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gstreamer/gst-libs/gst/audio/audio-resampler.c
Thibault Saunier 909baa2360 Pass the code through codespell
2019-08-30 13:05:36 +00:00

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/* GStreamer
* Copyright (C) <2015> Wim Taymans <wim.taymans@gmail.com>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <string.h>
#include <stdio.h>
#include <math.h>
#ifdef HAVE_ORC
#include <orc/orc.h>
#endif
#include "audio-resampler.h"
#include "audio-resampler-private.h"
#include "audio-resampler-macros.h"
#define MEM_ALIGN(m,a) ((gint8 *)((guintptr)((gint8 *)(m) + ((a)-1)) & ~((a)-1)))
#define ALIGN 16
#define TAPS_OVERREAD 16
GST_DEBUG_CATEGORY_STATIC (audio_resampler_debug);
#define GST_CAT_DEFAULT audio_resampler_debug
/**
* SECTION:gstaudioresampler
* @title: GstAudioResampler
* @short_description: Utility structure for resampler information
*
* #GstAudioResampler is a structure which holds the information
* required to perform various kinds of resampling filtering.
*
*/
static const gint oversample_qualities[] = {
4, 4, 4, 8, 8, 16, 16, 16, 16, 32, 32
};
typedef struct
{
gdouble cutoff;
gdouble downsample_cutoff_factor;
gdouble stopband_attenuation;
gdouble transition_bandwidth;
} KaiserQualityMap;
static const KaiserQualityMap kaiser_qualities[] = {
{0.860, 0.96511, 60, 0.7}, /* 8 taps */
{0.880, 0.96591, 65, 0.29}, /* 16 taps */
{0.910, 0.96923, 70, 0.145}, /* 32 taps */
{0.920, 0.97600, 80, 0.105}, /* 48 taps */
{0.940, 0.97979, 85, 0.087}, /* 64 taps default quality */
{0.940, 0.98085, 95, 0.077}, /* 80 taps */
{0.945, 0.99471, 100, 0.068}, /* 96 taps */
{0.950, 1.0, 105, 0.055}, /* 128 taps */
{0.960, 1.0, 110, 0.045}, /* 160 taps */
{0.968, 1.0, 115, 0.039}, /* 192 taps */
{0.975, 1.0, 120, 0.0305} /* 256 taps */
};
typedef struct
{
gint n_taps;
gdouble cutoff;
} BlackmanQualityMap;
static const BlackmanQualityMap blackman_qualities[] = {
{8, 0.5,},
{16, 0.6,},
{24, 0.72,},
{32, 0.8,},
{48, 0.85,}, /* default */
{64, 0.90,},
{80, 0.92,},
{96, 0.933,},
{128, 0.950,},
{148, 0.955,},
{160, 0.960,}
};
#define DEFAULT_RESAMPLER_METHOD GST_AUDIO_RESAMPLER_METHOD_KAISER
#define DEFAULT_QUALITY GST_AUDIO_RESAMPLER_QUALITY_DEFAULT
#define DEFAULT_OPT_CUBIC_B 1.0
#define DEFAULT_OPT_CUBIC_C 0.0
#define DEFAULT_OPT_FILTER_MODE GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO
#define DEFAULT_OPT_FILTER_MODE_THRESHOLD 1048576
#define DEFAULT_OPT_FILTER_INTERPOLATION GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC
#define DEFAULT_OPT_FILTER_OVERSAMPLE 8
#define DEFAULT_OPT_MAX_PHASE_ERROR 0.1
static gdouble
get_opt_double (GstStructure * options, const gchar * name, gdouble def)
{
gdouble res;
if (!options || !gst_structure_get_double (options, name, &res))
res = def;
return res;
}
static gint
get_opt_int (GstStructure * options, const gchar * name, gint def)
{
gint res;
if (!options || !gst_structure_get_int (options, name, &res))
res = def;
return res;
}
static gint
get_opt_enum (GstStructure * options, const gchar * name, GType type, gint def)
{
gint res;
if (!options || !gst_structure_get_enum (options, name, type, &res))
res = def;
return res;
}
#define GET_OPT_CUTOFF(options,def) get_opt_double(options, \
GST_AUDIO_RESAMPLER_OPT_CUTOFF,def)
#define GET_OPT_DOWN_CUTOFF_FACTOR(options,def) get_opt_double(options, \
GST_AUDIO_RESAMPLER_OPT_DOWN_CUTOFF_FACTOR, def)
#define GET_OPT_STOP_ATTENUATION(options,def) get_opt_double(options, \
GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION, def)
#define GET_OPT_TRANSITION_BANDWIDTH(options,def) get_opt_double(options, \
GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH, def)
#define GET_OPT_CUBIC_B(options) get_opt_double(options, \
GST_AUDIO_RESAMPLER_OPT_CUBIC_B, DEFAULT_OPT_CUBIC_B)
#define GET_OPT_CUBIC_C(options) get_opt_double(options, \
GST_AUDIO_RESAMPLER_OPT_CUBIC_C, DEFAULT_OPT_CUBIC_C)
#define GET_OPT_N_TAPS(options,def) get_opt_int(options, \
GST_AUDIO_RESAMPLER_OPT_N_TAPS, def)
#define GET_OPT_FILTER_MODE(options) get_opt_enum(options, \
GST_AUDIO_RESAMPLER_OPT_FILTER_MODE, GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE, \
DEFAULT_OPT_FILTER_MODE)
#define GET_OPT_FILTER_MODE_THRESHOLD(options) get_opt_int(options, \
GST_AUDIO_RESAMPLER_OPT_FILTER_MODE_THRESHOLD, DEFAULT_OPT_FILTER_MODE_THRESHOLD)
#define GET_OPT_FILTER_INTERPOLATION(options) get_opt_enum(options, \
GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION, GST_TYPE_AUDIO_RESAMPLER_FILTER_INTERPOLATION, \
DEFAULT_OPT_FILTER_INTERPOLATION)
#define GET_OPT_FILTER_OVERSAMPLE(options) get_opt_int(options, \
GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE, DEFAULT_OPT_FILTER_OVERSAMPLE)
#define GET_OPT_MAX_PHASE_ERROR(options) get_opt_double(options, \
GST_AUDIO_RESAMPLER_OPT_MAX_PHASE_ERROR, DEFAULT_OPT_MAX_PHASE_ERROR)
#include "dbesi0.c"
#define bessel dbesi0
static inline gdouble
get_linear_tap (gdouble x, gint n_taps)
{
gdouble res = GST_ROUND_UP_2 (n_taps) / 2 - fabs (x);
return res;
}
static inline gdouble
get_cubic_tap (gdouble x, gint n_taps, gdouble b, gdouble c)
{
gdouble res, a, a2, a3;
a = fabs (x * 4.0) / n_taps;
a2 = a * a;
a3 = a2 * a;
if (a <= 1.0)
res = ((12.0 - 9.0 * b - 6.0 * c) * a3 +
(-18.0 + 12.0 * b + 6.0 * c) * a2 + (6.0 - 2.0 * b)) / 6.0;
else if (a <= 2.0)
res = ((-b - 6.0 * c) * a3 +
(6.0 * b + 30.0 * c) * a2 +
(-12.0 * b - 48.0 * c) * a + (8.0 * b + 24.0 * c)) / 6.0;
else
res = 0.0;
return res;
}
static inline gdouble
get_blackman_nuttall_tap (gdouble x, gint n_taps, gdouble Fc)
{
gdouble s, y, w;
y = G_PI * x;
s = (y == 0.0 ? Fc : sin (y * Fc) / y);
w = 2.0 * y / n_taps + G_PI;
return s * (0.3635819 - 0.4891775 * cos (w) + 0.1365995 * cos (2 * w) -
0.0106411 * cos (3 * w));
}
static inline gdouble
get_kaiser_tap (gdouble x, gint n_taps, gdouble Fc, gdouble beta)
{
gdouble s, y, w;
y = G_PI * x;
s = (y == 0.0 ? Fc : sin (y * Fc) / y);
w = 2.0 * x / n_taps;
return s * bessel (beta * sqrt (MAX (1 - w * w, 0)));
}
#define MAKE_CONVERT_TAPS_INT_FUNC(type, precision) \
static void \
convert_taps_##type##_c (gdouble *tmp_taps, gpointer taps, \
gdouble weight, gint n_taps) \
{ \
gint64 one = (1LL << precision) - 1; \
type *t = taps; \
