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FFmpeg/libavfilter/af_anequalizer.c
Andreas Rheinhardt a04ad248a0 avfilter: Constify all AVFilters 
This is possible now that the next-API is gone.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 11:48:05 -03:00

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/*
* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
* Copyright (c) 2015 Paul B Mahol
*
* 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
*/
#include "libavutil/intreadwrite.h"
#include "libavutil/avstring.h"
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "libavutil/parseutils.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
#define FILTER_ORDER 4
enum FilterType {
BUTTERWORTH,
CHEBYSHEV1,
CHEBYSHEV2,
NB_TYPES
};
typedef struct FoSection {
double a0, a1, a2, a3, a4;
double b0, b1, b2, b3, b4;
double num[4];
double denum[4];
} FoSection;
typedef struct EqualizatorFilter {
int ignore;
int channel;
int type;
double freq;
double gain;
double width;
FoSection section[2];
} EqualizatorFilter;
typedef struct AudioNEqualizerContext {
const AVClass *class;
char *args;
char *colors;
int draw_curves;
int w, h;
double mag;
int fscale;
int nb_filters;
int nb_allocated;
EqualizatorFilter *filters;
AVFrame *video;
} AudioNEqualizerContext;
#define OFFSET(x) offsetof(AudioNEqualizerContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define V AV_OPT_FLAG_VIDEO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM
static const AVOption anequalizer_options[] = {
{ "params", NULL, OFFSET(args), AV_OPT_TYPE_STRING, {.str=""}, 0, 0, A|F },
{ "curves", "draw frequency response curves", OFFSET(draw_curves), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, V|F },
{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, V|F },
{ "mgain", "set max gain", OFFSET(mag), AV_OPT_TYPE_DOUBLE, {.dbl=60}, -900, 900, V|F },
{ "fscale", "set frequency scale", OFFSET(fscale), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, V|F, "fscale" },
{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, V|F, "fscale" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, V|F, "fscale" },
{ "colors", "set channels curves colors", OFFSET(colors), AV_OPT_TYPE_STRING, {.str = "red|green|blue|yellow|orange|lime|pink|magenta|brown" }, 0, 0, V|F },
{ NULL }
};
AVFILTER_DEFINE_CLASS(anequalizer);
static void draw_curves(AVFilterContext *ctx, AVFilterLink *inlink, AVFrame *out)
{
AudioNEqualizerContext *s = ctx->priv;
char *colors, *color, *saveptr = NULL;
int ch, i, n;
colors = av_strdup(s->colors);
if (!colors)
return;
memset(out->data[0], 0, s->h * out->linesize[0]);
for (ch = 0; ch < inlink->channels; ch++) {
uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff };
int prev_v = -1;
double f;
color = av_strtok(ch == 0 ? colors : NULL, " |", &saveptr);
if (color)
av_parse_color(fg, color, -1, ctx);
for (f = 0; f < s->w; f++) {
double zr, zi, zr2, zi2;
double Hr, Hi;
double Hmag = 1;
double w;
int v, y, x;
w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1);
zr = cos(w);
zr2 = zr * zr;
zi = -sin(w);
