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https://github.com/FFmpeg/FFmpeg.git
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365 lines
10 KiB
C
365 lines
10 KiB
C
/*
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* Copyright (c) 2002 Naoki Shibata
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* Copyright (c) 2017 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/opt.h"
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#include "libavutil/tx.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "filters.h"
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#include "internal.h"
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#define NBANDS 17
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#define M 15
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typedef struct EqParameter {
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float lower, upper, gain;
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} EqParameter;
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typedef struct SuperEqualizerContext {
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const AVClass *class;
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EqParameter params[NBANDS + 1];
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float gains[NBANDS + 1];
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float fact[M + 1];
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float aa;
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float iza;
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float *ires, *irest;
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float *fsamples, *fsamples_out;
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int winlen, tabsize;
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AVFrame *in, *out;
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AVTXContext *rdft, *irdft;
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av_tx_fn tx_fn, itx_fn;
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} SuperEqualizerContext;
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static const float bands[] = {
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65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
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1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
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};
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static float izero(SuperEqualizerContext *s, float x)
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{
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float ret = 1;
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int m;
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for (m = 1; m <= M; m++) {
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float t;
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t = pow(x / 2, m) / s->fact[m];
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ret += t*t;
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}
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return ret;
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}
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static float hn_lpf(int n, float f, float fs)
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{
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float t = 1 / fs;
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float omega = 2 * M_PI * f;
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if (n * omega * t == 0)
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return 2 * f * t;
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return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
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}
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static float hn_imp(int n)
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{
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return n == 0 ? 1.f : 0.f;
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}
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static float hn(int n, EqParameter *param, float fs)
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{
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float ret, lhn;
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int i;
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lhn = hn_lpf(n, param[0].upper, fs);
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ret = param[0].gain*lhn;
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for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
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float lhn2 = hn_lpf(n, param[i].upper, fs);
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ret += param[i].gain * (lhn2 - lhn);
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lhn = lhn2;
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}
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ret += param[i].gain * (hn_imp(n) - lhn);
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return ret;
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}
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static float alpha(float a)
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{
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if (a <= 21)
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return 0;
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if (a <= 50)
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return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
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return .1102f * (a - 8.7f);
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}
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static float win(SuperEqualizerContext *s, float n, int N)
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{
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return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
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}
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static void process_param(float *bc, EqParameter *param, float fs)
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{
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int i;
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for (i = 0; i <= NBANDS; i++) {
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param[i].lower = i == 0 ? 0 : bands[i - 1];
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param[i].upper = i == NBANDS ? fs : bands[i];
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param[i].gain = bc[i];
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}
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}
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static int equ_init(SuperEqualizerContext *s, int wb)
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{
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float scale = 1.f, iscale = 1.f;
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int i, j, ret;
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ret = av_tx_init(&s->rdft, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, 1 << wb, &scale, 0);
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if (ret < 0)
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return ret;
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ret = av_tx_init(&s->irdft, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, 1 << wb, &iscale, 0);
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if (ret < 0)
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return ret;
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s->aa = 96;
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s->winlen = (1 << (wb-1))-1;
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s->tabsize = 1 << wb;
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s->ires = av_calloc(s->tabsize + 2, sizeof(float));
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s->irest = av_calloc(s->tabsize, sizeof(float));
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s->fsamples = av_calloc(s->tabsize, sizeof(float));
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s->fsamples_out = av_calloc(s->tabsize + 2, sizeof(float));
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if (!s->ires || !s->irest || !s->fsamples || !s->fsamples_out)
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return AVERROR(ENOMEM);
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for (i = 0; i <= M; i++) {
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s->fact[i] = 1;
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for (j = 1; j <= i; j++)
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s->fact[i] *= j;
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}
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s->iza = izero(s, alpha(s->aa));
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return 0;
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}
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static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
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{
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const int winlen = s->winlen;
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const int tabsize = s->tabsize;
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int i;
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if (fs <= 0)
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return;
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process_param(lbc, param, fs);
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for (i = 0; i < winlen; i++)
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s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
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for (; i < tabsize; i++)
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s->irest[i] = 0;
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s->tx_fn(s->rdft, s->ires, s->irest, sizeof(float));
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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SuperEqualizerContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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const float *ires = s->ires;
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float *fsamples_out = s->fsamples_out;
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float *fsamples = s->fsamples;
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int ch, i;
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AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
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float *src, *dst, *ptr;
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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for (ch = 0; ch < in->ch_layout.nb_channels; ch++) {
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ptr = (float *)out->extended_data[ch];
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dst = (float *)s->out->extended_data[ch];
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src = (float *)in->extended_data[ch];
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for (i = 0; i < in->nb_samples; i++)
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fsamples[i] = src[i];
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for (; i < s->tabsize; i++)
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fsamples[i] = 0;
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s->tx_fn(s->rdft, fsamples_out, fsamples, sizeof(float));
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for (i = 0; i <= s->tabsize / 2; i++) {
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float re, im;
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re = ires[i*2 ] * fsamples_out[i*2] - ires[i*2+1] * fsamples_out[i*2+1];
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im = ires[i*2+1] * fsamples_out[i*2] + ires[i*2 ] * fsamples_out[i*2+1];
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fsamples_out[i*2 ] = re;
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fsamples_out[i*2+1] = im;
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}
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s->itx_fn(s->irdft, fsamples, fsamples_out, sizeof(float));
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for (i = 0; i < s->winlen; i++)
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dst[i] += fsamples[i] / s->tabsize;
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for (i = s->winlen; i < s->tabsize; i++)
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dst[i] = fsamples[i] / s->tabsize;
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for (i = 0; i < out->nb_samples; i++)
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ptr[i] = dst[i];
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for (i = 0; i < s->winlen; i++)
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dst[i] = dst[i+s->winlen];
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}
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out->pts = in->pts;
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static int activate(AVFilterContext *ctx)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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SuperEqualizerContext *s = ctx->priv;
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AVFrame *in = NULL;
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int ret;
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FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
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ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
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if (ret < 0)
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return ret;
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if (ret > 0)
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return filter_frame(inlink, in);
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FF_FILTER_FORWARD_STATUS(inlink, outlink);
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FF_FILTER_FORWARD_WANTED(outlink, inlink);
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return FFERROR_NOT_READY;
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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SuperEqualizerContext *s = ctx->priv;
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return equ_init(s, 14);
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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SuperEqualizerContext *s = ctx->priv;
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s->out = ff_get_audio_buffer(inlink, s->tabsize);
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if (!s->out)
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return AVERROR(ENOMEM);
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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SuperEqualizerContext *s = ctx->priv;
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make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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SuperEqualizerContext *s = ctx->priv;
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av_frame_free(&s->out);
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av_freep(&s->irest);
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av_freep(&s->ires);
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av_freep(&s->fsamples);
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av_freep(&s->fsamples_out);
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av_tx_uninit(&s->rdft);
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av_tx_uninit(&s->irdft);
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}
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static const AVFilterPad superequalizer_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_input,
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},
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};
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static const AVFilterPad superequalizer_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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};
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#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define OFFSET(x) offsetof(SuperEqualizerContext, x)
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static const AVOption superequalizer_options[] = {
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{ "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(superequalizer);
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const AVFilter ff_af_superequalizer = {
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.name = "superequalizer",
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.description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
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.priv_size = sizeof(SuperEqualizerContext),
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.priv_class = &superequalizer_class,
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.init = init,
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.activate = activate,
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.uninit = uninit,
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FILTER_INPUTS(superequalizer_inputs),
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FILTER_OUTPUTS(superequalizer_outputs),
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FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
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};
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