diff -r 71e347f905f2 -r 4a7fac7dd34a gst_plugins_good/gst/audiofx/audiodynamic.c
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/audiofx/audiodynamic.c	Fri Apr 16 15:15:52 2010 +0300
@@ -0,0 +1,711 @@
+/* 
+ * 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.
+ */
+
+/**
+ * SECTION:element-audiodynamic
+ *
+ * This element can act as a compressor or expander. A compressor changes the
+ * amplitude of all samples above a specific threshold with a specific ratio,
+ * a expander does the same for all samples below a specific threshold. If
+ * soft-knee mode is selected the ratio is applied smoothly.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink
+ * gst-launch audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
+ * ]|
+ * </refsect2>
+ */
+
+/* TODO: Implement attack and release parameters */
+
+#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 "audiodynamic.h"
+
+#define GST_CAT_DEFAULT gst_audio_dynamic_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("Dynamic range controller",
+    "Filter/Effect/Audio",
+    "Compressor and Expander",
+    "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+enum
+{
+  PROP_0,
+  PROP_CHARACTERISTICS,
+  PROP_MODE,
+  PROP_THRESHOLD,
+  PROP_RATIO
+};
+
+#define ALLOWED_CAPS \
+    "audio/x-raw-int,"                                                \
+    " depth=(int)16,"                                                 \
+    " width=(int)16,"                                                 \
+    " endianness=(int)BYTE_ORDER,"                                    \
+    " signed=(bool)TRUE,"                                             \
+    " rate=(int)[1,MAX],"                                             \
+    " channels=(int)[1,MAX]; "                                        \
+    "audio/x-raw-float,"                                              \
+    " width=(int)32,"                                                 \
+    " endianness=(int)BYTE_ORDER,"                                    \
+    " rate=(int)[1,MAX],"                                             \
+    " channels=(int)[1,MAX]"
+
+#define DEBUG_INIT(bla) \
+  GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0, "audiodynamic element");
+
+GST_BOILERPLATE_FULL (GstAudioDynamic, gst_audio_dynamic, GstAudioFilter,
+    GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,
+    GstRingBufferSpec * format);
+static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+
+static void
+gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
+    gint16 * data, guint num_samples);
+static void
+gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
+    filter, gfloat * data, guint num_samples);
+static void
+gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
+    gint16 * data, guint num_samples);
+static void
+gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
+    filter, gfloat * data, guint num_samples);
+static void gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic
+    * filter, gint16 * data, guint num_samples);
+static void
+gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
+    gfloat * data, guint num_samples);
+static void gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic
+    * filter, gint16 * data, guint num_samples);
+static void
+gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
+    gfloat * data, guint num_samples);
+
+static GstAudioDynamicProcessFunc process_functions[] = {
+  (GstAudioDynamicProcessFunc)
+      gst_audio_dynamic_transform_hard_knee_compressor_int,
+  (GstAudioDynamicProcessFunc)
+      gst_audio_dynamic_transform_hard_knee_compressor_float,
+  (GstAudioDynamicProcessFunc)
+      gst_audio_dynamic_transform_soft_knee_compressor_int,
+  (GstAudioDynamicProcessFunc)
+      gst_audio_dynamic_transform_soft_knee_compressor_float,
+  (GstAudioDynamicProcessFunc)
+      gst_audio_dynamic_transform_hard_knee_expander_int,
+  (GstAudioDynamicProcessFunc)
+      gst_audio_dynamic_transform_hard_knee_expander_float,
+  (GstAudioDynamicProcessFunc)
+      gst_audio_dynamic_transform_soft_knee_expander_int,
+  (GstAudioDynamicProcessFunc)