zi2 = zi * zi;
for (n = 0; n < s->nb_filters; n++) {
if (s->filters[n].channel != ch ||
s->filters[n].ignore)
continue;
for (i = 0; i < FILTER_ORDER / 2; i++) {
FoSection *S = &s->filters[n].section[i];
/* H *= (((((S->b4 * z + S->b3) * z + S->b2) * z + S->b1) * z + S->b0) /
((((S->a4 * z + S->a3) * z + S->a2) * z + S->a1) * z + S->a0)); */
Hr = S->b4*(1-8*zr2*zi2) + S->b2*(zr2-zi2) + zr*(S->b1+S->b3*(zr2-3*zi2))+ S->b0;
Hi = zi*(S->b3*(3*zr2-zi2) + S->b1 + 2*zr*(2*S->b4*(zr2-zi2) + S->b2));
Hmag *= hypot(Hr, Hi);
Hr = S->a4*(1-8*zr2*zi2) + S->a2*(zr2-zi2) + zr*(S->a1+S->a3*(zr2-3*zi2))+ S->a0;
Hi = zi*(S->a3*(3*zr2-zi2) + S->a1 + 2*zr*(2*S->a4*(zr2-zi2) + S->a2));
Hmag /= hypot(Hr, Hi);
}
}
v = av_clip((1. + -20 * log10(Hmag) / s->mag) * s->h / 2, 0, s->h - 1);
x = lrint(f);
if (prev_v == -1)
prev_v = v;
if (v <= prev_v) {
for (y = v; y <= prev_v; y++)
AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
} else {
for (y = prev_v; y <= v; y++)
AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
}
prev_v = v;
}
}
av_free(colors);
}
static int config_video(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioNEqualizerContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVFrame *out;
outlink->w = s->w;
outlink->h = s->h;
av_frame_free(&s->video);
s->video = out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
if (!out)
return AVERROR(ENOMEM);
outlink->sample_aspect_ratio = (AVRational){1,1};
draw_curves(ctx, inlink, out);
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioNEqualizerContext *s = ctx->priv;
AVFilterPad pad, vpad;
int ret;
pad = (AVFilterPad){
.name = "out0",
.type = AVMEDIA_TYPE_AUDIO,
};
ret = ff_insert_outpad(ctx, 0, &pad);
if (ret < 0)
return ret;
if (s->draw_curves) {
vpad = (AVFilterPad){
.name = "out1",
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video,
};
ret = ff_insert_outpad(ctx, 1, &vpad);
if (ret < 0)
return ret;
}
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AudioNEqualizerContext *s = ctx->priv;
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE };
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;
if (s->draw_curves) {
AVFilterLink *videolink = ctx->outputs[1];
formats = ff_make_format_list(pix_fmts);
if ((ret = ff_formats_ref(formats, &videolink->incfg.formats)) < 0)
return ret;
}
formats = ff_make_format_list(sample_fmts);
if ((ret = ff_formats_ref(formats, &inlink->outcfg.formats)) < 0 ||
(ret = ff_formats_ref(formats, &outlink->incfg.formats)) < 0)
return ret;
layouts = ff_all_channel_counts();
if ((ret = ff_channel_layouts_ref(layouts, &inlink->outcfg.channel_layouts)) < 0 ||
(ret = ff_channel_layouts_ref(layouts, &outlink->incfg.channel_layouts)) < 0)
return ret;
formats = ff_all_samplerates();
if ((ret = ff_formats_ref(formats, &inlink->outcfg.samplerates)) < 0 ||
(ret = ff_formats_ref(formats, &outlink->incfg.samplerates)) < 0)
return ret;
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioNEqualizerContext *s = ctx->priv;
av_frame_free(&s->video);
av_freep(&s->filters);
s->nb_filters = 0;
s->nb_allocated = 0;
}
