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FFmpeg/libavfilter/af_biquads.c

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/*
* Copyright (c) 2013 Paul B Mahol
* Copyright (c) 2006-2008 Rob Sykes <robs@users.sourceforge.net>
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* 2-pole filters designed by Robert Bristow-Johnson <rbj@audioimagination.com>
* see http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
*
* 1-pole filters based on code (c) 2000 Chris Bagwell <cbagwell@sprynet.com>
* Algorithms: Recursive single pole low/high pass filter
* Reference: The Scientist and Engineer's Guide to Digital Signal Processing
*
* low-pass: output[N] = input[N] * A + output[N-1] * B
* X = exp(-2.0 * pi * Fc)
* A = 1 - X
* B = X
* Fc = cutoff freq / sample rate
*
* Mimics an RC low-pass filter:
*
* ---/\/\/\/\----------->
* |
* --- C
* ---
* |
* |
* V
*
* high-pass: output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
* X = exp(-2.0 * pi * Fc)
* A0 = (1 + X) / 2
* A1 = -(1 + X) / 2
* B1 = X
* Fc = cutoff freq / sample rate
*
* Mimics an RC high-pass filter:
*
* || C
* ----||--------->
* || |
* <
* > R
* <
* |
* V
*/
#include "config_components.h"
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
enum FilterType {
biquad,
equalizer,
bass,
treble,
bandpass,
bandreject,
allpass,
highpass,
lowpass,
lowshelf,
highshelf,
2022-05-25 11:13:32 +02:00
tiltshelf,
};
enum WidthType {
NONE,
HERTZ,
OCTAVE,
QFACTOR,
SLOPE,
KHERTZ,
NB_WTYPE,
};
enum TransformType {
DI,
DII,
TDI,
TDII,
LATT,
SVF,
ZDF,
NB_TTYPE,
};
typedef struct ChanCache {
double i1, i2;
double o1, o2;
double ri1, ri2;
double ro1, ro2;
int clippings;
} ChanCache;
typedef struct BiquadsContext {
const AVClass *class;
enum FilterType filter_type;
int width_type;
int poles;
int csg;
int transform_type;
int precision;
int block_samples;
int bypass;
double gain;
double frequency;
double width;
double mix;
char *ch_layout_str;
AVChannelLayout ch_layout;
int normalize;
int order;
double a0, a1, a2;
double b0, b1, b2;
double oa0, oa1, oa2;
double ob0, ob1, ob2;
AVFrame *block[3];
ChanCache *cache;
int block_align;
int64_t pts;
int nb_samples;
void (*filter)(struct BiquadsContext *s, const void *ibuf, void *obuf, int len,
double *i1, double *i2, double *o1, double *o2,
double b0, double b1, double b2, double a0, double a1, double a2, int *clippings,
int disabled);
} BiquadsContext;
static int query_formats(AVFilterContext *ctx)
{
BiquadsContext *s = ctx->priv;
static const enum AVSampleFormat auto_sample_fmts[] = {
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S32P,
AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE
};
const enum AVSampleFormat *sample_fmts_list = sample_fmts;
int ret = ff_set_common_all_channel_counts(ctx);
if (ret < 0)
return ret;
switch (s->precision) {
case 0:
sample_fmts[0] = AV_SAMPLE_FMT_S16P;
break;
case 1:
sample_fmts[0] = AV_SAMPLE_FMT_S32P;
break;
case 2:
sample_fmts[0] = AV_SAMPLE_FMT_FLTP;
break;
case 3:
sample_fmts[0] = AV_SAMPLE_FMT_DBLP;
break;
default:
sample_fmts_list = auto_sample_fmts;
break;
}
ret = ff_set_common_formats_from_list(ctx, sample_fmts_list);
if (ret < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
#define BIQUAD_FILTER(name, type, min, max, need_clipping) \
static void biquad_## name (BiquadsContext *s, \
const void *input, void *output, int len, \
double *in1, double *in2, \
double *out1, double *out2, \
double b0, double b1, double b2, \
double a0, double a1, double a2, int *clippings, \
int disabled) \
{ \
const type *ibuf = input; \
type *obuf = output; \
double i1 = *in1; \
double i2 = *in2; \
double o1 = *out1; \
double o2 = *out2; \
double wet = s->mix; \
double dry = 1. - wet; \
double out; \
int i; \
a1 = -a1; \
a2 = -a2; \
\
for (i = 0; i+1 < len; i++) { \
o2 = i2 * b2 + i1 * b1 + ibuf[i] * b0 + o2 * a2 + o1 * a1; \
i2 = ibuf[i]; \
out = o2 * wet + i2 * dry; \
if (disabled) { \
obuf[i] = i2; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
i++; \
o1 = i1 * b2 + i2 * b1 + ibuf[i] * b0 + o1 * a2 + o2 * a1; \
i1 = ibuf[i]; \
out = o1 * wet + i1 * dry; \
if (disabled) { \
obuf[i] = i1; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
if (i < len) { \
double o0 = ibuf[i] * b0 + i1 * b1 + i2 * b2 + o1 * a1 + o2 * a2; \
i2 = i1; \
i1 = ibuf[i]; \
o2 = o1; \
o1 = o0; \
out = o0 * wet + i1 * dry; \
if (disabled) { \
obuf[i] = i1; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
*in1 = i1; \
*in2 = i2; \
*out1 = o1; \
*out2 = o2; \
}
BIQUAD_FILTER(s16, int16_t, INT16_MIN, INT16_MAX, 1)
BIQUAD_FILTER(s32, int32_t, INT32_MIN, INT32_MAX, 1)
BIQUAD_FILTER(flt, float, -1., 1., 0)
BIQUAD_FILTER(dbl, double, -1., 1., 0)
#define BIQUAD_DII_FILTER(name, type, min, max, need_clipping) \
static void biquad_dii_## name (BiquadsContext *s, \
const void *input, void *output, int len, \
double *z1, double *z2, \
double *unused1, double *unused2, \
double b0, double b1, double b2, \
double a0, double a1, double a2, int *clippings, \
int disabled) \
{ \
const type *ibuf = input; \
type *obuf = output; \
double w1 = *z1; \
double w2 = *z2; \
double wet = s->mix; \
double dry = 1. - wet; \
double in, out, w0; \
\
a1 = -a1; \
a2 = -a2; \
\
for (int i = 0; i < len; i++) { \
in = ibuf[i]; \
w0 = in + a1 * w1 + a2 * w2; \
out = b0 * w0 + b1 * w1 + b2 * w2; \
w2 = w1; \
w1 = w0; \
out = out * wet + in * dry; \
if (disabled) { \
obuf[i] = in; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
*z1 = w1; \
*z2 = w2; \
}
BIQUAD_DII_FILTER(s16, int16_t, INT16_MIN, INT16_MAX, 1)
BIQUAD_DII_FILTER(s32, int32_t, INT32_MIN, INT32_MAX, 1)
BIQUAD_DII_FILTER(flt, float, -1., 1., 0)
BIQUAD_DII_FILTER(dbl, double, -1., 1., 0)
#define BIQUAD_TDI_FILTER(name, type, min, max, need_clipping) \
static void biquad_tdi_## name (BiquadsContext *s, \
const void *input, void *output, int len, \
double *z1, double *z2, \
double *z3, double *z4, \
double b0, double b1, double b2, \
double a0, double a1, double a2, int *clippings, \