+  gst_audio_dynamic_transform_soft_knee_expander_float
+};
+
+enum
+{
+  CHARACTERISTICS_HARD_KNEE = 0,
+  CHARACTERISTICS_SOFT_KNEE
+};
+
+#define GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS (gst_audio_dynamic_characteristics_get_type ())
+static GType
+gst_audio_dynamic_characteristics_get_type (void)
+{
+  static GType gtype = 0;
+
+  if (gtype == 0) {
+    static const GEnumValue values[] = {
+      {CHARACTERISTICS_HARD_KNEE, "Hard Knee (default)",
+          "hard-knee"},
+      {CHARACTERISTICS_SOFT_KNEE, "Soft Knee (smooth)",
+          "soft-knee"},
+      {0, NULL, NULL}
+    };
+
+    gtype = g_enum_register_static ("GstAudioDynamicCharacteristics", values);
+  }
+  return gtype;
+}
+
+enum
+{
+  MODE_COMPRESSOR = 0,
+  MODE_EXPANDER
+};
+
+#define GST_TYPE_AUDIO_DYNAMIC_MODE (gst_audio_dynamic_mode_get_type ())
+static GType
+gst_audio_dynamic_mode_get_type (void)
+{
+  static GType gtype = 0;
+
+  if (gtype == 0) {
+    static const GEnumValue values[] = {
+      {MODE_COMPRESSOR, "Compressor (default)",
+          "compressor"},
+      {MODE_EXPANDER, "Expander", "expander"},
+      {0, NULL, NULL}
+    };
+
+    gtype = g_enum_register_static ("GstAudioDynamicMode", values);
+  }
+  return gtype;
+}
+
+static gboolean
+gst_audio_dynamic_set_process_function (GstAudioDynamic * filter)
+{
+  gint func_index;
+
+  func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;
+  func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;
+  func_index +=
+      (GST_AUDIO_FILTER (filter)->format.type == GST_BUFTYPE_FLOAT) ? 1 : 0;
+
+  if (func_index >= 0 && func_index < 8) {
+    filter->process = process_functions[func_index];
+    return TRUE;
+  }
+
+  return FALSE;
+}
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_dynamic_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_dynamic_class_init (GstAudioDynamicClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gobject_class->set_property = gst_audio_dynamic_set_property;
+  gobject_class->get_property = gst_audio_dynamic_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_CHARACTERISTICS,
+      g_param_spec_enum ("characteristics", "Characteristics",
+          "Selects whether the ratio should be applied smooth (soft-knee) "
+          "or hard (hard-knee).",
+          GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, CHARACTERISTICS_HARD_KNEE,
+          G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_MODE,
+      g_param_spec_enum ("mode", "Mode",
+          "Selects whether the filter should work on loud samples (compressor) or"
+          "quiet samples (expander).",
+          GST_TYPE_AUDIO_DYNAMIC_MODE, MODE_COMPRESSOR, G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_THRESHOLD,
+      g_param_spec_float ("threshold", "Threshold",
+          "Threshold until the filter is activated", 0.0, 1.0,
+          0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+  g_object_class_install_property (gobject_class, PROP_RATIO,
+      g_param_spec_float ("ratio", "Ratio",
+          "Ratio that should be applied", 0.0, G_MAXFLOAT,
+          1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+  GST_AUDIO_FILTER_CLASS (klass)->setup =
+      GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);
+  GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
+      GST_DEBUG_FUNCPTR (gst_audio_dynamic_transform_ip);
+}
+
+static void
+gst_audio_dynamic_init (GstAudioDynamic * filter, GstAudioDynamicClass * klass)
+{
+  filter->ratio = 1.0;
+  filter->threshold = 0.0;
+  filter->characteristics = CHARACTERISTICS_HARD_KNEE;
+  filter->mode = MODE_COMPRESSOR;
+  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
+  gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
+}
+
+static void
+gst_audio_dynamic_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
+
+  switch (prop_id) {
+    case PROP_CHARACTERISTICS:
+      filter->characteristics = g_value_get_enum (value);
+      gst_audio_dynamic_set_process_function (filter);
+      break;
+    case PROP_MODE:
+      filter->mode = g_value_get_enum (value);