gdouble multiplier = one; \
gint i, j; \
gdouble offset, l_offset, h_offset; \
gboolean exact = FALSE; \
/* Round to integer, but with an adjustable bias that we use to */ \
/* eliminate the DC error. */ \
l_offset = 0.0; \
h_offset = 1.0; \
offset = 0.5; \
for (i = 0; i < 32; i++) { \
gint64 sum = 0; \
for (j = 0; j < n_taps; j++) \
sum += floor (offset + tmp_taps[j] * multiplier / weight); \
if (sum == one) { \
exact = TRUE; \
break; \
} \
if (l_offset == h_offset) \
break; \
if (sum < one) { \
if (offset > l_offset) \
l_offset = offset; \
offset += (h_offset - l_offset) / 2; \
} else { \
if (offset < h_offset) \
h_offset = offset; \
offset -= (h_offset - l_offset) / 2; \
} \
} \
for (j = 0; j < n_taps; j++) \
t[j] = floor (offset + tmp_taps[j] * multiplier / weight); \
if (!exact) \
GST_WARNING ("can't find exact taps"); \
}
#define MAKE_CONVERT_TAPS_FLOAT_FUNC(type) \
static void \
convert_taps_##type##_c (gdouble *tmp_taps, gpointer taps, \
gdouble weight, gint n_taps) \
{ \
gint i; \
type *t = taps; \
for (i = 0; i < n_taps; i++) \
t[i] = tmp_taps[i] / weight; \
}
MAKE_CONVERT_TAPS_INT_FUNC (gint16, PRECISION_S16);
MAKE_CONVERT_TAPS_INT_FUNC (gint32, PRECISION_S32);
MAKE_CONVERT_TAPS_FLOAT_FUNC (gfloat);
MAKE_CONVERT_TAPS_FLOAT_FUNC (gdouble);
static ConvertTapsFunc convert_taps_funcs[] = {
convert_taps_gint16_c,
convert_taps_gint32_c,
convert_taps_gfloat_c,
convert_taps_gdouble_c
};
#define convert_taps_gint16 convert_taps_funcs[0]
#define convert_taps_gint32 convert_taps_funcs[1]
#define convert_taps_gfloat convert_taps_funcs[2]
#define convert_taps_gdouble convert_taps_funcs[3]
static void
make_taps (GstAudioResampler * resampler, gdouble * res, gdouble x, gint n_taps)
{
gdouble weight = 0.0, *tmp_taps = resampler->tmp_taps;
gint i;
switch (resampler->method) {
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
break;
case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
for (i = 0; i < n_taps; i++)
weight += tmp_taps[i] = get_linear_tap (x + i, resampler->n_taps);
break;
case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
for (i = 0; i < n_taps; i++)
weight += tmp_taps[i] = get_cubic_tap (x + i, resampler->n_taps,
resampler->b, resampler->c);
break;
case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
for (i = 0; i < n_taps; i++)
weight += tmp_taps[i] =
get_blackman_nuttall_tap (x + i,
resampler->n_taps, resampler->cutoff);
break;
case GST_AUDIO_RESAMPLER_METHOD_KAISER:
for (i = 0; i < n_taps; i++)
weight += tmp_taps[i] =
get_kaiser_tap (x + i, resampler->n_taps,
resampler->cutoff, resampler->kaiser_beta);
break;
}
resampler->convert_taps (tmp_taps, res, weight, n_taps);
}
#define MAKE_COEFF_LINEAR_INT_FUNC(type,type2,prec) \
static inline void \
make_coeff_##type##_linear (gint num, gint denom, type *icoeff) \
{ \
type x = ((gint64)num << prec) / denom; \
icoeff[0] = icoeff[2] = x; \
icoeff[1] = icoeff[3] = (type)(((type2)1 << prec)-1) - x; \
}
#define MAKE_COEFF_LINEAR_FLOAT_FUNC(type) \
static inline void \
make_coeff_##type##_linear (gint num, gint denom, type *icoeff) \
{ \
type x = (type)num / denom; \
icoeff[0] = icoeff[2] = x; \
icoeff[1] = icoeff[3] = (type)1.0 - x; \
}
MAKE_COEFF_LINEAR_INT_FUNC (gint16, gint32, PRECISION_S16);
MAKE_COEFF_LINEAR_INT_FUNC (gint32, gint64, PRECISION_S32);
MAKE_COEFF_LINEAR_FLOAT_FUNC (gfloat);
MAKE_COEFF_LINEAR_FLOAT_FUNC (gdouble);
#define MAKE_COEFF_CUBIC_INT_FUNC(type,type2,prec) \
static inline void \
make_coeff_##type##_cubic (gint num, gint denom, type *icoeff) \
{ \
type2 one = ((type2)1 << prec) - 1; \
type2 x = ((gint64) num << prec) / denom; \
type2 x2 = (x * x) >> prec; \
type2 x3 = (x2 * x) >> prec; \
icoeff[0] = (((x3 - x) << prec) / 6) >> prec; \
icoeff[1] = x + ((x2 - x3) >> 1); \
icoeff[3] = -(((x << prec) / 3) >> prec) + \
(x2 >> 1) - (((x3 << prec) / 6) >> prec); \
icoeff[2] = one - icoeff[0] - icoeff[1] - icoeff[3]; \
}
#define MAKE_COEFF_CUBIC_FLOAT_FUNC(type) \
static inline void \
make_coeff_##type##_cubic (gint num, gint denom, type *icoeff) \
{ \
type x = (type) num / denom, x2 = x * x, x3 = x2 * x; \
icoeff[0] = 0.16667f * (x3 - x); \
icoeff[1] = x + 0.5f * (x2 - x3); \
icoeff[3] = -0.33333f * x + 0.5f * x2 - 0.16667f * x3; \
icoeff[2] = (type)1.0 - icoeff[0] - icoeff[1] - icoeff[3]; \
}
MAKE_COEFF_CUBIC_INT_FUNC (gint16, gint32, PRECISION_S16);
MAKE_COEFF_CUBIC_INT_FUNC (gint32, gint64, PRECISION_S32);
MAKE_COEFF_CUBIC_FLOAT_FUNC (gfloat);
MAKE_COEFF_CUBIC_FLOAT_FUNC (gdouble);
#define INTERPOLATE_INT_LINEAR_FUNC(type,type2,prec,limit) \
static inline void \
interpolate_##type##_linear_c (gpointer op, const gpointer ap, \
gint len, const gpointer icp, gint astride) \
{ \
gint i; \
type *o = op, *a = ap, *ic = icp; \
type2 tmp, c0 = ic[0]; \
const type *c[2] = {(type*)((gint8*)a + 0*astride), \
(type*)((gint8*)a + 1*astride)}; \
\
for (i = 0; i < len; i++) { \
tmp = ((type2)c[0][i] - (type2)c[1][i]) * c0 + \
(((type2)c[1][i]) << (prec)); \
o[i] = (tmp + ((type2)1 << ((prec) - 1))) >> (prec); \
} \
}
#define INTERPOLATE_FLOAT_LINEAR_FUNC(type) \
static inline void \
interpolate_##type##_linear_c (gpointer op, const gpointer ap, \
gint len, const gpointer icp, gint astride) \
{ \
gint i; \
type *o = op, *a = ap, *ic = icp; \
type c0 = ic[0]; \
const type *c[2] = {(type*)((gint8*)a + 0*astride), \
(type*)((gint8*)a + 1*astride)}; \
\
for (i = 0; i < len; i++) { \
o[i] = (c[0][i] - c[1][i]) * c0 + c[1][i]; \
} \
}
INTERPOLATE_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
INTERPOLATE_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
INTERPOLATE_FLOAT_LINEAR_FUNC (gfloat);
INTERPOLATE_FLOAT_LINEAR_FUNC (gdouble);
#define INTERPOLATE_INT_CUBIC_FUNC(type,type2,prec,limit) \
static inline void \
interpolate_##type##_cubic_c (gpointer op, const gpointer ap, \
gint len, const gpointer icp, gint astride) \
{ \
gint i; \
type *o = op, *a = ap, *ic = icp; \
type2 tmp, c0 = ic[0], c1 = ic[1], c2 = ic[2], c3 = ic[3]; \
const type *c[4] = {(type*)((gint8*)a + 0*astride), \
(type*)((gint8*)a + 1*astride), \
(type*)((gint8*)a + 2*astride), \
(type*)((gint8*)a + 3*astride)}; \
\
for (i = 0; i < len; i++) { \
tmp = (type2)c[0][i] * c0 + (type2)c[1][i] * c1 + \
