gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements.

Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_mode_get_type),
(gst_audio_chebyshev_freq_band_base_init),
(gst_audio_chebyshev_freq_band_dispose),
(gst_audio_chebyshev_freq_band_class_init),
(gst_audio_chebyshev_freq_band_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_band_set_property),
(gst_audio_chebyshev_freq_band_get_property),
(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
(gst_audio_chebyshev_freq_band_start):
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_mode_get_type),
(gst_audio_chebyshev_freq_limit_base_init),
(gst_audio_chebyshev_freq_limit_dispose),
(gst_audio_chebyshev_freq_limit_class_init),
(gst_audio_chebyshev_freq_limit_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_limit_set_property),
(gst_audio_chebyshev_freq_limit_get_property),
(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
(gst_audio_chebyshev_freq_limit_start):
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Fixes #464800.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebyshevfreqband.c:
(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
* tests/check/elements/audiochebyshevfreqlimit.c:
(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
Add unit tests for the chebyshev filters.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
And add docs for the chebyshev filters. While doing
that also run make update in docs/plugins.

10 files changed, 3793 insertions, 4 deletions

diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am
index b5bf0bf9..61a8dcc8 100644
--- a/gst/audiofx/Makefile.am
+++ b/gst/audiofx/Makefile.am
@@ -7,7 +7,9 @@ libgstaudiofx_la_SOURCES = audiofx.c\
 	audiopanorama.c \
 	audioinvert.c \
 	audioamplify.c \
-	audiodynamic.c
+	audiodynamic.c \
+	audiochebyshevfreqlimit.c \
+	audiochebyshevfreqband.c
 
 # flags used to compile this plugin
 libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
@@ -18,12 +20,15 @@ libgstaudiofx_la_LIBADD = $(GST_LIBS) \
 	$(GST_BASE_LIBS) \
 	$(GST_CONTROLLER_LIBS) \
 	$(GST_PLUGINS_BASE_LIBS) \
-	-lgstaudio-$(GST_MAJORMINOR)
+	-lgstaudio-$(GST_MAJORMINOR) \
+	$(LIBM)
 libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 
 # headers we need but don't want installed
 noinst_HEADERS = audiopanorama.h \
 	audioinvert.h \
 	audioamplify.h \
-	audiodynamic.h
+	audiodynamic.h \
+	audiochebyshevfreqlimit.h \
+	audiochebyshevfreqband.c
 
diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c
new file mode 100644
index 00000000..d4730607
--- /dev/null
+++ b/gst/audiofx/audiochebband.c
@@ -0,0 +1,916 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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-audiochebyshevfreqband
+ * @short_description: Chebyshev band pass and band reject filter
+ *
+ * <refsect2>
+ * <para>
+ * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
+ * band. The number of poles and the ripple parameter control the rolloff.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </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 "audiochebyshevfreqband.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
+    "Filter/Effect/Audio",
+    "Chebyshev band pass and band reject filter",
+    "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+  PROP_MODE,
+  PROP_TYPE,
+  PROP_LOWER_FREQUENCY,
+  PROP_UPPER_FREQUENCY,
+  PROP_RIPPLE,
+  PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-float,"                                              \
+    " width = (int) { 32, 64 }, "                                     \
+    " endianness = (int) BYTE_ORDER,"                                 \
+    " rate = (int) [ 1, MAX ],"                                       \
+    " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
+    GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
+    GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqBand * filter,
+    gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqBand * filter,
+    gfloat * data, guint num_samples);
+
+enum
+{
+  MODE_BAND_PASS = 0,
+  MODE_BAND_REJECT
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_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 ("GstAudioChebyshevFreqBandMode", values);
+  }
+  return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_band_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstCaps *caps;
+
+  gst_element_class_set_details (element_class, &element_details);
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_band_dispose (GObject * object)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqBandChannelCtx *ctx;
+    gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
+    klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseTransformClass *trans_class;
+  GstAudioFilterClass *filter_class;
+
+  gobject_class = (GObjectClass *) klass;
+  trans_class = (GstBaseTransformClass *) klass;
+  filter_class = (GstAudioFilterClass *) klass;
+
+  gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property;
+  gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property;
+  gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose;
+
+  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_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_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, G_MAXFLOAT,
+          0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  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, G_MAXFLOAT,
+          0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  g_object_class_install_property (gobject_class, PROP_RIPPLE,
+      g_param_spec_float ("ripple", "Ripple",
+          "Amount of ripple (dB)", 0.0, G_MAXFLOAT,
+          0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  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));
+
+  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
+  trans_class->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
+  trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
+}
+
+static void
+gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
+    GstAudioChebyshevFreqBandClass * klass)
+{
+  filter->lower_frequency = filter->upper_frequency = 0.0;
+  filter->mode = MODE_BAND_PASS;
+  filter->type = 1;
+  filter->poles = 4;
+  filter->ripple = 0.25;
+  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+  filter->have_coeffs = FALSE;
+  filter->num_a = 0;
+  filter->num_b = 0;
+  filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqBand * 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 rz = 0.0, 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;
