diff -r 71e347f905f2 -r 4a7fac7dd34a gst_plugins_good/gst/audiofx/audiocheblimit.c
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/audiofx/audiocheblimit.c	Fri Apr 16 15:15:52 2010 +0300
@@ -0,0 +1,568 @@
+/* 
+ * 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
+ *
+ */
+
+/**
+ * SECTION:element-audiocheblimit
+ *
+ * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the
+ * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff.
+ *
+ * This element has the advantage over the windowed sinc lowpass and highpass 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.
+ * </para>
+ * <note><para>
+ * 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.
+ * </para></note>
+ * <para>
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiocheblimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiocheblimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiocheblimit mode=low-pass cutoff=800 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 "audiocheblimit.h"
+
+#define GST_CAT_DEFAULT gst_audio_cheb_limit_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+enum
+{
+  PROP_0,
+  PROP_MODE,
+  PROP_TYPE,
+  PROP_CUTOFF,
+  PROP_RIPPLE,
+  PROP_POLES
+};
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_cheb_limit_debug, "audiocheblimit", 0, "audiocheblimit element");
+
+GST_BOILERPLATE_FULL (GstAudioChebLimit,
+    gst_audio_cheb_limit, GstAudioFXBaseIIRFilter,
+    GST_TYPE_AUDIO_FX_BASE_IIR_FILTER, DEBUG_INIT);
+
+static void gst_audio_cheb_limit_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_cheb_limit_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+static void gst_audio_cheb_limit_finalize (GObject * object);
+
+static gboolean gst_audio_cheb_limit_setup (GstAudioFilter * filter,
+    GstRingBufferSpec * format);
+
+enum
+{
+  MODE_LOW_PASS = 0,
+  MODE_HIGH_PASS
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_cheb_limit_mode_get_type ())
+static GType
+gst_audio_cheb_limit_mode_get_type (void)
+{
+  static GType gtype = 0;
+
+  if (gtype == 0) {
+    static const GEnumValue values[] = {
+      {MODE_LOW_PASS, "Low pass (default)",
+          "low-pass"},
+      {MODE_HIGH_PASS, "High pass",
+          "high-pass"},
+      {0, NULL, NULL}
+    };
+
+    gtype = g_enum_register_static ("GstAudioChebLimitMode", values);
+  }
+  return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_cheb_limit_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+  gst_element_class_set_details_simple (element_class,
+      "Low pass & high pass filter",
+      "Filter/Effect/Audio",
+      "Chebyshev low pass and high pass filter",
+      "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+}
+
+static void
+gst_audio_cheb_limit_class_init (GstAudioChebLimitClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstAudioFilterClass *filter_class = (GstAudioFilterClass *) klass;
+
+  gobject_class->set_property = gst_audio_cheb_limit_set_property;
+  gobject_class->get_property = gst_audio_cheb_limit_get_property;
+  gobject_class->finalize = gst_audio_cheb_limit_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_LIMIT_MODE, MODE_LOW_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_CUTOFF,
+      g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (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 cutoff frequency of
+   * rate/4 32 poles are completely possible, with a cutoff frequency 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 even number",
+          2, 32, 4,
+          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
+
+  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_cheb_limit_setup);
+}
+
+static void
+gst_audio_cheb_limit_init (GstAudioChebLimit * filter,
+    GstAudioChebLimitClass * klass)
+{
+  filter->cutoff = 0.0;
+  filter->mode = MODE_LOW_PASS;
+  filter->type = 1;
+  filter->poles = 4;
+  filter->ripple = 0.25;
+
+  filter->lock = g_mutex_new ();
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebLimit * filter,
+    gint p, gdouble * a0, gdouble * a1, gdouble * a2,
+    gdouble * b1, gdouble * b2)
+{
+  gint np = filter->poles;
+  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 convert 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 lowpass
+   * or highpass.