(type2)c[2][i] * c2 + (type2)c[3][i] * c3; \
tmp = (tmp + ((type2)1 << ((prec) - 1))) >> (prec); \
o[i] = CLAMP (tmp, -(limit), (limit) - 1); \
} \
}
#define INTERPOLATE_FLOAT_CUBIC_FUNC(type) \
static inline void \
interpolate_##type##_cubic_c (gpointer op, const gpointer ap, \
gint len, const gpointer icp, gint astride) \
{ \
gint i; \
type *o = op, *a = ap, *ic = icp; \
type c0 = ic[0], c1 = ic[1], c2 = ic[2], c3 = ic[3]; \
const type *c[4] = {(type*)((gint8*)a + 0*astride), \
(type*)((gint8*)a + 1*astride), \
(type*)((gint8*)a + 2*astride), \
(type*)((gint8*)a + 3*astride)}; \
\
for (i = 0; i < len; i++) { \
o[i] = c[0][i] * c0 + c[1][i] * c1 + \
c[2][i] * c2 + c[3][i] * c3; \
} \
}
INTERPOLATE_INT_CUBIC_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
INTERPOLATE_INT_CUBIC_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
INTERPOLATE_FLOAT_CUBIC_FUNC (gfloat);
INTERPOLATE_FLOAT_CUBIC_FUNC (gdouble);
static InterpolateFunc interpolate_funcs[] = {
interpolate_gint16_linear_c,
interpolate_gint32_linear_c,
interpolate_gfloat_linear_c,
interpolate_gdouble_linear_c,
interpolate_gint16_cubic_c,
interpolate_gint32_cubic_c,
interpolate_gfloat_cubic_c,
interpolate_gdouble_cubic_c,
};
#define interpolate_gint16_linear interpolate_funcs[0]
#define interpolate_gint32_linear interpolate_funcs[1]
#define interpolate_gfloat_linear interpolate_funcs[2]
#define interpolate_gdouble_linear interpolate_funcs[3]
#define interpolate_gint16_cubic interpolate_funcs[4]
#define interpolate_gint32_cubic interpolate_funcs[5]
#define interpolate_gfloat_cubic interpolate_funcs[6]
#define interpolate_gdouble_cubic interpolate_funcs[7]
#define GET_TAPS_NEAREST_FUNC(type) \
static inline gpointer \
get_taps_##type##_nearest (GstAudioResampler * resampler, \
gint *samp_index, gint *samp_phase, type icoeff[4]) \
{ \
gint out_rate = resampler->out_rate; \
*samp_index += resampler->samp_inc; \
*samp_phase += resampler->samp_frac; \
if (*samp_phase >= out_rate) { \
*samp_phase -= out_rate; \
*samp_index += 1; \
} \
return NULL; \
}
GET_TAPS_NEAREST_FUNC (gint16);
GET_TAPS_NEAREST_FUNC (gint32);
GET_TAPS_NEAREST_FUNC (gfloat);
GET_TAPS_NEAREST_FUNC (gdouble);
#define get_taps_gint16_nearest get_taps_gint16_nearest
#define get_taps_gint32_nearest get_taps_gint32_nearest
#define get_taps_gfloat_nearest get_taps_gfloat_nearest
#define get_taps_gdouble_nearest get_taps_gdouble_nearest
#define GET_TAPS_FULL_FUNC(type) \
DECL_GET_TAPS_FULL_FUNC(type) \
{ \
gpointer res; \
gint out_rate = resampler->out_rate; \
gint n_phases = resampler->n_phases; \
gint phase = (n_phases == out_rate ? *samp_phase : \
((gint64)*samp_phase * n_phases) / out_rate); \
\
res = resampler->cached_phases[phase]; \
if (G_UNLIKELY (res == NULL)) { \
res = (gint8 *) resampler->cached_taps + \
phase * resampler->cached_taps_stride; \
switch (resampler->filter_interpolation) { \
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE: \
{ \
gdouble x; \
gint n_taps = resampler->n_taps; \
\
x = 1.0 - n_taps / 2 - (gdouble) phase / n_phases; \
make_taps (resampler, res, x, n_taps); \
break; \
} \
default: \
{ \
gint offset, pos, frac; \
gint oversample = resampler->oversample; \
gint taps_stride = resampler->taps_stride; \
gint n_taps = resampler->n_taps; \
type ic[4], *taps; \
\
pos = phase * oversample; \
offset = (oversample - 1) - pos / n_phases; \
frac = pos % n_phases; \
\
taps = (type *) ((gint8 *) resampler->taps + offset * taps_stride); \
\
switch (resampler->filter_interpolation) { \
default: \
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR: \
make_coeff_##type##_linear (frac, n_phases, ic); \
break; \
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC: \
make_coeff_##type##_cubic (frac, n_phases, ic); \
break; \
} \
resampler->interpolate (res, taps, n_taps, ic, taps_stride); \
} \
} \
resampler->cached_phases[phase] = res; \
} \
*samp_index += resampler->samp_inc; \
*samp_phase += resampler->samp_frac; \
if (*samp_phase >= out_rate) { \
*samp_phase -= out_rate; \
*samp_index += 1; \
} \
return res; \
}
GET_TAPS_FULL_FUNC (gint16);
GET_TAPS_FULL_FUNC (gint32);
GET_TAPS_FULL_FUNC (gfloat);
GET_TAPS_FULL_FUNC (gdouble);
#define GET_TAPS_INTERPOLATE_FUNC(type,inter) \
DECL_GET_TAPS_INTERPOLATE_FUNC (type, inter) \
{ \
gpointer res; \
gint out_rate = resampler->out_rate; \
gint offset, frac, pos; \
gint oversample = resampler->oversample; \
gint taps_stride = resampler->taps_stride; \
\
pos = *samp_phase * oversample; \
offset = (oversample - 1) - pos / out_rate; \
frac = pos % out_rate; \
\
res = (gint8 *) resampler->taps + offset * taps_stride; \
make_coeff_##type##_##inter (frac, out_rate, icoeff); \
\
*samp_index += resampler->samp_inc; \
*samp_phase += resampler->samp_frac; \
if (*samp_phase >= out_rate) { \
*samp_phase -= out_rate; \
*samp_index += 1; \
} \
return res; \
}
GET_TAPS_INTERPOLATE_FUNC (gint16, linear);
GET_TAPS_INTERPOLATE_FUNC (gint32, linear);
GET_TAPS_INTERPOLATE_FUNC (gfloat, linear);
GET_TAPS_INTERPOLATE_FUNC (gdouble, linear);
GET_TAPS_INTERPOLATE_FUNC (gint16, cubic);
GET_TAPS_INTERPOLATE_FUNC (gint32, cubic);
GET_TAPS_INTERPOLATE_FUNC (gfloat, cubic);
GET_TAPS_INTERPOLATE_FUNC (gdouble, cubic);
#define INNER_PRODUCT_NEAREST_FUNC(type) \
static inline void \
inner_product_##type##_nearest_1_c (type * o, const type * a, \
const type * b, gint len, const type *ic, gint bstride) \
{ \
*o = *a; \
}
INNER_PRODUCT_NEAREST_FUNC (gint16);
INNER_PRODUCT_NEAREST_FUNC (gint32);
INNER_PRODUCT_NEAREST_FUNC (gfloat);
INNER_PRODUCT_NEAREST_FUNC (gdouble);
#define INNER_PRODUCT_INT_FULL_FUNC(type,type2,prec,limit) \
static inline void \
inner_product_##type##_full_1_c (type * o, const type * a, \
const type * b, gint len, const type *ic, gint bstride) \
{ \
gint i; \
type2 res[4] = { 0, 0, 0, 0 }; \
\
for (i = 0; i < len; i += 4) { \
res[0] += (type2) a[i + 0] * (type2) b[i + 0]; \
res[1] += (type2) a[i + 1] * (type2) b[i + 1]; \
res[2] += (type2) a[i + 2] * (type2) b[i + 2]; \
res[3] += (type2) a[i + 3] * (type2) b[i + 3]; \
} \
res[0] = res[0] + res[1] + res[2] + res[3]; \
res[0] = (res[0] + ((type2)1 << ((prec) - 1))) >> (prec); \
*o = CLAMP (res[0], -(limit), (limit) - 1); \
}
INNER_PRODUCT_INT_FULL_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
