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713 lines
21 KiB
C
713 lines
21 KiB
C
/*
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* GStreamer
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* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-audiodynamic
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*
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* This element can act as a compressor or expander. A compressor changes the
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* amplitude of all samples above a specific threshold with a specific ratio,
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* a expander does the same for all samples below a specific threshold. If
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* soft-knee mode is selected the ratio is applied smoothly.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch-1.0 audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 ratio=0.5 ! alsasink
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* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 ratio=4.0 ! alsasink
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* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
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* ]|
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* </refsect2>
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*/
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/* TODO: Implement attack and release parameters */
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include "audiodynamic.h"
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#define GST_CAT_DEFAULT gst_audio_dynamic_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_CHARACTERISTICS,
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PROP_MODE,
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PROP_THRESHOLD,
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PROP_RATIO
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};
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#define ALLOWED_CAPS \
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"audio/x-raw," \
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" format=(string) {"GST_AUDIO_NE(S16)","GST_AUDIO_NE(F32)"}," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]," \
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" layout=(string) {interleaved, non-interleaved}"
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G_DEFINE_TYPE (GstAudioDynamic, gst_audio_dynamic, GST_TYPE_AUDIO_FILTER);
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static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,
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const GstAudioInfo * info);
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static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static void
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gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
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gint16 * data, guint num_samples);
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static void
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gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
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filter, gfloat * data, guint num_samples);
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static void
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gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
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gint16 * data, guint num_samples);
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static void
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gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
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filter, gfloat * data, guint num_samples);
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static void gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic
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* filter, gint16 * data, guint num_samples);
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static void
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gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
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gfloat * data, guint num_samples);
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static void gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic
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* filter, gint16 * data, guint num_samples);
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static void
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gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
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gfloat * data, guint num_samples);
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static const GstAudioDynamicProcessFunc process_functions[] = {
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_hard_knee_compressor_int,
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_hard_knee_compressor_float,
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_soft_knee_compressor_int,
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_soft_knee_compressor_float,
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_hard_knee_expander_int,
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_hard_knee_expander_float,
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_soft_knee_expander_int,
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(GstAudioDynamicProcessFunc)
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gst_audio_dynamic_transform_soft_knee_expander_float
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};
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enum
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{
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CHARACTERISTICS_HARD_KNEE = 0,
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CHARACTERISTICS_SOFT_KNEE
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};
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#define GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS (gst_audio_dynamic_characteristics_get_type ())
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static GType
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gst_audio_dynamic_characteristics_get_type (void)
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{
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static GType gtype = 0;
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if (gtype == 0) {
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static const GEnumValue values[] = {
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{CHARACTERISTICS_HARD_KNEE, "Hard Knee (default)",
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"hard-knee"},
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{CHARACTERISTICS_SOFT_KNEE, "Soft Knee (smooth)",
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"soft-knee"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAudioDynamicCharacteristics", values);
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}
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return gtype;
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}
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enum
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{
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MODE_COMPRESSOR = 0,
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MODE_EXPANDER
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};
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#define GST_TYPE_AUDIO_DYNAMIC_MODE (gst_audio_dynamic_mode_get_type ())
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static GType
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gst_audio_dynamic_mode_get_type (void)
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{
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static GType gtype = 0;
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if (gtype == 0) {
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static const GEnumValue values[] = {
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{MODE_COMPRESSOR, "Compressor (default)",
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"compressor"},
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{MODE_EXPANDER, "Expander", "expander"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAudioDynamicMode", values);
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}
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return gtype;
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}
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static void
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gst_audio_dynamic_set_process_function (GstAudioDynamic * filter,
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const GstAudioInfo * info)
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{
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gint func_index;
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func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;
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func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;
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func_index += (GST_AUDIO_INFO_FORMAT (info) == GST_AUDIO_FORMAT_F32) ? 1 : 0;
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g_assert (func_index >= 0 && func_index < G_N_ELEMENTS (process_functions));