+
+    rz = 0.0;
+    iz = cos (angle);
+    mag2 = rz * rz + iz * iz;
+    rz /= mag2;
+    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;
+    }
+  }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+    gdouble zi)
+{
+  gdouble sum_ar, sum_ai;
+  gdouble sum_br, sum_bi;
+  gdouble gain_r, gain_i;
+
+  gdouble sum_r_old;
+  gdouble sum_i_old;
+
+  gint i;
+
+  sum_ar = 0.0;
+  sum_ai = 0.0;
+  for (i = num_a; i >= 0; i--) {
+    sum_r_old = sum_ar;
+    sum_i_old = sum_ai;
+
+    sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+    sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+  }
+
+  sum_br = 0.0;
+  sum_bi = 0.0;
+  for (i = num_b; i >= 0; i--) {
+    sum_r_old = sum_br;
+    sum_i_old = sum_bi;
+
+    sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+    sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+  }
+  sum_br += 1.0;
+  sum_bi += 0.0;
+
+  gain_r =
+      (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+  gain_i =
+      (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+  return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqBand * filter)
+{
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqBandChannelCtx *ctx;
+    gint i;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+    GST_LOG_OBJECT (filter, "rate was not set yet");
+    return;
+  }
+
+  filter->have_coeffs = TRUE;
+
+  if (filter->upper_frequency <= filter->lower_frequency) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+    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;
+
+    filter->num_a = np + 1;
+    filter->a = a = g_new0 (gdouble, np + 5);
+    filter->num_b = np + 1;
+    filter->b = b = g_new0 (gdouble, np + 5);
+
+    filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+    for (i = 0; i < channels; i++) {
+      GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
+
+      ctx->x = g_new0 (gdouble, np + 1);
+      ctx->y = g_new0 (gdouble, np + 1);
+    }
+
+    /* 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 = calculate_gain (a, b, np, np, 1.0, 0.0);
+      gdouble gain2 = calculate_gain (a, b, np, np, -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 = calculate_gain (a, b, np, np, zr, zi);
+
+      for (i = 0; i <= np; i++) {
+        a[i] /= gain;
+      }
+    }
+
+    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 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+    {
+      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, zi;
+
+      zr = cos (w1);
+      zi = sin (w1);
+      GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+          20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+          (int) filter->lower_frequency);
+      zr = cos (w0);
+      zi = sin (w0);
+      GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+          20.0 * log10 (calculate_gain (a, b, np, np, 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 (calculate_gain (a, b, np, np, zr, zi)),
+          (int) filter->upper_frequency);
+    }
+    GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+        20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+        GST_AUDIO_FILTER (filter)->format.rate / 2);
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->mode = g_value_get_enum (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_TYPE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->type = g_value_get_int (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_LOWER_FREQUENCY:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->lower_frequency = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_UPPER_FREQUENCY:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->upper_frequency = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_RIPPLE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->ripple = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_POLES:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->poles = GST_ROUND_UP_4 (g_value_get_int (value));
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_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_chebyshev_freq_band_setup (GstAudioFilter * base,
+    GstRingBufferSpec * format)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+  gboolean ret = TRUE;
+
+  if (format->width == 32)
+    filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+        process_32;
+  else if (format->width == 64)
+    filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+        process_64;
+  else
+    ret = FALSE;
+
+  filter->have_coeffs = FALSE;
+
+  return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqBand * filter,
+    GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
+{
+  gdouble val = filter->a[0] * x0;
+  gint i, j;
+
+  for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+    val += filter->a[i] * ctx->x[j];
+    j--;
+    if (j < 0)
+      j = filter->num_a - 1;
+  }
+
+  for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+    val += filter->b[i] * ctx->y[j];
+    j--;
+    if (j < 0)
+      j = filter->num_b - 1;
+  }
+
+  if (ctx->x) {
+    ctx->x_pos++;
+    if (ctx->x_pos > filter->num_a - 1)
+      ctx->x_pos = 0;
+    ctx->x[ctx->x_pos] = x0;
+  }
+
+  if (ctx->y) {
+    ctx->y_pos++;
+    if (ctx->y_pos > filter->num_b - 1)
+      ctx->y_pos = 0;
+
+    ctx->y[ctx->y_pos] = val;
+  }
+
+  return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqBand * filter,
+    gdouble * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqBand * filter,
+    gfloat * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+  guint num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  if (!gst_buffer_is_writable (buf))
+    return GST_FLOW_OK;
+
+  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+  if (!filter->have_coeffs)
+    generate_coefficients (filter);
+
+  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+  GstAudioChebyshevFreqBandChannelCtx *ctx;
+  gint i;
+
+  /* Reset the history of input and output values if
+   * already existing */
+  if (channels && filter->channels) {
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      if (ctx->x)
+        memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+      if (ctx->y)
+        memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+    }
+  }
+  return TRUE;
+}
diff --git a/gst/audiofx/audiochebband.h b/gst/audiofx/audiochebband.h
new file mode 100644
index 00000000..e8c58074
--- /dev/null
+++ b/gst/audiofx/audiochebband.h
@@ -0,0 +1,79 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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.