static void butterworth_fo_section(FoSection *S, double beta,
double si, double g, double g0,
double D, double c0)
{
if (c0 == 1 || c0 == -1) {
S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
S->b1 = 2*c0*(g*g*beta*beta - g0*g0)/D;
S->b2 = (g*g*beta*beta - 2*g0*g*beta*si + g0*g0)/D;
S->b3 = 0;
S->b4 = 0;
S->a0 = 1;
S->a1 = 2*c0*(beta*beta - 1)/D;
S->a2 = (beta*beta - 2*beta*si + 1)/D;
S->a3 = 0;
S->a4 = 0;
} else {
S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
S->b1 = -4*c0*(g0*g0 + g*g0*si*beta)/D;
S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - g*g*beta*beta)/D;
S->b3 = -4*c0*(g0*g0 - g*g0*si*beta)/D;
S->b4 = (g*g*beta*beta - 2*g*g0*si*beta + g0*g0)/D;
S->a0 = 1;
S->a1 = -4*c0*(1 + si*beta)/D;
S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/D;
S->a3 = -4*c0*(1 - si*beta)/D;
S->a4 = (beta*beta - 2*si*beta + 1)/D;
}
}
static void butterworth_bp_filter(EqualizatorFilter *f,
int N, double w0, double wb,
double G, double Gb, double G0)
{
double g, c0, g0, beta;
double epsilon;
int r = N % 2;
int L = (N - r) / 2;
int i;
if (G == 0 && G0 == 0) {
f->section[0].a0 = 1;
f->section[0].b0 = 1;
f->section[1].a0 = 1;
f->section[1].b0 = 1;
return;
}
G = ff_exp10(G/20);
Gb = ff_exp10(Gb/20);
G0 = ff_exp10(G0/20);
epsilon = sqrt((G * G - Gb * Gb) / (Gb * Gb - G0 * G0));
g = pow(G, 1.0 / N);
g0 = pow(G0, 1.0 / N);
beta = pow(epsilon, -1.0 / N) * tan(wb/2);
c0 = cos(w0);
for (i = 1; i <= L; i++) {
double ui = (2.0 * i - 1) / N;
double si = sin(M_PI * ui / 2.0);
double Di = beta * beta + 2 * si * beta + 1;
butterworth_fo_section(&f->section[i - 1], beta, si, g, g0, Di, c0);
}
}
static void chebyshev1_fo_section(FoSection *S, double a,
double c, double tetta_b,
double g0, double si, double b,
double D, double c0)
{
if (c0 == 1 || c0 == -1) {
S->b0 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) + 2*g0*b*si*tetta_b*tetta_b + g0*g0)/D;
S->b1 = 2*c0*(tetta_b*tetta_b*(b*b+g0*g0*c*c) - g0*g0)/D;
S->b2 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) - 2*g0*b*si*tetta_b + g0*g0)/D;
S->b3 = 0;
S->b4 = 0;
S->a0 = 1;
S->a1 = 2*c0*(tetta_b*tetta_b*(a*a+c*c) - 1)/D;
S->a2 = (tetta_b*tetta_b*(a*a+c*c) - 2*a*si*tetta_b + 1)/D;
S->a3 = 0;
S->a4 = 0;
} else {
S->b0 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b + 2*g0*b*si*tetta_b + g0*g0)/D;
S->b1 = -4*c0*(g0*g0 + g0*b*si*tetta_b)/D;
S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - (b*b + g0*g0*c*c)*tetta_b*tetta_b)/D;
S->b3 = -4*c0*(g0*g0 - g0*b*si*tetta_b)/D;
S->b4 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b - 2*g0*b*si*tetta_b + g0*g0)/D;
S->a0 = 1;
S->a1 = -4*c0*(1 + a*si*tetta_b)/D;
S->a2 = 2*(1 + 2*c0*c0 - (a*a + c*c)*tetta_b*tetta_b)/D;
S->a3 = -4*c0*(1 - a*si*tetta_b)/D;
S->a4 = ((a*a + c*c)*tetta_b*tetta_b - 2*a*si*tetta_b + 1)/D;
}
}
static void chebyshev1_bp_filter(EqualizatorFilter *f,
int N, double w0, double wb,
double G, double Gb, double G0)
{
double a, b, c0, g0, alfa, beta, tetta_b;
double epsilon;
int r = N % 2;
int L = (N - r) / 2;
int i;
if (G == 0 && G0 == 0) {
f->section[0].a0 = 1;
f->section[0].b0 = 1;