INNER_PRODUCT_INT_FULL_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
#define INNER_PRODUCT_INT_LINEAR_FUNC(type,type2,prec,limit) \
static inline void \
inner_product_##type##_linear_1_c (type * o, const type * a, \
const type * b, gint len, const type *ic, gint bstride) \
{ \
gint i; \
type2 res[4] = { 0, 0, 0, 0 }, c0 = ic[0]; \
const type *c[2] = {(type*)((gint8*)b + 0*bstride), \
(type*)((gint8*)b + 1*bstride)}; \
\
for (i = 0; i < len; i += 2) { \
res[0] += (type2) a[i + 0] * (type2) c[0][i + 0]; \
res[1] += (type2) a[i + 0] * (type2) c[1][i + 0]; \
res[2] += (type2) a[i + 1] * (type2) c[0][i + 1]; \
res[3] += (type2) a[i + 1] * (type2) c[1][i + 1]; \
} \
res[0] = (res[0] + res[2]) >> (prec); \
res[1] = (res[1] + res[3]) >> (prec); \
res[0] = ((type2)(type)res[0] - (type2)(type)res[1]) * c0 + \
((type2)(type)res[1] << (prec)); \
res[0] = (res[0] + ((type2)1 << ((prec) - 1))) >> (prec); \
*o = CLAMP (res[0], -(limit), (limit) - 1); \
}
INNER_PRODUCT_INT_LINEAR_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
INNER_PRODUCT_INT_LINEAR_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
#define INNER_PRODUCT_INT_CUBIC_FUNC(type,type2,prec,limit) \
static inline void \
inner_product_##type##_cubic_1_c (type * o, const type * a, \
const type * b, gint len, const type *ic, gint bstride) \
{ \
gint i; \
type2 res[4] = { 0, 0, 0, 0 }; \
const type *c[4] = {(type*)((gint8*)b + 0*bstride), \
(type*)((gint8*)b + 1*bstride), \
(type*)((gint8*)b + 2*bstride), \
(type*)((gint8*)b + 3*bstride)}; \
\
for (i = 0; i < len; i++) { \
res[0] += (type2) a[i] * (type2) c[0][i]; \
res[1] += (type2) a[i] * (type2) c[1][i]; \
res[2] += (type2) a[i] * (type2) c[2][i]; \
res[3] += (type2) a[i] * (type2) c[3][i]; \
} \
res[0] = (type2)(type)(res[0] >> (prec)) * (type2) ic[0] + \
(type2)(type)(res[1] >> (prec)) * (type2) ic[1] + \
(type2)(type)(res[2] >> (prec)) * (type2) ic[2] + \
(type2)(type)(res[3] >> (prec)) * (type2) ic[3]; \
res[0] = (res[0] + ((type2)1 << ((prec) - 1))) >> (prec); \
*o = CLAMP (res[0], -(limit), (limit) - 1); \
}
INNER_PRODUCT_INT_CUBIC_FUNC (gint16, gint32, PRECISION_S16, (gint32) 1 << 15);
INNER_PRODUCT_INT_CUBIC_FUNC (gint32, gint64, PRECISION_S32, (gint64) 1 << 31);
#define INNER_PRODUCT_FLOAT_FULL_FUNC(type) \
static inline void \
inner_product_##type##_full_1_c (type * o, const type * a, \
const type * b, gint len, const type *ic, gint bstride) \
{ \
gint i; \
type res[4] = { 0.0, 0.0, 0.0, 0.0 }; \
\
for (i = 0; i < len; i += 4) { \
res[0] += a[i + 0] * b[i + 0]; \
res[1] += a[i + 1] * b[i + 1]; \
res[2] += a[i + 2] * b[i + 2]; \
res[3] += a[i + 3] * b[i + 3]; \
} \
*o = res[0] + res[1] + res[2] + res[3]; \
}
INNER_PRODUCT_FLOAT_FULL_FUNC (gfloat);
INNER_PRODUCT_FLOAT_FULL_FUNC (gdouble);
#define INNER_PRODUCT_FLOAT_LINEAR_FUNC(type) \
static inline void \
inner_product_##type##_linear_1_c (type * o, const type * a, \
const type * b, gint len, const type *ic, gint bstride) \
{ \
gint i; \
type res[4] = { 0.0, 0.0, 0.0, 0.0 }; \
const type *c[2] = {(type*)((gint8*)b + 0*bstride), \
(type*)((gint8*)b + 1*bstride)}; \
\
for (i = 0; i < len; i += 2) { \
res[0] += a[i + 0] * c[0][i + 0]; \
res[1] += a[i + 0] * c[1][i + 0]; \
res[2] += a[i + 1] * c[0][i + 1]; \
res[3] += a[i + 1] * c[1][i + 1]; \
} \
res[0] += res[2]; \
res[1] += res[3]; \
*o = (res[0] - res[1]) * ic[0] + res[1]; \
}
INNER_PRODUCT_FLOAT_LINEAR_FUNC (gfloat);
INNER_PRODUCT_FLOAT_LINEAR_FUNC (gdouble);
#define INNER_PRODUCT_FLOAT_CUBIC_FUNC(type) \
static inline void \
inner_product_##type##_cubic_1_c (type * o, const type * a, \
const type * b, gint len, const type *ic, gint bstride) \
{ \
gint i; \
type res[4] = { 0.0, 0.0, 0.0, 0.0 }; \
const type *c[4] = {(type*)((gint8*)b + 0*bstride), \
(type*)((gint8*)b + 1*bstride), \
(type*)((gint8*)b + 2*bstride), \
(type*)((gint8*)b + 3*bstride)}; \
\
for (i = 0; i < len; i++) { \
res[0] += a[i] * c[0][i]; \
res[1] += a[i] * c[1][i]; \
res[2] += a[i] * c[2][i]; \
res[3] += a[i] * c[3][i]; \
} \
*o = res[0] * ic[0] + res[1] * ic[1] + \
res[2] * ic[2] + res[3] * ic[3]; \
}
INNER_PRODUCT_FLOAT_CUBIC_FUNC (gfloat);
INNER_PRODUCT_FLOAT_CUBIC_FUNC (gdouble);
MAKE_RESAMPLE_FUNC_STATIC (gint16, nearest, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gint32, nearest, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gfloat, nearest, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gdouble, nearest, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gint16, full, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gint32, full, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gfloat, full, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gdouble, full, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gint16, linear, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gint32, linear, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gfloat, linear, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gdouble, linear, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gint16, cubic, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gint32, cubic, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gfloat, cubic, 1, c);
MAKE_RESAMPLE_FUNC_STATIC (gdouble, cubic, 1, c);
static ResampleFunc resample_funcs[] = {
resample_gint16_nearest_1_c,
resample_gint32_nearest_1_c,
resample_gfloat_nearest_1_c,
resample_gdouble_nearest_1_c,
resample_gint16_full_1_c,
resample_gint32_full_1_c,
resample_gfloat_full_1_c,
resample_gdouble_full_1_c,
resample_gint16_linear_1_c,
resample_gint32_linear_1_c,
resample_gfloat_linear_1_c,
resample_gdouble_linear_1_c,
resample_gint16_cubic_1_c,
resample_gint32_cubic_1_c,
resample_gfloat_cubic_1_c,
resample_gdouble_cubic_1_c,
};
#define resample_gint16_nearest_1 resample_funcs[0]
#define resample_gint32_nearest_1 resample_funcs[1]
#define resample_gfloat_nearest_1 resample_funcs[2]
#define resample_gdouble_nearest_1 resample_funcs[3]
#define resample_gint16_full_1 resample_funcs[4]
#define resample_gint32_full_1 resample_funcs[5]
#define resample_gfloat_full_1 resample_funcs[6]
#define resample_gdouble_full_1 resample_funcs[7]
#define resample_gint16_linear_1 resample_funcs[8]
