Fix up to use the newly ported (actually working) GstAudioFilter.

Original commit message from CVS:
* configure.ac:
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init),
(gst_iir_equalizer_class_init), (gst_iir_equalizer_init),
(setup_filter), (gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup),
(plugin_init):
* gst/equalizer/gstiirequalizer.h:
Fix up to use the newly ported (actually working) GstAudioFilter.
Bump core/base requirements to CVS for this.
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/equalizer-test.c: (check_bus),
(equalizer_set_band_value), (equalizer_set_all_band_values),
(equalizer_set_band_value_and_wait),
(equalizer_set_all_band_values_and_wait), (do_slider_fiddling),
(main):
Add brain-dead interactive test for equalizer.

8 files changed, 503 insertions, 152 deletions

diff --git a/ChangeLog b/ChangeLog
index c4483199..2c29a06c 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,26 @@
+2007-02-03  Tim-Philipp Müller  <tim at centricular dot net>
+
+	* configure.ac:
+	* gst/equalizer/Makefile.am:
+	* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_base_init),
+	(gst_iir_equalizer_class_init), (gst_iir_equalizer_init),
+	(setup_filter), (gst_iir_equalizer_compute_frequencies),
+	(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
+	(gst_iir_equalizer_transform_ip), (gst_iir_equalizer_setup),
+	(plugin_init):
+	* gst/equalizer/gstiirequalizer.h:
+	  Fix up to use the newly ported (actually working) GstAudioFilter.
+	  Bump core/base requirements to CVS for this.
+
+	* tests/icles/.cvsignore:
+	* tests/icles/Makefile.am:
+	* tests/icles/equalizer-test.c: (check_bus),
+	(equalizer_set_band_value), (equalizer_set_all_band_values),
+	(equalizer_set_band_value_and_wait),
+	(equalizer_set_all_band_values_and_wait), (do_slider_fiddling),
+	(main):
+	  Add brain-dead interactive test for equalizer.
+
 2007-02-02  Tim-Philipp Müller  <tim at centricular dot net>
 
 	* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_class_init),
diff --git a/configure.ac b/configure.ac
index c10f193c..61255482 100644
--- a/configure.ac
+++ b/configure.ac
@@ -42,8 +42,8 @@ dnl AS_LIBTOOL_TAGS([CXX])
 AM_PROG_LIBTOOL
 
 dnl *** required versions of GStreamer stuff ***
-GST_REQ=0.10.10.1
-GSTPB_REQ=0.10.10.1
+GST_REQ=0.10.11.1
+GSTPB_REQ=0.10.11.1
 
 dnl *** autotools stuff ****
 
diff --git a/gst/equalizer/Makefile.am b/gst/equalizer/Makefile.am
index ed26e295..1ef12c94 100644
--- a/gst/equalizer/Makefile.am
+++ b/gst/equalizer/Makefile.am
@@ -1,6 +1,8 @@
 plugin_LTLIBRARIES = libgstequalizer.la
 
-libgstequalizer_la_SOURCES = gstiirequalizer.c
+libgstequalizer_la_SOURCES = gstiirequalizer.c gstiirequalizer.h
 libgstequalizer_la_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
 libgstequalizer_la_LIBADD = $(GST_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR)
 libgstequalizer_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_HEADERS = gstiirequalizer.h
diff --git a/gst/equalizer/gstiirequalizer.c b/gst/equalizer/gstiirequalizer.c
index fd8f3781..48dfac42 100644
--- a/gst/equalizer/gstiirequalizer.c
+++ b/gst/equalizer/gstiirequalizer.c
@@ -23,157 +23,91 @@
 