int disabled) \
{ \
const type *ibuf = input; \
type *obuf = output; \
double s1 = *z1; \
double s2 = *z2; \
double s3 = *z3; \
double s4 = *z4; \
double wet = s->mix; \
double dry = 1. - wet; \
double in, out; \
\
a1 = -a1; \
a2 = -a2; \
\
for (int i = 0; i < len; i++) { \
double t1, t2, t3, t4; \
in = ibuf[i] + s1; \
t1 = in * a1 + s2; \
t2 = in * a2; \
t3 = in * b1 + s4; \
t4 = in * b2; \
out = b0 * in + s3; \
out = out * wet + in * dry; \
s1 = t1; s2 = t2; s3 = t3; s4 = t4; \
if (disabled) { \
obuf[i] = in; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
\
*z1 = s1; \
*z2 = s2; \
*z3 = s3; \
*z4 = s4; \
}
BIQUAD_TDI_FILTER(s16, int16_t, INT16_MIN, INT16_MAX, 1)
BIQUAD_TDI_FILTER(s32, int32_t, INT32_MIN, INT32_MAX, 1)
BIQUAD_TDI_FILTER(flt, float, -1., 1., 0)
BIQUAD_TDI_FILTER(dbl, double, -1., 1., 0)
#define BIQUAD_TDII_FILTER(name, type, min, max, need_clipping) \
static void biquad_tdii_## name (BiquadsContext *s, \
const void *input, void *output, int len, \
double *z1, double *z2, \
double *unused1, double *unused2, \
double b0, double b1, double b2, \
double a0, double a1, double a2, int *clippings, \
int disabled) \
{ \
const type *ibuf = input; \
type *obuf = output; \
double w1 = *z1; \
double w2 = *z2; \
double wet = s->mix; \
double dry = 1. - wet; \
double in, out; \
\
a1 = -a1; \
a2 = -a2; \
\
for (int i = 0; i < len; i++) { \
in = ibuf[i]; \
out = b0 * in + w1; \
w1 = b1 * in + w2 + a1 * out; \
w2 = b2 * in + a2 * out; \
out = out * wet + in * dry; \
if (disabled) { \
obuf[i] = in; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
*z1 = w1; \
*z2 = w2; \
}
BIQUAD_TDII_FILTER(s16, int16_t, INT16_MIN, INT16_MAX, 1)
BIQUAD_TDII_FILTER(s32, int32_t, INT32_MIN, INT32_MAX, 1)
BIQUAD_TDII_FILTER(flt, float, -1., 1., 0)
BIQUAD_TDII_FILTER(dbl, double, -1., 1., 0)
#define BIQUAD_LATT_FILTER(name, type, min, max, need_clipping) \
static void biquad_latt_## name (BiquadsContext *s, \
const void *input, void *output, int len, \
double *z1, double *z2, \
double *unused1, double *unused2, \
double v0, double v1, double v2, \
double unused, double k0, double k1, \
int *clippings, \
int disabled) \
{ \
const type *ibuf = input; \
type *obuf = output; \
double s0 = *z1; \
double s1 = *z2; \
double wet = s->mix; \
double dry = 1. - wet; \
double in, out; \
double t0, t1; \
\
for (int i = 0; i < len; i++) { \
out = 0.; \
in = ibuf[i]; \
t0 = in - k1 * s0; \
t1 = t0 * k1 + s0; \
out += t1 * v2; \
\
t0 = t0 - k0 * s1; \
t1 = t0 * k0 + s1; \
out += t1 * v1; \
\
out += t0 * v0; \
s0 = t1; \
s1 = t0; \
\
out = out * wet + in * dry; \
if (disabled) { \
obuf[i] = in; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
*z1 = s0; \
*z2 = s1; \
}
BIQUAD_LATT_FILTER(s16, int16_t, INT16_MIN, INT16_MAX, 1)
BIQUAD_LATT_FILTER(s32, int32_t, INT32_MIN, INT32_MAX, 1)
BIQUAD_LATT_FILTER(flt, float, -1., 1., 0)
BIQUAD_LATT_FILTER(dbl, double, -1., 1., 0)
#define BIQUAD_SVF_FILTER(name, type, min, max, need_clipping) \
static void biquad_svf_## name (BiquadsContext *s, \
const void *input, void *output, int len, \
double *y0, double *y1, \
double *unused1, double *unused2, \
double b0, double b1, double b2, \
double a0, double a1, double a2, int *clippings, \
int disabled) \
{ \
const type *ibuf = input; \
type *obuf = output; \
double s0 = *y0; \
double s1 = *y1; \
double wet = s->mix; \
double dry = 1. - wet; \
double in, out; \
double t0, t1; \
\
for (int i = 0; i < len; i++) { \
in = ibuf[i]; \
out = b2 * in + s0; \
t0 = b0 * in + a1 * s0 + s1; \
t1 = b1 * in + a2 * s0; \
s0 = t0; \
s1 = t1; \
\
out = out * wet + in * dry; \
if (disabled) { \
obuf[i] = in; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
*y0 = s0; \
*y1 = s1; \
}
BIQUAD_SVF_FILTER(s16, int16_t, INT16_MIN, INT16_MAX, 1)
BIQUAD_SVF_FILTER(s32, int32_t, INT32_MIN, INT32_MAX, 1)
BIQUAD_SVF_FILTER(flt, float, -1., 1., 0)
BIQUAD_SVF_FILTER(dbl, double, -1., 1., 0)
#define BIQUAD_ZDF_FILTER(name, type, min, max, need_clipping) \
static void biquad_zdf_## name (BiquadsContext *s, \
const void *input, void *output, int len, \
double *y0, double *y1, \
double *unused1, double *unused2, \
double m0, double m1, double m2, \
double a0, double a1, double a2, int *clippings, \
int disabled) \
{ \
const type *ibuf = input; \
type *obuf = output; \
double b0 = *y0; \
double b1 = *y1; \
double wet = s->mix; \
double dry = 1. - wet; \
double out; \
\
for (int i = 0; i < len; i++) { \
const double in = ibuf[i]; \
const double v0 = in; \
const double v3 = v0 - b1; \
const double v1 = a0 * b0 + a1 * v3; \
const double v2 = b1 + a1 * b0 + a2 * v3; \
\
b0 = 2. * v1 - b0; \
b1 = 2. * v2 - b1; \
\
out = m0 * v0 + m1 * v1 + m2 * v2; \
out = out * wet + in * dry; \
if (disabled) { \
obuf[i] = in; \
} else if (need_clipping && out < min) { \
(*clippings)++; \
obuf[i] = min; \
} else if (need_clipping && out > max) { \
(*clippings)++; \
obuf[i] = max; \
} else { \
obuf[i] = out; \
} \
} \
*y0 = b0; \
*y1 = b1; \
}
BIQUAD_ZDF_FILTER(s16, int16_t, INT16_MIN, INT16_MAX, 1)
BIQUAD_ZDF_FILTER(s32, int32_t, INT32_MIN, INT32_MAX, 1)
BIQUAD_ZDF_FILTER(flt, float, -1., 1., 0)
BIQUAD_ZDF_FILTER(dbl, double, -1., 1., 0)
static void convert_dir2latt(BiquadsContext *s)
{
double k0, k1, v0, v1, v2;
k1 = s->a2;
k0 = s->a1 / (1. + k1);
v2 = s->b2;
v1 = s->b1 - v2 * s->a1;
v0 = s->b0 - v1 * k0 - v2 * k1;
s->a1 = k0;
s->a2 = k1;
s->b0 = v0;
s->b1 = v1;
s->b2 = v2;
}
static void convert_dir2svf(BiquadsContext *s)
{
double a[2];
double b[3];
a[0] = -s->a1;
a[1] = -s->a2;
b[0] = s->b1 - s->a1 * s->b0;
b[1] = s->b2 - s->a2 * s->b0;
b[2] = s->b0;
s->a1 = a[0];
s->a2 = a[1];
s->b0 = b[0];
s->b1 = b[1];
s->b2 = b[2];
}
static double convert_width2qfactor(double width,
double frequency,
double gain,
double sample_rate,
int width_type)
{
double w0 = 2. * M_PI * frequency / sample_rate;
double A = ff_exp10(gain / 40.);
double ret;
switch (width_type) {
case NONE:
case QFACTOR:
ret = width;