+ */
+
+#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND            (gst_audio_chebyshev_freq_band_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand;
+typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass;
+
+typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint);
+
+typedef struct
+{
+  gdouble *x;
+  gint x_pos;
+  gdouble *y;
+  gint y_pos;
+} GstAudioChebyshevFreqBandChannelCtx;
+
+struct _GstAudioChebyshevFreqBand
+{
+  GstAudioFilter audiofilter;
+
+  gint mode;
+  gint type;
+  gint poles;
+  gfloat lower_frequency;
+  gfloat upper_frequency;
+  gfloat ripple;
+
+  /* < private > */
+  GstAudioChebyshevFreqBandProcessFunc process;
+
+  gboolean have_coeffs;
+  gdouble *a;
+  gint num_a;
+  gdouble *b;
+  gint num_b;
+  GstAudioChebyshevFreqBandChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqBandClass
+{
+  GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_band_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */
diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c
new file mode 100644
index 00000000..872b277d
--- /dev/null
+++ b/gst/audiofx/audiocheblimit.c
@@ -0,0 +1,816 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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-audiochebyshevfreqlimit
+ * @short_description: Chebyshev low pass and high pass filter
+ *
+ * <refsect2>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </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 "audiochebyshevfreqlimit.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
+    "Filter/Effect/Audio",
+    "Chebyshev low pass and high pass filter",
+    "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+  PROP_MODE,
+  PROP_TYPE,
+  PROP_CUTOFF,
+  PROP_RIPPLE,
+  PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-float,"                                              \
+    " width = (int) { 32, 64 }, "                                     \
+    " endianness = (int) BYTE_ORDER,"                                 \
+    " rate = (int) [ 1, MAX ],"                                       \
+    " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
+    gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
+    DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
+    GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqLimit * filter,
+    gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqLimit * filter,
+    gfloat * data, guint num_samples);
+
+enum
+{
+  MODE_LOW_PASS = 0,
+  MODE_HIGH_PASS
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_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 ("GstAudioChebyshevFreqLimitMode", values);
+  }
+  return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstCaps *caps;
+
+  gst_element_class_set_details (element_class, &element_details);
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_dispose (GObject * object)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqLimitChannelCtx *ctx;
+    gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
+    klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseTransformClass *trans_class;
+  GstAudioFilterClass *filter_class;
+
+  gobject_class = (GObjectClass *) klass;
+  trans_class = (GstBaseTransformClass *) klass;
+  filter_class = (GstAudioFilterClass *) klass;
+
+  gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
+  gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
+  gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
+
+  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_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_object_class_install_property (gobject_class, PROP_CUTOFF,
+      g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
+          G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  g_object_class_install_property (gobject_class, PROP_RIPPLE,
+      g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
+          G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  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));
+
+  filter_class->setup =
+      GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
+  trans_class->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
+  trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
+    GstAudioChebyshevFreqLimitClass * klass)
+{
+  filter->cutoff = 0.0;
+  filter->mode = MODE_LOW_PASS;
+  filter->type = 1;
+  filter->poles = 4;
+  filter->ripple = 0.25;
+  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+  filter->have_coeffs = FALSE;
+  filter->num_a = 0;
+  filter->num_b = 0;
+  filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqLimit * 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 rz = 0.0, 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;
+
+    rz = 0.0;
+    iz = cos (angle);
+    mag2 = rz * rz + iz * iz;
+    rz /= mag2;
+    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;
+    }
+  }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+    gdouble zi)
+{
+  gdouble sum_ar, sum_ai;
+  gdouble sum_br, sum_bi;
+  gdouble gain_r, gain_i;
+
+  gdouble sum_r_old;
+  gdouble sum_i_old;
+
+  gint i;
+
+  sum_ar = 0.0;
+  sum_ai = 0.0;
+  for (i = num_a; i >= 0; i--) {
+    sum_r_old = sum_ar;
+    sum_i_old = sum_ai;
+
+    sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+    sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+  }
+
+  sum_br = 0.0;
+  sum_bi = 0.0;
+  for (i = num_b; i >= 0; i--) {
+    sum_r_old = sum_br;
+    sum_i_old = sum_bi;
+
+    sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+    sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+  }
+  sum_br += 1.0;
+  sum_bi += 0.0;
+
+  gain_r =
+      (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+  gain_i =
+      (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+  return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqLimit * filter)
+{
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqLimitChannelCtx *ctx;
+    gint i;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    GST_LOG_OBJECT (filter, "rate was not set yet");
+    return;
+  }
+
+  filter->have_coeffs = TRUE;
+
+  if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
+    return;
+  } else if (filter->cutoff <= 0.0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    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;
+
+    filter->num_a = np + 1;
+    filter->a = a = g_new0 (gdouble, np + 3);
+    filter->num_b = np + 1;
