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GST-Tensordecoder-ov_ep/subprojects/gst-plugins-bad/gst/freeverb/gstfreeverb.c

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/*
* GStreamer
* Copyright (C) 2011 Stefan Sauer <ensonic@users.sf.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/*
* Freeverb
*
* Written by Jezar at Dreampoint, June 2000
* http://www.dreampoint.co.uk
* This code is public domain
*
* Translated to C by Peter Hanappe, Mai 2001
* Transformed into a GStreamer plugin by Stefan Sauer, Nov 2011
*/
/**
* SECTION:element-freeverb
* @title: freeverb
*
* Reverberation/room effect.
*
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc wave=saw ! freeverb ! autoaudiosink
* gst-launch-1.0 filesrc location="melo1.ogg" ! decodebin ! audioconvert ! freeverb ! autoaudiosink
* ]|
*
*/
/* FIXME:
* - add mono-to-mono, then we might also need stereo-to-mono ?
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <math.h>
#include <stdlib.h>
#include <string.h>
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include "gstfreeverb.h"
#define GST_CAT_DEFAULT gst_freeverb_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
enum
{
PROP_0,
PROP_ROOM_SIZE,
PROP_DAMPING,
PROP_PAN_WIDTH,
PROP_LEVEL
};
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ], "
"layout = (string) interleaved")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { " GST_AUDIO_NE (F32) ", " GST_AUDIO_NE (S16) "}, "
"rate = (int) [ 1, MAX ], " "channels = (int) 2, "
"layout = (string) interleaved")
);
static void gst_freeverb_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_freeverb_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_freeverb_finalize (GObject * object);
static gboolean gst_freeverb_get_unit_size (GstBaseTransform * base,
GstCaps * caps, gsize * size);
static GstCaps *gst_freeverb_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter);
static gboolean gst_freeverb_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_freeverb_transform (GstBaseTransform * base,
GstBuffer * inbuf, GstBuffer * outbuf);
static gboolean gst_freeverb_transform_m2s_int (GstFreeverb * filter,
gint16 * idata, gint16 * odata, guint num_samples);
static gboolean gst_freeverb_transform_s2s_int (GstFreeverb * filter,
gint16 * idata, gint16 * odata, guint num_samples);
static gboolean gst_freeverb_transform_m2s_float (GstFreeverb * filter,
gfloat * idata, gfloat * odata, guint num_samples);
static gboolean gst_freeverb_transform_s2s_float (GstFreeverb * filter,
gfloat * idata, gfloat * odata, guint num_samples);
/* Table with processing functions: [channels][format] */
static const GstFreeverbProcessFunc process_functions[2][2] = {
{
(GstFreeverbProcessFunc) gst_freeverb_transform_m2s_int,
(GstFreeverbProcessFunc) gst_freeverb_transform_m2s_float,
},
{
(GstFreeverbProcessFunc) gst_freeverb_transform_s2s_int,
(GstFreeverbProcessFunc) gst_freeverb_transform_s2s_float,
}
};
/***************************************************************
*
* REVERB
*/
/* Denormalising:
*
* Another method fixes the problem cheaper: Use a small DC-offset in
* the filter calculations. Now the signals converge not against 0,
* but against the offset. The constant offset is invisible from the
* outside world (i.e. it does not appear at the output. There is a
* very small turn-on transient response, which should not cause
* problems.
