--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/audiofx/audiocheblimit.c Fri Apr 16 15:15:52 2010 +0300
@@ -0,0 +1,568 @@
+/*
+ * GStreamer
+ * Copyright (C) 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * 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
+ *
+ * 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.
+ * </para>
+ * <note><para>
+ * 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.
+ * </para></note>
+ * <para>
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * 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
+ * ]|
+ * </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 "math_compat.h"
+
+#include "audiocheblimit.h"
+
+#define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+ PROP_0,
+ PROP_MODE,
+ PROP_TYPE,
+ PROP_CUTOFF,
+ PROP_RIPPLE,
+ PROP_POLES
+};
+
+#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, GstAudioFXBaseIIRFilter,
+ GST_TYPE_AUDIO_FX_BASE_IIR_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 void gst_audio_cheb_limit_finalize (GObject * object);
+
+static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter,
+ GstRingBufferSpec * format);
+
+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);
+
+ gst_element_class_set_details_simple (element_class,
+ "Low pass & high pass filter",
+ "Filter/Effect/Audio",
+ "Chebyshev low pass and high pass filter",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+}
+
+static void
+gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
+{
+ GObjectClass *gobject_class = (GObjectClass *) klass;
+ GstAudioFilterClass *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->finalize = gst_audio_cheb_limit_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_LIMIT_MODE, MODE_LOW_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_CUTOFF,
+ g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (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 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 | G_PARAM_STATIC_STRINGS));
+
+ filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup);
+}
+
+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;
+
+ filter->lock = g_mutex_new ();
+}
+
+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 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;
+
+ 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 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;
+ }
+ }
+}
+
+static void
+generate_coefficients (GstAudioChebLimit * 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->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
+ gdouble *a = g_new0 (gdouble, 1);
+
+ a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+ gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
+ (filter), a, 1, NULL, 0);
+ GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
+ return;
+ } else if (filter->cutoff <= 0.0) {
+ gdouble *a = g_new0 (gdouble, 1);
+
+ a[0] = (filter->mode == MODE_LOW_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, "cutoff is lower than zero");
+ return;
+ }
+
+ /* Calculate coefficients for the chebyshev filter */
+ {
+ gint np = filter->poles;
+ gdouble *a, *b;
+ gint i, p;
+
+ a = g_new0 (gdouble, np + 3);
+ b = g_new0 (gdouble, np + 3);
+
+ /* 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 =
+ gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
+ 1.0, 0.0);
+ else
+ gain =
+ gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
+ -1.0, 0.0);
+
+ 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, 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 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
+ np + 1, 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 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1,
+ b, np + 1, zr, zi)), (int) filter->cutoff);
+ }
+#endif
+
+ GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+ 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_limit_finalize (GObject * object)
+{
+ GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
+
+ g_mutex_free (filter->lock);
+ filter->lock = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+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:
+ 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_CUTOFF:
+ g_mutex_lock (filter->lock);
+ filter->cutoff = 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_2 (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_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);
+
+ generate_coefficients (filter);
+
+ return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
+}