+    filter->b = b = g_new0 (gdouble, np + 3);
+
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    for (i = 0; i < channels; i++) {
+      GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
+
+      ctx->x = g_new0 (gdouble, np + 1);
+      ctx->y = g_new0 (gdouble, np + 1);
+    }
+
+    /* 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 = calculate_gain (a, b, np, np, 1.0, 0.0);
+      else
+        gain = calculate_gain (a, b, np, np, -1.0, 0.0);
+
+      for (i = 0; i <= np; i++) {
+        a[i] /= gain;
+      }
+    }
+
+    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 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+    {
+      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 (calculate_gain (a, b, np, np, zr, zi)),
+          (int) filter->cutoff);
+    }
+    GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+        20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+        GST_AUDIO_FILTER (filter)->format.rate / 2);
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->mode = g_value_get_enum (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_TYPE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->type = g_value_get_int (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_CUTOFF:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->cutoff = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_RIPPLE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->ripple = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_POLES:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_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_chebyshev_freq_limit_setup (GstAudioFilter * base,
+    GstRingBufferSpec * format)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+  gboolean ret = TRUE;
+
+  if (format->width == 32)
+    filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+        process_32;
+  else if (format->width == 64)
+    filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+        process_64;
+  else
+    ret = FALSE;
+
+  filter->have_coeffs = FALSE;
+
+  return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqLimit * filter,
+    GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
+{
+  gdouble val = filter->a[0] * x0;
+  gint i, j;
+
+  for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+    val += filter->a[i] * ctx->x[j];
+    j--;
+    if (j < 0)
+      j = filter->num_a - 1;
+  }
+
+  for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+    val += filter->b[i] * ctx->y[j];
+    j--;
+    if (j < 0)
+      j = filter->num_b - 1;
+  }
+
+  if (ctx->x) {
+    ctx->x_pos++;
+    if (ctx->x_pos > filter->num_a - 1)
+      ctx->x_pos = 0;
+    ctx->x[ctx->x_pos] = x0;
+  }
+
+  if (ctx->y) {
+    ctx->y_pos++;
+    if (ctx->y_pos > filter->num_b - 1)
+      ctx->y_pos = 0;
+
+    ctx->y[ctx->y_pos] = val;
+  }
+
+  return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqLimit * filter,
+    gdouble * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqLimit * filter,
+    gfloat * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+  guint num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  if (!gst_buffer_is_writable (buf))
+    return GST_FLOW_OK;
+
+  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+  if (!filter->have_coeffs)
+    generate_coefficients (filter);
+
+  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+  return GST_FLOW_OK;
+}
+
+
+static gboolean
+gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+  GstAudioChebyshevFreqLimitChannelCtx *ctx;
+  gint i;
+
+  /* Reset the history of input and output values if
+   * already existing */
+  if (channels && filter->channels) {
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      if (ctx->x)
+        memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+      if (ctx->y)
+        memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+    }
+  }
+  return TRUE;
+}
diff --git a/gst/audiofx/audiocheblimit.h b/gst/audiofx/audiocheblimit.h
new file mode 100644
index 00000000..4c87ba8e
--- /dev/null
+++ b/gst/audiofx/audiocheblimit.h
@@ -0,0 +1,78 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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.
+ */
+
+#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT            (gst_audio_chebyshev_freq_limit_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit;
+typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass;
+
+typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint);
+
+typedef struct
+{
+  gdouble *x;
+  gint x_pos;
+  gdouble *y;
+  gint y_pos;
+} GstAudioChebyshevFreqLimitChannelCtx;
+
+struct _GstAudioChebyshevFreqLimit
+{
+  GstAudioFilter audiofilter;
+
+  gint mode;
+  gint type;
+  gint poles;
+  gfloat cutoff;
+  gfloat ripple;
+
+  /* < private > */
+  GstAudioChebyshevFreqLimitProcessFunc process;
+
+  gboolean have_coeffs;
+  gdouble *a;
+  gint num_a;
+  gdouble *b;
+  gint num_b;
+  GstAudioChebyshevFreqLimitChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqLimitClass
+{
+  GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_limit_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */
diff --git a/gst/audiofx/audiochebyshevfreqband.c b/gst/audiofx/audiochebyshevfreqband.c
new file mode 100644
index 00000000..d4730607
--- /dev/null
+++ b/gst/audiofx/audiochebyshevfreqband.c
@@ -0,0 +1,916 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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-audiochebyshevfreqband
+ * @short_description: Chebyshev band pass and band reject filter
+ *
+ * <refsect2>
+ * <para>
+ * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency
+ * band. The number of poles and the ripple parameter control the rolloff.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </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 "audiochebyshevfreqband.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand",
+    "Filter/Effect/Audio",
+    "Chebyshev band pass and band reject filter",
+    "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+  PROP_MODE,
+  PROP_TYPE,
+  PROP_LOWER_FREQUENCY,
+  PROP_UPPER_FREQUENCY,
+  PROP_RIPPLE,
+  PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-float,"                                              \
+    " width = (int) { 32, 64 }, "                                     \
+    " endianness = (int) BYTE_ORDER,"                                 \