*/
//#define DC_OFFSET 0
#define DC_OFFSET 1e-8
//#define DC_OFFSET 0.001f
/* all pass filter */
typedef struct _freeverb_allpass
{
gfloat feedback;
gfloat *buffer;
gint bufsize;
gint bufidx;
} freeverb_allpass;
static void
freeverb_allpass_setbuffer (freeverb_allpass * allpass, gint size)
{
allpass->bufidx = 0;
allpass->buffer = g_new (gfloat, size);
allpass->bufsize = size;
}
static void
freeverb_allpass_release (freeverb_allpass * allpass)
{
g_free (allpass->buffer);
}
static void
freeverb_allpass_init (freeverb_allpass * allpass)
{
gint i, len = allpass->bufsize;
gfloat *buf = allpass->buffer;
for (i = 0; i < len; i++) {
buf[i] = (gfloat) DC_OFFSET; /* this is not 100 % correct. */
}
}
static void
freeverb_allpass_setfeedback (freeverb_allpass * allpass, gfloat val)
{
allpass->feedback = val;
}
/*
static gfloat
freeverb_allpass_getfeedback(freeverb_allpass* allpass)
{
return allpass->feedback;
}*/
#define freeverb_allpass_process(_allpass, _input_1) \
{ \
gfloat output; \
gfloat bufout; \
bufout = _allpass.buffer[_allpass.bufidx]; \
output = bufout-_input_1; \
_allpass.buffer[_allpass.bufidx] = _input_1 + (bufout * _allpass.feedback); \
if (++_allpass.bufidx >= _allpass.bufsize) { \
_allpass.bufidx = 0; \
} \
_input_1 = output; \
}
/* comb filter */
typedef struct _freeverb_comb
{
gfloat feedback;
gfloat filterstore;
gfloat damp1;
gfloat damp2;
gfloat *buffer;
gint bufsize;
gint bufidx;
} freeverb_comb;
static void
freeverb_comb_setbuffer (freeverb_comb * comb, gint size)
{
comb->filterstore = 0;
comb->bufidx = 0;
comb->buffer = g_new (gfloat, size);
comb->bufsize = size;
}
static void
freeverb_comb_release (freeverb_comb * comb)
{
g_free (comb->buffer);
}
static void
freeverb_comb_init (freeverb_comb * comb)
{
gint i, len = comb->bufsize;
gfloat *buf = comb->buffer;
for (i = 0; i < len; i++) {
buf[i] = (gfloat) DC_OFFSET; /* This is not 100 % correct. */
}
}
static void
freeverb_comb_setdamp (freeverb_comb * comb, gfloat val)
{
comb->damp1 = val;
comb->damp2 = 1 - val;
}
/*
static gfloat
freeverb_comb_getdamp(freeverb_comb* comb)
{
return comb->damp1;
}*/
static void
freeverb_comb_setfeedback (freeverb_comb * comb, gfloat val)
{
comb->feedback = val;
}
/*
static gfloat
freeverb_comb_getfeedback(freeverb_comb* comb)
{
return comb->feedback;
}*/
#define freeverb_comb_process(_comb, _input_1, _output) \
{ \
gfloat _tmp = _comb.buffer[_comb.bufidx]; \
_comb.filterstore = (_tmp * _comb.damp2) + (_comb.filterstore * _comb.damp1); \
_comb.buffer[_comb.bufidx] = _input_1 + (_comb.filterstore * _comb.feedback); \
if (++_comb.bufidx >= _comb.bufsize) { \
_comb.bufidx = 0; \
} \
_output += _tmp; \
}
#define numcombs 8
#define numallpasses 4
#define fixedgain 0.015f
#define scalewet 1.0f
#define scaledry 1.0f
#define scaledamp 1.0f
#define scaleroom 0.28f
#define offsetroom 0.7f
#define stereospread 23
/* These values assume 44.1KHz sample rate
* they will need scaling for 96KHz (or other) sample rates.
* The values were obtained by listening tests.
