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66f6e12888
Add private replacements for deprecated functions such as g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly to avoid the deprecation warnings. We'll change these over to the new API once we depend on glib >= 2.32.
668 lines
21 KiB
C
668 lines
21 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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* Transformation from lowpass to bandpass/bandreject:
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* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm
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* http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm
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*
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*/
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/**
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* SECTION:element-audiochebband
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*
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* Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
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* band. 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 bandpass and bandreject 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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*
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* <note>
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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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* </note>
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*
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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 ! audiochebband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
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* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
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* gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=4000 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 <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 <gst/controller/gstcontroller.h>
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#include <math.h>
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#include "math_compat.h"
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#include "audiochebband.h"
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#include "gst/glib-compat-private.h"
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#define GST_CAT_DEFAULT gst_audio_cheb_band_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_LOWER_FREQUENCY,
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PROP_UPPER_FREQUENCY,
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PROP_RIPPLE,
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PROP_POLES
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};
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0, "audiochebband element");
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GST_BOILERPLATE_FULL (GstAudioChebBand, gst_audio_cheb_band,
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GstAudioFXBaseIIRFilter, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
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static void gst_audio_cheb_band_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_band_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static void gst_audio_cheb_band_finalize (GObject * object);
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static gboolean gst_audio_cheb_band_setup (GstAudioFilter * filter,
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GstRingBufferSpec * format);
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enum
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{
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MODE_BAND_PASS = 0,
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MODE_BAND_REJECT
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};
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#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_cheb_band_mode_get_type ())
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static GType
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gst_audio_cheb_band_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_BAND_PASS, "Band pass (default)",
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"band-pass"},
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{MODE_BAND_REJECT, "Band reject",
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"band-reject"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAudioChebBandMode", 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_band_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_set_details_simple (element_class,
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"Band pass & band reject filter", "Filter/Effect/Audio",
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"Chebyshev band pass and band reject filter",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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}
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static void
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gst_audio_cheb_band_class_init (GstAudioChebBandClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
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gobject_class->set_property = gst_audio_cheb_band_set_property;
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gobject_class->get_property = gst_audio_cheb_band_get_property;
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gobject_class->finalize = gst_audio_cheb_band_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", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE,
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MODE_BAND_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_LOWER_FREQUENCY,
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g_param_spec_float ("lower-frequency", "Lower frequency",
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"Start frequency of the band (Hz)", 0.0, 100000.0,
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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_UPPER_FREQUENCY,
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g_param_spec_float ("upper-frequency", "Upper frequency",
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"Stop frequency of the band (Hz)", 0.0, 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 frequencies near
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* rate/4 32 poles are completely possible, with frequencies 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 multiply of four",
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4, 32, 4,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_setup);
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}
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static void
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gst_audio_cheb_band_init (GstAudioChebBand * filter,
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GstAudioChebBandClass * klass)
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{
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filter->lower_frequency = filter->upper_frequency = 0.0;
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filter->mode = MODE_BAND_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 (GstAudioChebBand * filter,
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gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,
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gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)
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{
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gint np = filter->poles / 2;
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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 move 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 bandpass
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* or band reject.