+    " rate = (int) [ 1, MAX ],"                                       \
+    " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band,
+    GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_band_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_band_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter,
+    GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqBand * filter,
+    gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqBand * filter,
+    gfloat * data, guint num_samples);
+
+enum
+{
+  MODE_BAND_PASS = 0,
+  MODE_BAND_REJECT
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_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 ("GstAudioChebyshevFreqBandMode", values);
+  }
+  return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_band_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstCaps *caps;
+
+  gst_element_class_set_details (element_class, &element_details);
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_band_dispose (GObject * object)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqBandChannelCtx *ctx;
+    gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass *
+    klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseTransformClass *trans_class;
+  GstAudioFilterClass *filter_class;
+
+  gobject_class = (GObjectClass *) klass;
+  trans_class = (GstBaseTransformClass *) klass;
+  filter_class = (GstAudioFilterClass *) klass;
+
+  gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property;
+  gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property;
+  gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose;
+
+  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_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_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, G_MAXFLOAT,
+          0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  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, G_MAXFLOAT,
+          0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  g_object_class_install_property (gobject_class, PROP_RIPPLE,
+      g_param_spec_float ("ripple", "Ripple",
+          "Amount of ripple (dB)", 0.0, G_MAXFLOAT,
+          0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  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));
+
+  filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup);
+  trans_class->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip);
+  trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start);
+}
+
+static void
+gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter,
+    GstAudioChebyshevFreqBandClass * klass)
+{
+  filter->lower_frequency = filter->upper_frequency = 0.0;
+  filter->mode = MODE_BAND_PASS;
+  filter->type = 1;
+  filter->poles = 4;
+  filter->ripple = 0.25;
+  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+  filter->have_coeffs = FALSE;
+  filter->num_a = 0;
+  filter->num_b = 0;
+  filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqBand * 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 rz = 0.0, 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;
+
+    rz = 0.0;
+    iz = cos (angle);
+    mag2 = rz * rz + iz * iz;
+    rz /= mag2;
+    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;
+    }
+  }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+    gdouble zi)
+{
+  gdouble sum_ar, sum_ai;
+  gdouble sum_br, sum_bi;
+  gdouble gain_r, gain_i;
+
+  gdouble sum_r_old;
+  gdouble sum_i_old;
+
+  gint i;
+
+  sum_ar = 0.0;
+  sum_ai = 0.0;
+  for (i = num_a; i >= 0; i--) {
+    sum_r_old = sum_ar;
+    sum_i_old = sum_ai;
+
+    sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+    sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+  }
+
+  sum_br = 0.0;
+  sum_bi = 0.0;
+  for (i = num_b; i >= 0; i--) {
+    sum_r_old = sum_br;
+    sum_i_old = sum_bi;
+
+    sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+    sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+  }
+  sum_br += 1.0;
+  sum_bi += 0.0;
+
+  gain_r =
+      (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+  gain_i =
+      (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+  return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqBand * filter)
+{
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqBandChannelCtx *ctx;
+    gint i;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+    GST_LOG_OBJECT (filter, "rate was not set yet");
+    return;
+  }
+
+  filter->have_coeffs = TRUE;
+
+  if (filter->upper_frequency <= filter->lower_frequency) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+    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;
+
+    filter->num_a = np + 1;
+    filter->a = a = g_new0 (gdouble, np + 5);
+    filter->num_b = np + 1;
+    filter->b = b = g_new0 (gdouble, np + 5);
+
+    filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels);
+    for (i = 0; i < channels; i++) {
+      GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i];
+
+      ctx->x = g_new0 (gdouble, np + 1);
+      ctx->y = g_new0 (gdouble, np + 1);
+    }
+
+    /* 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 = calculate_gain (a, b, np, np, 1.0, 0.0);
+      gdouble gain2 = calculate_gain (a, b, np, np, -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 = calculate_gain (a, b, np, np, zr, zi);
+
+      for (i = 0; i <= np; i++) {
+        a[i] /= gain;
+      }
+    }
+
+    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 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+    {
+      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, zi;
+
+      zr = cos (w1);
+      zi = sin (w1);
+      GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+          20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)),
+          (int) filter->lower_frequency);
+      zr = cos (w0);
+      zi = sin (w0);
+      GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+          20.0 * log10 (calculate_gain (a, b, np, np, 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 (calculate_gain (a, b, np, np, zr, zi)),
+          (int) filter->upper_frequency);
+    }
+    GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz",
+        20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+        GST_AUDIO_FILTER (filter)->format.rate / 2);