 #include <math.h>
 #include <string.h>
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-#include <gst/audio/gstaudiofilter.h>
-
-typedef struct _GstIirEqualizer GstIirEqualizer;
-typedef struct _GstIirEqualizerClass GstIirEqualizerClass;
-
-#define GST_TYPE_IIR_EQUALIZER \
-  (gst_iir_equalizer_get_type())
-#define GST_IIR_EQUALIZER(obj) \
-  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_IIR_EQUALIZER,GstIirEqualizer))
-#define GST_IIR_EQUALIZER_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_IIR_EQUALIZER,GstIirEqualizerClass))
-#define GST_IS_IIR_EQUALIZER(obj) \
-  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_IIR_EQUALIZER))
-#define GST_IS_IIR_EQUALIZER_CLASS(klass) \
-  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_IIR_EQUALIZER))
-
-#define LOWEST_FREQ (20.0)
-#define HIGHEST_FREQ (20000.0)
-
-typedef void (*ProcessFunc) (GstIirEqualizer * equ, guint8 * data, guint size,
-    guint channels);
-
-typedef struct
-{
-  gdouble alpha;                /* IIR coefficients for outputs */
-  gdouble beta;                 /* IIR coefficients for inputs */
-  gdouble gamma;                /* IIR coefficients for inputs */
-} SecondOrderFilter;
 
-struct _GstIirEqualizer
-{
-  GstAudioFilter audiofilter;
-
-  /* properties */
-  guint freq_count;
-  gdouble bandwidth;
-  gdouble *freqs;
-  gdouble *values;
-
-  /* data */
-  SecondOrderFilter *filter;
-  gpointer history;
-  ProcessFunc process;
-  guint history_size;
-};
+#include "gstiirequalizer.h"
 
-struct _GstIirEqualizerClass
-{
-  GstAudioFilterClass audiofilter_class;
-};
+#define GST_EQUALIZER_TRANSFORM_LOCK(eq) \
+    g_mutex_lock (GST_BASE_TRANSFORM(eq)->transform_lock)
+
+#define GST_EQUALIZER_TRANSFORM_UNLOCK(eq) \
+    g_mutex_unlock (GST_BASE_TRANSFORM(eq)->transform_lock)
 
 enum
 {
   ARG_0,
-  ARG_BANDS,
-  ARG_BANDWIDTH,
+  ARG_NUM_BANDS,
+  ARG_BAND_WIDTH,
   ARG_BAND_VALUES
 };
 
-static void gst_iir_equalizer_base_init (gpointer g_class);
-static void gst_iir_equalizer_class_init (gpointer g_class,
-    gpointer class_data);
-static void gst_iir_equalizer_init (GTypeInstance * instance, gpointer g_class);
 static void gst_iir_equalizer_finalize (GObject * object);
-
 static void gst_iir_equalizer_set_property (GObject * object,
     guint prop_id, const GValue * value, GParamSpec * pspec);
 static void gst_iir_equalizer_get_property (GObject * object,
     guint prop_id, GValue * value, GParamSpec * pspec);
 
-static void gst_iir_equalizer_setup (GstAudioFilter * iir_equalizer);
-static void gst_iir_equalizer_filter_inplace (GstAudioFilter *
-    iir_equalizer, GstBuffer * buf);
-
-static GstAudioFilterClass *parent_class;
-
-GType
-gst_iir_equalizer_get_type (void)
-{
-  static GType iir_equalizer_type = 0;
-
-  if (!iir_equalizer_type) {
-    static const GTypeInfo iir_equalizer_info = {
-      sizeof (GstIirEqualizerClass),
-      gst_iir_equalizer_base_init,
-      NULL,
-      gst_iir_equalizer_class_init,
-      NULL,
-      gst_iir_equalizer_init,
-      sizeof (GstIirEqualizer),
-      0,
-      NULL,
-    };
-
-    iir_equalizer_type = g_type_register_static (GST_TYPE_AUDIO_FILTER,
-        "GstIirEqualizer", &iir_equalizer_info, 0);
-  }
-  return iir_equalizer_type;
-}
+static gboolean gst_iir_equalizer_setup (GstAudioFilter * filter,
+    GstRingBufferSpec * fmt);
+static GstFlowReturn gst_iir_equalizer_transform_ip (GstBaseTransform * btrans,
+    GstBuffer * buf);
+
+GST_DEBUG_CATEGORY_STATIC (equalizer_debug);
+#define GST_CAT_DEFAULT equalizer_debug
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-int,"                                                \
+    " depth=(int)16,"                                                 \
+    " width=(int)16,"                                                 \
+    " endianness=(int)BYTE_ORDER,"                                    \
+    " signed=(bool)TRUE,"                                             \
+    " rate=(int)[1000,MAX],"                                          \
+    " channels=(int)[1,MAX]; "                                        \
+    "audio/x-raw-float,"                                              \
+    " width=(int)32,"                                                 \
+    " endianness=(int)BYTE_ORDER,"                                    \
+    " rate=(int)[1000,MAX],"                                          \
+    " channels=(int)[1,MAX]"
+
+GST_BOILERPLATE (GstIirEqualizer, gst_iir_equalizer, GstAudioFilter,
+    GST_TYPE_AUDIO_FILTER);
 