break;
case HERTZ:
ret = frequency / width;
break;
case KHERTZ:
ret = frequency / (width * 1000.);
break;
case OCTAVE:
ret = 1. / (2. * sinh(log(2.) / 2. * width * w0 / sin(w0)));
break;
case SLOPE:
ret = 1. / sqrt((A + 1. / A) * (1. / width - 1.) + 2.);
break;
default:
av_assert0(0);
break;
}
return ret;
}
static void convert_dir2zdf(BiquadsContext *s, int sample_rate)
{
double Q = convert_width2qfactor(s->width, s->frequency, s->gain, sample_rate, s->width_type);
double g, k, A;
double a[3];
double m[3];
switch (s->filter_type) {
case biquad:
a[0] = s->oa0;
a[1] = s->oa1;
a[2] = s->oa2;
m[0] = s->ob0;
m[1] = s->ob1;
m[2] = s->ob2;
break;
case equalizer:
A = ff_exp10(s->gain / 40.);
g = tan(M_PI * s->frequency / sample_rate);
k = 1. / (Q * A);
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 1.;
m[1] = k * (A * A - 1.);
m[2] = 0.;
break;
case bass:
case lowshelf:
A = ff_exp10(s->gain / 40.);
g = tan(M_PI * s->frequency / sample_rate) / sqrt(A);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 1.;
m[1] = k * (A - 1.);
m[2] = A * A - 1.;
break;
2022-05-25 11:13:32 +02:00
case tiltshelf:
A = ff_exp10(s->gain / 20.);
g = tan(M_PI * s->frequency / sample_rate) / sqrt(A);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 1./ A;
m[1] = k * (A - 1.) / A;
m[2] = (A * A - 1.) / A;
break;
case treble:
case highshelf:
A = ff_exp10(s->gain / 40.);
g = tan(M_PI * s->frequency / sample_rate) / sqrt(A);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = A * A;
m[1] = k * (1. - A) * A;
m[2] = 1. - A * A;
break;
case bandpass:
g = tan(M_PI * s->frequency / sample_rate);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 0.;
m[1] = s->csg ? 1. : 2.;
m[2] = 0.;
break;
case bandreject:
g = tan(M_PI * s->frequency / sample_rate);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 1.;
m[1] = -k;
m[2] = 0.;
break;
case lowpass:
g = tan(M_PI * s->frequency / sample_rate);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 0.;
m[1] = 0.;
m[2] = 1.;
break;
case highpass:
g = tan(M_PI * s->frequency / sample_rate);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 1.;
m[1] = -k;
m[2] = -1.;
break;
case allpass:
g = tan(M_PI * s->frequency / sample_rate);
k = 1. / Q;
a[0] = 1. / (1. + g * (g + k));
a[1] = g * a[0];
a[2] = g * a[1];
m[0] = 1.;
m[1] = -2. * k;
m[2] = 0.;
break;
default:
av_assert0(0);
}
s->a0 = a[0];
s->a1 = a[1];
s->a2 = a[2];
s->b0 = m[0];
s->b1 = m[1];
s->b2 = m[2];
}
static int config_filter(AVFilterLink *outlink, int reset)
{
AVFilterContext *ctx = outlink->src;
BiquadsContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
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double gain = s->gain * ((s->filter_type == tiltshelf) + 1.);
double A = ff_exp10(gain / 40);
double w0 = 2 * M_PI * s->frequency / inlink->sample_rate;
double K = tan(w0 / 2.);
double alpha, beta;
s->bypass = (((w0 > M_PI || w0 <= 0.) && reset) || (s->width <= 0.)) && (s->filter_type != biquad);
if (s->bypass) {
av_log(ctx, AV_LOG_WARNING, "Invalid frequency and/or width!\n");
return 0;
}
if ((w0 > M_PI || w0 <= 0.) && (s->filter_type != biquad))
return AVERROR(EINVAL);
switch (s->width_type) {
case NONE:
alpha = 0.0;
break;
case HERTZ:
alpha = sin(w0) / (2 * s->frequency / s->width);
break;
case KHERTZ:
alpha = sin(w0) / (2 * s->frequency / (s->width * 1000));
break;
case OCTAVE:
alpha = sin(w0) * sinh(log(2.) / 2 * s->width * w0 / sin(w0));
break;
case QFACTOR:
alpha = sin(w0) / (2 * s->width);
break;
case SLOPE:
alpha = sin(w0) / 2 * sqrt((A + 1 / A) * (1 / s->width - 1) + 2);
break;
default:
av_assert0(0);
}
beta = 2 * sqrt(A);
switch (s->filter_type) {
case biquad:
s->a0 = s->oa0;
s->a1 = s->oa1;
s->a2 = s->oa2;
s->b0 = s->ob0;
s->b1 = s->ob1;
s->b2 = s->ob2;
break;
case equalizer:
s->a0 = 1 + alpha / A;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha / A;
s->b0 = 1 + alpha * A;
s->b1 = -2 * cos(w0);
s->b2 = 1 - alpha * A;
break;
case bass:
beta = sqrt((A * A + 1) - (A - 1) * (A - 1));
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case tiltshelf:
case lowshelf:
if (s->poles == 1) {
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double A = ff_exp10(gain / 20);
double ro = -sin(w0 / 2. - M_PI_4) / sin(w0 / 2. + M_PI_4);
double n = (A + 1) / (A - 1);
double alpha1 = A == 1. ? 0. : n - FFSIGN(n) * sqrt(n * n - 1);
double beta0 = ((1 + A) + (1 - A) * alpha1) * 0.5;
double beta1 = ((1 - A) + (1 + A) * alpha1) * 0.5;
s->a0 = 1 + ro * alpha1;
s->a1 = -ro - alpha1;
s->a2 = 0;
s->b0 = beta0 + ro * beta1;
s->b1 = -beta1 - ro * beta0;
s->b2 = 0;
} else {
s->a0 = (A + 1) + (A - 1) * cos(w0) + beta * alpha;
s->a1 = -2 * ((A - 1) + (A + 1) * cos(w0));
s->a2 = (A + 1) + (A - 1) * cos(w0) - beta * alpha;
s->b0 = A * ((A + 1) - (A - 1) * cos(w0) + beta * alpha);
s->b1 = 2 * A * ((A - 1) - (A + 1) * cos(w0));
s->b2 = A * ((A + 1) - (A - 1) * cos(w0) - beta * alpha);
}
break;
case treble:
beta = sqrt((A * A + 1) - (A - 1) * (A - 1));
case highshelf:
if (s->poles == 1) {
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double A = ff_exp10(gain / 20);
double ro = sin(w0 / 2. - M_PI_4) / sin(w0 / 2. + M_PI_4);
double n = (A + 1) / (A - 1);
double alpha1 = A == 1. ? 0. : n - FFSIGN(n) * sqrt(n * n - 1);
double beta0 = ((1 + A) + (1 - A) * alpha1) * 0.5;
double beta1 = ((1 - A) + (1 + A) * alpha1) * 0.5;
s->a0 = 1 + ro * alpha1;
s->a1 = ro + alpha1;
s->a2 = 0;
s->b0 = beta0 + ro * beta1;
s->b1 = beta1 + ro * beta0;
s->b2 = 0;
} else {
s->a0 = (A + 1) - (A - 1) * cos(w0) + beta * alpha;
s->a1 = 2 * ((A - 1) - (A + 1) * cos(w0));
s->a2 = (A + 1) - (A - 1) * cos(w0) - beta * alpha;
s->b0 = A * ((A + 1) + (A - 1) * cos(w0) + beta * alpha);
s->b1 =-2 * A * ((A - 1) + (A + 1) * cos(w0));
s->b2 = A * ((A + 1) + (A - 1) * cos(w0) - beta * alpha);
}
break;
case bandpass:
if (s->csg) {
s->a0 = 1 + alpha;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha;
s->b0 = sin(w0) / 2;
s->b1 = 0;
s->b2 = -sin(w0) / 2;
} else {