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->mode = g_value_get_enum (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_TYPE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->type = g_value_get_int (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_LOWER_FREQUENCY:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->lower_frequency = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_UPPER_FREQUENCY:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->upper_frequency = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_RIPPLE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->ripple = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_POLES:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->poles = GST_ROUND_UP_4 (g_value_get_int (value));
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_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_chebyshev_freq_band_setup (GstAudioFilter * base,
+    GstRingBufferSpec * format)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+  gboolean ret = TRUE;
+
+  if (format->width == 32)
+    filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+        process_32;
+  else if (format->width == 64)
+    filter->process = (GstAudioChebyshevFreqBandProcessFunc)
+        process_64;
+  else
+    ret = FALSE;
+
+  filter->have_coeffs = FALSE;
+
+  return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqBand * filter,
+    GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0)
+{
+  gdouble val = filter->a[0] * x0;
+  gint i, j;
+
+  for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+    val += filter->a[i] * ctx->x[j];
+    j--;
+    if (j < 0)
+      j = filter->num_a - 1;
+  }
+
+  for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+    val += filter->b[i] * ctx->y[j];
+    j--;
+    if (j < 0)
+      j = filter->num_b - 1;
+  }
+
+  if (ctx->x) {
+    ctx->x_pos++;
+    if (ctx->x_pos > filter->num_a - 1)
+      ctx->x_pos = 0;
+    ctx->x[ctx->x_pos] = x0;
+  }
+
+  if (ctx->y) {
+    ctx->y_pos++;
+    if (ctx->y_pos > filter->num_b - 1)
+      ctx->y_pos = 0;
+
+    ctx->y[ctx->y_pos] = val;
+  }
+
+  return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqBand * filter,
+    gdouble * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqBand * filter,
+    gfloat * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+  guint num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  if (!gst_buffer_is_writable (buf))
+    return GST_FLOW_OK;
+
+  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+  if (!filter->have_coeffs)
+    generate_coefficients (filter);
+
+  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_audio_chebyshev_freq_band_start (GstBaseTransform * base)
+{
+  GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base);
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+  GstAudioChebyshevFreqBandChannelCtx *ctx;
+  gint i;
+
+  /* Reset the history of input and output values if
+   * already existing */
+  if (channels && filter->channels) {
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      if (ctx->x)
+        memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+      if (ctx->y)
+        memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+    }
+  }
+  return TRUE;
+}
diff --git a/gst/audiofx/audiochebyshevfreqband.h b/gst/audiofx/audiochebyshevfreqband.h
new file mode 100644
index 00000000..e8c58074
--- /dev/null
+++ b/gst/audiofx/audiochebyshevfreqband.h
@@ -0,0 +1,79 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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.
+ */
+
+#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND            (gst_audio_chebyshev_freq_band_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND))
+#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass))
+typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand;
+typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass;
+
+typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint);
+
+typedef struct
+{
+  gdouble *x;
+  gint x_pos;
+  gdouble *y;
+  gint y_pos;
+} GstAudioChebyshevFreqBandChannelCtx;
+
+struct _GstAudioChebyshevFreqBand
+{
+  GstAudioFilter audiofilter;
+
+  gint mode;
+  gint type;
+  gint poles;
+  gfloat lower_frequency;
+  gfloat upper_frequency;
+  gfloat ripple;
+
+  /* < private > */
+  GstAudioChebyshevFreqBandProcessFunc process;
+
+  gboolean have_coeffs;
+  gdouble *a;
+  gint num_a;
+  gdouble *b;
+  gint num_b;
+  GstAudioChebyshevFreqBandChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqBandClass
+{
+  GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_band_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */
diff --git a/gst/audiofx/audiochebyshevfreqlimit.c b/gst/audiofx/audiochebyshevfreqlimit.c
new file mode 100644
index 00000000..872b277d
--- /dev/null
+++ b/gst/audiofx/audiochebyshevfreqlimit.c
@@ -0,0 +1,816 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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-audiochebyshevfreqlimit
+ * @short_description: Chebyshev low pass and high pass filter
+ *
+ * <refsect2>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * 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.
+ * </para>
+ * <para>
+ * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink
+ * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </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 "audiochebyshevfreqlimit.h"
+
+#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit",
+    "Filter/Effect/Audio",
+    "Chebyshev low pass and high pass filter",
+    "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+  PROP_MODE,
+  PROP_TYPE,
+  PROP_CUTOFF,
+  PROP_RIPPLE,
+  PROP_POLES
+};
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-float,"                                              \
+    " width = (int) { 32, 64 }, "                                     \
+    " endianness = (int) BYTE_ORDER,"                                 \
+    " rate = (int) [ 1, MAX ],"                                       \
+    " channels = (int) [ 1, MAX ]"
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element");
+
+GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit,
+    gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER,
+    DEBUG_INIT);
+
+static void gst_audio_chebyshev_freq_limit_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_audio_chebyshev_freq_limit_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter,
+    GstRingBufferSpec * format);