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filter->process = process_functions[func_index];
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}
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/* GObject vmethod implementations */
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static void
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gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstCaps *caps;
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GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0,
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"audiodynamic element");
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gobject_class->set_property = gst_audio_dynamic_set_property;
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gobject_class->get_property = gst_audio_dynamic_get_property;
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g_object_class_install_property (gobject_class, PROP_CHARACTERISTICS,
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g_param_spec_enum ("characteristics", "Characteristics",
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"Selects whether the ratio should be applied smooth (soft-knee) "
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"or hard (hard-knee).",
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GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, CHARACTERISTICS_HARD_KNEE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MODE,
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g_param_spec_enum ("mode", "Mode",
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"Selects whether the filter should work on loud samples (compressor) or"
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"quiet samples (expander).",
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GST_TYPE_AUDIO_DYNAMIC_MODE, MODE_COMPRESSOR,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_THRESHOLD,
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g_param_spec_float ("threshold", "Threshold",
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"Threshold until the filter is activated", 0.0, 1.0,
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0.0,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RATIO,
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g_param_spec_float ("ratio", "Ratio",
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"Ratio that should be applied", 0.0, G_MAXFLOAT,
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1.0,
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G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
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gst_element_class_set_static_metadata (gstelement_class,
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"Dynamic range controller", "Filter/Effect/Audio",
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"Compressor and Expander", "Sebastian Dröge <slomo@circular-chaos.org>");
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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GST_AUDIO_FILTER_CLASS (klass)->setup =
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GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_dynamic_transform_ip);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_ip_on_passthrough = FALSE;
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}
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static void
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gst_audio_dynamic_init (GstAudioDynamic * filter)
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{
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filter->ratio = 1.0;
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filter->threshold = 0.0;
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filter->characteristics = CHARACTERISTICS_HARD_KNEE;
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filter->mode = MODE_COMPRESSOR;
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gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
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}
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static void
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gst_audio_dynamic_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
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switch (prop_id) {
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case PROP_CHARACTERISTICS:
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filter->characteristics = g_value_get_enum (value);
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gst_audio_dynamic_set_process_function (filter,
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GST_AUDIO_FILTER_INFO (filter));
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break;
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case PROP_MODE:
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filter->mode = g_value_get_enum (value);
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gst_audio_dynamic_set_process_function (filter,
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GST_AUDIO_FILTER_INFO (filter));
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break;
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case PROP_THRESHOLD:
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filter->threshold = g_value_get_float (value);
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break;
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case PROP_RATIO:
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filter->ratio = g_value_get_float (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_dynamic_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
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switch (prop_id) {
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case PROP_CHARACTERISTICS:
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g_value_set_enum (value, filter->characteristics);
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break;
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case PROP_MODE:
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g_value_set_enum (value, filter->mode);
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break;
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case PROP_THRESHOLD:
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g_value_set_float (value, filter->threshold);
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break;
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case PROP_RATIO:
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g_value_set_float (value, filter->ratio);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/* GstAudioFilter vmethod implementations */
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static gboolean
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gst_audio_dynamic_setup (GstAudioFilter * base, const GstAudioInfo * info)
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{
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GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
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gst_audio_dynamic_set_process_function (filter, info);
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return TRUE;
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}
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static void
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gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
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gint16 * data, guint num_samples)
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{
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glong val;
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glong thr_p = filter->threshold * G_MAXINT16;
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glong thr_n = filter->threshold * G_MININT16;
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/* Nothing to do for us if ratio is 1.0 or if the threshold
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* equals 1.0. */
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if (filter->threshold == 1.0 || filter->ratio == 1.0)
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return;
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for (; num_samples; num_samples--) {
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val = *data;
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if (val > thr_p) {
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val = thr_p + (val - thr_p) * filter->ratio;
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} else if (val < thr_n) {
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val = thr_n + (val - thr_n) * filter->ratio;
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}
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*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
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}
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}
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static void
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gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
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filter, gfloat * data, guint num_samples)
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{
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gdouble val, threshold = filter->threshold;
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/* Nothing to do for us if ratio == 1.0.