s->a0 = 1 + alpha;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha;
s->b0 = alpha;
s->b1 = 0;
s->b2 = -alpha;
}
break;
case bandreject:
s->a0 = 1 + alpha;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha;
s->b0 = 1;
s->b1 = -2 * cos(w0);
s->b2 = 1;
break;
case lowpass:
if (s->poles == 1) {
s->a0 = 1;
s->a1 = -exp(-w0);
s->a2 = 0;
s->b0 = 1 + s->a1;
s->b1 = 0;
s->b2 = 0;
} else {
s->a0 = 1 + alpha;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha;
s->b0 = (1 - cos(w0)) / 2;
s->b1 = 1 - cos(w0);
s->b2 = (1 - cos(w0)) / 2;
}
break;
case highpass:
if (s->poles == 1) {
s->a0 = 1;
s->a1 = -exp(-w0);
s->a2 = 0;
s->b0 = (1 - s->a1) / 2;
s->b1 = -s->b0;
s->b2 = 0;
} else {
s->a0 = 1 + alpha;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha;
s->b0 = (1 + cos(w0)) / 2;
s->b1 = -(1 + cos(w0));
s->b2 = (1 + cos(w0)) / 2;
}
break;
case allpass:
switch (s->order) {
case 1:
s->a0 = 1.;
s->a1 = -(1. - K) / (1. + K);
s->a2 = 0.;
s->b0 = s->a1;
s->b1 = s->a0;
s->b2 = 0.;
break;
case 2:
s->a0 = 1 + alpha;
s->a1 = -2 * cos(w0);
s->a2 = 1 - alpha;
s->b0 = 1 - alpha;
s->b1 = -2 * cos(w0);
s->b2 = 1 + alpha;
break;
}
break;
default:
av_assert0(0);
}
av_log(ctx, AV_LOG_VERBOSE, "a=%f %f %f:b=%f %f %f\n", s->a0, s->a1, s->a2, s->b0, s->b1, s->b2);
s->a1 /= s->a0;
s->a2 /= s->a0;
s->b0 /= s->a0;
s->b1 /= s->a0;
s->b2 /= s->a0;
s->a0 /= s->a0;
if (s->normalize && fabs(s->b0 + s->b1 + s->b2) > 1e-6) {
double factor = (s->a0 + s->a1 + s->a2) / (s->b0 + s->b1 + s->b2);
s->b0 *= factor;
s->b1 *= factor;
s->b2 *= factor;
}
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switch (s->filter_type) {
case tiltshelf:
s->b0 /= A;
s->b1 /= A;
s->b2 /= A;
break;
}
s->cache = av_realloc_f(s->cache, sizeof(ChanCache), inlink->ch_layout.nb_channels);
if (!s->cache)
return AVERROR(ENOMEM);
if (reset)
memset(s->cache, 0, sizeof(ChanCache) * inlink->ch_layout.nb_channels);
if (reset && s->block_samples > 0) {
for (int i = 0; i < 3; i++) {
s->block[i] = ff_get_audio_buffer(outlink, s->block_samples * 2);
if (!s->block[i])
return AVERROR(ENOMEM);
av_samples_set_silence(s->block[i]->extended_data, 0, s->block_samples * 2,
s->block[i]->ch_layout.nb_channels, s->block[i]->format);
}
}
switch (s->transform_type) {
case DI:
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
s->filter = biquad_s16;
break;
case AV_SAMPLE_FMT_S32P:
s->filter = biquad_s32;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter = biquad_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter = biquad_dbl;
break;
default: av_assert0(0);
}
break;
case DII:
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
s->filter = biquad_dii_s16;
break;
case AV_SAMPLE_FMT_S32P:
s->filter = biquad_dii_s32;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter = biquad_dii_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter = biquad_dii_dbl;
break;
default: av_assert0(0);
}
break;
case TDI:
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
s->filter = biquad_tdi_s16;
break;
case AV_SAMPLE_FMT_S32P:
s->filter = biquad_tdi_s32;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter = biquad_tdi_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter = biquad_tdi_dbl;
break;
default: av_assert0(0);
}
break;
case TDII:
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
s->filter = biquad_tdii_s16;
break;
case AV_SAMPLE_FMT_S32P:
s->filter = biquad_tdii_s32;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter = biquad_tdii_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter = biquad_tdii_dbl;
break;
default: av_assert0(0);
}
break;
case LATT:
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
s->filter = biquad_latt_s16;
break;
case AV_SAMPLE_FMT_S32P:
s->filter = biquad_latt_s32;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter = biquad_latt_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter = biquad_latt_dbl;
break;
default: av_assert0(0);
}
break;
case SVF:
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
s->filter = biquad_svf_s16;
break;
case AV_SAMPLE_FMT_S32P:
s->filter = biquad_svf_s32;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter = biquad_svf_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter = biquad_svf_dbl;
break;
default: av_assert0(0);
}
break;
case ZDF:
switch (inlink->format) {
case AV_SAMPLE_FMT_S16P:
s->filter = biquad_zdf_s16;
break;
case AV_SAMPLE_FMT_S32P:
s->filter = biquad_zdf_s32;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter = biquad_zdf_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->filter = biquad_zdf_dbl;
break;
default: av_assert0(0);
}
break;
default:
av_assert0(0);
}
s->block_align = av_get_bytes_per_sample(inlink->format);
if (s->transform_type == LATT)
convert_dir2latt(s);
else if (s->transform_type == SVF)
convert_dir2svf(s);
else if (s->transform_type == ZDF)
convert_dir2zdf(s, inlink->sample_rate);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
return config_filter(outlink, 1);
}
typedef struct ThreadData {
AVFrame *in, *out;
int eof;
} ThreadData;
static void reverse_samples(AVFrame *out, AVFrame *in, int p,
int oo, int io, int nb_samples)
{
switch (out->format) {
case AV_SAMPLE_FMT_S16P: {
const int16_t *src = ((const int16_t *)in->extended_data[p]) + io;
int16_t *dst = ((int16_t *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
case AV_SAMPLE_FMT_S32P: {
const int32_t *src = ((const int32_t *)in->extended_data[p]) + io;
int32_t *dst = ((int32_t *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
case AV_SAMPLE_FMT_FLTP: {
const float *src = ((const float *)in->extended_data[p]) + io;
float *dst = ((float *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
case AV_SAMPLE_FMT_DBLP: {
const double *src = ((const double *)in->extended_data[p]) + io;
double *dst = ((double *)out->extended_data[p]) + oo;
for (int i = 0, j = nb_samples - 1; i < nb_samples; i++, j--)
dst[i] = src[j];
}
break;
}
}