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base);
+
+static void process_64 (GstAudioChebyshevFreqLimit * filter,
+    gdouble * data, guint num_samples);
+static void process_32 (GstAudioChebyshevFreqLimit * filter,
+    gfloat * data, guint num_samples);
+
+enum
+{
+  MODE_LOW_PASS = 0,
+  MODE_HIGH_PASS
+};
+
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ())
+static GType
+gst_audio_chebyshev_freq_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 ("GstAudioChebyshevFreqLimitMode", values);
+  }
+  return gtype;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_chebyshev_freq_limit_base_init (gpointer klass)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+  GstCaps *caps;
+
+  gst_element_class_set_details (element_class, &element_details);
+
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+      caps);
+  gst_caps_unref (caps);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_dispose (GObject * object)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqLimitChannelCtx *ctx;
+    gint i, channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass *
+    klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseTransformClass *trans_class;
+  GstAudioFilterClass *filter_class;
+
+  gobject_class = (GObjectClass *) klass;
+  trans_class = (GstBaseTransformClass *) klass;
+  filter_class = (GstAudioFilterClass *) klass;
+
+  gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property;
+  gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property;
+  gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose;
+
+  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_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_object_class_install_property (gobject_class, PROP_CUTOFF,
+      g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0,
+          G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  g_object_class_install_property (gobject_class, PROP_RIPPLE,
+      g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0,
+          G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+  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));
+
+  filter_class->setup =
+      GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup);
+  trans_class->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip);
+  trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start);
+}
+
+static void
+gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter,
+    GstAudioChebyshevFreqLimitClass * klass)
+{
+  filter->cutoff = 0.0;
+  filter->mode = MODE_LOW_PASS;
+  filter->type = 1;
+  filter->poles = 4;
+  filter->ripple = 0.25;
+  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+
+  filter->have_coeffs = FALSE;
+  filter->num_a = 0;
+  filter->num_b = 0;
+  filter->channels = NULL;
+}
+
+static void
+generate_biquad_coefficients (GstAudioChebyshevFreqLimit * 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 rz = 0.0, 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;
+
+    rz = 0.0;
+    iz = cos (angle);
+    mag2 = rz * rz + iz * iz;
+    rz /= mag2;
+    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;
+    }
+  }
+}
+
+/* Evaluate the transfer function that corresponds to the IIR
+ * coefficients at zr + zi*I and return the magnitude */
+static gdouble
+calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr,
+    gdouble zi)
+{
+  gdouble sum_ar, sum_ai;
+  gdouble sum_br, sum_bi;
+  gdouble gain_r, gain_i;
+
+  gdouble sum_r_old;
+  gdouble sum_i_old;
+
+  gint i;
+
+  sum_ar = 0.0;
+  sum_ai = 0.0;
+  for (i = num_a; i >= 0; i--) {
+    sum_r_old = sum_ar;
+    sum_i_old = sum_ai;
+
+    sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i];
+    sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0;
+  }
+
+  sum_br = 0.0;
+  sum_bi = 0.0;
+  for (i = num_b; i >= 0; i--) {
+    sum_r_old = sum_br;
+    sum_i_old = sum_bi;
+
+    sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i];
+    sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0;
+  }
+  sum_br += 1.0;
+  sum_bi += 0.0;
+
+  gain_r =
+      (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+  gain_i =
+      (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi);
+
+  return (sqrt (gain_r * gain_r + gain_i * gain_i));
+}
+
+static void
+generate_coefficients (GstAudioChebyshevFreqLimit * filter)
+{
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+
+  if (filter->a) {
+    g_free (filter->a);
+    filter->a = NULL;
+  }
+
+  if (filter->b) {
+    g_free (filter->b);
+    filter->b = NULL;
+  }
+
+  if (filter->channels) {
+    GstAudioChebyshevFreqLimitChannelCtx *ctx;
+    gint i;
+
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      g_free (ctx->x);
+      g_free (ctx->y);
+    }
+
+    g_free (filter->channels);
+    filter->channels = NULL;
+  }
+
+  if (GST_AUDIO_FILTER (filter)->format.rate == 0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    GST_LOG_OBJECT (filter, "rate was not set yet");
+    return;
+  }
+
+  filter->have_coeffs = TRUE;
+
+  if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency");
+    return;
+  } else if (filter->cutoff <= 0.0) {
+    filter->num_a = 1;
+    filter->a = g_new0 (gdouble, 1);
+    filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0;
+    filter->num_b = 0;
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    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;
+
+    filter->num_a = np + 1;
+    filter->a = a = g_new0 (gdouble, np + 3);
+    filter->num_b = np + 1;
+    filter->b = b = g_new0 (gdouble, np + 3);
+
+    filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels);
+    for (i = 0; i < channels; i++) {
+      GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i];
+
+      ctx->x = g_new0 (gdouble, np + 1);
+      ctx->y = g_new0 (gdouble, np + 1);
+    }
+
+    /* 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 = calculate_gain (a, b, np, np, 1.0, 0.0);
+      else
+        gain = calculate_gain (a, b, np, np, -1.0, 0.0);
+
+      for (i = 0; i <= np; i++) {
+        a[i] /= gain;
+      }
+    }
+
+    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 (calculate_gain (a, b, np, np, 1.0, 0.0)));
+    {
+      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 (calculate_gain (a, b, np, np, zr, zi)),
+          (int) filter->cutoff);
+    }
+    GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz",