+      gst_audio_dynamic_set_process_function (filter);
+      break;
+    case PROP_THRESHOLD:
+      filter->threshold = g_value_get_float (value);
+      break;
+    case PROP_RATIO:
+      filter->ratio = g_value_get_float (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_dynamic_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
+
+  switch (prop_id) {
+    case PROP_CHARACTERISTICS:
+      g_value_set_enum (value, filter->characteristics);
+      break;
+    case PROP_MODE:
+      g_value_set_enum (value, filter->mode);
+      break;
+    case PROP_THRESHOLD:
+      g_value_set_float (value, filter->threshold);
+      break;
+    case PROP_RATIO:
+      g_value_set_float (value, filter->ratio);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_dynamic_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+{
+  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
+  gboolean ret = TRUE;
+
+  ret = gst_audio_dynamic_set_process_function (filter);
+
+  return ret;
+}
+
+static void
+gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
+    gint16 * data, guint num_samples)
+{
+  glong val;
+  glong thr_p = filter->threshold * G_MAXINT16;
+  glong thr_n = filter->threshold * G_MININT16;
+
+  /* Nothing to do for us if ratio is 1.0 or if the threshold
+   * equals 1.0. */
+  if (filter->threshold == 1.0 || filter->ratio == 1.0)
+    return;
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val > thr_p) {
+      val = thr_p + (val - thr_p) * filter->ratio;
+    } else if (val < thr_n) {
+      val = thr_n + (val - thr_n) * filter->ratio;
+    }
+    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
+  }
+}
+
+static void
+gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
+    filter, gfloat * data, guint num_samples)
+{
+  gdouble val, threshold = filter->threshold;
+
+  /* Nothing to do for us if ratio == 1.0.
+   * As float values can be above 1.0 we have to do something
+   * if threshold is greater than 1.0. */
+  if (filter->ratio == 1.0)
+    return;
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val > threshold) {
+      val = threshold + (val - threshold) * filter->ratio;
+    } else if (val < -threshold) {
+      val = -threshold + (val + threshold) * filter->ratio;
+    }
+    *data++ = (gfloat) val;
+  }
+}
+
+static void
+gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
+    gint16 * data, guint num_samples)
+{
+  glong val;
+  glong thr_p = filter->threshold * G_MAXINT16;
+  glong thr_n = filter->threshold * G_MININT16;
+  gdouble a_p, b_p, c_p;
+  gdouble a_n, b_n, c_n;
+
+  /* Nothing to do for us if ratio is 1.0 or if the threshold
+   * equals 1.0. */
+  if (filter->threshold == 1.0 || filter->ratio == 1.0)
+    return;
+
+  /* We build a 2nd degree polynomial here for
+   * values greater than threshold or small than
+   * -threshold with:
+   * f(t) = t, f'(t) = 1, f'(m) = r
+   * =>
+   * a = (1-r)/(2*(t-m))
+   * b = (r*t - m)/(t-m)
+   * c = t * (1 - b - a*t)
+   * f(x) = ax^2 + bx + c
+   */
+
+  /* shouldn't happen because this would only be the case
+   * for threshold == 1.0 which we catch above */
+  g_assert (thr_p - G_MAXINT16 != 0);
+  g_assert (thr_n - G_MININT != 0);
+
+  a_p = (1 - filter->ratio) / (2 * (thr_p - G_MAXINT16));
+  b_p = (filter->ratio * thr_p - G_MAXINT16) / (thr_p - G_MAXINT16);
+  c_p = thr_p * (1 - b_p - a_p * thr_p);
+  a_n = (1 - filter->ratio) / (2 * (thr_n - G_MININT16));
+  b_n = (filter->ratio * thr_n - G_MININT16) / (thr_n - G_MININT16);
+  c_n = thr_n * (1 - b_n - a_n * thr_n);
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val > thr_p) {
+      val = a_p * val * val + b_p * val + c_p;
+    } else if (val < thr_n) {
+      val = a_n * val * val + b_n * val + c_n;
+    }
+    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
+  }
+}
+
+static void
+gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
+    filter, gfloat * data, guint num_samples)
+{
+  gdouble val;
+  gdouble threshold = filter->threshold;
+  gdouble a_p, b_p, c_p;
+  gdouble a_n, b_n, c_n;
+
+  /* Nothing to do for us if ratio == 1.0.