*/
#define combtuningL1 1116
#define combtuningR1 (1116 + stereospread)
#define combtuningL2 1188
#define combtuningR2 (1188 + stereospread)
#define combtuningL3 1277
#define combtuningR3 (1277 + stereospread)
#define combtuningL4 1356
#define combtuningR4 (1356 + stereospread)
#define combtuningL5 1422
#define combtuningR5 (1422 + stereospread)
#define combtuningL6 1491
#define combtuningR6 (1491 + stereospread)
#define combtuningL7 1557
#define combtuningR7 (1557 + stereospread)
#define combtuningL8 1617
#define combtuningR8 (1617 + stereospread)
#define allpasstuningL1 556
#define allpasstuningR1 (556 + stereospread)
#define allpasstuningL2 441
#define allpasstuningR2 (441 + stereospread)
#define allpasstuningL3 341
#define allpasstuningR3 (341 + stereospread)
#define allpasstuningL4 225
#define allpasstuningR4 (225 + stereospread)
struct _GstFreeverbPrivate
{
gfloat roomsize;
gfloat damp;
gfloat wet, wet1, wet2, dry;
gfloat width;
gfloat gain;
/*
The following are all declared inline
to remove the need for dynamic allocation
with its subsequent error-checking messiness
*/
/* Comb filters */
freeverb_comb combL[numcombs];
freeverb_comb combR[numcombs];
/* Allpass filters */
freeverb_allpass allpassL[numallpasses];
freeverb_allpass allpassR[numallpasses];
};
G_DEFINE_TYPE_WITH_CODE (GstFreeverb, gst_freeverb, GST_TYPE_BASE_TRANSFORM,
G_ADD_PRIVATE (GstFreeverb)
G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
GST_ELEMENT_REGISTER_DEFINE (freeverb, "freeverb",
GST_RANK_NONE, GST_TYPE_FREEVERB);
static void
freeverb_revmodel_init (GstFreeverb * filter)
{
GstFreeverbPrivate *priv = filter->priv;
gint i;
for (i = 0; i < numcombs; i++) {
freeverb_comb_init (&priv->combL[i]);
freeverb_comb_init (&priv->combR[i]);
}
for (i = 0; i < numallpasses; i++) {
freeverb_allpass_init (&priv->allpassL[i]);
freeverb_allpass_init (&priv->allpassR[i]);
}
}
static void
freeverb_revmodel_free (GstFreeverb * filter)
{
GstFreeverbPrivate *priv = filter->priv;
gint i;
for (i = 0; i < numcombs; i++) {
freeverb_comb_release (&priv->combL[i]);
freeverb_comb_release (&priv->combR[i]);
}
for (i = 0; i < numallpasses; i++) {
freeverb_allpass_release (&priv->allpassL[i]);
freeverb_allpass_release (&priv->allpassR[i]);
}
}
/* GObject vmethod implementations */
static void
gst_freeverb_class_init (GstFreeverbClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
GST_DEBUG_CATEGORY_INIT (gst_freeverb_debug, "freeverb", 0,
"freeverb element");
gobject_class = (GObjectClass *) klass;
element_class = (GstElementClass *) klass;
gobject_class->set_property = gst_freeverb_set_property;
gobject_class->get_property = gst_freeverb_get_property;
gobject_class->finalize = gst_freeverb_finalize;
g_object_class_install_property (gobject_class, PROP_ROOM_SIZE,
g_param_spec_float ("room-size", "Room size",
"Size of the simulated room", 0.0, 1.0, 0.5,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_DAMPING,
g_param_spec_float ("damping", "Damping", "Damping of high frequencies",
0.0, 1.0, 0.2f,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PAN_WIDTH,
g_param_spec_float ("width", "Width", "Stereo panorama width", 0.0, 1.0,
1.0,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_LEVEL,
g_param_spec_float ("level", "Level", "dry/wet level", 0.0, 1.0, 0.5,
G_PARAM_CONSTRUCT | G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE |
G_PARAM_STATIC_STRINGS));
gst_element_class_set_static_metadata (element_class,
"Reverberation/room effect", "Filter/Effect/Audio",
"Add reverberation to audio streams",
"Stefan Sauer <ensonic@users.sf.net>");
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_add_static_pad_template (element_class, &sink_template);
