/* * 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, 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 * @title: 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. * * > 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-1.0 audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink * gst-launch-1.0 audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink * ]| * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include <gst/gst.h> #include <gst/base/gstbasetransform.h> #include <gst/audio/audio.h> #include <gst/audio/gstaudiofilter.h> #include <math.h> #include "math_compat.h" #include "audiocheblimit.h" #include "gst/glib-compat-private.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 gst_audio_cheb_limit_parent_class parent_class G_DEFINE_TYPE (GstAudioChebLimit, gst_audio_cheb_limit, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER); GST_ELEMENT_REGISTER_DEFINE (audiocheblimit, "audiocheblimit", GST_RANK_NONE, GST_TYPE_AUDIO_CHEB_LIMIT); 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, const GstAudioInfo * info); 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_class_init (GstAudioChebLimitClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass; GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element"); 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)); gst_element_class_set_static_metadata (gstelement_class, "Low pass & high pass filter", "Filter/Effect/Audio", "Chebyshev low pass and high pass filter", "Sebastian Dröge <sebastian.droege@collabora.co.uk>"); filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup); gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, 0); } static void gst_audio_cheb_limit_init (GstAudioChebLimit * filter) { filter->cutoff = 0.0; filter->mode = MODE_LOW_PASS; filter->type = 1; filter->poles = 4; filter->ripple = 0.25; g_mutex_init (&filter->lock); } static void generate_biquad_coefficients (GstAudioChebLimit * filter, gint p, gint rate, gdouble * b0, gdouble * b1, gdouble * b2, gdouble * a1, gdouble * a2) { 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 = (G_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 = G_PI / (np * 2.0) + ((p - 1) * G_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 * G_PI * (filter->cutoff / 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; *b0 = (x0 + k * (-x1 + k * x2)) / d; *b1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d; *b2 = (x0 * k * k - x1 * k + x2) / d; *a1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d; *a2 = (-k * k - y1 * k + y2) / d; if (filter->mode == MODE_HIGH_PASS) { *a1 = -*a1; *b1 = -*b1; } } } static void generate_coefficients (GstAudioChebLimit * filter, const GstAudioInfo * info) { gint rate; if (info) { rate = GST_AUDIO_INFO_RATE (info); } else { rate = GST_AUDIO_FILTER_RATE (filter); } GST_LOG_OBJECT (filter, "cutoff %f", filter->cutoff); if (rate == 0) { gdouble *a = g_new0 (gdouble, 1); gdouble *b = g_new0 (gdouble, 1); a[0] = 1.0; b[0] = 1.0; gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER (filter), a, 1, b, 1); GST_LOG_OBJECT (filter, "rate was not set yet"); return; } if (filter->cutoff >= rate / 2.0) { gdouble *a = g_new0 (gdouble, 1); gdouble *b = g_new0 (gdouble, 1); a[0] = 1.0; b[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, b, 1); GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency"); return; } else if (filter->cutoff <= 0.0) { gdouble *a = g_new0 (gdouble, 1); gdouble *b = g_new0 (gdouble, 1); a[0] = 1.0; b[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, b, 1); 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 b0, b1, b2, a1, a2; gdouble *ta = g_new0 (gdouble, np + 3); gdouble *tb = g_new0 (gdouble, np + 3); generate_biquad_coefficients (filter, p, rate, &b0, &b1, &b2, &a1, &a2); 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++) { b[i] = b0 * tb[i] + b1 * tb[i - 1] + b2 * tb[i - 2]; a[i] = ta[i] - a1 * ta[i - 1] - a2 * ta[i - 2]; } g_free (ta); g_free (tb); } /* Move coefficients to the beginning of the array to move from * the transfer function's coefficients to the difference * equation's coefficients */ 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++) { b[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 * G_PI * (filter->cutoff / 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)), rate); } } static void gst_audio_cheb_limit_finalize (GObject * object) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object); g_mutex_clear (&filter->lock); 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, NULL); g_mutex_unlock (&filter->lock); break; case PROP_TYPE: g_mutex_lock (&filter->lock); filter->type = g_value_get_int (value); generate_coefficients (filter, NULL); g_mutex_unlock (&filter->lock); break; case PROP_CUTOFF: g_mutex_lock (&filter->lock); filter->cutoff = g_value_get_float (value); generate_coefficients (filter, NULL); g_mutex_unlock (&filter->lock); break; case PROP_RIPPLE: g_mutex_lock (&filter->lock); filter->ripple = g_value_get_float (value); generate_coefficients (filter, NULL); 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, NULL); 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, const GstAudioInfo * info) { GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base); generate_coefficients (filter, info); return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, info); }