f->section[1].a0 = 1;
f->section[1].b0 = 1;
return;
}
G = ff_exp10(G/20);
Gb = ff_exp10(Gb/20);
G0 = ff_exp10(G0/20);
epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
g0 = pow(G0,1.0/N);
alfa = pow(1.0/epsilon + sqrt(1 + 1/(epsilon*epsilon)), 1.0/N);
beta = pow(G/epsilon + Gb * sqrt(1 + 1/(epsilon*epsilon)), 1.0/N);
a = 0.5 * (alfa - 1.0/alfa);
b = 0.5 * (beta - g0*g0*(1/beta));
tetta_b = tan(wb/2);
c0 = cos(w0);
for (i = 1; i <= L; i++) {
double ui = (2.0*i-1.0)/N;
double ci = cos(M_PI*ui/2.0);
double si = sin(M_PI*ui/2.0);
double Di = (a*a + ci*ci)*tetta_b*tetta_b + 2.0*a*si*tetta_b + 1;
chebyshev1_fo_section(&f->section[i - 1], a, ci, tetta_b, g0, si, b, Di, c0);
}
}
static void chebyshev2_fo_section(FoSection *S, double a,
double c, double tetta_b,
double g, double si, double b,
double D, double c0)
{
if (c0 == 1 || c0 == -1) {
S->b0 = (g*g*tetta_b*tetta_b + 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;
S->b1 = 2*c0*(g*g*tetta_b*tetta_b - b*b - g*g*c*c)/D;
S->b2 = (g*g*tetta_b*tetta_b - 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;
S->b3 = 0;
S->b4 = 0;
S->a0 = 1;
S->a1 = 2*c0*(tetta_b*tetta_b - a*a - c*c)/D;
S->a2 = (tetta_b*tetta_b - 2*tetta_b*a*si + a*a + c*c)/D;
S->a3 = 0;
S->a4 = 0;
} else {
S->b0 = (g*g*tetta_b*tetta_b + 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;
S->b1 = -4*c0*(b*b + g*g*c*c + g*b*si*tetta_b)/D;
S->b2 = 2*((b*b + g*g*c*c)*(1 + 2*c0*c0) - g*g*tetta_b*tetta_b)/D;
S->b3 = -4*c0*(b*b + g*g*c*c - g*b*si*tetta_b)/D;
S->b4 = (g*g*tetta_b*tetta_b - 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;
S->a0 = 1;
S->a1 = -4*c0*(a*a + c*c + a*si*tetta_b)/D;
S->a2 = 2*((a*a + c*c)*(1 + 2*c0*c0) - tetta_b*tetta_b)/D;
S->a3 = -4*c0*(a*a + c*c - a*si*tetta_b)/D;
S->a4 = (tetta_b*tetta_b - 2*a*si*tetta_b + a*a + c*c)/D;
}
}
static void chebyshev2_bp_filter(EqualizatorFilter *f,
int N, double w0, double wb,
double G, double Gb, double G0)
{
double a, b, c0, tetta_b;
double epsilon, g, eu, ew;
int r = N % 2;
int L = (N - r) / 2;
int i;
if (G == 0 && G0 == 0) {
f->section[0].a0 = 1;
f->section[0].b0 = 1;
f->section[1].a0 = 1;
f->section[1].b0 = 1;
return;
}
G = ff_exp10(G/20);
Gb = ff_exp10(Gb/20);
G0 = ff_exp10(G0/20);
epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
g = pow(G, 1.0 / N);
eu = pow(epsilon + sqrt(1 + epsilon*epsilon), 1.0/N);
ew = pow(G0*epsilon + Gb*sqrt(1 + epsilon*epsilon), 1.0/N);
a = (eu - 1.0/eu)/2.0;
b = (ew - g*g/ew)/2.0;
tetta_b = tan(wb/2);
c0 = cos(w0);
for (i = 1; i <= L; i++) {
double ui = (2.0 * i - 1.0)/N;
double ci = cos(M_PI * ui / 2.0);
double si = sin(M_PI * ui / 2.0);
double Di = tetta_b*tetta_b + 2*a*si*tetta_b + a*a + ci*ci;
chebyshev2_fo_section(&f->section[i - 1], a, ci, tetta_b, g, si, b, Di, c0);
}
}
static double butterworth_compute_bw_gain_db(double gain)
{
double bw_gain = 0;
if (gain <= -6)
bw_gain = gain + 3;
else if(gain > -6 && gain < 6)
bw_gain = gain * 0.5;
else if(gain >= 6)
bw_gain = gain - 3;
return bw_gain;
}
static double chebyshev1_compute_bw_gain_db(double gain)
{
double bw_gain = 0;
if (gain <= -6)