 static void
 gst_iir_equalizer_base_init (gpointer g_class)
 {
-  static const GstElementDetails iir_equalizer_details =
+  GstAudioFilterClass *audiofilter_class = GST_AUDIO_FILTER_CLASS (g_class);
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+  const GstElementDetails iir_equalizer_details =
       GST_ELEMENT_DETAILS ("Equalizer",
       "Filter/Effect/Audio",
       "Direct Form IIR equalizer",
       "Benjamin Otte <otte@gnome.org>");
-  GstIirEqualizerClass *klass = (GstIirEqualizerClass *) g_class;
-  GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
   GstCaps *caps;
 
   gst_element_class_set_details (element_class, &iir_equalizer_details);
 
-  caps = gst_caps_from_string ("audio/x-raw-int, depth=(int)16, width=(int)16, "
-      "endianness=(int)BYTE_ORDER, signed=(bool)TRUE, "
-      "rate=(int)[1000,MAX], channels=(int)[1,6];"
-      "audio/x-raw-float, width=(int)32, endianness=(int)BYTE_ORDER,"
-      "rate=(int)[1000,MAX], channels=(int)[1,6]");
-  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (g_class),
-      caps);
+  caps = gst_caps_from_string (ALLOWED_CAPS);
+  gst_audio_filter_class_add_pad_templates (audiofilter_class, caps);
   gst_caps_unref (caps);
 }
 
 static void
-gst_iir_equalizer_class_init (gpointer g_class, gpointer class_data)
+gst_iir_equalizer_class_init (GstIirEqualizerClass * klass)
 {
-  GObjectClass *gobject_class;
-  GstElementClass *gstelement_class;
-  GstIirEqualizerClass *klass;
-  GstAudioFilterClass *audio_filter_class;
-
-  klass = (GstIirEqualizerClass *) g_class;
-  gobject_class = (GObjectClass *) klass;
-  gstelement_class = (GstElementClass *) klass;
-  audio_filter_class = (GstAudioFilterClass *) g_class;
+  GstAudioFilterClass *audio_filter_class = (GstAudioFilterClass *) klass;
+  GstBaseTransformClass *btrans_class = (GstBaseTransformClass *) klass;
+  GObjectClass *gobject_class = (GObjectClass *) klass;
 
   gobject_class->set_property = gst_iir_equalizer_set_property;
   gobject_class->get_property = gst_iir_equalizer_get_property;
   gobject_class->finalize = gst_iir_equalizer_finalize;
 
-  parent_class = g_type_class_peek_parent (g_class);
-
-  g_object_class_install_property (gobject_class, ARG_BANDS,
-      g_param_spec_uint ("bands", "bands", "number of different bands to use",
-          2, 64, 15, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
-  g_object_class_install_property (gobject_class, ARG_BANDWIDTH,
-      g_param_spec_double ("bandwidth", "bandwidth",
-          "bandwidth calculated as distance between bands * this value", 0.1,
+  g_object_class_install_property (gobject_class, ARG_NUM_BANDS,
+      g_param_spec_uint ("num-bands", "num-bands",
+          "number of different bands to use", 2, 64, 15,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+  g_object_class_install_property (gobject_class, ARG_BAND_WIDTH,
+      g_param_spec_double ("band-width", "band-width",
+          "band width calculated as distance between bands * this value", 0.1,
           5.0, 1.0, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
   g_object_class_install_property (gobject_class, ARG_BAND_VALUES,
       g_param_spec_value_array ("band-values", "band values",
@@ -184,12 +118,13 @@ gst_iir_equalizer_class_init (gpointer g_class, gpointer class_data)
           G_PARAM_WRITABLE));
 
   audio_filter_class->setup = gst_iir_equalizer_setup;
-  audio_filter_class->filter_inplace = gst_iir_equalizer_filter_inplace;
+  btrans_class->transform_ip = gst_iir_equalizer_transform_ip;
 }
 