#define resample_gint32_linear_1 resample_funcs[9]
#define resample_gfloat_linear_1 resample_funcs[10]
#define resample_gdouble_linear_1 resample_funcs[11]
#define resample_gint16_cubic_1 resample_funcs[12]
#define resample_gint32_cubic_1 resample_funcs[13]
#define resample_gfloat_cubic_1 resample_funcs[14]
#define resample_gdouble_cubic_1 resample_funcs[15]
#if defined HAVE_ORC && !defined DISABLE_ORC
# if defined (HAVE_ARM_NEON)
# define CHECK_NEON
# include "audio-resampler-neon.h"
# endif
# if defined (__i386__) || defined (__x86_64__)
# define CHECK_X86
# include "audio-resampler-x86.h"
# endif
#endif
static void
audio_resampler_init (void)
{
static gsize init_gonce = 0;
if (g_once_init_enter (&init_gonce)) {
GST_DEBUG_CATEGORY_INIT (audio_resampler_debug, "audio-resampler", 0,
"audio-resampler object");
#if defined HAVE_ORC && !defined DISABLE_ORC
orc_init ();
{
OrcTarget *target = orc_target_get_default ();
gint i;
if (target) {
const gchar *name;
unsigned int flags = orc_target_get_default_flags (target);
for (i = -1; i < 32; ++i) {
if (i == -1) {
name = orc_target_get_name (target);
GST_DEBUG ("target %s, default flags %08x", name, flags);
} else if (flags & (1U << i)) {
name = orc_target_get_flag_name (target, i);
GST_DEBUG ("target flag %s", name);
} else
name = NULL;
if (name) {
#ifdef CHECK_X86
audio_resampler_check_x86 (name);
#endif
#ifdef CHECK_NEON
audio_resampler_check_neon (name);
#endif
}
}
}
}
#endif
g_once_init_leave (&init_gonce, 1);
}
}
#define MAKE_DEINTERLEAVE_FUNC(type) \
static void \
deinterleave_ ##type (GstAudioResampler * resampler, gpointer sbuf[], \
gpointer in[], gsize in_frames) \
{ \
gint i, c, channels = resampler->channels; \
gsize samples_avail = resampler->samples_avail; \
for (c = 0; c < channels; c++) { \
type *s = (type *) sbuf[c] + samples_avail; \
if (G_UNLIKELY (in == NULL)) { \
for (i = 0; i < in_frames; i++) \
s[i] = 0; \
} else { \
type *ip = (type *) in[0] + c; \
for (i = 0; i < in_frames; i++, ip += channels) \
s[i] = *ip; \
} \
} \
}
MAKE_DEINTERLEAVE_FUNC (gint16);
MAKE_DEINTERLEAVE_FUNC (gint32);
MAKE_DEINTERLEAVE_FUNC (gfloat);
MAKE_DEINTERLEAVE_FUNC (gdouble);
static DeinterleaveFunc deinterleave_funcs[] = {
deinterleave_gint16,
deinterleave_gint32,
deinterleave_gfloat,
deinterleave_gdouble
};
static void
copy_func (GstAudioResampler * resampler, gpointer sbuf[],
gpointer in[], gsize in_frames)
{
gint c, channels = resampler->channels;
gsize samples_avail = resampler->samples_avail;
for (c = 0; c < channels; c++) {
guint8 *s = ((guint8 *) sbuf[c]) + (samples_avail * resampler->bps);
if (G_UNLIKELY (in == NULL)) {
memset (s, 0, in_frames * resampler->bps);
} else {
memcpy (s, in[c], in_frames * resampler->bps);
}
}
}
static void
calculate_kaiser_params (GstAudioResampler * resampler)
{
gdouble A, B, dw, tr_bw, Fc;
gint n;
const KaiserQualityMap *q = &kaiser_qualities[DEFAULT_QUALITY];
/* default cutoff */
Fc = q->cutoff;
if (resampler->out_rate < resampler->in_rate)
Fc *= q->downsample_cutoff_factor;
Fc = GET_OPT_CUTOFF (resampler->options, Fc);
A = GET_OPT_STOP_ATTENUATION (resampler->options, q->stopband_attenuation);
tr_bw =
GET_OPT_TRANSITION_BANDWIDTH (resampler->options,
q->transition_bandwidth);
GST_LOG ("Fc %f, A %f, tr_bw %f", Fc, A, tr_bw);
/* calculate Beta */
if (A > 50)
B = 0.1102 * (A - 8.7);
else if (A >= 21)
B = 0.5842 * pow (A - 21, 0.4) + 0.07886 * (A - 21);
else
B = 0.0;
/* calculate transition width in radians */
dw = 2 * G_PI * (tr_bw);
/* order of the filter */
n = (A - 8.0) / (2.285 * dw);
resampler->kaiser_beta = B;
resampler->n_taps = n + 1;
resampler->cutoff = Fc;
GST_LOG ("using Beta %f n_taps %d cutoff %f", resampler->kaiser_beta,
resampler->n_taps, resampler->cutoff);
}
static void
alloc_taps_mem (GstAudioResampler * resampler, gint bps, gint n_taps,
gint n_phases)
{
if (resampler->alloc_taps >= n_taps && resampler->alloc_phases >= n_phases)
return;
GST_DEBUG ("allocate bps %d n_taps %d n_phases %d", bps, n_taps, n_phases);
resampler->tmp_taps =
g_realloc_n (resampler->tmp_taps, n_taps, sizeof (gdouble));
resampler->taps_stride = GST_ROUND_UP_32 (bps * (n_taps + TAPS_OVERREAD));
g_free (resampler->taps_mem);
resampler->taps_mem =
g_malloc0 (n_phases * resampler->taps_stride + ALIGN - 1);
resampler->taps = MEM_ALIGN ((gint8 *) resampler->taps_mem, ALIGN);
resampler->alloc_taps = n_taps;
resampler->alloc_phases = n_phases;
}
static void
alloc_cache_mem (GstAudioResampler * resampler, gint bps, gint n_taps,
gint n_phases)
{
gsize phases_size;
resampler->tmp_taps =
g_realloc_n (resampler->tmp_taps, n_taps, sizeof (gdouble));
resampler->cached_taps_stride =
GST_ROUND_UP_32 (bps * (n_taps + TAPS_OVERREAD));
phases_size = sizeof (gpointer) * n_phases;
g_free (resampler->cached_taps_mem);
resampler->cached_taps_mem =
g_malloc0 (phases_size + n_phases * resampler->cached_taps_stride +
ALIGN - 1);
resampler->cached_taps =
MEM_ALIGN ((gint8 *) resampler->cached_taps_mem + phases_size, ALIGN);
resampler->cached_phases = resampler->cached_taps_mem;
}
static void
setup_functions (GstAudioResampler * resampler)
{
gint index, fidx;
index = resampler->format_index;
if (resampler->in_rate == resampler->out_rate)
resampler->resample = resample_funcs[index];
else {
switch (resampler->filter_interpolation) {
default:
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE:
fidx = 0;
break;
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
GST_DEBUG ("using linear interpolation for filter coefficients");
fidx = 0;
break;
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
GST_DEBUG ("using cubic interpolation for filter coefficients");
fidx = 4;
break;
}
GST_DEBUG ("using filter interpolate function %d", index + fidx);
resampler->interpolate = interpolate_funcs[index + fidx];
switch (resampler->method) {
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
GST_DEBUG ("using nearest filter function");
break;
default:
index += 4;
switch (resampler->filter_mode) {
default:
case GST_AUDIO_RESAMPLER_FILTER_MODE_FULL:
GST_DEBUG ("using full filter function");
break;
case GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED:
index += 4 + fidx;
GST_DEBUG ("using interpolated filter function");
break;
}