+        20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)),
+        GST_AUDIO_FILTER (filter)->format.rate / 2);
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object);
+
+  switch (prop_id) {
+    case PROP_MODE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->mode = g_value_get_enum (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_TYPE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->type = g_value_get_int (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_CUTOFF:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->cutoff = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_RIPPLE:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->ripple = g_value_get_float (value);
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    case PROP_POLES:
+      GST_BASE_TRANSFORM_LOCK (filter);
+      filter->poles = GST_ROUND_UP_2 (g_value_get_int (value));
+      generate_coefficients (filter);
+      GST_BASE_TRANSFORM_UNLOCK (filter);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_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_chebyshev_freq_limit_setup (GstAudioFilter * base,
+    GstRingBufferSpec * format)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+  gboolean ret = TRUE;
+
+  if (format->width == 32)
+    filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+        process_32;
+  else if (format->width == 64)
+    filter->process = (GstAudioChebyshevFreqLimitProcessFunc)
+        process_64;
+  else
+    ret = FALSE;
+
+  filter->have_coeffs = FALSE;
+
+  return ret;
+}
+
+static inline gdouble
+process (GstAudioChebyshevFreqLimit * filter,
+    GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0)
+{
+  gdouble val = filter->a[0] * x0;
+  gint i, j;
+
+  for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) {
+    val += filter->a[i] * ctx->x[j];
+    j--;
+    if (j < 0)
+      j = filter->num_a - 1;
+  }
+
+  for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) {
+    val += filter->b[i] * ctx->y[j];
+    j--;
+    if (j < 0)
+      j = filter->num_b - 1;
+  }
+
+  if (ctx->x) {
+    ctx->x_pos++;
+    if (ctx->x_pos > filter->num_a - 1)
+      ctx->x_pos = 0;
+    ctx->x[ctx->x_pos] = x0;
+  }
+
+  if (ctx->y) {
+    ctx->y_pos++;
+    if (ctx->y_pos > filter->num_b - 1)
+      ctx->y_pos = 0;
+
+    ctx->y[ctx->y_pos] = val;
+  }
+
+  return val;
+}
+
+static void
+process_64 (GstAudioChebyshevFreqLimit * filter,
+    gdouble * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+static void
+process_32 (GstAudioChebyshevFreqLimit * filter,
+    gfloat * data, guint num_samples)
+{
+  gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels;
+  gdouble val;
+
+  for (i = 0; i < num_samples / channels; i++) {
+    for (j = 0; j < channels; j++) {
+      val = process (filter, &filter->channels[j], *data);
+      *data++ = val;
+    }
+  }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+  guint num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  if (!gst_buffer_is_writable (buf))
+    return GST_FLOW_OK;
+
+  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+  if (!filter->have_coeffs)
+    generate_coefficients (filter);
+
+  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+  return GST_FLOW_OK;
+}
+
+
+static gboolean
+gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base)
+{
+  GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base);
+  gint channels = GST_AUDIO_FILTER (filter)->format.channels;
+  GstAudioChebyshevFreqLimitChannelCtx *ctx;
+  gint i;
+
+  /* Reset the history of input and output values if
+   * already existing */
+  if (channels && filter->channels) {
+    for (i = 0; i < channels; i++) {
+      ctx = &filter->channels[i];
+      if (ctx->x)
+        memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble));
+      if (ctx->y)
+        memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble));
+    }
+  }
+  return TRUE;
+}
diff --git a/gst/audiofx/audiochebyshevfreqlimit.h b/gst/audiofx/audiochebyshevfreqlimit.h
new file mode 100644
index 00000000..4c87ba8e
--- /dev/null
+++ b/gst/audiofx/audiochebyshevfreqlimit.h
@@ -0,0 +1,78 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * 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.
+ */
+
+#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT            (gst_audio_chebyshev_freq_limit_get_type())
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT))
+#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass))
+typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit;
+typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass;
+
+typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint);
+
+typedef struct
+{
+  gdouble *x;
+  gint x_pos;
+  gdouble *y;
+  gint y_pos;
+} GstAudioChebyshevFreqLimitChannelCtx;
+
+struct _GstAudioChebyshevFreqLimit
+{
+  GstAudioFilter audiofilter;
+
+  gint mode;
+  gint type;
+  gint poles;
+  gfloat cutoff;
+  gfloat ripple;
+
+  /* < private > */
+  GstAudioChebyshevFreqLimitProcessFunc process;
+
+  gboolean have_coeffs;
+  gdouble *a;
+  gint num_a;
+  gdouble *b;
+  gint num_b;
+  GstAudioChebyshevFreqLimitChannelCtx *channels;
+};
+
+struct _GstAudioChebyshevFreqLimitClass
+{
+  GstAudioFilterClass parent;
+};
+
+GType gst_audio_chebyshev_freq_limit_get_type (void);
+
+G_END_DECLS
+#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */
diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c
index e6d84d24..2c198f32 100644
--- a/gst/audiofx/audiofx.c
+++ b/gst/audiofx/audiofx.c
@@ -29,6 +29,8 @@
 #include "audioinvert.h"
 #include "audioamplify.h"
 #include "audiodynamic.h"
+#include "audiochebyshevfreqlimit.h"
+#include "audiochebyshevfreqband.h"
 
 /* entry point to initialize the plug-in
  * initialize the plug-in itself
@@ -48,7 +50,11 @@ plugin_init (GstPlugin * plugin)
       gst_element_register (plugin, "audioamplify", GST_RANK_NONE,
           GST_TYPE_AUDIO_AMPLIFY) &&
       gst_element_register (plugin, "audiodynamic", GST_RANK_NONE,
-          GST_TYPE_AUDIO_DYNAMIC));
+          GST_TYPE_AUDIO_DYNAMIC) &&
+      gst_element_register (plugin, "audiochebyshevfreqlimit", GST_RANK_NONE,
+          GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT) &&
+      gst_element_register (plugin, "audiochebyshevfreqband", GST_RANK_NONE,
+          GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND));
 }
 
 GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,