 static void
-gst_iir_equalizer_init (GTypeInstance * instance, gpointer g_class)
+gst_iir_equalizer_init (GstIirEqualizer * eq, GstIirEqualizerClass * g_class)
 {
+  /* nothing to do here */
 }
 
 static void
@@ -220,7 +155,7 @@ setup_filter (GstIirEqualizer * equ, SecondOrderFilter * filter, gdouble gain,
     gdouble frequency)
 {
   gdouble q = pow (HIGHEST_FREQ / LOWEST_FREQ,
-      1.0 / (equ->freq_count - 1)) * equ->bandwidth;
+      1.0 / (equ->freq_count - 1)) * equ->band_width;
   gdouble theta = frequency * 2 * M_PI;
 
   filter->beta = (q - theta / 2) / (2 * q + theta);
@@ -257,21 +192,22 @@ gst_iir_equalizer_compute_frequencies (GstIirEqualizer * equ, guint band_count)
     memset (equ->filter + sizeof (SecondOrderFilter) * old_count, 0,
         sizeof (SecondOrderFilter) * (band_count - old_count));
   }
+
+  /* free + alloc = no memcpy */
+  g_free (equ->history);
   equ->history =
-      g_realloc (equ->history,
-      equ->history_size * audio->channels * band_count);
-  memset (equ->history, 0, equ->history_size * audio->channels * band_count);
+      g_malloc0 (equ->history_size * audio->format.channels * band_count);
   equ->freqs[0] = LOWEST_FREQ;
   for (i = 1; i < band_count; i++) {
     equ->freqs[i] = equ->freqs[i - 1] * step;
   }
 
-  if (audio->rate) {
+  if (audio->format.rate > 0) {
     guint i;
 
     for (i = 0; i < band_count; i++) {
       setup_filter (equ, &equ->filter[i], arg_to_scale (equ->values[i]),
-          equ->freqs[i] / audio->rate);
+          equ->freqs[i] / audio->format.rate);
     }
   }
 }
@@ -282,20 +218,21 @@ gst_iir_equalizer_set_property (GObject * object, guint prop_id,
 {
   GstIirEqualizer *equ = GST_IIR_EQUALIZER (object);
 
+  GST_EQUALIZER_TRANSFORM_LOCK (equ);
   GST_OBJECT_LOCK (equ);
   switch (prop_id) {
-    case ARG_BANDS:
+    case ARG_NUM_BANDS:
       gst_iir_equalizer_compute_frequencies (equ, g_value_get_uint (value));
       break;
-    case ARG_BANDWIDTH:
-      if (g_value_get_double (value) != equ->bandwidth) {
-        equ->bandwidth = g_value_get_double (value);
-        if (GST_AUDIO_FILTER (equ)->rate) {
+    case ARG_BAND_WIDTH:
+      if (g_value_get_double (value) != equ->band_width) {
+        equ->band_width = g_value_get_double (value);
+        if (GST_AUDIO_FILTER (equ)->format.rate) {
           guint i;
 
           for (i = 0; i < equ->freq_count; i++) {
             setup_filter (equ, &equ->filter[i], arg_to_scale (equ->values[i]),
-                equ->freqs[i] / GST_AUDIO_FILTER (equ)->rate);
+                equ->freqs[i] / GST_AUDIO_FILTER (equ)->format.rate);
           }
         }
       }
@@ -319,7 +256,7 @@ gst_iir_equalizer_set_property (GObject * object, guint prop_id,
           if (new_val != equ->values[i]) {
             equ->values[i] = new_val;
             setup_filter (equ, &equ->filter[i], arg_to_scale (new_val),
-                equ->freqs[i] / GST_AUDIO_FILTER (equ)->rate);
+                equ->freqs[i] / GST_AUDIO_FILTER (equ)->format.rate);
           }
         }
       }
@@ -330,6 +267,7 @@ gst_iir_equalizer_set_property (GObject * object, guint prop_id,
       break;
   }
   GST_OBJECT_UNLOCK (equ);
+  GST_EQUALIZER_TRANSFORM_UNLOCK (equ);
 }
 
 static void
@@ -338,19 +276,21 @@ gst_iir_equalizer_get_property (GObject * object, guint prop_id,
 {
   GstIirEqualizer *equ = GST_IIR_EQUALIZER (object);
 