bw_gain = gain + 1;
else if(gain > -6 && gain < 6)
bw_gain = gain * 0.9;
else if(gain >= 6)
bw_gain = gain - 1;
return bw_gain;
}
static double chebyshev2_compute_bw_gain_db(double gain)
{
double bw_gain = 0;
if (gain <= -6)
bw_gain = -3;
else if(gain > -6 && gain < 6)
bw_gain = gain * 0.3;
else if(gain >= 6)
bw_gain = 3;
return bw_gain;
}
static inline double hz_2_rad(double x, double fs)
{
return 2 * M_PI * x / fs;
}
static void equalizer(EqualizatorFilter *f, double sample_rate)
{
double w0 = hz_2_rad(f->freq, sample_rate);
double wb = hz_2_rad(f->width, sample_rate);
double bw_gain;
switch (f->type) {
case BUTTERWORTH:
bw_gain = butterworth_compute_bw_gain_db(f->gain);
butterworth_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
break;
case CHEBYSHEV1:
bw_gain = chebyshev1_compute_bw_gain_db(f->gain);
chebyshev1_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
break;
case CHEBYSHEV2:
bw_gain = chebyshev2_compute_bw_gain_db(f->gain);
chebyshev2_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
break;
}
}
static int add_filter(AudioNEqualizerContext *s, AVFilterLink *inlink)
{
equalizer(&s->filters[s->nb_filters], inlink->sample_rate);
if (s->nb_filters >= s->nb_allocated - 1) {
EqualizatorFilter *filters;
filters = av_calloc(s->nb_allocated, 2 * sizeof(*s->filters));
if (!filters)
return AVERROR(ENOMEM);
memcpy(filters, s->filters, sizeof(*s->filters) * s->nb_allocated);
av_free(s->filters);
s->filters = filters;
s->nb_allocated *= 2;
}
s->nb_filters++;
return 0;
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioNEqualizerContext *s = ctx->priv;
char *args = av_strdup(s->args);
char *saveptr = NULL;
int ret = 0;
if (!args)
return AVERROR(ENOMEM);
s->nb_allocated = 32 * inlink->channels;
s->filters = av_calloc(inlink->channels, 32 * sizeof(*s->filters));
if (!s->filters) {
s->nb_allocated = 0;
av_free(args);
return AVERROR(ENOMEM);
}
while (1) {
char *arg = av_strtok(s->nb_filters == 0 ? args : NULL, "|", &saveptr);
if (!arg)
break;
s->filters[s->nb_filters].type = 0;
if (sscanf(arg, "c%d f=%lf w=%lf g=%lf t=%d", &s->filters[s->nb_filters].channel,
&s->filters[s->nb_filters].freq,
&s->filters[s->nb_filters].width,
&s->filters[s->nb_filters].gain,
&s->filters[s->nb_filters].type) != 5 &&
sscanf(arg, "c%d f=%lf w=%lf g=%lf", &s->filters[s->nb_filters].channel,
&s->filters[s->nb_filters].freq,
&s->filters[s->nb_filters].width,
&s->filters[s->nb_filters].gain) != 4 ) {
av_free(args);
return AVERROR(EINVAL);
}
if (s->filters[s->nb_filters].freq < 0 ||
s->filters[s->nb_filters].freq > inlink->sample_rate / 2.0)
s->filters[s->nb_filters].ignore = 1;
if (s->filters[s->nb_filters].channel < 0 ||
s->filters[s->nb_filters].channel >= inlink->channels)
s->filters[s->nb_filters].ignore = 1;
s->filters[s->nb_filters].type = av_clip(s->filters[s->nb_filters].type, 0, NB_TYPES - 1);
ret = add_filter(s, inlink);
if (ret < 0)
break;
}
av_free(args);
return ret;
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
AudioNEqualizerContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
int ret = AVERROR(ENOSYS);
