = filter->poles; - gdouble *a, *b; - gint i, p; - - a = g_new0 (gdouble, np + 5); - b = g_new0 (gdouble, np + 5); - - /* Calculate transfer function coefficients */ - a[4] = 1.0; - b[4] = 1.0; - - for (p = 1; p <= np / 4; p++) { - gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4; - gdouble *ta = g_new0 (gdouble, np + 5); - gdouble *tb = g_new0 (gdouble, np + 5); - - generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1, - &b2, &b3, &b4); - - memcpy (ta, a, sizeof (gdouble) * (np + 5)); - memcpy (tb, b, sizeof (gdouble) * (np + 5)); - - /* add the new coefficients for the new two poles - * to the cascade by multiplication of the transfer - * functions */ - for (i = 4; i < np + 5; i++) { - a[i] = - a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] + - a4 * ta[i - 4]; - b[i] = - tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] - - b4 * tb[i - 4]; - } - 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[4] = 0.0; - for (i = 0; i <= np; i++) { - a[i] = a[i + 4]; - b[i] = -b[i + 4]; - } - - /* Normalize to unity gain at frequency 0 and frequency - * 0.5 for bandreject and unity gain at band center frequency - * for bandpass */ - if (filter->mode == MODE_BAND_REJECT) { - /* gain is sqrt(H(0)*H(0.5)) */ - - gdouble gain1 = - gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1, - 1.0, 0.0); - gdouble gain2 = - gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1, - -1.0, 0.0); - - gain1 = sqrt (gain1 * gain2); - - for (i = 0; i <= np; i++) { - a[i] /= gain1; - } - } else { - /* gain is H(wc), wc = center frequency */ - - gdouble w1 = - 2.0 * M_PI * (filter->lower_frequency / - GST_AUDIO_FILTER (filter)->format.rate); - gdouble w2 = - 2.0 * M_PI * (filter->upper_frequency / - GST_AUDIO_FILTER (filter)->format.rate); - gdouble w0 = (w2 + w1) / 2.0; - gdouble zr = cos (w0), zi = sin (w0); - gdouble gain = - gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1, zr, - zi); - - for (i = 0; i <= np; i++) { - a[i] /= gain; - } - } - - gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER - (filter), a, np + 1, b, np + 1); - - GST_LOG_OBJECT (filter, - "Generated IIR coefficients for the Chebyshev filter"); - GST_LOG_OBJECT (filter, - "mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB", - (filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject", - filter->type, filter->poles, filter->lower_frequency, - filter->upper_frequency, filter->ripple); - - GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz", - 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, - np + 1, 1.0, 0.0))); - { - gdouble w1 = - 2.0 * M_PI * (filter->lower_frequency / - GST_AUDIO_FILTER (filter)->format.rate); - gdouble w2 = - 2.0 * M_PI * (filter->upper_frequency / - GST_AUDIO_FILTER (filter)->format.rate); - gdouble w0 = (w2 + w1) / 2.0; - gdouble zr, zi; - - zr = cos (w1); - zi = sin (w1); - GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", - 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, - b, np + 1, zr, zi)), (int) filter->lower_frequency); - zr = cos (w0); - zi = sin (w0); - GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", - 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, - b, np + 1, zr, zi)), - (int) ((filter->lower_frequency + filter->upper_frequency) / 2.0)); - zr = cos (w2); - zi = sin (w2); - GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", - 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, - b, np + 1, zr, zi)), (int) filter->upper_frequency); - } - GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", - 20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, - np + 1, -1.0, 0.0)), - GST_AUDIO_FILTER (filter)->format.rate / 2); - } -} - -static void -gst_audio_cheb_band_finalize (GObject * object) -{ - GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object); - - g_mutex_free (filter->lock); - filter->lock = NULL; - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static void -gst_audio_cheb_band_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object); - - switch (prop_id) { - case PROP_MODE: - g_mutex_lock (filter->lock); - filter->mode = g_value_get_enum (value); - generate_coefficients (filter); - g_mutex_unlock (filter->lock); - break; - case PROP_TYPE: - g_mutex_lock (filter->lock); - filter->type = g_value_get_int (value); - generate_coefficients (filter); - g_mutex_unlock (filter->lock); - break; - case PROP_LOWER_FREQUENCY: - g_mutex_lock (filter->lock); - filter->lower_frequency = g_value_get_float (value); - generate_coefficients (filter); - g_mutex_unlock (filter->lock); - break; - case PROP_UPPER_FREQUENCY: - g_mutex_lock (filter->lock); - filter->upper_frequency = g_value_get_float (value); - generate_coefficients (filter); - g_mutex_unlock (filter->lock); - break; - case PROP_RIPPLE: - g_mutex_lock (filter->lock); - filter->ripple = g_value_get_float (value); - generate_coefficients (filter); - g_mutex_unlock (filter->lock); - break; - case PROP_POLES: - g_mutex_lock (filter->lock); - filter->poles = GST_ROUND_UP_4 (g_value_get_int (value)); - generate_coefficients (filter); - g_mutex_unlock (filter->lock); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_audio_cheb_band_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (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_LOWER_FREQUENCY: - g_value_set_float (value, filter->lower_frequency); - break; - case PROP_UPPER_FREQUENCY: - g_value_set_float (value, filter->upper_frequency); - 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_cheb_band_setup (GstAudioFilter * base, GstRingBufferSpec * format) -{ - GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base); - - generate_coefficients (filter); - - return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format); -}