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FFmpeg/libavfilter/af_anequalizer.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

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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/mem.h"
#include "libavutil/opt.h"
#include "libavutil/parseutils.h"
#include "avfilter.h"
#include "formats.h"
#include "internal.h"
#include "audio.h"
#include "video.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, .unit = "fscale" },
{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, V|F, .unit = "fscale" },
{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, V|F, .unit = "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->ch_layout.nb_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_append_outpad(ctx, &pad);
if (ret < 0)
return ret;
if (s->draw_curves) {
vpad = (AVFilterPad){
.name = "out1",
.type = AVMEDIA_TYPE_VIDEO,
.config_props = config_video,
};
ret = ff_append_outpad(ctx, &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->ch_layout.nb_channels;
s->filters = av_calloc(inlink->ch_layout.nb_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->ch_layout.nb_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->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (buf->ch_layout.nb_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)
ff_filter_execute(ctx, filter_channels, buf, NULL,
FFMIN(inlink->ch_layout.nb_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,
.flags = AVFILTERPAD_FLAG_NEEDS_WRITABLE,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
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,
FILTER_INPUTS(inputs),
.outputs = NULL,
FILTER_QUERY_FUNC(query_formats),
.process_command = process_command,
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS |
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
};
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