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+/*
+ * 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>
+ * 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>
+ * <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));
+ g_object_class_install_property (gobject_class, PROP_CUTOFF,
+ g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
+ G_MAXFLOAT, 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,
+ G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+ 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;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqLimit * filter,
+ gdouble * 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;
+ }
+ }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqLimit * filter,
+ gfloat * 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;
+ }
+ }
+}
+
+/* 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;
+}