static int filter_channel(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AVFilterLink *inlink = ctx->inputs[0];
ThreadData *td = arg;
AVFrame *buf = td->in;
AVFrame *out_buf = td->out;
BiquadsContext *s = ctx->priv;
const int start = (buf->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (buf->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
int ch;
for (ch = start; ch < end; ch++) {
enum AVChannel channel = av_channel_layout_channel_from_index(&inlink->ch_layout, ch);
if (av_channel_layout_index_from_channel(&s->ch_layout, channel) < 0) {
if (buf != out_buf)
memcpy(out_buf->extended_data[ch], buf->extended_data[ch],
buf->nb_samples * s->block_align);
continue;
}
if (!s->block_samples) {
s->filter(s, buf->extended_data[ch], out_buf->extended_data[ch], buf->nb_samples,
&s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2,
s->b0, s->b1, s->b2, s->a0, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
} else if (td->eof) {
memcpy(out_buf->extended_data[ch], s->block[1]->extended_data[ch] + s->block_align * s->block_samples,
s->nb_samples * s->block_align);
} else {
memcpy(s->block[0]->extended_data[ch] + s->block_align * s->block_samples, buf->extended_data[ch],
buf->nb_samples * s->block_align);
memset(s->block[0]->extended_data[ch] + s->block_align * (s->block_samples + buf->nb_samples),
0, (s->block_samples - buf->nb_samples) * s->block_align);
s->filter(s, s->block[0]->extended_data[ch], s->block[1]->extended_data[ch], s->block_samples,
&s->cache[ch].i1, &s->cache[ch].i2, &s->cache[ch].o1, &s->cache[ch].o2,
s->b0, s->b1, s->b2, s->a0, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
s->cache[ch].ri1 = s->cache[ch].i1;
s->cache[ch].ri2 = s->cache[ch].i2;
s->cache[ch].ro1 = s->cache[ch].o1;
s->cache[ch].ro2 = s->cache[ch].o2;
s->filter(s, s->block[0]->extended_data[ch] + s->block_samples * s->block_align,
s->block[1]->extended_data[ch] + s->block_samples * s->block_align,
s->block_samples,
&s->cache[ch].ri1, &s->cache[ch].ri2, &s->cache[ch].ro1, &s->cache[ch].ro2,
s->b0, s->b1, s->b2, s->a0, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
reverse_samples(s->block[2], s->block[1], ch, 0, 0, 2 * s->block_samples);
s->cache[ch].ri1 = 0.;
s->cache[ch].ri2 = 0.;
s->cache[ch].ro1 = 0.;
s->cache[ch].ro2 = 0.;
s->filter(s, s->block[2]->extended_data[ch], s->block[2]->extended_data[ch], 2 * s->block_samples,
&s->cache[ch].ri1, &s->cache[ch].ri2, &s->cache[ch].ro1, &s->cache[ch].ro2,
s->b0, s->b1, s->b2, s->a0, s->a1, s->a2, &s->cache[ch].clippings, ctx->is_disabled);
reverse_samples(s->block[1], s->block[2], ch, 0, 0, 2 * s->block_samples);
memcpy(out_buf->extended_data[ch], s->block[1]->extended_data[ch],
s->block_samples * s->block_align);
memmove(s->block[0]->extended_data[ch], s->block[0]->extended_data[ch] + s->block_align * s->block_samples,
s->block_samples * s->block_align);
}
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf, int eof)
{
AVFilterContext *ctx = inlink->dst;
BiquadsContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
Merge commit '7e350379f87e7f74420b4813170fe808e2313911' * commit '7e350379f87e7f74420b4813170fe808e2313911': lavfi: switch to AVFrame. Conflicts: doc/filters.texi libavfilter/af_ashowinfo.c libavfilter/audio.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/buffersink.c libavfilter/buffersrc.c libavfilter/buffersrc.h libavfilter/f_select.c libavfilter/f_setpts.c libavfilter/fifo.c libavfilter/split.c libavfilter/src_movie.c libavfilter/version.h libavfilter/vf_aspect.c libavfilter/vf_bbox.c libavfilter/vf_blackframe.c libavfilter/vf_delogo.c libavfilter/vf_drawbox.c libavfilter/vf_drawtext.c libavfilter/vf_fade.c libavfilter/vf_fieldorder.c libavfilter/vf_fps.c libavfilter/vf_frei0r.c libavfilter/vf_gradfun.c libavfilter/vf_hqdn3d.c libavfilter/vf_lut.c libavfilter/vf_overlay.c libavfilter/vf_pad.c libavfilter/vf_scale.c libavfilter/vf_showinfo.c libavfilter/vf_transpose.c libavfilter/vf_vflip.c libavfilter/vf_yadif.c libavfilter/video.c libavfilter/vsrc_testsrc.c libavfilter/yadif.h Following are notes about the merge authorship and various technical details. Michael Niedermayer: * Main merge operation, notably avfilter.c and video.c * Switch to AVFrame: - afade - anullsrc - apad - aresample - blackframe - deshake - idet - il - mandelbrot - mptestsrc - noise - setfield - smartblur - tinterlace * various merge changes and fixes in: - ashowinfo - blackdetect - field - fps - select - testsrc - yadif Nicolas George: * Switch to AVFrame: - make rawdec work with refcounted frames. Adapted from commit 759001c534287a96dc96d1e274665feb7059145d by Anton Khirnov. Also, fix the use of || instead of | in a flags check. - make buffer sink and src, audio and video work all together Clément Bœsch: * Switch to AVFrame: - aevalsrc - alphaextract - blend - cellauto - colormatrix - concat - earwax - ebur128 - edgedetect - geq - histeq - histogram - hue - kerndeint - life - movie - mp (with the help of Michael) - overlay - pad - pan - pp - pp - removelogo - sendcmd - showspectrum - showwaves - silencedetect - stereo3d - subtitles - super2xsai - swapuv - thumbnail - tile Hendrik Leppkes: * Switch to AVFrame: - aconvert - amerge - asetnsamples - atempo - biquads Matthieu Bouron: * Switch to AVFrame - alphamerge - decimate - volumedetect Stefano Sabatini: * Switch to AVFrame: - astreamsync - flite - framestep Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Nicolas George <nicolas.george@normalesup.org> Signed-off-by: Clément Bœsch <ubitux@gmail.com> Signed-off-by: Hendrik Leppkes <h.leppkes@gmail.com> Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com> Signed-off-by: Stefano Sabatini <stefasab@gmail.com> Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-03-10 03:30:30 +03:00
AVFrame *out_buf;
ThreadData td;
int ch, ret, drop = 0;
if (s->bypass)
return ff_filter_frame(outlink, buf);
ret = av_channel_layout_copy(&s->ch_layout, &inlink->ch_layout);
if (ret < 0) {
av_frame_free(&buf);
return ret;
}
