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9041a588f9
... in that it will now call subclass with info on proposed audio format without having set that info already in base class. As such, subclass can not rely on audio format info being available there.
576 lines
17 KiB
C
576 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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g_mutex_init (&filter->lock);
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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, const GstAudioInfo * info)
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{
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gint rate;
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if (info) {
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rate = GST_AUDIO_INFO_RATE (info);
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} else {
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rate = GST_AUDIO_FILTER_RATE (filter);
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}
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GST_LOG_OBJECT (filter, "cutoff %f", filter->cutoff);
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if (rate == 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 >= rate / 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",
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(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_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);
|
|
}
|