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* As float values can be above 1.0 we have to do something
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* if threshold is greater than 1.0. */
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if (filter->ratio == 1.0)
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return;
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for (; num_samples; num_samples--) {
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val = *data;
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if (val > threshold) {
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val = threshold + (val - threshold) * filter->ratio;
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} else if (val < -threshold) {
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val = -threshold + (val + threshold) * filter->ratio;
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}
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*data++ = (gfloat) val;
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}
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}
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static void
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gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
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gint16 * data, guint num_samples)
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{
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glong val;
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glong thr_p = filter->threshold * G_MAXINT16;
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glong thr_n = filter->threshold * G_MININT16;
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gdouble a_p, b_p, c_p;
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gdouble a_n, b_n, c_n;
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/* Nothing to do for us if ratio is 1.0 or if the threshold
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* equals 1.0. */
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if (filter->threshold == 1.0 || filter->ratio == 1.0)
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return;
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/* We build a 2nd degree polynomial here for
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* values greater than threshold or small than
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* -threshold with:
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* f(t) = t, f'(t) = 1, f'(m) = r
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* =>
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* a = (1-r)/(2*(t-m))
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* b = (r*t - m)/(t-m)
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* c = t * (1 - b - a*t)
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* f(x) = ax^2 + bx + c
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*/
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/* shouldn't happen because this would only be the case
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* for threshold == 1.0 which we catch above */
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g_assert (thr_p - G_MAXINT16 != 0);
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g_assert (thr_n - G_MININT != 0);
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a_p = (1 - filter->ratio) / (2 * (thr_p - G_MAXINT16));
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b_p = (filter->ratio * thr_p - G_MAXINT16) / (thr_p - G_MAXINT16);
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c_p = thr_p * (1 - b_p - a_p * thr_p);
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a_n = (1 - filter->ratio) / (2 * (thr_n - G_MININT16));
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b_n = (filter->ratio * thr_n - G_MININT16) / (thr_n - G_MININT16);
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c_n = thr_n * (1 - b_n - a_n * thr_n);
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for (; num_samples; num_samples--) {
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val = *data;
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if (val > thr_p) {
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val = a_p * val * val + b_p * val + c_p;
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} else if (val < thr_n) {
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val = a_n * val * val + b_n * val + c_n;
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}
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*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
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}
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}
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static void
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gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
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filter, gfloat * data, guint num_samples)
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{
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gdouble val;
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gdouble threshold = filter->threshold;
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gdouble a_p, b_p, c_p;
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gdouble a_n, b_n, c_n;
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/* Nothing to do for us if ratio == 1.0.
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* As float values can be above 1.0 we have to do something
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* if threshold is greater than 1.0. */
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if (filter->ratio == 1.0)
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return;
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/* We build a 2nd degree polynomial here for
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* values greater than threshold or small than
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* -threshold with:
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* f(t) = t, f'(t) = 1, f'(m) = r
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* =>
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* a = (1-r)/(2*(t-m))
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* b = (r*t - m)/(t-m)
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* c = t * (1 - b - a*t)
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* f(x) = ax^2 + bx + c
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*/
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/* FIXME: If treshold is the same as the maximum
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* we need to raise it a bit to prevent
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* division by zero. */
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if (threshold == 1.0)
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threshold = 1.0 + 0.00001;
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a_p = (1.0 - filter->ratio) / (2.0 * (threshold - 1.0));
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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;
|
|
gdouble r2;
|
|
|
|
/* 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 */
|
|
r2 = filter->ratio * filter->ratio;
|
|
a_p = (1.0 - r2) / (4.0 * thr_p);
|
|
b_p = (1.0 + r2) / 2.0;
|
|
c_p = thr_p * (1.0 - b_p - a_p * thr_p);
|
|
a_n = (1.0 - r2) / (4.0 * thr_n);
|
|
b_n = (1.0 + r2) / 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;
|
|
gdouble r2;
|
|
|
|
/* 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 */
|
|
r2 = filter->ratio * filter->ratio;
|
|
a_p = (1.0 - r2) / (4.0 * threshold);
|
|
b_p = (1.0 + r2) / 2.0;
|
|
c_p = threshold * (1.0 - b_p - a_p * threshold);
|
|
a_n = (1.0 - r2) / (-4.0 * threshold);
|
|
b_n = (1.0 + r2) / 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;
|
|
GstClockTime timestamp, stream_time;
|
|
GstMapInfo map;
|
|
|
|
timestamp = GST_BUFFER_TIMESTAMP (buf);
|
|
stream_time =
|
|
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
|
|
|
|
GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (timestamp));
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (stream_time))
|
|
gst_object_sync_values (GST_OBJECT (filter), stream_time);
|
|
|
|
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
|
|
return GST_FLOW_OK;
|
|
|
|
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
|
|
num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);
|
|
|
|
filter->process (filter, map.data, num_samples);
|
|
|
|
gst_buffer_unmap (buf, &map);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|