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*
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* For bandpass substitute z^(-1) with:
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*
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* -2 -1
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* -z + alpha * z - beta
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* ----------------------------
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* -2 -1
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* beta * z - alpha * z + 1
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*
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* alpha = (2*a*b)/(1+b)
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* beta = (b-1)/(b+1)
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* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
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* b = tan(1/2) * cot((w1 - w0)/2)
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*
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* For bandreject substitute z^(-1) with:
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*
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* -2 -1
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* z - alpha * z + beta
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* ----------------------------
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* -2 -1
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* beta * z - alpha * z + 1
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*
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* alpha = (2*a)/(1+b)
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* beta = (1-b)/(1+b)
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* a = cos((w1 + w0)/2) / cos((w1 - w0)/2)
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* b = tan(1/2) * tan((w1 - w0)/2)
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*
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*/
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{
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gdouble a, b, d;
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gdouble alpha, beta;
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gdouble w0 =
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2.0 * G_PI * (filter->lower_frequency /
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GST_AUDIO_FILTER (filter)->format.rate);
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gdouble w1 =
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2.0 * G_PI * (filter->upper_frequency /
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GST_AUDIO_FILTER (filter)->format.rate);
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if (filter->mode == MODE_BAND_PASS) {
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a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
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b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0);
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alpha = (2.0 * a * b) / (1.0 + b);
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beta = (b - 1.0) / (b + 1.0);
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d = 1.0 + beta * (y1 - beta * y2);
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*a0 = (x0 + beta * (-x1 + beta * x2)) / d;
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*a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d;
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*a2 =
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(-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) +
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alpha * alpha * (x0 - x1 + x2)) / d;
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*a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d;
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*a4 = (beta * (beta * x0 - x1) + x2) / d;
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*b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d;
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*b2 =
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(-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) +
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2.0 * beta * (-1.0 + y2)) / d;
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*b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d;
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*b4 = (-beta * beta - beta * y1 + y2) / d;
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} else {
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a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);
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b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0);
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alpha = (2.0 * a) / (1.0 + b);
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beta = (1.0 - b) / (1.0 + b);
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d = -1.0 + beta * (beta * y2 + y1);
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*a0 = (-x0 - beta * x1 - beta * beta * x2) / d;
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*a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d;
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*a2 =
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(-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) -
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alpha * alpha * (x0 + x1 + x2)) / d;
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*a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d;
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*a4 = (-beta * beta * x0 - beta * x1 - x2) / d;
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*b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d;
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*b2 =
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-(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) +
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alpha * alpha * (-1.0 + y1 + y2)) / d;
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*b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d;
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*b4 = -(-beta * beta + beta * y1 + y2) / d;
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}
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}
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}
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static void
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generate_coefficients (GstAudioChebBand * filter)