break;
}
GST_DEBUG ("using resample function %d", index);
resampler->resample = resample_funcs[index];
}
}
static void
resampler_calculate_taps (GstAudioResampler * resampler)
{
gint bps;
gint n_taps, oversample;
gint in_rate, out_rate;
gboolean scale = TRUE, sinc_table = FALSE;
GstAudioResamplerFilterInterpolation filter_interpolation;
switch (resampler->method) {
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
resampler->n_taps = 2;
scale = FALSE;
break;
case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
resampler->n_taps = GET_OPT_N_TAPS (resampler->options, 2);
break;
case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
resampler->n_taps = GET_OPT_N_TAPS (resampler->options, 4);
resampler->b = GET_OPT_CUBIC_B (resampler->options);
resampler->c = GET_OPT_CUBIC_C (resampler->options);;
break;
case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
{
const BlackmanQualityMap *q = &blackman_qualities[DEFAULT_QUALITY];
resampler->n_taps = GET_OPT_N_TAPS (resampler->options, q->n_taps);
resampler->cutoff = GET_OPT_CUTOFF (resampler->options, q->cutoff);
sinc_table = TRUE;
break;
}
case GST_AUDIO_RESAMPLER_METHOD_KAISER:
calculate_kaiser_params (resampler);
sinc_table = TRUE;
break;
}
in_rate = resampler->in_rate;
out_rate = resampler->out_rate;
if (out_rate < in_rate && scale) {
resampler->cutoff = resampler->cutoff * out_rate / in_rate;
resampler->n_taps =
gst_util_uint64_scale_int (resampler->n_taps, in_rate, out_rate);
}
if (sinc_table) {
resampler->n_taps = GST_ROUND_UP_8 (resampler->n_taps);
resampler->filter_mode = GET_OPT_FILTER_MODE (resampler->options);
resampler->filter_threshold =
GET_OPT_FILTER_MODE_THRESHOLD (resampler->options);
filter_interpolation = GET_OPT_FILTER_INTERPOLATION (resampler->options);
} else {
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_FULL;
filter_interpolation = GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE;
}
/* calculate oversampling for interpolated filter */
if (filter_interpolation != GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE) {
gint mult = 2;
oversample = GET_OPT_FILTER_OVERSAMPLE (resampler->options);
while (oversample > 1) {
if (mult * out_rate >= in_rate)
break;
mult *= 2;
oversample >>= 1;
}
switch (filter_interpolation) {
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
oversample *= 11;
break;
default:
break;
}
} else {
oversample = 1;
}
resampler->oversample = oversample;
n_taps = resampler->n_taps;
bps = resampler->bps;
GST_LOG ("using n_taps %d cutoff %f oversample %d", n_taps, resampler->cutoff,
oversample);
if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO) {
if (out_rate <= oversample
&& !(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_VARIABLE_RATE)) {
/* don't interpolate if we need to calculate at least the same amount
* of filter coefficients than the full table case */
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_FULL;
GST_DEBUG ("automatically selected full filter, %d <= %d", out_rate,
oversample);
} else if (bps * n_taps * out_rate < resampler->filter_threshold) {
/* switch to full filter when memory is below threshold */
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_FULL;
GST_DEBUG ("automatically selected full filter, memory %d <= %d",
bps * n_taps * out_rate, resampler->filter_threshold);
} else {
GST_DEBUG ("automatically selected interpolated filter");
resampler->filter_mode = GST_AUDIO_RESAMPLER_FILTER_MODE_INTERPOLATED;
}
}
/* interpolated table but no interpolation given, assume default */
if (resampler->filter_mode != GST_AUDIO_RESAMPLER_FILTER_MODE_FULL &&
filter_interpolation == GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE)
filter_interpolation = DEFAULT_OPT_FILTER_INTERPOLATION;
resampler->filter_interpolation = filter_interpolation;
if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_FULL &&
resampler->method != GST_AUDIO_RESAMPLER_METHOD_NEAREST) {
GST_DEBUG ("setting up filter cache");
resampler->n_phases = out_rate;
alloc_cache_mem (resampler, bps, n_taps, out_rate);
}
if (resampler->filter_interpolation !=
GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_NONE) {
gint i, isize;
gdouble x;
gpointer taps;
switch (resampler->filter_interpolation) {
default:
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR:
GST_DEBUG ("using linear interpolation to build filter");
isize = 2;
break;
case GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC:
GST_DEBUG ("using cubic interpolation to build filter");
isize = 4;
break;
}
alloc_taps_mem (resampler, bps, n_taps, oversample + isize);
for (i = 0; i < oversample + isize; i++) {
x = -(n_taps / 2) + i / (gdouble) oversample;
taps = (gint8 *) resampler->taps + i * resampler->taps_stride;
make_taps (resampler, taps, x, n_taps);
}
}
}
#define PRINT_TAPS(type,print) \
G_STMT_START { \
type sum = 0.0, *taps; \
type icoeff[4]; \
gint samp_index = 0, samp_phase = i; \
\
taps = get_taps_##type##_full (resampler, &samp_index,\
&samp_phase, icoeff); \
\
for (j = 0; j < n_taps; j++) { \
type tap = taps[j]; \
fprintf (stderr, "\t%" print " ", tap); \
sum += tap; \
} \
fprintf (stderr, "\t: sum %" print "\n", sum); \
} G_STMT_END
static void
resampler_dump (GstAudioResampler * resampler)
{
#if 0
gint i, n_taps, out_rate;
gint64 a;
out_rate = resampler->out_rate;
n_taps = resampler->n_taps;
fprintf (stderr, "out size %d, max taps %d\n", out_rate, n_taps);
a = g_get_monotonic_time ();
for (i = 0; i < out_rate; i++) {
gint j;
//fprintf (stderr, "%u: %d %d\t ", i, t->sample_inc, t->next_phase);
switch (resampler->format) {
case GST_AUDIO_FORMAT_F64:
PRINT_TAPS (gdouble, "f");
break;
case GST_AUDIO_FORMAT_F32:
PRINT_TAPS (gfloat, "f");
break;
case GST_AUDIO_FORMAT_S32:
PRINT_TAPS (gint32, "d");
break;
case GST_AUDIO_FORMAT_S16:
PRINT_TAPS (gint16, "d");
break;
default:
break;
}
}
fprintf (stderr, "time %" G_GUINT64_FORMAT "\n", g_get_monotonic_time () - a);
#endif
}
/**
* gst_audio_resampler_options_set_quality:
* @method: a #GstAudioResamplerMethod
* @quality: the quality
* @in_rate: the input rate
* @out_rate: the output rate
* @options: a #GstStructure
*
* Set the parameters for resampling from @in_rate to @out_rate using @method
* for @quality in @options.