+  GST_EQUALIZER_TRANSFORM_LOCK (equ);
   GST_OBJECT_LOCK (equ);
   switch (prop_id) {
-    case ARG_BANDS:
+    case ARG_NUM_BANDS:
       g_value_set_uint (value, equ->freq_count);
       break;
-    case ARG_BANDWIDTH:
-      g_value_set_double (value, equ->bandwidth);
+    case ARG_BAND_WIDTH:
+      g_value_set_double (value, equ->band_width);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
   }
   GST_OBJECT_UNLOCK (equ);
+  GST_EQUALIZER_TRANSFORM_UNLOCK (equ);
 }
 
 /* start of code that is type specific */
@@ -411,37 +351,47 @@ guint size, guint channels)                                             \
 CREATE_OPTIMIZED_FUNCTIONS (gint16, gint, -32768, 32767);
 CREATE_OPTIMIZED_FUNCTIONS (gfloat, gfloat, -1.0, 1.0);
 
-static void
-gst_iir_equalizer_filter_inplace (GstAudioFilter * filter, GstBuffer * buf)
+static GstFlowReturn
+gst_iir_equalizer_transform_ip (GstBaseTransform * btrans, GstBuffer * buf)
 {
-  GstIirEqualizer *equ = GST_IIR_EQUALIZER (filter);
+  GstAudioFilter *filter = GST_AUDIO_FILTER (btrans);
+  GstIirEqualizer *equ = GST_IIR_EQUALIZER (btrans);
+
+  if (G_UNLIKELY (filter->format.channels < 1 || equ->process == NULL))
+    return GST_FLOW_NOT_NEGOTIATED;
 
-  GST_OBJECT_LOCK (equ);
   equ->process (equ, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
-      filter->channels);
-  GST_OBJECT_UNLOCK (equ);
+      filter->format.channels);
+
+  return GST_FLOW_OK;
 }
 
-static void
-gst_iir_equalizer_setup (GstAudioFilter * audio)
+static gboolean
+gst_iir_equalizer_setup (GstAudioFilter * audio, GstRingBufferSpec * fmt)
 {
   GstIirEqualizer *equ = GST_IIR_EQUALIZER (audio);
 
-  if (audio->width == 16) {
-    equ->history_size = history_size_gint16;
-    equ->process = gst_iir_equ_process_gint16;
-  } else if (audio->width == 32) {
-    equ->history_size = history_size_gfloat;
-    equ->process = gst_iir_equ_process_gfloat;
-  } else {
-    g_assert_not_reached ();
+  switch (fmt->width) {
+    case 16:
+      equ->history_size = history_size_gint16;
+      equ->process = gst_iir_equ_process_gint16;
+      break;
+    case 32:
+      equ->history_size = history_size_gfloat;
+      equ->process = gst_iir_equ_process_gfloat;
+      break;
+    default:
+      return FALSE;
   }
   gst_iir_equalizer_compute_frequencies (equ, equ->freq_count);
+  return TRUE;
 }
 