GST_BASE_TRANSFORM_CLASS (klass)->get_unit_size =
GST_DEBUG_FUNCPTR (gst_freeverb_get_unit_size);
GST_BASE_TRANSFORM_CLASS (klass)->transform_caps =
GST_DEBUG_FUNCPTR (gst_freeverb_transform_caps);
GST_BASE_TRANSFORM_CLASS (klass)->set_caps =
GST_DEBUG_FUNCPTR (gst_freeverb_set_caps);
GST_BASE_TRANSFORM_CLASS (klass)->transform =
GST_DEBUG_FUNCPTR (gst_freeverb_transform);
}
static void
gst_freeverb_init (GstFreeverb * filter)
{
filter->priv = gst_freeverb_get_instance_private (filter);
gst_audio_info_init (&filter->info);
filter->process = NULL;
gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
freeverb_revmodel_init (filter);
}
static void
gst_freeverb_finalize (GObject * object)
{
GstFreeverb *filter = GST_FREEVERB (object);
freeverb_revmodel_free (filter);
G_OBJECT_CLASS (gst_freeverb_parent_class)->finalize (object);
}
static gboolean
gst_freeverb_set_process_function (GstFreeverb * filter, GstAudioInfo * info)
{
gint channel_index, format_index;
const GstAudioFormatInfo *finfo = info->finfo;
/* set processing function */
channel_index = GST_AUDIO_INFO_CHANNELS (info) - 1;
if (channel_index > 1 || channel_index < 0) {
filter->process = NULL;
return FALSE;
}
format_index = GST_AUDIO_FORMAT_INFO_IS_FLOAT (finfo) ? 1 : 0;
filter->process = process_functions[channel_index][format_index];
return TRUE;
}
static void
gst_freeverb_init_rev_model (GstFreeverb * filter)
{
gfloat srfactor = GST_AUDIO_INFO_RATE (&filter->info) / 44100.0f;
GstFreeverbPrivate *priv = filter->priv;
freeverb_revmodel_free (filter);
priv->gain = fixedgain;
freeverb_comb_setbuffer (&priv->combL[0], combtuningL1 * srfactor);
freeverb_comb_setbuffer (&priv->combR[0], combtuningR1 * srfactor);
freeverb_comb_setbuffer (&priv->combL[1], combtuningL2 * srfactor);
freeverb_comb_setbuffer (&priv->combR[1], combtuningR2 * srfactor);
freeverb_comb_setbuffer (&priv->combL[2], combtuningL3 * srfactor);
freeverb_comb_setbuffer (&priv->combR[2], combtuningR3 * srfactor);
freeverb_comb_setbuffer (&priv->combL[3], combtuningL4 * srfactor);
freeverb_comb_setbuffer (&priv->combR[3], combtuningR4 * srfactor);
freeverb_comb_setbuffer (&priv->combL[4], combtuningL5 * srfactor);
freeverb_comb_setbuffer (&priv->combR[4], combtuningR5 * srfactor);
freeverb_comb_setbuffer (&priv->combL[5], combtuningL6 * srfactor);
freeverb_comb_setbuffer (&priv->combR[5], combtuningR6 * srfactor);
freeverb_comb_setbuffer (&priv->combL[6], combtuningL7 * srfactor);
freeverb_comb_setbuffer (&priv->combR[6], combtuningR7 * srfactor);
freeverb_comb_setbuffer (&priv->combL[7], combtuningL8 * srfactor);
freeverb_comb_setbuffer (&priv->combR[7], combtuningR8 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[0], allpasstuningL1 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[0], allpasstuningR1 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[1], allpasstuningL2 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[1], allpasstuningR2 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[2], allpasstuningL3 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[2], allpasstuningR3 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassL[3], allpasstuningL4 * srfactor);
freeverb_allpass_setbuffer (&priv->allpassR[3], allpasstuningR4 * srfactor);
/* clear buffers */
freeverb_revmodel_init (filter);
/* set default values */
freeverb_allpass_setfeedback (&priv->allpassL[0], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[0], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassL[1], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[1], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassL[2], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[2], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassL[3], 0.5f);