if (!strcmp(cmd, "change")) {
double freq, width, gain;
int filter;
if (sscanf(args, "%d|f=%lf|w=%lf|g=%lf", &filter, &freq, &width, &gain) != 4)
return AVERROR(EINVAL);
if (filter < 0 || filter >= s->nb_filters)
return AVERROR(EINVAL);
if (freq < 0 || freq > inlink->sample_rate / 2.0)
return AVERROR(EINVAL);
s->filters[filter].freq = freq;
s->filters[filter].width = width;
s->filters[filter].gain = gain;
equalizer(&s->filters[filter], inlink->sample_rate);
if (s->draw_curves)
draw_curves(ctx, inlink, s->video);
ret = 0;
}
return ret;
}
static inline double section_process(FoSection *S, double in)
{
double out;
out = S->b0 * in;
out+= S->b1 * S->num[0] - S->denum[0] * S->a1;
out+= S->b2 * S->num[1] - S->denum[1] * S->a2;
out+= S->b3 * S->num[2] - S->denum[2] * S->a3;
out+= S->b4 * S->num[3] - S->denum[3] * S->a4;
S->num[3] = S->num[2];
S->num[2] = S->num[1];
S->num[1] = S->num[0];
S->num[0] = in;
S->denum[3] = S->denum[2];
S->denum[2] = S->denum[1];
S->denum[1] = S->denum[0];
S->denum[0] = out;
return out;
}
static double process_sample(FoSection *s1, double in)
{
double p0 = in, p1;
int i;
for (i = 0; i < FILTER_ORDER / 2; i++) {
p1 = section_process(&s1[i], p0);
p0 = p1;
}
return p1;
}
static int filter_channels(AVFilterContext *ctx, void *arg,
int jobnr, int nb_jobs)
{
AudioNEqualizerContext *s = ctx->priv;
AVFrame *buf = arg;
const int start = (buf->channels * jobnr) / nb_jobs;
const int end = (buf->channels * (jobnr+1)) / nb_jobs;
for (int i = 0; i < s->nb_filters; i++) {
EqualizatorFilter *f = &s->filters[i];
double *bptr;
if (f->gain == 0. || f->ignore)
continue;
if (f->channel < start ||
f->channel >= end)
continue;
bptr = (double *)buf->extended_data[f->channel];
for (int n = 0; n < buf->nb_samples; n++) {
double sample = bptr[n];
sample = process_sample(f->section, sample);
bptr[n] = sample;
}
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
{
AVFilterContext *ctx = inlink->dst;
AudioNEqualizerContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
if (!ctx->is_disabled)
ctx->internal->execute(ctx, filter_channels, buf, NULL, FFMIN(inlink->channels,
ff_filter_get_nb_threads(ctx)));
if (s->draw_curves) {
AVFrame *clone;
const int64_t pts = buf->pts +
av_rescale_q(buf->nb_samples, (AVRational){ 1, inlink->sample_rate },
outlink->time_base);
int ret;
s->video->pts = pts;
clone = av_frame_clone(s->video);
if (!clone)
return AVERROR(ENOMEM);
ret = ff_filter_frame(ctx->outputs[1], clone);
if (ret < 0)
return ret;
}
return ff_filter_frame(outlink, buf);
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
.needs_writable = 1,
},
{ NULL }
};
const AVFilter ff_af_anequalizer = {
.name = "anequalizer",
.description = NULL_IF_CONFIG_SMALL("Apply high-order audio parametric multi band equalizer."),
.priv_size = sizeof(AudioNEqualizerContext),
.priv_class = &anequalizer_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.inputs = inputs,
.outputs = NULL,
.process_command = process_command,
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
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