if (strcmp(s->ch_layout_str, "all"))
av_channel_layout_from_string(&s->ch_layout,
s->ch_layout_str);
if (av_frame_is_writable(buf) && s->block_samples == 0) {
out_buf = buf;
} else {
out_buf = ff_get_audio_buffer(outlink, s->block_samples > 0 ? s->block_samples : buf->nb_samples);
if (!out_buf) {
av_frame_free(&buf);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out_buf, buf);
}
if (s->block_samples > 0 && s->pts == AV_NOPTS_VALUE)
drop = 1;
td.in = buf;
td.out = out_buf;
td.eof = eof;
ff_filter_execute(ctx, filter_channel, &td, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
for (ch = 0; ch < outlink->ch_layout.nb_channels; ch++) {
if (s->cache[ch].clippings > 0)
av_log(ctx, AV_LOG_WARNING, "Channel %d clipping %d times. Please reduce gain.\n",
ch, s->cache[ch].clippings);
s->cache[ch].clippings = 0;
}
if (s->block_samples > 0) {
int nb_samples = buf->nb_samples;
int64_t pts = buf->pts;
out_buf->pts = s->pts;
out_buf->nb_samples = s->nb_samples;
s->pts = pts;
s->nb_samples = nb_samples;
}
if (buf != out_buf)
Merge commit '7e350379f87e7f74420b4813170fe808e2313911' * commit '7e350379f87e7f74420b4813170fe808e2313911': lavfi: switch to AVFrame. Conflicts: doc/filters.texi libavfilter/af_ashowinfo.c libavfilter/audio.c libavfilter/avfilter.c libavfilter/avfilter.h libavfilter/buffersink.c libavfilter/buffersrc.c libavfilter/buffersrc.h libavfilter/f_select.c libavfilter/f_setpts.c libavfilter/fifo.c libavfilter/split.c libavfilter/src_movie.c libavfilter/version.h libavfilter/vf_aspect.c libavfilter/vf_bbox.c libavfilter/vf_blackframe.c libavfilter/vf_delogo.c libavfilter/vf_drawbox.c libavfilter/vf_drawtext.c libavfilter/vf_fade.c libavfilter/vf_fieldorder.c libavfilter/vf_fps.c libavfilter/vf_frei0r.c libavfilter/vf_gradfun.c libavfilter/vf_hqdn3d.c libavfilter/vf_lut.c libavfilter/vf_overlay.c libavfilter/vf_pad.c libavfilter/vf_scale.c libavfilter/vf_showinfo.c libavfilter/vf_transpose.c libavfilter/vf_vflip.c libavfilter/vf_yadif.c libavfilter/video.c libavfilter/vsrc_testsrc.c libavfilter/yadif.h Following are notes about the merge authorship and various technical details. Michael Niedermayer: * Main merge operation, notably avfilter.c and video.c * Switch to AVFrame: - afade - anullsrc - apad - aresample - blackframe - deshake - idet - il - mandelbrot - mptestsrc - noise - setfield - smartblur - tinterlace * various merge changes and fixes in: - ashowinfo - blackdetect - field - fps - select - testsrc - yadif Nicolas George: * Switch to AVFrame: - make rawdec work with refcounted frames. Adapted from commit 759001c534287a96dc96d1e274665feb7059145d by Anton Khirnov. Also, fix the use of || instead of | in a flags check. - make buffer sink and src, audio and video work all together Clément Bœsch: * Switch to AVFrame: - aevalsrc - alphaextract - blend - cellauto - colormatrix - concat - earwax - ebur128 - edgedetect - geq - histeq - histogram - hue - kerndeint - life - movie - mp (with the help of Michael) - overlay - pad - pan - pp - pp - removelogo - sendcmd - showspectrum - showwaves - silencedetect - stereo3d - subtitles - super2xsai - swapuv - thumbnail - tile Hendrik Leppkes: * Switch to AVFrame: - aconvert - amerge - asetnsamples - atempo - biquads Matthieu Bouron: * Switch to AVFrame - alphamerge - decimate - volumedetect Stefano Sabatini: * Switch to AVFrame: - astreamsync - flite - framestep Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Nicolas George <nicolas.george@normalesup.org> Signed-off-by: Clément Bœsch <ubitux@gmail.com> Signed-off-by: Hendrik Leppkes <h.leppkes@gmail.com> Signed-off-by: Matthieu Bouron <matthieu.bouron@gmail.com> Signed-off-by: Stefano Sabatini <stefasab@gmail.com> Merged-by: Michael Niedermayer <michaelni@gmx.at>
2013-03-10 03:30:30 +03:00
av_frame_free(&buf);
if (!drop)
return ff_filter_frame(outlink, out_buf);
else {
av_frame_free(&out_buf);
ff_filter_set_ready(ctx, 10);
return 0;
}
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
BiquadsContext *s = ctx->priv;
AVFrame *in = NULL;
int64_t pts;
int status;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (s->block_samples > 0) {
ret = ff_inlink_consume_samples(inlink, s->block_samples, s->block_samples, &in);
} else {
ret = ff_inlink_consume_frame(inlink, &in);
}
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, in, 0);
if (s->block_samples > 0 && ff_inlink_queued_samples(inlink) >= s->block_samples) {
ff_filter_set_ready(ctx, 10);
return 0;
}
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (s->block_samples > 0) {
AVFrame *in = ff_get_audio_buffer(outlink, s->block_samples);
if (!in)
return AVERROR(ENOMEM);
ret = filter_frame(inlink, in, 1);
}
ff_outlink_set_status(outlink, status, pts);
return ret;
}
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
AVFilterLink *outlink = ctx->outputs[0];
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
return config_filter(outlink, 0);
}
static av_cold void uninit(AVFilterContext *ctx)
{
BiquadsContext *s = ctx->priv;
for (int i = 0; i < 3; i++)
av_frame_free(&s->block[i]);
av_freep(&s->cache);
av_channel_layout_uninit(&s->ch_layout);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
#define OFFSET(x) offsetof(BiquadsContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define DEFINE_BIQUAD_FILTER_2(name_, description_, priv_class_) \
static av_cold int name_##_init(AVFilterContext *ctx) \
{ \
BiquadsContext *s = ctx->priv; \
s->filter_type = name_; \
s->pts = AV_NOPTS_VALUE; \
return 0; \
} \
\
const AVFilter ff_af_##name_ = { \
.name = #name_, \
.description = NULL_IF_CONFIG_SMALL(description_), \
.priv_class = &priv_class_##_class, \
.priv_size = sizeof(BiquadsContext), \
.init = name_##_init, \
.activate = activate, \
.uninit = uninit, \
2021-08-12 13:05:31 +02:00
FILTER_INPUTS(inputs), \
FILTER_OUTPUTS(outputs), \
