--- a/gst_plugins_good/gst/audiofx/audiocheblimit.c Fri Jul 09 16:26:45 2010 -0500
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,568 +0,0 @@
-/*
- * 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);
-}