 static gboolean
 plugin_init (GstPlugin * plugin)
 {
+  GST_DEBUG_CATEGORY_INIT (equalizer_debug, "equalizer", 0, "equalizer");
+
   return gst_element_register (plugin, "equalizer", GST_RANK_NONE,
       GST_TYPE_IIR_EQUALIZER);
 }
diff --git a/gst/equalizer/gstiirequalizer.h b/gst/equalizer/gstiirequalizer.h
new file mode 100644
index 00000000..2193369b
--- /dev/null
+++ b/gst/equalizer/gstiirequalizer.h
@@ -0,0 +1,77 @@
+/* GStreamer IIR equalizer
+ * Copyright (C) <2004> Benjamin Otte <otte@gnome.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_IIR_EQUALIZER__
+#define __GST_IIR_EQUALIZER__
+
+#include <gst/audio/gstaudiofilter.h>
+#include <gst/audio/gstringbuffer.h>
+
+typedef struct _GstIirEqualizer GstIirEqualizer;
+typedef struct _GstIirEqualizerClass GstIirEqualizerClass;
+
+#define GST_TYPE_IIR_EQUALIZER \
+  (gst_iir_equalizer_get_type())
+#define GST_IIR_EQUALIZER(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_IIR_EQUALIZER,GstIirEqualizer))
+#define GST_IIR_EQUALIZER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_IIR_EQUALIZER,GstIirEqualizerClass))
+#define GST_IS_IIR_EQUALIZER(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_IIR_EQUALIZER))
+#define GST_IS_IIR_EQUALIZER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_IIR_EQUALIZER))
+
+#define LOWEST_FREQ (20.0)
+#define HIGHEST_FREQ (20000.0)
+
+typedef void (*ProcessFunc) (GstIirEqualizer * eq, guint8 * data, guint size,
+    guint channels);
+
+typedef struct
+{
+  gdouble alpha;                /* IIR coefficients for outputs */
+  gdouble beta;                 /* IIR coefficients for inputs */
+  gdouble gamma;                /* IIR coefficients for inputs */
+} SecondOrderFilter;
+
+struct _GstIirEqualizer
+{
+  GstAudioFilter audiofilter;
+
+  /*< private >*/
+
+  /* properties */
+  guint freq_count;
+  gdouble band_width;
+  gdouble *freqs;
+  gdouble *values;
+
+  /* data */
+  SecondOrderFilter *filter;
+  gpointer history;
+  ProcessFunc process;
+  guint history_size;
+};
+
+struct _GstIirEqualizerClass
+{
+  GstAudioFilterClass audiofilter_class;
+};
+
+#endif /* __GST_IIR_EQUALIZER__ */
diff --git a/tests/icles/.gitignore b/tests/icles/.gitignore
index 364303c6..bb39cc6c 100644
--- a/tests/icles/.gitignore
+++ b/tests/icles/.gitignore
@@ -1,3 +1,4 @@
+equalizer-test
 pitch-test
 v4l2src-test
 videocrop-test
diff --git a/tests/icles/Makefile.am b/tests/icles/Makefile.am
index 26e0cbc3..b57880e4 100644
--- a/tests/icles/Makefile.am
+++ b/tests/icles/Makefile.am
@@ -16,4 +16,9 @@ videocrop_test_CFLAGS  = $(GST_CFLAGS)
 videocrop_test_LDADD   = $(GST_LIBS)
 videocrop_test_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 
-noinst_PROGRAMS = $(GST_SOUNDTOUCH_TESTS) videocrop-test
+equalizer_test_SOURCES = equalizer-test.c
+equalizer_test_CFLAGS  = $(GST_CFLAGS)
+equalizer_test_LDADD   = $(GST_LIBS)
+equalizer_test_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+noinst_PROGRAMS = $(GST_SOUNDTOUCH_TESTS) equalizer-test videocrop-test
diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c
new file mode 100644
index 00000000..c7603982
--- /dev/null
+++ b/tests/icles/equalizer-test.c
@@ -0,0 +1,293 @@
+/* GStreamer test for the equalizer element
+ * Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <gst/gst.h>
+
+#include <stdlib.h>
+#include <math.h>
+
+GST_DEBUG_CATEGORY_STATIC (equalizer_test_debug);
+#define GST_CAT_DEFAULT equalizer_test_debug
+
+static GstBus *pipeline_bus;
+
+static gboolean
+check_bus (GstClockTime max_wait_time)
+{
+  GstMessage *msg;
+
+  msg = gst_bus_poll (pipeline_bus, GST_MESSAGE_ERROR | GST_MESSAGE_EOS,
+      max_wait_time);