+   * As float values can be above 1.0 we have to do something
+   * if threshold is greater than 1.0. */
+  if (filter->ratio == 1.0)
+    return;
+
+  /* We build a 2nd degree polynomial here for
+   * values greater than threshold or small than
+   * -threshold with:
+   * f(t) = t, f'(t) = 1, f'(m) = r
+   * =>
+   * a = (1-r)/(2*(t-m))
+   * b = (r*t - m)/(t-m)
+   * c = t * (1 - b - a*t)
+   * f(x) = ax^2 + bx + c
+   */
+
+  /* FIXME: If treshold is the same as the maximum
+   * we need to raise it a bit to prevent
+   * division by zero. */
+  if (threshold == 1.0)
+    threshold = 1.0 + 0.00001;
+
+  a_p = (1.0 - filter->ratio) / (2.0 * (threshold - 1.0));
+  b_p = (filter->ratio * threshold - 1.0) / (threshold - 1.0);
+  c_p = threshold * (1.0 - b_p - a_p * threshold);
+  a_n = (1.0 - filter->ratio) / (2.0 * (-threshold + 1.0));
+  b_n = (-filter->ratio * threshold + 1.0) / (-threshold + 1.0);
+  c_n = -threshold * (1.0 - b_n + a_n * threshold);
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val > 1.0) {
+      val = 1.0 + (val - 1.0) * filter->ratio;
+    } else if (val > threshold) {
+      val = a_p * val * val + b_p * val + c_p;
+    } else if (val < -1.0) {
+      val = -1.0 + (val + 1.0) * filter->ratio;
+    } else if (val < -threshold) {
+      val = a_n * val * val + b_n * val + c_n;
+    }
+    *data++ = (gfloat) val;
+  }
+}
+
+static void
+gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic * filter,
+    gint16 * data, guint num_samples)
+{
+  glong val;
+  glong thr_p = filter->threshold * G_MAXINT16;
+  glong thr_n = filter->threshold * G_MININT16;
+  gdouble zero_p, zero_n;
+
+  /* Nothing to do for us here if threshold equals 0.0
+   * or ratio equals 1.0 */
+  if (filter->threshold == 0.0 || filter->ratio == 1.0)
+    return;
+
+  /* zero crossing of our function */
+  if (filter->ratio != 0.0) {
+    zero_p = thr_p - thr_p / filter->ratio;
+    zero_n = thr_n - thr_n / filter->ratio;
+  } else {
+    zero_p = zero_n = 0.0;
+  }
+
+  if (zero_p < 0.0)
+    zero_p = 0.0;
+  if (zero_n > 0.0)
+    zero_n = 0.0;
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val < thr_p && val > zero_p) {
+      val = filter->ratio * val + thr_p * (1 - filter->ratio);
+    } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
+      val = 0;
+    } else if (val > thr_n && val < zero_n) {
+      val = filter->ratio * val + thr_n * (1 - filter->ratio);
+    }
+    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
+  }
+}
+
+static void
+gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
+    gfloat * data, guint num_samples)
+{
+  gdouble val, threshold = filter->threshold, zero;
+
+  /* Nothing to do for us here if threshold equals 0.0
+   * or ratio equals 1.0 */
+  if (filter->threshold == 0.0 || filter->ratio == 1.0)
+    return;
+
+  /* zero crossing of our function */
+  if (filter->ratio != 0.0)
+    zero = threshold - threshold / filter->ratio;
+  else
+    zero = 0.0;
+
+  if (zero < 0.0)
+    zero = 0.0;
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val < threshold && val > zero) {
+      val = filter->ratio * val + threshold * (1.0 - filter->ratio);
+    } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
+      val = 0.0;
+    } else if (val > -threshold && val < -zero) {
+      val = filter->ratio * val - threshold * (1.0 - filter->ratio);
+    }
+    *data++ = (gfloat) val;
+  }
+}
+
+static void
+gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic * filter,
+    gint16 * data, guint num_samples)
+{
+  glong val;
+  glong thr_p = filter->threshold * G_MAXINT16;
+  glong thr_n = filter->threshold * G_MININT16;
+  gdouble zero_p, zero_n;
+  gdouble a_p, b_p, c_p;
+  gdouble a_n, b_n, c_n;
+