*/
void
gst_audio_resampler_options_set_quality (GstAudioResamplerMethod method,
guint quality, gint in_rate, gint out_rate, GstStructure * options)
{
g_return_if_fail (options != NULL);
g_return_if_fail (quality <= GST_AUDIO_RESAMPLER_QUALITY_MAX);
g_return_if_fail (in_rate > 0 && out_rate > 0);
switch (method) {
case GST_AUDIO_RESAMPLER_METHOD_NEAREST:
break;
case GST_AUDIO_RESAMPLER_METHOD_LINEAR:
gst_structure_set (options,
GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, 2, NULL);
break;
case GST_AUDIO_RESAMPLER_METHOD_CUBIC:
gst_structure_set (options,
GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, 4,
GST_AUDIO_RESAMPLER_OPT_CUBIC_B, G_TYPE_DOUBLE, DEFAULT_OPT_CUBIC_B,
GST_AUDIO_RESAMPLER_OPT_CUBIC_C, G_TYPE_DOUBLE, DEFAULT_OPT_CUBIC_C,
NULL);
break;
case GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL:
{
const BlackmanQualityMap *map = &blackman_qualities[quality];
gst_structure_set (options,
GST_AUDIO_RESAMPLER_OPT_N_TAPS, G_TYPE_INT, map->n_taps,
GST_AUDIO_RESAMPLER_OPT_CUTOFF, G_TYPE_DOUBLE, map->cutoff, NULL);
break;
}
case GST_AUDIO_RESAMPLER_METHOD_KAISER:
{
const KaiserQualityMap *map = &kaiser_qualities[quality];
gdouble cutoff;
cutoff = map->cutoff;
if (out_rate < in_rate)
cutoff *= map->downsample_cutoff_factor;
gst_structure_set (options,
GST_AUDIO_RESAMPLER_OPT_CUTOFF, G_TYPE_DOUBLE, cutoff,
GST_AUDIO_RESAMPLER_OPT_STOP_ATTENUATION, G_TYPE_DOUBLE,
map->stopband_attenuation,
GST_AUDIO_RESAMPLER_OPT_TRANSITION_BANDWIDTH, G_TYPE_DOUBLE,
map->transition_bandwidth, NULL);
break;
}
}
gst_structure_set (options,
GST_AUDIO_RESAMPLER_OPT_FILTER_OVERSAMPLE, G_TYPE_INT,
oversample_qualities[quality], NULL);
}
/**
* gst_audio_resampler_new:
* @method: a #GstAudioResamplerMethod
* @flags: #GstAudioResamplerFlags
* @format: the #GstAudioFormat
* @channels: the number of channels
* @in_rate: input rate
* @out_rate: output rate
* @options: extra options
*
* Make a new resampler.
*
* Returns: (skip) (transfer full): The new #GstAudioResampler, or
* %NULL on failure.
*/
GstAudioResampler *
gst_audio_resampler_new (GstAudioResamplerMethod method,
GstAudioResamplerFlags flags,
GstAudioFormat format, gint channels,
gint in_rate, gint out_rate, GstStructure * options)
{
gboolean non_interleaved_in, non_interleaved_out;
GstAudioResampler *resampler;
const GstAudioFormatInfo *info;
GstStructure *def_options = NULL;
g_return_val_if_fail (method >= GST_AUDIO_RESAMPLER_METHOD_NEAREST
&& method <= GST_AUDIO_RESAMPLER_METHOD_KAISER, NULL);
g_return_val_if_fail (format == GST_AUDIO_FORMAT_S16 ||
format == GST_AUDIO_FORMAT_S32 || format == GST_AUDIO_FORMAT_F32 ||
format == GST_AUDIO_FORMAT_F64, NULL);
g_return_val_if_fail (channels > 0, NULL);
g_return_val_if_fail (in_rate > 0, NULL);
g_return_val_if_fail (out_rate > 0, NULL);
audio_resampler_init ();
resampler = g_slice_new0 (GstAudioResampler);
resampler->method = method;
resampler->flags = flags;
resampler->format = format;
resampler->channels = channels;
switch (format) {
case GST_AUDIO_FORMAT_S16:
resampler->format_index = 0;
break;
case GST_AUDIO_FORMAT_S32:
resampler->format_index = 1;
break;
case GST_AUDIO_FORMAT_F32:
resampler->format_index = 2;
break;
case GST_AUDIO_FORMAT_F64:
resampler->format_index = 3;
break;
default:
g_assert_not_reached ();
break;
}
info = gst_audio_format_get_info (format);
resampler->bps = GST_AUDIO_FORMAT_INFO_WIDTH (info) / 8;
resampler->sbuf = g_malloc0 (sizeof (gpointer) * channels);
non_interleaved_in =
(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_IN);
non_interleaved_out =
(resampler->flags & GST_AUDIO_RESAMPLER_FLAG_NON_INTERLEAVED_OUT);
/* we resample each channel separately */
resampler->blocks = resampler->channels;
resampler->inc = 1;
resampler->ostride = non_interleaved_out ? 1 : resampler->channels;
resampler->deinterleave = non_interleaved_in ?
copy_func : deinterleave_funcs[resampler->format_index];
resampler->convert_taps = convert_taps_funcs[resampler->format_index];
GST_DEBUG ("method %d, bps %d, channels %d", method, resampler->bps,
resampler->channels);
if (options == NULL) {
options = def_options =
gst_structure_new_empty ("GstAudioResampler.options");
gst_audio_resampler_options_set_quality (DEFAULT_RESAMPLER_METHOD,
GST_AUDIO_RESAMPLER_QUALITY_DEFAULT, in_rate, out_rate, options);
}
gst_audio_resampler_update (resampler, in_rate, out_rate, options);
gst_audio_resampler_reset (resampler);
if (def_options)
gst_structure_free (def_options);
return resampler;
}
/* make the buffers to hold the (deinterleaved) samples */
static inline gpointer *
get_sample_bufs (GstAudioResampler * resampler, gsize need)
{
if (G_LIKELY (resampler->samples_len < need)) {
gint c, blocks = resampler->blocks;
gsize bytes, to_move = 0;
gint8 *ptr, *samples;
GST_LOG ("realloc %d -> %d", (gint) resampler->samples_len, (gint) need);
bytes = GST_ROUND_UP_N (need * resampler->bps * resampler->inc, ALIGN);
samples = g_malloc0 (blocks * bytes + ALIGN - 1);
ptr = MEM_ALIGN (samples, ALIGN);
/* if we had some data, move history */
if (resampler->samples_len > 0)
to_move = resampler->samples_avail * resampler->bps * resampler->inc;
/* set up new pointers */
for (c = 0; c < blocks; c++) {
memcpy (ptr + (c * bytes), resampler->sbuf[c], to_move);
resampler->sbuf[c] = ptr + (c * bytes);
}
g_free (resampler->samples);
resampler->samples = samples;
resampler->samples_len = need;
}
return resampler->sbuf;
}
/**
* gst_audio_resampler_reset:
* @resampler: a #GstAudioResampler
*
* Reset @resampler to the state it was when it was first created, discarding
* all sample history.
*/
void
gst_audio_resampler_reset (GstAudioResampler * resampler)
{
g_return_if_fail (resampler != NULL);
if (resampler->samples) {
gsize bytes;
gint c, blocks, bpf;
bpf = resampler->bps * resampler->inc;
bytes = (resampler->n_taps / 2) * bpf;
blocks = resampler->blocks;
for (c = 0; c < blocks; c++)
memset (resampler->sbuf[c], 0, bytes);
}
/* half of the filter is filled with 0 */
resampler->samp_index = 0;
resampler->samples_avail = resampler->n_taps / 2 - 1;
}
/**
* gst_audio_resampler_update:
* @resampler: a #GstAudioResampler
* @in_rate: new input rate
* @out_rate: new output rate
* @options: new options or %NULL
*
* Update the resampler parameters for @resampler. This function should
* not be called concurrently with any other function on @resampler.
*
* When @in_rate or @out_rate is 0, its value is unchanged.
*
* When @options is %NULL, the previously configured options are reused.
