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56353e24d2
Most signal processing texts, including MATLAB, use the following convention for IIR filter coefficients: a_0 y[n] + a_1 y[n-1] + ... + a_M y[n-M] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N] Usually, a_0 is set to 1 because the coefficients can always be rescaled, giving y[n] = b_0 x[n] + b_1 x[n-1] + ... + b[N] x[n-N] - a_1 y[n-1] - ... - a_M y[n-M] The convention that was previously used by audiofxbaseiirfilter and derived class had the a and b coefficients swapped, and did not have the minus signs. This change makes the audiofx plugin use the more common convention described above.
567 lines
17 KiB
C
567 lines
17 KiB
C
/*
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* GStreamer
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* Copyright (C) 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/*
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* Chebyshev type 1 filter design based on
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* "The Scientist and Engineer's Guide to DSP", Chapter 20.
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* http://www.dspguide.com/
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*
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* For type 2 and Chebyshev filters in general read
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* http://en.wikipedia.org/wiki/Chebyshev_filter
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*
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*/
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/**
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* SECTION:element-audiocheblimit
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*
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* Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
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* cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
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*
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* This element has the advantage over the windowed sinc lowpass and highpass filter that it is
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* much faster and produces almost as good results. It's only disadvantages are the highly
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* non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.
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*
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* For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.
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* some frequencies in the passband will be amplified by that value. A higher ripple value will allow
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* a faster rolloff.
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*
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* For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will
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* be at most this value. A lower ripple value will allow a faster rolloff.
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*
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* As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
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* </para>
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* <note><para>
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* Be warned that a too large number of poles can produce noise. The most poles are possible with
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* a cutoff frequency at a quarter of the sampling rate.
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* </para></note>
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* <para>
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
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* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
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* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
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* ]|
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <math.h>
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#include "math_compat.h"
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#include "audiocheblimit.h"
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#include "gst/glib-compat-private.h"
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#define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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enum
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{
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PROP_0,
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PROP_MODE,
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PROP_TYPE,
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PROP_CUTOFF,
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PROP_RIPPLE,
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PROP_POLES
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};
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#define gst_audio_cheb_limit_parent_class parent_class
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G_DEFINE_TYPE (GstAudioChebLimit,
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gst_audio_cheb_limit, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER);
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static void gst_audio_cheb_limit_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_audio_cheb_limit_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static void gst_audio_cheb_limit_finalize (GObject * object);
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static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter,
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const GstAudioInfo * info);
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enum
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{
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MODE_LOW_PASS = 0,
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MODE_HIGH_PASS
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};
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#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_cheb_limit_mode_get_type ())
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static GType
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gst_audio_cheb_limit_mode_get_type (void)
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{
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static GType gtype = 0;
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if (gtype == 0) {
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static const GEnumValue values[] = {
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{MODE_LOW_PASS, "Low pass (default)",
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"low-pass"},
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{MODE_HIGH_PASS, "High pass",
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"high-pass"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAudioChebLimitMode", values);
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}
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return gtype;
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}
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/* GObject vmethod implementations */
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static void
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gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
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GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0,
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"audiocheblimit element");
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gobject_class->set_property = gst_audio_cheb_limit_set_property;
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gobject_class->get_property = gst_audio_cheb_limit_get_property;
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gobject_class->finalize = gst_audio_cheb_limit_finalize;
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g_object_class_install_property (gobject_class, PROP_MODE,
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g_param_spec_enum ("mode", "Mode",
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"Low pass or high pass mode",
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GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TYPE,
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g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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/* FIXME: Don't use the complete possible range but restrict the upper boundary
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* so automatically generated UIs can use a slider without */