+  /* Nothing to do for us here if threshold equals 0.0
+   * or ratio equals 1.0 */
+  if (filter->threshold == 0.0 || filter->ratio == 1.0)
+    return;
+
+  /* zero crossing of our function */
+  zero_p = (thr_p * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
+  zero_n = (thr_n * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
+
+  if (zero_p < 0.0)
+    zero_p = 0.0;
+  if (zero_n > 0.0)
+    zero_n = 0.0;
+
+  /* shouldn't happen as this would only happen
+   * with threshold == 0.0 */
+  g_assert (thr_p != 0);
+  g_assert (thr_n != 0);
+
+  /* We build a 2n degree polynomial here for values between
+   * 0 and threshold or 0 and -threshold with:
+   * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
+   * z between 0 and t
+   * =>
+   * a = (1 - r^2) / (4 * t)
+   * b = (1 + r^2) / 2
+   * c = t * (1.0 - b - a*t)
+   * f(x) = ax^2 + bx + c */
+  a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_p);
+  b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;
+  c_p = thr_p * (1.0 - b_p - a_p * thr_p);
+  a_n = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_n);
+  b_n = (1.0 + filter->ratio * filter->ratio) / 2.0;
+  c_n = thr_n * (1.0 - b_n - a_n * thr_n);
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val < thr_p && val > zero_p) {
+      val = a_p * val * val + b_p * val + c_p;
+    } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
+      val = 0;
+    } else if (val > thr_n && val < zero_n) {
+      val = a_n * val * val + b_n * val + c_n;
+    }
+    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
+  }
+}
+
+static void
+gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
+    gfloat * data, guint num_samples)
+{
+  gdouble val;
+  gdouble threshold = filter->threshold;
+  gdouble zero;
+  gdouble a_p, b_p, c_p;
+  gdouble a_n, b_n, c_n;
+
+  /* Nothing to do for us here if threshold equals 0.0
+   * or ratio equals 1.0 */
+  if (filter->threshold == 0.0 || filter->ratio == 1.0)
+    return;
+
+  /* zero crossing of our function */
+  zero = (threshold * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
+
+  if (zero < 0.0)
+    zero = 0.0;
+
+  /* shouldn't happen as this only happens with
+   * threshold == 0.0 */
+  g_assert (threshold != 0.0);
+
+  /* We build a 2n degree polynomial here for values between
+   * 0 and threshold or 0 and -threshold with:
+   * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
+   * z between 0 and t
+   * =>
+   * a = (1 - r^2) / (4 * t)
+   * b = (1 + r^2) / 2
+   * c = t * (1.0 - b - a*t)
+   * f(x) = ax^2 + bx + c */
+  a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * threshold);
+  b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;
+  c_p = threshold * (1.0 - b_p - a_p * threshold);
+  a_n = (1.0 - filter->ratio * filter->ratio) / (-4.0 * threshold);
+  b_n = (1.0 + filter->ratio * filter->ratio) / 2.0;
+  c_n = -threshold * (1.0 - b_n + a_n * threshold);
+
+  for (; num_samples; num_samples--) {
+    val = *data;
+
+    if (val < threshold && val > zero) {
+      val = a_p * val * val + b_p * val + c_p;
+    } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
+      val = 0.0;
+    } else if (val > -threshold && val < -zero) {
+      val = a_n * val * val + b_n * val + c_n;
+    }
+    *data++ = (gfloat) val;
+  }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
+  guint num_samples =
+      GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+  if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+    gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+  if (gst_base_transform_is_passthrough (base) ||
+      G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
+    return GST_FLOW_OK;
+
+  filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+  return GST_FLOW_OK;
+}