*
* Returns: %TRUE if the new parameters could be set
*/
gboolean
gst_audio_resampler_update (GstAudioResampler * resampler,
gint in_rate, gint out_rate, GstStructure * options)
{
gint gcd, samp_phase, old_n_taps;
gdouble max_error;
g_return_val_if_fail (resampler != NULL, FALSE);
if (in_rate <= 0)
in_rate = resampler->in_rate;
if (out_rate <= 0)
out_rate = resampler->out_rate;
if (resampler->out_rate > 0) {
GST_INFO ("old phase %d/%d", resampler->samp_phase, resampler->out_rate);
samp_phase =
gst_util_uint64_scale_int (resampler->samp_phase, out_rate,
resampler->out_rate);
} else
samp_phase = 0;
gcd = gst_util_greatest_common_divisor (in_rate, out_rate);
max_error = GET_OPT_MAX_PHASE_ERROR (resampler->options);
if (max_error < 1.0e-8) {
GST_INFO ("using exact phase divider");
gcd = gst_util_greatest_common_divisor (gcd, samp_phase);
} else {
while (gcd > 1) {
gdouble ph1 = (gdouble) samp_phase / out_rate;
gint factor = 2;
/* reduce the factor until we have a phase error of less than 10% */
gdouble ph2 = (gdouble) (samp_phase / gcd) / (out_rate / gcd);
if (fabs (ph1 - ph2) < max_error)
break;
while (gcd % factor != 0)
factor++;
gcd /= factor;
GST_INFO ("divide by factor %d, gcd %d", factor, gcd);
}
}
GST_INFO ("phase %d out_rate %d, in_rate %d, gcd %d", samp_phase, out_rate,
in_rate, gcd);
resampler->samp_phase = samp_phase /= gcd;
resampler->in_rate = in_rate /= gcd;
resampler->out_rate = out_rate /= gcd;
GST_INFO ("new phase %d/%d", resampler->samp_phase, resampler->out_rate);
resampler->samp_inc = in_rate / out_rate;
resampler->samp_frac = in_rate % out_rate;
if (options) {
GST_INFO ("have new options, reconfigure filter");
if (resampler->options)
gst_structure_free (resampler->options);
resampler->options = gst_structure_copy (options);
old_n_taps = resampler->n_taps;
resampler_calculate_taps (resampler);
resampler_dump (resampler);
if (old_n_taps > 0 && old_n_taps != resampler->n_taps) {
gpointer *sbuf;
gint i, bpf, bytes, soff, doff, diff;
sbuf = get_sample_bufs (resampler, resampler->n_taps);
bpf = resampler->bps * resampler->inc;
bytes = resampler->samples_avail * bpf;
soff = doff = resampler->samp_index * bpf;
diff = ((gint) resampler->n_taps - old_n_taps) / 2;
GST_DEBUG ("taps %d->%d, %d", old_n_taps, resampler->n_taps, diff);
if (diff < 0) {
/* diff < 0, decrease taps, adjust source */
soff += -diff * bpf;
bytes -= -diff * bpf;
} else {
/* diff > 0, increase taps, adjust dest */
doff += diff * bpf;
}
/* now shrink or enlarge the history buffer, when we enlarge we
* just leave the old samples in there. FIXME, probably do something better
* like mirror or fill with zeroes. */
for (i = 0; i < resampler->blocks; i++)
memmove ((gint8 *) sbuf[i] + doff, (gint8 *) sbuf[i] + soff, bytes);
resampler->samples_avail += diff;
}
} else if (resampler->filter_mode == GST_AUDIO_RESAMPLER_FILTER_MODE_FULL) {
GST_DEBUG ("setting up filter cache");
resampler->n_phases = resampler->out_rate;
alloc_cache_mem (resampler, resampler->bps, resampler->n_taps,
resampler->n_phases);
}
setup_functions (resampler);
return TRUE;
}
/**
* gst_audio_resampler_free:
* @resampler: a #GstAudioResampler
*
* Free a previously allocated #GstAudioResampler @resampler.
*
* Since: 1.6
*/
void
gst_audio_resampler_free (GstAudioResampler * resampler)
{
g_return_if_fail (resampler != NULL);
g_free (resampler->cached_taps_mem);
g_free (resampler->taps_mem);
g_free (resampler->tmp_taps);
g_free (resampler->samples);
g_free (resampler->sbuf);
if (resampler->options)
gst_structure_free (resampler->options);
g_slice_free (GstAudioResampler, resampler);
}
/**
* gst_audio_resampler_get_out_frames:
* @resampler: a #GstAudioResampler
* @in_frames: number of input frames
*
* Get the number of output frames that would be currently available when
* @in_frames are given to @resampler.
*
* Returns: The number of frames that would be available after giving
* @in_frames as input to @resampler.
*/
gsize
gst_audio_resampler_get_out_frames (GstAudioResampler * resampler,
gsize in_frames)
{
gsize need, avail, out;
g_return_val_if_fail (resampler != NULL, 0);
need = resampler->n_taps + resampler->samp_index + resampler->skip;
avail = resampler->samples_avail + in_frames;
GST_LOG ("need %d = %d + %d + %d, avail %d = %d + %d", (gint) need,
resampler->n_taps, resampler->samp_index, resampler->skip,
(gint) avail, (gint) resampler->samples_avail, (gint) in_frames);
if (avail < need)
return 0;
out = (avail - need) * resampler->out_rate;
if (out < resampler->samp_phase)
return 0;
out = ((out - resampler->samp_phase) / resampler->in_rate) + 1;
GST_LOG ("out %d = ((%d * %d - %d) / %d) + 1", (gint) out,
(gint) (avail - need), resampler->out_rate, resampler->samp_phase,
resampler->in_rate);
return out;
}
/**
* gst_audio_resampler_get_in_frames:
* @resampler: a #GstAudioResampler
* @out_frames: number of input frames
*
* Get the number of input frames that would currently be needed
* to produce @out_frames from @resampler.
*
* Returns: The number of input frames needed for producing
* @out_frames of data from @resampler.
*/
gsize
gst_audio_resampler_get_in_frames (GstAudioResampler * resampler,
gsize out_frames)
{
gsize in_frames;
g_return_val_if_fail (resampler != NULL, 0);
in_frames =
(resampler->samp_phase +
out_frames * resampler->samp_frac) / resampler->out_rate;
in_frames += out_frames * resampler->samp_inc;
return in_frames;
}
/**
* gst_audio_resampler_get_max_latency:
* @resampler: a #GstAudioResampler
*
* Get the maximum number of input samples that the resampler would
* need before producing output.
*
* Returns: the latency of @resampler as expressed in the number of
* frames.
*/
gsize
gst_audio_resampler_get_max_latency (GstAudioResampler * resampler)
{
g_return_val_if_fail (resampler != NULL, 0);
return resampler->n_taps / 2;
}
/**
* gst_audio_resampler_resample:
* @resampler: a #GstAudioResampler
* @in: input samples
* @in_frames: number of input frames
* @out: output samples
* @out_frames: number of output frames
*
* Perform resampling on @in_frames frames in @in and write @out_frames to @out.
*
* In case the samples are interleaved, @in and @out must point to an
* array with a single element pointing to a block of interleaved samples.
*
* If non-interleaved samples are used, @in and @out must point to an
* array with pointers to memory blocks, one for each channel.
*
* @in may be %NULL, in which case @in_frames of silence samples are pushed
* into the resampler.
*
* This function always produces @out_frames of output and consumes @in_frames of
* input. Use gst_audio_resampler_get_out_frames() and
* gst_audio_resampler_get_in_frames() to make sure @in_frames and @out_frames
* are matching and @in and @out point to enough memory.
*/
void
gst_audio_resampler_resample (GstAudioResampler * resampler,
gpointer in[], gsize in_frames, gpointer out[], gsize out_frames)
{
gsize samples_avail;
gsize need, consumed;
gpointer *sbuf;
/* do sample skipping */
if (G_UNLIKELY (resampler->skip >= in_frames)) {
/* we need tp skip all input */
resampler->skip -= in_frames;
return;
}
/* skip the last samples by advancing the sample index */
resampler->samp_index += resampler->skip;
samples_avail = resampler->samples_avail;
/* make sure we have enough space to copy our samples */
sbuf = get_sample_bufs (resampler, in_frames + samples_avail);
/* copy/deinterleave the samples */
resampler->deinterleave (resampler, sbuf, in, in_frames);
/* update new amount of samples in our buffer */
resampler->samples_avail = samples_avail += in_frames;
need = resampler->n_taps + resampler->samp_index;
if (G_UNLIKELY (samples_avail < need)) {
/* not enough samples to start */
return;
}
/* resample all channels */
resampler->resample (resampler, sbuf, samples_avail, out, out_frames,
&consumed);
GST_LOG ("in %" G_GSIZE_FORMAT ", avail %" G_GSIZE_FORMAT ", consumed %"
G_GSIZE_FORMAT, in_frames, samples_avail, consumed);
/* update pointers */
if (G_LIKELY (consumed > 0)) {
gssize left = samples_avail - consumed;
if (left > 0) {
/* we consumed part of our samples */
resampler->samples_avail = left;
} else {
/* we consumed all our samples, empty our buffers */
resampler->samples_avail = 0;
resampler->skip = -left;
}
}
}
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