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g_object_class_install_property (gobject_class, PROP_CUTOFF,
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g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
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100000.0, 0.0,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RIPPLE,
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g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
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200.0, 0.25,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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/* FIXME: What to do about this upper boundary? With a cutoff frequency of
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* rate/4 32 poles are completely possible, with a cutoff frequency very low
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* or very high 16 poles already produces only noise */
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g_object_class_install_property (gobject_class, PROP_POLES,
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g_param_spec_int ("poles", "Poles",
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"Number of poles to use, will be rounded up to the next even number",
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2, 32, 4,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_details_simple (gstelement_class,
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"Low pass & high pass filter",
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"Filter/Effect/Audio",
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"Chebyshev low pass and high pass filter",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup);
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}
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static void
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gst_audio_cheb_limit_init (GstAudioChebLimit * filter)
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{
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filter->cutoff = 0.0;
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filter->mode = MODE_LOW_PASS;
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filter->type = 1;
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filter->poles = 4;
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filter->ripple = 0.25;
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filter->lock = g_mutex_new ();
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}
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static void
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generate_biquad_coefficients (GstAudioChebLimit * filter,
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gint p, gdouble * b0, gdouble * b1, gdouble * b2,
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gdouble * a1, gdouble * a2)
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{
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gint np = filter->poles;
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gdouble ripple = filter->ripple;
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/* pole location in s-plane */
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gdouble rp, ip;
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/* zero location in s-plane */
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gdouble iz = 0.0;
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/* transfer function coefficients for the z-plane */
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gdouble x0, x1, x2, y1, y2;
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gint type = filter->type;
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/* Calculate pole location for lowpass at frequency 1 */
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{
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gdouble angle = (G_PI / 2.0) * (2.0 * p - 1) / np;
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rp = -sin (angle);
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ip = cos (angle);
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}
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/* If we allow ripple, move the pole from the unit
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* circle to an ellipse and keep cutoff at frequency 1 */
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if (ripple > 0 && type == 1) {
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gdouble es, vx;
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es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
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vx = (1.0 / np) * asinh (1.0 / es);
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rp = rp * sinh (vx);
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ip = ip * cosh (vx);
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} else if (type == 2) {
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gdouble es, vx;
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es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
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vx = (1.0 / np) * asinh (es);
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rp = rp * sinh (vx);
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ip = ip * cosh (vx);
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}
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/* Calculate inverse of the pole location to convert from
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* type I to type II */
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if (type == 2) {
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gdouble mag2 = rp * rp + ip * ip;
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rp /= mag2;
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ip /= mag2;
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}
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/* Calculate zero location for frequency 1 on the
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* unit circle for type 2 */
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if (type == 2) {
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gdouble angle = G_PI / (np * 2.0) + ((p - 1) * G_PI) / (np);
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gdouble mag2;
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iz = cos (angle);
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mag2 = iz * iz;
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iz /= mag2;
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}
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/* Convert from s-domain to z-domain by
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* using the bilinear Z-transform, i.e.
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* substitute s by (2/t)*((z-1)/(z+1))
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* with t = 2 * tan(0.5).
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*/
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if (type == 1) {
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gdouble t, m, d;
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t = 2.0 * tan (0.5);
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m = rp * rp + ip * ip;
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d = 4.0 - 4.0 * rp * t + m * t * t;
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x0 = (t * t) / d;
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x1 = 2.0 * x0;
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x2 = x0;
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y1 = (8.0 - 2.0 * m * t * t) / d;
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y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
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} else {
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gdouble t, m, d;
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t = 2.0 * tan (0.5);
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m = rp * rp + ip * ip;
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d = 4.0 - 4.0 * rp * t + m * t * t;
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x0 = (t * t * iz * iz + 4.0) / d;
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x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;
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x2 = x0;
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y1 = (8.0 - 2.0 * m * t * t) / d;
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y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;
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}
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/* Convert from lowpass at frequency 1 to either lowpass
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* or highpass.
