/* 

 * GStreamer

 * Copyright (C) 2007-2009 Sebastian Dröge <sebastian.droege@collabora.co.uk>

 *

 * This library is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Library General Public

 * License as published by the Free Software Foundation; either

 * version 2 of the License, or (at your option) any later version.

 *

 * This library is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Library General Public License for more details.

 *

 * You should have received a copy of the GNU Library General Public

 * License along with this library; if not, write to the

 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,

 * Boston, MA 02111-1307, USA.

 */

/* 

 * Chebyshev type 1 filter design based on

 * "The Scientist and Engineer's Guide to DSP", Chapter 20.

 * http://www.dspguide.com/

 *

 * For type 2 and Chebyshev filters in general read

 * http://en.wikipedia.org/wiki/Chebyshev_filter

 *

 * Transformation from lowpass to bandpass/bandreject:

 * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm

 * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm

 * 

 */

/**

 * SECTION:element-audiochebband

 *

 * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency

 * band. The number of poles and the ripple parameter control the rolloff.

 *

 * This element has the advantage over the windowed sinc bandpass and bandreject filter that it is

 * much faster and produces almost as good results. It's only disadvantages are the highly

 * non-linear phase and the slower rolloff compared to a windowed sinc filter with a large kernel.

 *

 * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e.

 * some frequencies in the passband will be amplified by that value. A higher ripple value will allow

 * a faster rolloff.

 *

 * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will

 * be at most this value. A lower ripple value will allow a faster rolloff.

 *

 * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.

 *

 * <note>

 * Be warned that a too large number of poles can produce noise. The most poles are possible with

 * a cutoff frequency at a quarter of the sampling rate.

 * </note>

 *

 * <refsect2>

 * <title>Example launch line</title>

 * |[

 * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink

 * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink

 * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink

 * ]|

 * </refsect2>

 */

#ifdef HAVE_CONFIG_H

#include "config.h"

#endif

#include <gst/gst.h>

#include <gst/base/gstbasetransform.h>

#include <gst/audio/audio.h>

#include <gst/audio/gstaudiofilter.h>

#include <gst/controller/gstcontroller.h>

#include <math.h>

#include "math_compat.h"

#include "audiochebband.h"

#define GST_CAT_DEFAULT gst_audio_cheb_band_debug

GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

enum

{

  PROP_0,

  PROP_MODE,

  PROP_TYPE,

  PROP_LOWER_FREQUENCY,

  PROP_UPPER_FREQUENCY,

  PROP_RIPPLE,

  PROP_POLES

};

#define DEBUG_INIT(bla) \

  GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_band_debug, "audiochebband", 0, "audiochebband element");

GST_BOILERPLATE_FULL (GstAudioChebBand, gst_audio_cheb_band,

    GstAudioFXBaseIIRFilter, GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);

static void gst_audio_cheb_band_set_property (GObject * object,

    guint prop_id, const GValue * value, GParamSpec * pspec);

static void gst_audio_cheb_band_get_property (GObject * object,

    guint prop_id, GValue * value, GParamSpec * pspec);

static void gst_audio_cheb_band_finalize (GObject * object);

static gboolean gst_audio_cheb_band_setup (GstAudioFilter * filter,

    GstRingBufferSpec * format);

enum

{

  MODE_BAND_PASS = 0,

  MODE_BAND_REJECT

};

#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_cheb_band_mode_get_type ())

static GType

gst_audio_cheb_band_mode_get_type (void)

{

  static GType gtype = 0;

  if (gtype == 0) {

    static const GEnumValue values[] = {

      {MODE_BAND_PASS, "Band pass (default)",

          "band-pass"},

      {MODE_BAND_REJECT, "Band reject",

          "band-reject"},

      {0, NULL, NULL}

    };

    gtype = g_enum_register_static ("GstAudioChebBandMode", values);

  }

  return gtype;

}

/* GObject vmethod implementations */

static void

gst_audio_cheb_band_base_init (gpointer klass)

{

  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

  gst_element_class_set_details_simple (element_class,

      "Band pass & band reject filter", "Filter/Effect/Audio",

      "Chebyshev band pass and band reject filter",

      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");

}

static void

gst_audio_cheb_band_class_init (GstAudioChebBandClass * klass)