+   *
+   * For lowpass substitute z^(-1) with:
+   *  -1
+   * z   - k
+   * ------------
+   *          -1
+   * 1 - k * z
+   *
+   * k = sin((1-w)/2) / sin((1+w)/2)
+   *
+   * For highpass substitute z^(-1) with:
+   *
+   *   -1
+   * -z   - k
+   * ------------
+   *          -1
+   * 1 + k * z
+   *
+   * k = -cos((1+w)/2) / cos((1-w)/2)
+   *
+   */
+  {
+    gdouble k, d;
+    gdouble omega =
+        2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate);
+
+    if (filter->mode == MODE_LOW_PASS)
+      k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0);
+    else
+      k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0);
+
+    d = 1.0 + y1 * k - y2 * k * k;
+    *a0 = (x0 + k * (-x1 + k * x2)) / d;
+    *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d;
+    *a2 = (x0 * k * k - x1 * k + x2) / d;
+    *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d;
+    *b2 = (-k * k - y1 * k + y2) / d;
+
+    if (filter->mode == MODE_HIGH_PASS) {
+      *a1 = -*a1;
+      *b1 = -*b1;
+    }
+  }
+}
+
+static void
+generate_coefficients (GstAudioChebLimit * 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->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
+    gdouble *a = g_new0 (gdouble, 1);
+
+    a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+    gst_audio_fx_base_iir_filter_set_coefficients (GST_AUDIO_FX_BASE_IIR_FILTER
+        (filter), a, 1, NULL, 0);
+    GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
+    return;
+  } else if (filter->cutoff <= 0.0) {
+    gdouble *a = g_new0 (gdouble, 1);
+
+    a[0] = (filter->mode == MODE_LOW_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, "cutoff is lower than zero");
+    return;
+  }
+
+  /* Calculate coefficients for the chebyshev filter */
+  {
+    gint np = filter->poles;
+    gdouble *a, *b;
+    gint i, p;
+
+    a = g_new0 (gdouble, np + 3);
+    b = g_new0 (gdouble, np + 3);
+
+    /* Calculate transfer function coefficients */
+    a[2] = 1.0;
+    b[2] = 1.0;
+
+    for (p = 1; p <= np / 2; p++) {
+      gdouble a0, a1, a2, b1, b2;
+      gdouble *ta = g_new0 (gdouble, np + 3);
+      gdouble *tb = g_new0 (gdouble, np + 3);
+
+      generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2);
+
+      memcpy (ta, a, sizeof (gdouble) * (np + 3));
+      memcpy (tb, b, sizeof (gdouble) * (np + 3));
+
+      /* add the new coefficients for the new two poles
+       * to the cascade by multiplication of the transfer
+       * functions */
+      for (i = 2; i < np + 3; i++) {
+        a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2];
+        b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2];
+      }
+      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[2] = 0.0;
+    for (i = 0; i <= np; i++) {
+      a[i] = a[i + 2];
+      b[i] = -b[i + 2];
+    }
+
+    /* Normalize to unity gain at frequency 0 for lowpass
+     * and frequency 0.5 for highpass */
+    {
+      gdouble gain;
+
+      if (filter->mode == MODE_LOW_PASS)
+        gain =
+            gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
+            1.0, 0.0);
+      else
+        gain =
+            gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b, np + 1,
+            -1.0, 0.0);
+
+      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, cutoff: %.2f Hz, ripple: %.2f dB",
+        (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass",
+        filter->type, filter->poles, filter->cutoff, filter->ripple);
+    GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz",
+        20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
+                np + 1, 1.0, 0.0)));
+
+#ifndef GST_DISABLE_GST_DEBUG
+    {
+      gdouble wc =
+          2.0 * M_PI * (filter->cutoff /
+          GST_AUDIO_FILTER (filter)->format.rate);
+      gdouble zr = cos (wc), zi = sin (wc);
+
+      GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+          20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1,
+                  b, np + 1, zr, zi)), (int) filter->cutoff);
+    }
+#endif
+
+    GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+        20.0 * log10 (gst_audio_fx_base_iir_filter_calculate_gain (a, np + 1, b,
+                np + 1, -1.0, 0.0)),
+        GST_AUDIO_FILTER (filter)->format.rate / 2);
+  }
+}
+
+static void
+gst_audio_cheb_limit_finalize (GObject * object)
+{
+  GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
+
+  g_mutex_free (filter->lock);
+  filter->lock = NULL;
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_audio_cheb_limit_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      g_mutex_lock (filter->lock);
+      filter->mode = g_value_get_enum (value);
+      generate_coefficients (filter);
+      g_mutex_unlock (filter->lock);
+      break;
+    case PROP_TYPE:
+      g_mutex_lock (filter->lock);
+      filter->type = g_value_get_int (value);
+      generate_coefficients (filter);
+      g_mutex_unlock (filter->lock);
+      break;
+    case PROP_CUTOFF:
+      g_mutex_lock (filter->lock);
+      filter->cutoff = g_value_get_float (value);
+      generate_coefficients (filter);
+      g_mutex_unlock (filter->lock);
+      break;
+    case PROP_RIPPLE:
+      g_mutex_lock (filter->lock);
+      filter->ripple = g_value_get_float (value);
+      generate_coefficients (filter);
+      g_mutex_unlock (filter->lock);
+      break;
+    case PROP_POLES:
+      g_mutex_lock (filter->lock);
+      filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
+      generate_coefficients (filter);
+      g_mutex_unlock (filter->lock);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_cheb_limit_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      g_value_set_enum (value, filter->mode);
+      break;
+    case PROP_TYPE:
+      g_value_set_int (value, filter->type);
+      break;
+    case PROP_CUTOFF:
+      g_value_set_float (value, filter->cutoff);
+      break;
+    case PROP_RIPPLE:
+      g_value_set_float (value, filter->ripple);
+      break;
+    case PROP_POLES:
+      g_value_set_int (value, filter->poles);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_cheb_limit_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+{
+  GstAudioChebLimit *filter = GST_AUDIO_CHEB_LIMIT (base);
+
+  generate_coefficients (filter);
+
+  return GST_AUDIO_FILTER_CLASS (parent_class)->setup (base, format);
+}