+
+  if (msg == NULL)
+    return FALSE;
+
+  if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR) {
+    GError *err = NULL;
+    gchar *debug = NULL;
+
+    g_assert (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_ERROR);
+    gst_message_parse_error (msg, &err, &debug);
+    GST_ERROR ("ERROR: %s [%s]", err->message, debug);
+    g_print ("\n===========> ERROR: %s\n%s\n\n", err->message, debug);
+    g_error_free (err);
+    g_free (debug);
+  }
+
+  if (GST_MESSAGE_TYPE (msg) == GST_MESSAGE_EOS) {
+    g_print ("\n === EOS ===\n\n");
+  }
+
+  gst_message_unref (msg);
+  return TRUE;
+}
+
+static void
+equalizer_set_band_value (GstElement * eq, GValueArray * arr, guint band,
+    gdouble val)
+{
+  g_value_set_double (g_value_array_get_nth (arr, band), val);
+  g_object_set (eq, "band-values", arr, NULL);
+  g_print ("Band %2d: %.2f\n", band, val);
+}
+
+static void
+equalizer_set_all_band_values (GstElement * eq, GValueArray * arr, gdouble val)
+{
+  gint i;
+
+  for (i = 0; i < arr->n_values; ++i) {
+    g_value_set_double (g_value_array_get_nth (arr, i), val);
+  }
+  g_object_set (eq, "band-values", arr, NULL);
+  g_print ("All bands: %.2f\n", val);
+}
+
+static gboolean
+equalizer_set_band_value_and_wait (GstElement * eq, GValueArray * arr,
+    guint band, gdouble val)
+{
+  equalizer_set_band_value (eq, arr, band, val);
+  return check_bus (100 * GST_MSECOND);
+}
+
+static gboolean
+equalizer_set_all_band_values_and_wait (GstElement * eq, GValueArray * arr,
+    gdouble val)
+{
+  equalizer_set_all_band_values (eq, arr, val);
+  return check_bus (100 * GST_MSECOND);
+}
+
+static void
+do_slider_fiddling (GstElement * playbin, GstElement * eq)
+{
+  GValueArray *arr;
+  gboolean stop;
+  guint num_bands, i;
+  gdouble d, step = 0.2;
+
+  stop = FALSE;
+
+  g_object_get (eq, "num-bands", &num_bands, NULL);
+
+  g_print ("%u bands.\n", num_bands);
+
+  arr = g_value_array_new (num_bands);
+  for (i = 0; i < num_bands; ++i) {
+    GValue val = { 0, };
+
+    g_value_init (&val, G_TYPE_DOUBLE);
+    g_value_set_double (&val, 0.0);
+    g_value_array_append (arr, &val);
+  }
+
+  g_object_set (eq, "band-values", arr, NULL);
+
+  while (!stop) {
+    for (i = 0; !stop && i < num_bands; ++i) {
+      d = 0.0;
+      while (!stop && d <= 1.0) {
+        stop = equalizer_set_band_value_and_wait (eq, arr, i, d);
+        d += step;
+      }
+      d = 1.0;
+      while (!stop && d >= -1.0) {
+        stop = equalizer_set_band_value_and_wait (eq, arr, i, d);
+        d -= step;
+      }
+      d = -1.0;
+      while (!stop && d <= 0.0) {
+        stop = equalizer_set_band_value_and_wait (eq, arr, i, d);
+        d += step;
+      }
+    }
+
+    d = 0.0;
+    while (!stop && d <= 1.0) {
+      stop = equalizer_set_all_band_values_and_wait (eq, arr, d);
+      d += step;
+    }
+    d = 1.0;
+    while (!stop && d >= -1.0) {
+      stop = equalizer_set_all_band_values_and_wait (eq, arr, d);
+      d -= step;
+    }
+    d = -1.0;
+    while (!stop && d <= 0.0) {
+      stop = equalizer_set_all_band_values_and_wait (eq, arr, d);
+      d += step;
+    }
+  }
+
+  g_value_array_free (arr);
+}
+
+int
+main (int argc, char **argv)
+{
+  gchar *opt_audiosink_str = NULL;
+  gchar **filenames = NULL;
+  const GOptionEntry test_goptions[] = {
+    {"audiosink", '\0', 0, G_OPTION_ARG_STRING, &opt_audiosink_str,
+        "audiosink to use (default: autoaudiosink)", NULL},
+    {G_OPTION_REMAINING, 0, 0, G_OPTION_ARG_FILENAME_ARRAY, &filenames, NULL},
+    {NULL, '\0', 0, 0, NULL, NULL, NULL}
+  };
+  GOptionContext *ctx;
+  GError *opt_err = NULL;
