--- a/gst_plugins_good/gst/audiofx/audiodynamic.c Tue Jul 06 14:35:10 2010 +0300
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,711 +0,0 @@
-/*
- * 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;
-}