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FFmpeg/libavfilter/af_superequalizer.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

366 lines
10 KiB
C

/*
* Copyright (c) 2002 Naoki Shibata
* Copyright (c) 2017 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/mem.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
#define NBANDS 17
#define M 15
typedef struct EqParameter {
float lower, upper, gain;
} EqParameter;
typedef struct SuperEqualizerContext {
const AVClass *class;
EqParameter params[NBANDS + 1];
float gains[NBANDS + 1];
float fact[M + 1];
float aa;
float iza;
float *ires, *irest;
float *fsamples, *fsamples_out;
int winlen, tabsize;
AVFrame *in, *out;
AVTXContext *rdft, *irdft;
av_tx_fn tx_fn, itx_fn;
} SuperEqualizerContext;
static const float bands[] = {
65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
};
static float izero(SuperEqualizerContext *s, float x)
{
float ret = 1;
int m;
for (m = 1; m <= M; m++) {
float t;
t = pow(x / 2, m) / s->fact[m];
ret += t*t;
}
return ret;
}
static float hn_lpf(int n, float f, float fs)
{
float t = 1 / fs;
float omega = 2 * M_PI * f;
if (n * omega * t == 0)
return 2 * f * t;
return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
}
static float hn_imp(int n)
{
return n == 0 ? 1.f : 0.f;
}
static float hn(int n, EqParameter *param, float fs)
{
float ret, lhn;
int i;
lhn = hn_lpf(n, param[0].upper, fs);
ret = param[0].gain*lhn;
for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
float lhn2 = hn_lpf(n, param[i].upper, fs);
ret += param[i].gain * (lhn2 - lhn);
lhn = lhn2;
}
ret += param[i].gain * (hn_imp(n) - lhn);
return ret;
}
static float alpha(float a)
{
if (a <= 21)
return 0;
if (a <= 50)
return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
return .1102f * (a - 8.7f);
}
static float win(SuperEqualizerContext *s, float n, int N)
{
return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
}
static void process_param(float *bc, EqParameter *param, float fs)
{
int i;
for (i = 0; i <= NBANDS; i++) {
param[i].lower = i == 0 ? 0 : bands[i - 1];
param[i].upper = i == NBANDS ? fs : bands[i];
param[i].gain = bc[i];
}
}
static int equ_init(SuperEqualizerContext *s, int wb)
{
float scale = 1.f, iscale = 1.f;
int i, j, ret;
ret = av_tx_init(&s->rdft, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, 1 << wb, &scale, 0);
if (ret < 0)
return ret;
ret = av_tx_init(&s->irdft, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, 1 << wb, &iscale, 0);
if (ret < 0)
return ret;
s->aa = 96;
s->winlen = (1 << (wb-1))-1;
s->tabsize = 1 << wb;
s->ires = av_calloc(s->tabsize + 2, sizeof(float));
s->irest = av_calloc(s->tabsize, sizeof(float));
s->fsamples = av_calloc(s->tabsize, sizeof(float));
s->fsamples_out = av_calloc(s->tabsize + 2, sizeof(float));
if (!s->ires || !s->irest || !s->fsamples || !s->fsamples_out)
return AVERROR(ENOMEM);
for (i = 0; i <= M; i++) {
s->fact[i] = 1;
for (j = 1; j <= i; j++)
s->fact[i] *= j;
}
s->iza = izero(s, alpha(s->aa));
return 0;
}
static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
{
const int winlen = s->winlen;
const int tabsize = s->tabsize;
int i;
if (fs <= 0)
return;
process_param(lbc, param, fs);
for (i = 0; i < winlen; i++)
s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
for (; i < tabsize; i++)
s->irest[i] = 0;
s->tx_fn(s->rdft, s->ires, s->irest, sizeof(float));
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
SuperEqualizerContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
const float *ires = s->ires;
float *fsamples_out = s->fsamples_out;
float *fsamples = s->fsamples;
int ch, i;
AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
float *src, *dst, *ptr;
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
for (ch = 0; ch < in->ch_layout.nb_channels; ch++) {
ptr = (float *)out->extended_data[ch];
dst = (float *)s->out->extended_data[ch];
src = (float *)in->extended_data[ch];
for (i = 0; i < in->nb_samples; i++)
fsamples[i] = src[i];
for (; i < s->tabsize; i++)
fsamples[i] = 0;
s->tx_fn(s->rdft, fsamples_out, fsamples, sizeof(float));
for (i = 0; i <= s->tabsize / 2; i++) {
float re, im;
re = ires[i*2 ] * fsamples_out[i*2] - ires[i*2+1] * fsamples_out[i*2+1];
im = ires[i*2+1] * fsamples_out[i*2] + ires[i*2 ] * fsamples_out[i*2+1];
fsamples_out[i*2 ] = re;
fsamples_out[i*2+1] = im;
}
s->itx_fn(s->irdft, fsamples, fsamples_out, sizeof(AVComplexFloat));
for (i = 0; i < s->winlen; i++)
dst[i] += fsamples[i] / s->tabsize;
for (i = s->winlen; i < s->tabsize; i++)
dst[i] = fsamples[i] / s->tabsize;
for (i = 0; i < out->nb_samples; i++)
ptr[i] = dst[i];
for (i = 0; i < s->winlen; i++)
dst[i] = dst[i+s->winlen];
}
out->pts = in->pts;
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
SuperEqualizerContext *s = ctx->priv;
AVFrame *in = NULL;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, in);
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static av_cold int init(AVFilterContext *ctx)
{
SuperEqualizerContext *s = ctx->priv;
return equ_init(s, 14);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
SuperEqualizerContext *s = ctx->priv;
s->out = ff_get_audio_buffer(inlink, s->tabsize);
if (!s->out)
return AVERROR(ENOMEM);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
SuperEqualizerContext *s = ctx->priv;
make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
SuperEqualizerContext *s = ctx->priv;
av_frame_free(&s->out);
av_freep(&s->irest);
av_freep(&s->ires);
av_freep(&s->fsamples);
av_freep(&s->fsamples_out);
av_tx_uninit(&s->rdft);
av_tx_uninit(&s->irdft);
}
static const AVFilterPad superequalizer_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
};
static const AVFilterPad superequalizer_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define OFFSET(x) offsetof(SuperEqualizerContext, x)
static const AVOption superequalizer_options[] = {
{ "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
{ NULL }
};
AVFILTER_DEFINE_CLASS(superequalizer);
const AVFilter ff_af_superequalizer = {
.name = "superequalizer",
.description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
.priv_size = sizeof(SuperEqualizerContext),
.priv_class = &superequalizer_class,
.init = init,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(superequalizer_inputs),
FILTER_OUTPUTS(superequalizer_outputs),
FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
};