freeverb_allpass_setfeedback (&priv->allpassR[3], 0.5f);
}
static void
gst_freeverb_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstFreeverb *filter = GST_FREEVERB (object);
GstFreeverbPrivate *priv = filter->priv;
gint i;
switch (prop_id) {
case PROP_ROOM_SIZE:
filter->room_size = g_value_get_float (value);
priv->roomsize = (filter->room_size * scaleroom) + offsetroom;
for (i = 0; i < numcombs; i++) {
freeverb_comb_setfeedback (&priv->combL[i], priv->roomsize);
freeverb_comb_setfeedback (&priv->combR[i], priv->roomsize);
}
break;
case PROP_DAMPING:
filter->damping = g_value_get_float (value);
priv->damp = filter->damping * scaledamp;
for (i = 0; i < numcombs; i++) {
freeverb_comb_setdamp (&priv->combL[i], priv->damp);
freeverb_comb_setdamp (&priv->combR[i], priv->damp);
}
break;
case PROP_PAN_WIDTH:
filter->pan_width = g_value_get_float (value);
priv->width = filter->pan_width;
priv->wet1 = priv->wet * (priv->width / 2.0f + 0.5f);
priv->wet2 = priv->wet * ((1.0f - priv->width) / 2.0f);
break;
case PROP_LEVEL:
filter->level = g_value_get_float (value);
priv->wet = filter->level * scalewet;
priv->dry = (1.0 - filter->level) * scaledry;
priv->wet1 = priv->wet * (priv->width / 2.0f + 0.5f);
priv->wet2 = priv->wet * ((1.0f - priv->width) / 2.0f);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_freeverb_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstFreeverb *filter = GST_FREEVERB (object);
switch (prop_id) {
case PROP_ROOM_SIZE:
g_value_set_float (value, filter->room_size);
break;
case PROP_DAMPING:
g_value_set_float (value, filter->damping);
break;
case PROP_PAN_WIDTH:
g_value_set_float (value, filter->pan_width);
break;
case PROP_LEVEL:
g_value_set_float (value, filter->level);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* GstBaseTransform vmethod implementations */
static gboolean
gst_freeverb_get_unit_size (GstBaseTransform * base, GstCaps * caps,
gsize * size)
{
GstAudioInfo info;
g_assert (size);
if (!gst_audio_info_from_caps (&info, caps))
return FALSE;
*size = GST_AUDIO_INFO_BPF (&info);
GST_INFO_OBJECT (base, "unit size: %" G_GSIZE_FORMAT, *size);
return TRUE;
}
static GstCaps *
gst_freeverb_transform_caps (GstBaseTransform * base,
GstPadDirection direction, GstCaps * caps, GstCaps * filter)
{
GstCaps *res;
GstStructure *structure;
gint i;
/* replace the channel property with our range. */
res = gst_caps_copy (caps);
for (i = 0; i < gst_caps_get_size (res); i++) {
structure = gst_caps_get_structure (res, i);
if (direction == GST_PAD_SRC) {
GST_INFO_OBJECT (base, "[%d] allow 1-2 channels", i);
gst_structure_set (structure, "channels", GST_TYPE_INT_RANGE, 1, 2, NULL);
} else {
GST_INFO_OBJECT (base, "[%d] allow 2 channels", i);
gst_structure_set (structure, "channels", G_TYPE_INT, 2, NULL);
}
gst_structure_remove_field (structure, "channel-mask");
}
GST_DEBUG_OBJECT (base, "transformed %" GST_PTR_FORMAT, res);
if (filter) {
GstCaps *intersection;
GST_DEBUG_OBJECT (base, "Using filter caps %" GST_PTR_FORMAT, filter);
intersection =
gst_caps_intersect_full (filter, res, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (res);
res = intersection;
GST_DEBUG_OBJECT (base, "Intersection %" GST_PTR_FORMAT, res);
}
return res;
}
static gboolean
gst_freeverb_set_caps (GstBaseTransform * base, GstCaps * incaps,
GstCaps * outcaps)
{
GstFreeverb *filter = GST_FREEVERB (base);
GstAudioInfo info;
/*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */
if (!gst_audio_info_from_caps (&info, incaps))
goto no_format;
GST_DEBUG ("try to process %d input with %d channels",
