662 lines
		
	
	
		
			21 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			662 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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| /* 
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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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| /**
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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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| 
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| #ifdef HAVE_CONFIG_H
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| #include "config.h"
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| #endif
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| 
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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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| 
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| #include <math.h>
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| 
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| #include "math_compat.h"
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| 
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| #include "audiochebband.h"
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| 
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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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| 
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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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| 
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| #define gst_audio_cheb_band_parent_class parent_class
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| G_DEFINE_TYPE (GstAudioChebBand, gst_audio_cheb_band,
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|     GST_TYPE_AUDIO_FX_BASE_IIR_FILTER);
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| 
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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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| 
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| static gboolean gst_audio_cheb_band_setup (GstAudioFilter * filter,
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|     GstRingBufferSpec * format);
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| 
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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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| 
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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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| 
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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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| 
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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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| 
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| /* GObject vmethod implementations */
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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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|   GstElementClass *gstelement_class = (GstElementClass *) klass;
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|   GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
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| 
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|   GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0,
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|       "audiochebband element");
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| 
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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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| 
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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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| 
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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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| 
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|   gst_element_class_set_details_simple (gstelement_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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|   filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_setup);
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| }
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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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| {
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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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| 
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|   filter->lock = g_mutex_new ();
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| }
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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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| 
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|   /* pole location in s-plane */
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|   gdouble rp, ip;
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| 
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|   /* zero location in s-plane */
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|   gdouble iz = 0.0;
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| 
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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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| 
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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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| 
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|     rp = -sin (angle);
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|     ip = cos (angle);
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|   }
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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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| 
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|     es = sqrt (pow (10.0, ripple / 10.0) - 1.0);
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| 
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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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| 
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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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| 
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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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| 
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|     rp /= mag2;
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|     ip /= mag2;
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|   }
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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|       d = 1.0 + beta * (y1 - beta * y2);
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| 
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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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| 
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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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| 
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|       d = -1.0 + beta * (beta * y2 + y1);
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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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| 
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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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|   }
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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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| 
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|     a = g_new0 (gdouble, np + 5);
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|     b = g_new0 (gdouble, np + 5);
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| 
 | |
|     /* 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);
 | |
| }
 |