+
+  GstStateChangeReturn ret;
+  GstElement *playbin, *sink, *bin, *eq, *auconv;
+  GstPad *eq_sinkpad;
+  gchar *uri;
+
+  if (!g_thread_supported ())
+    g_thread_init (NULL);
+
+  GST_DEBUG_CATEGORY_INIT (equalizer_test_debug, "equalizertest", 0, "eqtest");
+
+  /* command line option parsing */
+  ctx = g_option_context_new ("FILENAME");
+  g_option_context_add_group (ctx, gst_init_get_option_group ());
+  g_option_context_add_main_entries (ctx, test_goptions, NULL);
+
+  if (!g_option_context_parse (ctx, &argc, &argv, &opt_err)) {
+    g_error ("Error parsing command line options: %s", opt_err->message);
+    return -1;
+  }
+
+  if (filenames == NULL || *filenames == NULL) {
+    g_printerr ("Please specify a file to play back\n");
+    return -1;
+  }
+
+  playbin = gst_element_factory_make ("playbin", "playbin");
+  if (playbin == NULL) {
+    g_error ("Couldn't create 'playbin' element");
+    return -1;
+  }
+
+  if (opt_audiosink_str) {
+    g_print ("Trying audiosink '%s' ...", opt_audiosink_str);
+    sink = gst_element_factory_make (opt_audiosink_str, "sink");
+    g_print ("%s\n", (sink) ? "ok" : "element couldn't be created");
+  } else {
+    sink = NULL;
+  }
+  if (sink == NULL) {
+    g_print ("Trying audiosink '%s' ...", "autoaudiosink");
+    sink = gst_element_factory_make ("autoaudiosink", "sink");
+    g_print ("%s\n", (sink) ? "ok" : "element couldn't be created");
+  }
+  if (sink == NULL) {
+    g_print ("Trying audiosink '%s' ...", "alsasink");
+    sink = gst_element_factory_make ("alsasink", "sink");
+    g_print ("%s\n", (sink) ? "ok" : "element couldn't be created");
+  }
+  if (sink == NULL) {
+    g_print ("Trying audiosink '%s' ...", "osssink");
+    sink = gst_element_factory_make ("osssink", "sink");
+    g_print ("%s\n", (sink) ? "ok" : "element couldn't be created");
+  }
+
+  g_assert (sink != NULL);
+
+  bin = gst_bin_new ("ausinkbin");
+  g_assert (bin != NULL);
+
+  eq = gst_element_factory_make ("equalizer", "equalizer");
+  g_assert (eq != NULL);
+
+  auconv = gst_element_factory_make ("audioconvert", "eqauconv");
+  g_assert (auconv != NULL);
+
+  gst_bin_add_many (GST_BIN (bin), eq, auconv, sink, NULL);
+
+  if (!gst_element_link (eq, auconv))
+    g_error ("Failed to link equalizer to audioconvert");
+
+  if (!gst_element_link (auconv, sink))
+    g_error ("Failed to link audioconvert to audio sink");
+
+
+  eq_sinkpad = gst_element_get_static_pad (eq, "sink");
+  g_assert (eq_sinkpad != NULL);
+
+  gst_element_add_pad (bin, gst_ghost_pad_new (NULL, eq_sinkpad));
+  gst_object_unref (eq_sinkpad);
+
+  g_object_set (playbin, "audio-sink", bin, NULL);
+
+  /* won't work: uri = gst_uri_construct ("file", filenames[0]); */
+  uri = g_strdup_printf ("file://%s", filenames[0]);
+  g_object_set (playbin, "uri", uri, NULL);
+  g_free (uri);
+
+  pipeline_bus = GST_ELEMENT_BUS (playbin);
+
+  ret = gst_element_set_state (playbin, GST_STATE_PLAYING);
+  if (ret == GST_STATE_CHANGE_FAILURE) {
+    g_printerr ("Failed to set playbin to PLAYING\n");
+    check_bus (1 * GST_SECOND);
+    return -1;
+  }
+
+  ret = gst_element_get_state (playbin, NULL, NULL, 5 * GST_SECOND);
+  if (ret == GST_STATE_CHANGE_ASYNC) {
+    g_printerr ("Failed to go to PLAYING in 5 seconds, bailing out\n");
+    return -1;
+  } else if (ret != GST_STATE_CHANGE_SUCCESS) {
+    g_printerr ("State change to PLAYING failed\n");
+    check_bus (1 * GST_SECOND);
+    return -1;
+  }
+
+  g_print ("Playing ...\n");
+  do_slider_fiddling (playbin, eq);
+
+  gst_element_set_state (playbin, GST_STATE_NULL);
+  gst_object_unref (playbin);
+
+  return 0;
+}