GST_AUDIO_INFO_FORMAT (&info), GST_AUDIO_INFO_CHANNELS (&info));
if (!gst_freeverb_set_process_function (filter, &info))
goto no_format;
filter->info = info;
gst_freeverb_init_rev_model (filter);
filter->drained = FALSE;
GST_INFO_OBJECT (base, "model configured");
return TRUE;
no_format:
{
GST_DEBUG ("invalid caps");
return FALSE;
}
}
static gboolean
gst_freeverb_transform_m2s_int (GstFreeverb * filter,
gint16 * idata, gint16 * odata, guint num_samples)
{
GstFreeverbPrivate *priv = filter->priv;
gint i, k;
gfloat out_l1, out_r1, input_1;
gfloat out_l2, out_r2, input_2;
gboolean drained = TRUE;
for (k = 0; k < num_samples; k++) {
out_l1 = out_r1 = 0.0;
/* The original Freeverb code expects a stereo signal and 'input_1'
* is set to the sum of the left and right input_1 sample. Since
* this code works on a mono signal, 'input_1' is set to twice the
* input_1 sample. */
input_2 = (gfloat) * idata++;
input_1 = (2.0f * input_2 + DC_OFFSET) * priv->gain;
/* Accumulate comb filters in parallel */
for (i = 0; i < numcombs; i++) {
freeverb_comb_process (priv->combL[i], input_1, out_l1);
freeverb_comb_process (priv->combR[i], input_1, out_r1);
}
/* Feed through allpasses in series */
for (i = 0; i < numallpasses; i++) {
freeverb_allpass_process (priv->allpassL[i], out_l1);
freeverb_allpass_process (priv->allpassR[i], out_r1);
}
/* Remove the DC offset */
out_l1 -= (gfloat) DC_OFFSET;
out_r1 -= (gfloat) DC_OFFSET;
/* Calculate output */
out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2 * priv->dry;
out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2 * priv->dry;
out_l2 = CLAMP (out_l2, G_MININT16, G_MAXINT16);
out_r2 = CLAMP (out_r2, G_MININT16, G_MAXINT16);
*odata++ = (gint16) out_l2;
*odata++ = (gint16) out_r2;
if (abs ((gint16) out_l2) > 0 || abs ((gint16) out_r2) > 0)
drained = FALSE;
}
return drained;
}
static gboolean
gst_freeverb_transform_s2s_int (GstFreeverb * filter,
gint16 * idata, gint16 * odata, guint num_samples)
{
GstFreeverbPrivate *priv = filter->priv;
gint i, k;
gfloat out_l1, out_r1, input_1l, input_1r;
gfloat out_l2, out_r2, input_2l, input_2r;
gboolean drained = TRUE;
for (k = 0; k < num_samples; k++) {
out_l1 = out_r1 = 0.0;
input_2l = (gfloat) * idata++;
input_2r = (gfloat) * idata++;
input_1l = (input_2l + DC_OFFSET) * priv->gain;
input_1r = (input_2r + DC_OFFSET) * priv->gain;
/* Accumulate comb filters in parallel */
for (i = 0; i < numcombs; i++) {
freeverb_comb_process (priv->combL[i], input_1l, out_l1);
freeverb_comb_process (priv->combR[i], input_1r, out_r1);
}
/* Feed through allpasses in series */
for (i = 0; i < numallpasses; i++) {
freeverb_allpass_process (priv->allpassL[i], out_l1);
freeverb_allpass_process (priv->allpassR[i], out_r1);
}
/* Remove the DC offset */
out_l1 -= (gfloat) DC_OFFSET;
out_r1 -= (gfloat) DC_OFFSET;
/* Calculate output */
out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2l * priv->dry;
out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2r * priv->dry;
out_l2 = CLAMP (out_l2, G_MININT16, G_MAXINT16);
out_r2 = CLAMP (out_r2, G_MININT16, G_MAXINT16);
*odata++ = (gint16) out_l2;
*odata++ = (gint16) out_r2;
if (abs ((gint16) out_l2) > 0 || abs ((gint16) out_r2) > 0)
drained = FALSE;
}
return drained;
}
static gboolean
gst_freeverb_transform_m2s_float (GstFreeverb * filter,
gfloat * idata, gfloat * odata, guint num_samples)
{
GstFreeverbPrivate *priv = filter->priv;
gint i, k;
gfloat out_l1, out_r1, input_1;
gfloat out_l2, out_r2, input_2;
gboolean drained = TRUE;
for (k = 0; k < num_samples; k++) {
out_l1 = out_r1 = 0.0;
/* The original Freeverb code expects a stereo signal and 'input_1'
* is set to the sum of the left and right input_1 sample. Since