{

  GObjectClass *gobject_class = (GObjectClass *) klass;

  GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;

  gobject_class->set_property = gst_audio_cheb_band_set_property;

  gobject_class->get_property = gst_audio_cheb_band_get_property;

  gobject_class->finalize = gst_audio_cheb_band_finalize;

  g_object_class_install_property (gobject_class, PROP_MODE,

      g_param_spec_enum ("mode", "Mode",

          "Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE,

          MODE_BAND_PASS,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_TYPE,

      g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  /* FIXME: Don't use the complete possible range but restrict the upper boundary

   * so automatically generated UIs can use a slider without */

  g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY,

      g_param_spec_float ("lower-frequency", "Lower frequency",

          "Start frequency of the band (Hz)", 0.0, 100000.0,

          0.0,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY,

      g_param_spec_float ("upper-frequency", "Upper frequency",

          "Stop frequency of the band (Hz)", 0.0, 100000.0, 0.0,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RIPPLE,

      g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,

          200.0, 0.25,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  /* FIXME: What to do about this upper boundary? With a frequencies near

   * rate/4 32 poles are completely possible, with frequencies very low

   * or very high 16 poles already produces only noise */

  g_object_class_install_property (gobject_class, PROP_POLES,

      g_param_spec_int ("poles", "Poles",

          "Number of poles to use, will be rounded up to the next multiply of four",

          4, 32, 4,

          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_band_setup);

}

static void

gst_audio_cheb_band_init (GstAudioChebBand * filter,

    GstAudioChebBandClass * klass)

{

  filter->lower_frequency = filter->upper_frequency = 0.0;

  filter->mode = MODE_BAND_PASS;

  filter->type = 1;

  filter->poles = 4;

  filter->ripple = 0.25;

  filter->lock = g_mutex_new ();

}

static void

generate_biquad_coefficients (GstAudioChebBand * filter,

    gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3,

    gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4)

{

  gint np = filter->poles / 2;

  gdouble ripple = filter->ripple;

  /* pole location in s-plane */

  gdouble rp, ip;

  /* zero location in s-plane */

  gdouble iz = 0.0;

  /* transfer function coefficients for the z-plane */

  gdouble x0, x1, x2, y1, y2;

  gint type = filter->type;

  /* Calculate pole location for lowpass at frequency 1 */

  {

    gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np;

    rp = -sin (angle);

    ip = cos (angle);

  }

  /* If we allow ripple, move the pole from the unit

   * circle to an ellipse and keep cutoff at frequency 1 */

  if (ripple > 0 && type == 1) {

    gdouble es, vx;

    es = sqrt (pow (10.0, ripple / 10.0) - 1.0);

    vx = (1.0 / np) * asinh (1.0 / es);

    rp = rp * sinh (vx);

    ip = ip * cosh (vx);

  } else if (type == 2) {

    gdouble es, vx;

    es = sqrt (pow (10.0, ripple / 10.0) - 1.0);

    vx = (1.0 / np) * asinh (es);

    rp = rp * sinh (vx);

    ip = ip * cosh (vx);

  }

  /* Calculate inverse of the pole location to move from

   * type I to type II */

  if (type == 2) {

    gdouble mag2 = rp * rp + ip * ip;

    rp /= mag2;

    ip /= mag2;

  }

  /* Calculate zero location for frequency 1 on the

   * unit circle for type 2 */

  if (type == 2) {

    gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np);

    gdouble mag2;

    iz = cos (angle);

    mag2 = iz * iz;

    iz /= mag2;

  }

  /* Convert from s-domain to z-domain by

   * using the bilinear Z-transform, i.e.

   * substitute s by (2/t)*((z-1)/(z+1))

   * with t = 2 * tan(0.5).

   */

  if (type == 1) {

    gdouble t, m, d;

    t = 2.0 * tan (0.5);

    m = rp * rp + ip * ip;

    d = 4.0 - 4.0 * rp * t + m * t * t;

    x0 = (t * t) / d;

    x1 = 2.0 * x0;

    x2 = x0;

    y1 = (8.0 - 2.0 * m * t * t) / d;

    y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;

  } else {

    gdouble t, m, d;

    t = 2.0 * tan (0.5);

    m = rp * rp + ip * ip;

    d = 4.0 - 4.0 * rp * t + m * t * t;

    x0 = (t * t * iz * iz + 4.0) / d;

    x1 = (-8.0 + 2.0 * iz * iz * t * t) / d;

    x2 = x0;

    y1 = (8.0 - 2.0 * m * t * t) / d;

    y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d;

  }

  /* Convert from lowpass at frequency 1 to either bandpass

   * or band reject.