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*
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* For lowpass substitute z^(-1) with:
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* -1
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* z - k
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* ------------
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* -1
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* 1 - k * z
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*
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* k = sin((1-w)/2) / sin((1+w)/2)
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*
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* For highpass substitute z^(-1) with:
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*
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* -1
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* -z - k
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* ------------
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* -1
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* 1 + k * z
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*
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* k = -cos((1+w)/2) / cos((1-w)/2)
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*
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*/
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{
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gdouble k, d;
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gdouble omega =
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2.0 * G_PI * (filter->cutoff / GST_AUDIO_FILTER_RATE (filter));
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if (filter->mode == MODE_LOW_PASS)
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k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
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else
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k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
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d = 1.0 + y1 * k - y2 * k * k;
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*b0 = (x0 + k * (-x1 + k * x2)) / d;
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*b1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
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*b2 = (x0 * k * k - x1 * k + x2) / d;
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*a1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
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*a2 = (-k * k - y1 * k + y2) / d;
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if (filter->mode == MODE_HIGH_PASS) {
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*a1 = -*a1;
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*b1 = -*b1;
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}
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}
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}
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static void
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generate_coefficients (GstAudioChebLimit * filter)
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{
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if (GST_AUDIO_FILTER_RATE (filter) == 0) {
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gdouble *a = g_new0 (gdouble, 1);
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gdouble *b = g_new0 (gdouble, 1);
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a[0] = 1.0;
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b[0] = 1.0;
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gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
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(filter), a, 1, b, 1);
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GST_LOG_OBJECT (filter, "rate was not set yet");
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return;
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}
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if (filter->cutoff >= GST_AUDIO_FILTER_RATE (filter) / 2.0) {
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gdouble *a = g_new0 (gdouble, 1);
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gdouble *b = g_new0 (gdouble, 1);
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a[0] = 1.0;
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b[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
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gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
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(filter), a, 1, b, 1);
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GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
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return;
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} else if (filter->cutoff <= 0.0) {
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gdouble *a = g_new0 (gdouble, 1);
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gdouble *b = g_new0 (gdouble, 1);
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a[0] = 1.0;
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b[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
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gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
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(filter), a, 1, b, 1);
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GST_LOG_OBJECT (filter, "cutoff is lower than zero");
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return;
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}
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/* Calculate coefficients for the chebyshev filter */
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{
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gint np = filter->poles;
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gdouble *a, *b;
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gint i, p;
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a = g_new0 (gdouble, np + 3);
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b = g_new0 (gdouble, np + 3);
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/* Calculate transfer function coefficients */
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a[2] = 1.0;
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b[2] = 1.0;
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for (p = 1; p <= np / 2; p++) {
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gdouble b0, b1, b2, a1, a2;
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gdouble *ta = g_new0 (gdouble, np + 3);
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gdouble *tb = g_new0 (gdouble, np + 3);
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generate_biquad_coefficients (filter, p, &b0, &b1, &b2, &a1, &a2);
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memcpy (ta, a, sizeof (gdouble) * (np + 3));
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memcpy (tb, b, sizeof (gdouble) * (np + 3));
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/* add the new coefficients for the new two poles
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* to the cascade by multiplication of the transfer
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* functions */
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for (i = 2; i < np + 3; i++) {
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b[i] = b0 * tb[i] + b1 * tb[i - 1] + b2 * tb[i - 2];
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a[i] = ta[i] - a1 * ta[i - 1] - a2 * ta[i - 2];
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}
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g_free (ta);
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g_free (tb);
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}
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/* Move coefficients to the beginning of the array to move from
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* the transfer function's coefficients to the difference
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* equation's coefficients */
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for (i = 0; i <= np; i++) {
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a[i] = a[i + 2];
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b[i] = b[i + 2];
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}
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/* Normalize to unity gain at frequency 0 for lowpass
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* and frequency 0.5 for highpass */
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{
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gdouble gain;
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if (filter->mode == MODE_LOW_PASS)
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gain =
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gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
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1.0, 0.0);
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else
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gain =
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gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
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-1.0, 0.0);
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for (i = 0; i <= np; i++) {
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b[i] /= gain;
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}
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}
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gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
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(filter), a, np + 1, b, np + 1);
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GST_LOG_OBJECT (filter,
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"Generated IIR coefficients for the Chebyshev filter");
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GST_LOG_OBJECT (filter,
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"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 / GST_AUDIO_FILTER_RATE (filter));
|
|
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_RATE (filter) / 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, const GstAudioInfo * info)
|
|
{
|
|
GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
|
|
|
|
generate_coefficients (filter);
|
|
|
|
return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, info);
|
|
}
|