/* * GStreamer * Copyright (C) 2007 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 * */ /** * SECTION:element-audiocheblimit * @short_description: Chebyshev low pass and high pass filter * * * * 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. * * * 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. * * * 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 ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink * * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include #include #include "math_compat.h" #include "audiocheblimit.h" #define GST_CAT_DEFAULT gst_audio_cheb_limit_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); static const GstElementDetails element_details = GST_ELEMENT_DETAILS ("Low pass & high pass filter", "Filter/Effect/Audio", "Chebyshev low pass and high pass filter", "Sebastian Dröge "); /* 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_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element"); GST_BOILERPLATE_FULL (GstAudioChebLimit, gst_audio_cheb_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); static void gst_audio_cheb_limit_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_cheb_limit_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter, GstRingBufferSpec * format); static GstFlowReturn gst_audio_cheb_limit_transform_ip (GstBaseTransform * base, GstBuffer * buf); static gboolean gst_audio_cheb_limit_start (GstBaseTransform * base); static void process_64 (GstAudioChebLimit * filter, gdouble * data, guint num_samples); static void process_32 (GstAudioChebLimit * filter, gfloat * data, guint num_samples); enum { MODE_LOW_PASS = 0, MODE_HIGH_PASS }; #define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_cheb_limit_mode_get_type ()) static GType gst_audio_cheb_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 ("GstAudioChebLimitMode", values); } return gtype; } /* GObject vmethod implementations */ static void gst_audio_cheb_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_cheb_limit_dispose (GObject * object) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_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) { GstAudioChebLimitChannelCtx *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_cheb_limit_class_init (GstAudioChebLimitClass * 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_cheb_limit_set_property; gobject_class->get_property = gst_audio_cheb_limit_get_property; gobject_class->dispose = gst_audio_cheb_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_cheb_limit_setup); trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_transform_ip); trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_start); } static void gst_audio_cheb_limit_init (GstAudioChebLimit * filter, GstAudioChebLimitClass * 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 (GstAudioChebLimit * 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 (GstAudioChebLimit * 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) { GstAudioChebLimitChannelCtx *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 (GstAudioChebLimitChannelCtx, 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 (GstAudioChebLimitChannelCtx, 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 (GstAudioChebLimitChannelCtx, 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 (GstAudioChebLimitChannelCtx, channels); for (i = 0; i < channels; i++) { GstAudioChebLimitChannelCtx *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))); #ifndef GST_DISABLE_GST_DEBUG { 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); } #endif 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_cheb_limit_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_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_cheb_limit_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_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_cheb_limit_setup (GstAudioFilter * base, GstRingBufferSpec * format) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base); gboolean ret = TRUE; if (format->width == 32) filter->process = (GstAudioChebLimitProcessFunc) process_32; else if (format->width == 64) filter->process = (GstAudioChebLimitProcessFunc) process_64; else ret = FALSE; filter->have_coeffs = FALSE; return ret; } static inline gdouble process (GstAudioChebLimit * filter, GstAudioChebLimitChannelCtx * 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 (GstAudioChebLimit * 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_cheb_limit_transform_ip (GstBaseTransform * base, GstBuffer * buf) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base); guint num_samples = GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); if (gst_base_transform_is_passthrough (base)) return GST_FLOW_OK; if (!filter->have_coeffs) generate_coefficients (filter); filter->process (filter, GST_BUFFER_DATA (buf), num_samples); return GST_FLOW_OK; } static gboolean gst_audio_cheb_limit_start (GstBaseTransform * base) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base); gint channels = GST_AUDIO_FILTER (filter)->format.channels; GstAudioChebLimitChannelCtx *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; }