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{
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if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
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gdouble *a = g_new0 (gdouble, 1);
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a[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, NULL, 0);
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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->upper_frequency <= filter->lower_frequency) {
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gdouble *a = g_new0 (gdouble, 1);
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a[0] = (filter->mode == MODE_BAND_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, NULL, 0);
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GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");
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return;
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}
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if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) {
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filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2;
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GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency");
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}
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if (filter->lower_frequency < 0.0) {
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filter->lower_frequency = 0.0;
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GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0");
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|
}
|
|
|
|
/* Calculate coefficients for the chebyshev filter */
|
|
{
|
|
gint np = filter->poles;
|
|
gdouble *a, *b;
|
|
gint i, p;
|
|
|
|
a = g_new0 (gdouble, np + 5);
|
|
b = g_new0 (gdouble, np + 5);
|
|
|
|
/* Calculate transfer function coefficients */
|
|
a[4] = 1.0;
|
|
b[4] = 1.0;
|
|
|
|
for (p = 1; p <= np / 4; p++) {
|
|
gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4;
|
|
gdouble *ta = g_new0 (gdouble, np + 5);
|
|
gdouble *tb = g_new0 (gdouble, np + 5);
|
|
|
|
generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1,
|
|
&b2, &b3, &b4);
|
|
|
|
memcpy (ta, a, sizeof (gdouble) * (np + 5));
|
|
memcpy (tb, b, sizeof (gdouble) * (np + 5));
|
|
|
|
/* add the new coefficients for the new two poles
|
|
* to the cascade by multiplication of the transfer
|
|
* functions */
|
|
for (i = 4; i < np + 5; i++) {
|
|
a[i] =
|
|
a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] +
|
|
a4 * ta[i - 4];
|
|
b[i] =
|
|
tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] -
|
|
b4 * tb[i - 4];
|
|
}
|
|
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[4] = 0.0;
|
|
for (i = 0; i <= np; i++) {
|
|
a[i] = a[i + 4];
|
|
b[i] = -b[i + 4];
|
|
}
|
|
|
|
/* Normalize to unity gain at frequency 0 and frequency
|
|
* 0.5 for bandreject and unity gain at band center frequency
|
|
* for bandpass */
|
|
if (filter->mode == MODE_BAND_REJECT) {
|
|
/* gain is sqrt(H(0)*H(0.5)) */
|
|
|
|
gdouble gain1 =
|
|
gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
|
|
1.0, 0.0);
|
|
gdouble gain2 =
|
|
gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
|
|
-1.0, 0.0);
|
|
|
|
gain1 = sqrt (gain1 * gain2);
|
|
|
|
for (i = 0; i <= np; i++) {
|
|
a[i] /= gain1;
|
|
}
|
|
} else {
|
|
/* gain is H(wc), wc = center frequency */
|
|
|
|
gdouble w1 =
|
|
2.0 * G_PI * (filter->lower_frequency /
|
|
GST_AUDIO_FILTER (filter)->format.rate);
|
|
gdouble w2 =
|
|
2.0 * G_PI * (filter->upper_frequency /
|
|
GST_AUDIO_FILTER (filter)->format.rate);
|
|
gdouble w0 = (w2 + w1) / 2.0;
|
|
gdouble zr = cos (w0), zi = sin (w0);
|
|
gdouble gain =
|
|
gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1, zr,
|
|
zi);
|
|
|
|
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, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB",
|
|
(filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject",
|
|
filter->type, filter->poles, filter->lower_frequency,
|
|
filter->upper_frequency, filter->ripple);
|
|
|
|
GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz",
|
|
20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
|
|
np + 1, 1.0, 0.0)));
|
|
{
|
|
gdouble w1 =
|
|
2.0 * G_PI * (filter->lower_frequency /
|
|
GST_AUDIO_FILTER (filter)->format.rate);
|
|
gdouble w2 =
|
|
2.0 * G_PI * (filter->upper_frequency /
|
|
GST_AUDIO_FILTER (filter)->format.rate);
|
|
gdouble w0 = (w2 + w1) / 2.0;
|
|
gdouble zr, zi;
|
|
|
|
zr = cos (w1);
|
|
zi = sin (w1);
|
|
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
|
|
20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1,
|
|
b, np + 1, zr, zi)), (int) filter->lower_frequency);
|
|
zr = cos (w0);
|
|
zi = sin (w0);
|
|
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
|
|
20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1,
|
|
b, np + 1, zr, zi)),
|
|
(int) ((filter->lower_frequency + filter->upper_frequency) / 2.0));
|
|
zr = cos (w2);
|
|
zi = sin (w2);
|
|
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
|
|
20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1,
|
|
b, np + 1, zr, zi)), (int) filter->upper_frequency);
|
|
}
|
|
GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
|
|
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_band_finalize (GObject * object)
|
|
{
|
|
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (object);
|
|
|
|
g_mutex_free (filter->lock);
|
|
filter->lock = NULL;
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
gst_audio_cheb_band_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (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_LOWER_FREQUENCY:
|
|
g_mutex_lock (filter->lock);
|
|
filter->lower_frequency = g_value_get_float (value);
|
|
generate_coefficients (filter);
|
|
g_mutex_unlock (filter->lock);
|
|
break;
|
|
case PROP_UPPER_FREQUENCY:
|
|
g_mutex_lock (filter->lock);
|
|
filter->upper_frequency = 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_4 (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_band_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (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_LOWER_FREQUENCY:
|
|
g_value_set_float (value, filter->lower_frequency);
|
|
break;
|
|
case PROP_UPPER_FREQUENCY:
|
|
g_value_set_float (value, filter->upper_frequency);
|
|
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_band_setup (GstAudioFilter * base, GstRingBufferSpec * format)
|
|
{
|
|
GstAudioChebBand *filter = GST_AUDIO_CHEB_BAND (base);
|
|
|
|
generate_coefficients (filter);
|
|
|
|
return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
|
|
}
|