avfilter: Replace query_formats callback with union of list and callback If one looks at the many query_formats callbacks in existence, one will immediately recognize that there is one type of default callback for video and a slightly different default callback for audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);" for video with a filter-specific pix_fmts list. For audio, it is the same with a filter-specific sample_fmts list together with ff_set_common_all_samplerates() and ff_set_common_all_channel_counts(). This commit allows to remove the boilerplate query_formats callbacks by replacing said callback with a union consisting the old callback and pointers for pixel and sample format arrays. For the not uncommon case in which these lists only contain a single entry (besides the sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also added to the union to store them directly in the AVFilter, thereby avoiding a relocation. The state of said union will be contained in a new, dedicated AVFilter field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t in order to create a hole for this new field; this is no problem, as the maximum of all the nb_inputs is four; for nb_outputs it is only two). The state's default value coincides with the earlier default of query_formats being unset, namely that the filter accepts all formats (and also sample rates and channel counts/layouts for audio) provided that these properties agree coincide for all inputs and outputs. By using different union members for audio and video filters the type-unsafety of using the same functions for audio and video lists will furthermore be more confined to formats.c than before. When the new fields are used, they will also avoid allocations: Currently something nearly equivalent to ff_default_query_formats() is called after every successful call to a query_formats callback; yet in the common case that the newly allocated AVFilterFormats are not used at all (namely if there are no free links) these newly allocated AVFilterFormats are freed again without ever being used. Filters no longer using the callback will not exhibit this any more. Reviewed-by: Paul B Mahol <onemda@gmail.com> Reviewed-by: Nicolas George <george@nsup.org> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-09-27 12:07:35 +02:00
FILTER_QUERY_FUNC(query_formats), \
.process_command = process_command, \
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, \
}
#define DEFINE_BIQUAD_FILTER(name, description) \
AVFILTER_DEFINE_CLASS(name); \
DEFINE_BIQUAD_FILTER_2(name, description, name)
#define WIDTH_OPTION(x) \
{"width", "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=x}, 0, 99999, FLAGS}, \
{"w", "set width", OFFSET(width), AV_OPT_TYPE_DOUBLE, {.dbl=x}, 0, 99999, FLAGS}
#define WIDTH_TYPE_OPTION(x) \
{"width_type", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=x}, HERTZ, NB_WTYPE-1, FLAGS, "width_type"}, \
{"t", "set filter-width type", OFFSET(width_type), AV_OPT_TYPE_INT, {.i64=x}, HERTZ, NB_WTYPE-1, FLAGS, "width_type"}, \
{"h", "Hz", 0, AV_OPT_TYPE_CONST, {.i64=HERTZ}, 0, 0, FLAGS, "width_type"}, \
{"q", "Q-Factor", 0, AV_OPT_TYPE_CONST, {.i64=QFACTOR}, 0, 0, FLAGS, "width_type"}, \
{"o", "octave", 0, AV_OPT_TYPE_CONST, {.i64=OCTAVE}, 0, 0, FLAGS, "width_type"}, \
{"s", "slope", 0, AV_OPT_TYPE_CONST, {.i64=SLOPE}, 0, 0, FLAGS, "width_type"}, \
{"k", "kHz", 0, AV_OPT_TYPE_CONST, {.i64=KHERTZ}, 0, 0, FLAGS, "width_type"}
#define MIX_CHANNELS_NORMALIZE_OPTION(x, y, z) \
{"mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=x}, 0, 1, FLAGS}, \
{"m", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=x}, 0, 1, FLAGS}, \
{"channels", "set channels to filter", OFFSET(ch_layout_str), AV_OPT_TYPE_STRING, {.str=y}, 0, 0, FLAGS}, \
{"c", "set channels to filter", OFFSET(ch_layout_str), AV_OPT_TYPE_STRING, {.str=y}, 0, 0, FLAGS}, \
{"normalize", "normalize coefficients", OFFSET(normalize), AV_OPT_TYPE_BOOL, {.i64=z}, 0, 1, FLAGS}, \
{"n", "normalize coefficients", OFFSET(normalize), AV_OPT_TYPE_BOOL, {.i64=z}, 0, 1, FLAGS}
#define TRANSFORM_OPTION(x) \
{"transform", "set transform type", OFFSET(transform_type), AV_OPT_TYPE_INT, {.i64=x}, 0, NB_TTYPE-1, AF, "transform_type"}, \
{"a", "set transform type", OFFSET(transform_type), AV_OPT_TYPE_INT, {.i64=x}, 0, NB_TTYPE-1, AF, "transform_type"}, \
{"di", "direct form I", 0, AV_OPT_TYPE_CONST, {.i64=DI}, 0, 0, AF, "transform_type"}, \
{"dii", "direct form II", 0, AV_OPT_TYPE_CONST, {.i64=DII}, 0, 0, AF, "transform_type"}, \
{"tdi", "transposed direct form I", 0, AV_OPT_TYPE_CONST, {.i64=TDI}, 0, 0, AF, "transform_type"}, \
{"tdii", "transposed direct form II", 0, AV_OPT_TYPE_CONST, {.i64=TDII}, 0, 0, AF, "transform_type"}, \
{"latt", "lattice-ladder form", 0, AV_OPT_TYPE_CONST, {.i64=LATT}, 0, 0, AF, "transform_type"}, \
{"svf", "state variable filter form", 0, AV_OPT_TYPE_CONST, {.i64=SVF}, 0, 0, AF, "transform_type"}, \
{"zdf", "zero-delay filter form", 0, AV_OPT_TYPE_CONST, {.i64=ZDF}, 0, 0, AF, "transform_type"}
#define PRECISION_OPTION(x) \
{"precision", "set filtering precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=x}, -1, 3, AF, "precision"}, \
{"r", "set filtering precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=x}, -1, 3, AF, "precision"}, \
{"auto", "automatic", 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, AF, "precision"}, \
{"s16", "signed 16-bit", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "precision"}, \
{"s32", "signed 32-bit", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "precision"}, \
{"f32", "floating-point single", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "precision"}, \
{"f64", "floating-point double", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, AF, "precision"}
#define BLOCKSIZE_OPTION(x) \
{"blocksize", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=x}, 0, 32768, AF}, \
{"b", "set the block size", OFFSET(block_samples), AV_OPT_TYPE_INT, {.i64=x}, 0, 32768, AF}
#if CONFIG_EQUALIZER_FILTER
static const AVOption equalizer_options[] = {