* this code works on a mono signal, 'input_1' is set to twice the
* input_1 sample. */
input_2 = *idata++;
input_1 = (2.0f * input_2 + DC_OFFSET) * priv->gain;
/* Accumulate comb filters in parallel */
for (i = 0; i < numcombs; i++) {
freeverb_comb_process (priv->combL[i], input_1, out_l1);
freeverb_comb_process (priv->combR[i], input_1, out_r1);
}
/* Feed through allpasses in series */
for (i = 0; i < numallpasses; i++) {
freeverb_allpass_process (priv->allpassL[i], out_l1);
freeverb_allpass_process (priv->allpassR[i], out_r1);
}
/* Remove the DC offset */
out_l1 -= (gfloat) DC_OFFSET;
out_r1 -= (gfloat) DC_OFFSET;
/* Calculate output */
out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2 * priv->dry;
out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2 * priv->dry;
*odata++ = out_l2;
*odata++ = out_r2;
if (fabs (out_l2) > 0 || fabs (out_r2) > 0)
drained = FALSE;
}
return drained;
}
static gboolean
gst_freeverb_transform_s2s_float (GstFreeverb * filter,
gfloat * idata, gfloat * odata, guint num_samples)
{
GstFreeverbPrivate *priv = filter->priv;
gint i, k;
gfloat out_l1, out_r1, input_1l, input_1r;
gfloat out_l2, out_r2, input_2l, input_2r;
gboolean drained = TRUE;
for (k = 0; k < num_samples; k++) {
out_l1 = out_r1 = 0.0;
input_2l = *idata++;
input_2r = *idata++;
input_1l = (input_2l + DC_OFFSET) * priv->gain;
input_1r = (input_2r + DC_OFFSET) * priv->gain;
/* Accumulate comb filters in parallel */
for (i = 0; i < numcombs; i++) {
freeverb_comb_process (priv->combL[i], input_1l, out_l1);
freeverb_comb_process (priv->combR[i], input_1r, out_r1);
}
/* Feed through allpasses in series */
for (i = 0; i < numallpasses; i++) {
freeverb_allpass_process (priv->allpassL[i], out_l1);
freeverb_allpass_process (priv->allpassR[i], out_r1);
}
/* Remove the DC offset */
out_l1 -= (gfloat) DC_OFFSET;
out_r1 -= (gfloat) DC_OFFSET;
/* Calculate output */
out_l2 = out_l1 * priv->wet1 + out_r1 * priv->wet2 + input_2l * priv->dry;
out_r2 = out_r1 * priv->wet1 + out_l1 * priv->wet2 + input_2r * priv->dry;
*odata++ = out_l2;
*odata++ = out_r2;
if (fabs (out_l2) > 0 || fabs (out_r2) > 0)
drained = FALSE;
}
return drained;
}
/* this function does the actual processing
*/
static GstFlowReturn
gst_freeverb_transform (GstBaseTransform * base, GstBuffer * inbuf,
GstBuffer * outbuf)
{
GstFreeverb *filter = GST_FREEVERB (base);
guint num_samples;
GstClockTime timestamp;
GstMapInfo inmap, outmap;
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
timestamp =
gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
num_samples = outmap.size / (2 * GST_AUDIO_INFO_BPS (&filter->info));
GST_DEBUG_OBJECT (filter, "processing %u samples at %" GST_TIME_FORMAT,
num_samples, GST_TIME_ARGS (timestamp));
if (GST_CLOCK_TIME_IS_VALID (timestamp))
gst_object_sync_values (GST_OBJECT (filter), timestamp);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_DISCONT))) {
filter->drained = FALSE;
}
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))) {
if (filter->drained) {
memset (outmap.data, 0, outmap.size);
}
} else {
filter->drained = FALSE;
}
if (!filter->drained) {
filter->drained =
filter->process (filter, inmap.data, outmap.data, num_samples);
}
if (filter->drained) {
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
}
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
return GST_FLOW_OK;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return GST_ELEMENT_REGISTER (freeverb, plugin);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
freeverb,
"Reverberation/room effect",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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