   *

   * For bandpass substitute z^(-1) with:

   *

   *   -2            -1

   * -z   + alpha * z   - beta

   * ----------------------------

   *         -2            -1

   * beta * z   - alpha * z   + 1

   *

   * alpha = (2*a*b)/(1+b)

   * beta = (b-1)/(b+1)

   * a = cos((w1 + w0)/2) / cos((w1 - w0)/2)

   * b = tan(1/2) * cot((w1 - w0)/2)

   *

   * For bandreject substitute z^(-1) with:

   * 

   *  -2            -1

   * z   - alpha * z   + beta

   * ----------------------------

   *         -2            -1

   * beta * z   - alpha * z   + 1

   *

   * alpha = (2*a)/(1+b)

   * beta = (1-b)/(1+b)

   * a = cos((w1 + w0)/2) / cos((w1 - w0)/2)

   * b = tan(1/2) * tan((w1 - w0)/2)

   *

   */

  {

    gdouble a, b, d;

    gdouble alpha, beta;

    gdouble w0 =

        2.0 * M_PI * (filter->lower_frequency /

        GST_AUDIO_FILTER (filter)->format.rate);

    gdouble w1 =

        2.0 * M_PI * (filter->upper_frequency /

        GST_AUDIO_FILTER (filter)->format.rate);

    if (filter->mode == MODE_BAND_PASS) {

      a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);

      b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0);

      alpha = (2.0 * a * b) / (1.0 + b);

      beta = (b - 1.0) / (b + 1.0);

      d = 1.0 + beta * (y1 - beta * y2);

      *a0 = (x0 + beta * (-x1 + beta * x2)) / d;

      *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d;

      *a2 =

          (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) +

          alpha * alpha * (x0 - x1 + x2)) / d;

      *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d;

      *a4 = (beta * (beta * x0 - x1) + x2) / d;

      *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d;

      *b2 =

          (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) +

          2.0 * beta * (-1.0 + y2)) / d;

      *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d;

      *b4 = (-beta * beta - beta * y1 + y2) / d;

    } else {

      a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0);

      b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0);

      alpha = (2.0 * a) / (1.0 + b);

      beta = (1.0 - b) / (1.0 + b);

      d = -1.0 + beta * (beta * y2 + y1);

      *a0 = (-x0 - beta * x1 - beta * beta * x2) / d;

      *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d;

      *a2 =

          (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) -

          alpha * alpha * (x0 + x1 + x2)) / d;

      *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d;

      *a4 = (-beta * beta * x0 - beta * x1 - x2) / d;

      *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d;

      *b2 =

          -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) +

          alpha * alpha * (-1.0 + y1 + y2)) / d;

      *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d;

      *b4 = -(-beta * beta + beta * y1 + y2) / d;

    }

  }

}

static void

generate_coefficients (GstAudioChebBand * filter)

{

  if (GST_AUDIO_FILTER (filter)->format.rate == 0) {

    gdouble *a = g_new0 (gdouble, 1);

    a[0] = 1.0;

    gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER

        (filter), a, 1, NULL, 0);

    GST_LOG_OBJECT (filter, "rate was not set yet");

    return;

  }

  if (filter->upper_frequency <= filter->lower_frequency) {

    gdouble *a = g_new0 (gdouble, 1);

    a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;

    gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER

        (filter), a, 1, NULL, 0);

    GST_LOG_OBJECT (filter, "frequency band had no or negative dimension");

    return;

  }

  if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) {

    filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2;

    GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency");

  }

  if (filter->lower_frequency < 0.0) {

    filter->lower_frequency = 0.0;

    GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0");

  }

  /* 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 * M_PI * (filter->lower_frequency /

          GST_AUDIO_FILTER (filter)->format.rate);

      gdouble w2 =

          2.0 * M_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,