{"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 999999, FLAGS},
{"f", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(1.0),
{"gain", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
{"g", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
DEFINE_BIQUAD_FILTER(equalizer, "Apply two-pole peaking equalization (EQ) filter.");
#endif /* CONFIG_EQUALIZER_FILTER */
#if CONFIG_BASS_FILTER || CONFIG_LOWSHELF_FILTER
static const AVOption bass_lowshelf_options[] = {
{"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 0, 999999, FLAGS},
{"f", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(0.5),
{"gain", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
{"g", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
{"poles", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
{"p", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
AVFILTER_DEFINE_CLASS_EXT(bass_lowshelf, "bass/lowshelf", bass_lowshelf_options);
#if CONFIG_BASS_FILTER
DEFINE_BIQUAD_FILTER_2(bass, "Boost or cut lower frequencies.", bass_lowshelf);
#endif /* CONFIG_BASS_FILTER */
#if CONFIG_LOWSHELF_FILTER
DEFINE_BIQUAD_FILTER_2(lowshelf, "Apply a low shelf filter.", bass_lowshelf);
#endif /* CONFIG_LOWSHELF_FILTER */
#endif /* CONFIG_BASS_FILTER || CONFIG LOWSHELF_FILTER */
2022-05-25 11:13:32 +02:00
#if CONFIG_TREBLE_FILTER || CONFIG_HIGHSHELF_FILTER || CONFIG_TILTSHELF_FILTER
static const AVOption treble_highshelf_options[] = {
{"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
{"f", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(0.5),
{"gain", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
{"g", "set gain", OFFSET(gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -900, 900, FLAGS},
{"poles", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
{"p", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
2022-05-25 11:13:32 +02:00
AVFILTER_DEFINE_CLASS_EXT(treble_highshelf, "treble/high/tiltshelf",
treble_highshelf_options);
#if CONFIG_TREBLE_FILTER
DEFINE_BIQUAD_FILTER_2(treble, "Boost or cut upper frequencies.", treble_highshelf);
#endif /* CONFIG_TREBLE_FILTER */
#if CONFIG_HIGHSHELF_FILTER
DEFINE_BIQUAD_FILTER_2(highshelf, "Apply a high shelf filter.", treble_highshelf);
#endif /* CONFIG_HIGHSHELF_FILTER */
2022-05-25 11:13:32 +02:00
#if CONFIG_TILTSHELF_FILTER
DEFINE_BIQUAD_FILTER_2(tiltshelf, "Apply a tilt shelf filter.", treble_highshelf);
#endif
#endif /* CONFIG_TREBLE_FILTER || CONFIG_HIGHSHELF_FILTER || CONFIG_TILTSHELF_FILTER */
#if CONFIG_BANDPASS_FILTER
static const AVOption bandpass_options[] = {
{"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
{"f", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(0.5),
{"csg", "use constant skirt gain", OFFSET(csg), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS},
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
DEFINE_BIQUAD_FILTER(bandpass, "Apply a two-pole Butterworth band-pass filter.");
#endif /* CONFIG_BANDPASS_FILTER */
#if CONFIG_BANDREJECT_FILTER
static const AVOption bandreject_options[] = {
{"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
{"f", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(0.5),
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
DEFINE_BIQUAD_FILTER(bandreject, "Apply a two-pole Butterworth band-reject filter.");
#endif /* CONFIG_BANDREJECT_FILTER */
#if CONFIG_LOWPASS_FILTER
static const AVOption lowpass_options[] = {
{"frequency", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=500}, 0, 999999, FLAGS},
{"f", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=500}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(0.707),
{"poles", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
{"p", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
DEFINE_BIQUAD_FILTER(lowpass, "Apply a low-pass filter with 3dB point frequency.");
#endif /* CONFIG_LOWPASS_FILTER */
#if CONFIG_HIGHPASS_FILTER
static const AVOption highpass_options[] = {
{"frequency", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
{"f", "set frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(0.707),
{"poles", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
{"p", "set number of poles", OFFSET(poles), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, AF},
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
DEFINE_BIQUAD_FILTER(highpass, "Apply a high-pass filter with 3dB point frequency.");
#endif /* CONFIG_HIGHPASS_FILTER */
#if CONFIG_ALLPASS_FILTER
static const AVOption allpass_options[] = {
{"frequency", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
{"f", "set central frequency", OFFSET(frequency), AV_OPT_TYPE_DOUBLE, {.dbl=3000}, 0, 999999, FLAGS},
WIDTH_TYPE_OPTION(QFACTOR),
WIDTH_OPTION(0.707),
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
{"order", "set filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, FLAGS},
{"o", "set filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=2}, 1, 2, FLAGS},
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
{NULL}
};
DEFINE_BIQUAD_FILTER(allpass, "Apply a two-pole all-pass filter.");
#endif /* CONFIG_ALLPASS_FILTER */
#if CONFIG_BIQUAD_FILTER
static const AVOption biquad_options[] = {
{"a0", NULL, OFFSET(oa0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, INT32_MIN, INT32_MAX, FLAGS},
{"a1", NULL, OFFSET(oa1), AV_OPT_TYPE_DOUBLE, {.dbl=0}, INT32_MIN, INT32_MAX, FLAGS},
{"a2", NULL, OFFSET(oa2), AV_OPT_TYPE_DOUBLE, {.dbl=0}, INT32_MIN, INT32_MAX, FLAGS},
{"b0", NULL, OFFSET(ob0), AV_OPT_TYPE_DOUBLE, {.dbl=0}, INT32_MIN, INT32_MAX, FLAGS},
{"b1", NULL, OFFSET(ob1), AV_OPT_TYPE_DOUBLE, {.dbl=0}, INT32_MIN, INT32_MAX, FLAGS},
{"b2", NULL, OFFSET(ob2), AV_OPT_TYPE_DOUBLE, {.dbl=0}, INT32_MIN, INT32_MAX, FLAGS},
MIX_CHANNELS_NORMALIZE_OPTION(1, "all", 0),
TRANSFORM_OPTION(DI),
PRECISION_OPTION(-1),
BLOCKSIZE_OPTION(0),
{NULL}
};
DEFINE_BIQUAD_FILTER(biquad, "Apply a biquad IIR filter with the given coefficients.");
#endif /* CONFIG_BIQUAD_FILTER */