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https://github.com/FFmpeg/FFmpeg.git
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avfilter: add superequalizer filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol
parent
d790f18ac0
commit
ca5cf84655
@ -20,6 +20,7 @@ version <next>:
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- sofalizer filter switched to libmysofa
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- Gremlin Digital Video demuxer and decoder
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- headphone audio filter
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- superequalizer audio filter
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version 3.3:
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- CrystalHD decoder moved to new decode API
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@ -3835,6 +3835,49 @@ channels. Default is 0.3.
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Set level of input signal of original channel. Default is 0.8.
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@end table
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@section superequalizer
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Apply 18th band equalizer.
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The filter accpets the following options:
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@table @option
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@item 1b
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Set 65Hz band gain.
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@item 2b
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Set 92Hz band gain.
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@item 3b
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Set 131Hz band gain.
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@item 4b
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Set 185Hz band gain.
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@item 5b
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Set 262Hz band gain.
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@item 6b
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Set 370Hz band gain.
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@item 7b
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Set 523Hz band gain.
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@item 8b
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Set 740Hz band gain.
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@item 9b
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Set 1047Hz band gain.
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@item 10b
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Set 1480Hz band gain.
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@item 11b
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Set 2093Hz band gain.
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@item 12b
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Set 2960Hz band gain.
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@item 13b
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Set 4186Hz band gain.
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@item 14b
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Set 5920Hz band gain.
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@item 15b
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Set 8372Hz band gain.
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@item 16b
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Set 11840Hz band gain.
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@item 17b
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Set 16744Hz band gain.
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@item 18b
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Set 20000Hz band gain.
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@end table
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@section surround
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Apply audio surround upmix filter.
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@ -109,6 +109,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
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OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o
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OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
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OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
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OBJS-$(CONFIG_SUPEREQUALIZER_FILTER) += af_superequalizer.o
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OBJS-$(CONFIG_SURROUND_FILTER) += af_surround.o
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OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o
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OBJS-$(CONFIG_TREMOLO_FILTER) += af_tremolo.o
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368
libavfilter/af_superequalizer.c
Normal file
368
libavfilter/af_superequalizer.c
Normal file
@ -0,0 +1,368 @@
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/*
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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 "libavcodec/avfft.h"
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#include "audio.h"
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#include "avfilter.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;
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int winlen, tabsize;
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AVFrame *in, *out;
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RDFTContext *rdft, *irdft;
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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 - 1 ? 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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int i,j;
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s->rdft = av_rdft_init(wb, DFT_R2C);
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s->irdft = av_rdft_init(wb, IDFT_C2R);
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if (!s->rdft || !s->irdft)
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return AVERROR(ENOMEM);
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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, 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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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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float *nires;
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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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av_rdft_calc(s->rdft, s->irest);
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nires = s->ires;
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for (i = 0; i < tabsize; i++)
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nires[i] = s->irest[i];
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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 = s->fsamples;
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int ch, i;
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AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
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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->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 < s->winlen; 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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av_rdft_calc(s->rdft, fsamples);
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fsamples[0] = ires[0] * fsamples[0];
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fsamples[1] = ires[1] * fsamples[1];
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for (i = 1; i < s->tabsize / 2; i++) {
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float re, im;
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re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
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im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1];
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fsamples[i*2 ] = re;
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fsamples[i*2+1] = im;
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}
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av_rdft_calc(s->irdft, fsamples);
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for (i = 0; i < s->winlen; i++)
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dst[i] += fsamples[i] / s->tabsize * 2;
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for (i = s->winlen; i < s->tabsize; i++)
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dst[i] = fsamples[i] / s->tabsize * 2;
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for (i = 0; i < s->winlen; 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 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 query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if ((ret = ff_set_common_formats(ctx, formats)) < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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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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inlink->partial_buf_size =
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inlink->min_samples =
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inlink->max_samples = s->winlen;
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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_freep(&s->irest);
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av_freep(&s->ires);
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av_freep(&s->fsamples);
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av_rdft_end(s->rdft);
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av_rdft_end(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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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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{ NULL }
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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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{ NULL }
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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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AVFilter ff_af_superequalizer = {
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.name = "superequalizer",
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.description = NULL_IF_CONFIG_SMALL("Apply 18-th band equalization filter."),
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.priv_size = sizeof(SuperEqualizerContext),
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.priv_class = &superequalizer_class,
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.query_formats = query_formats,
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.init = init,
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.uninit = uninit,
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.inputs = superequalizer_inputs,
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.outputs = superequalizer_outputs,
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};
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@ -122,6 +122,7 @@ static void register_all(void)
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REGISTER_FILTER(SOFALIZER, sofalizer, af);
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REGISTER_FILTER(STEREOTOOLS, stereotools, af);
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REGISTER_FILTER(STEREOWIDEN, stereowiden, af);
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REGISTER_FILTER(SUPEREQUALIZER, superequalizer, af);
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REGISTER_FILTER(SURROUND, surround, af);
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REGISTER_FILTER(TREBLE, treble, af);
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REGISTER_FILTER(TREMOLO, tremolo, af);
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||||
@ -30,7 +30,7 @@
|
||||
#include "libavutil/version.h"
|
||||
|
||||
#define LIBAVFILTER_VERSION_MAJOR 6
|
||||
#define LIBAVFILTER_VERSION_MINOR 92
|
||||
#define LIBAVFILTER_VERSION_MINOR 93
|
||||
#define LIBAVFILTER_VERSION_MICRO 100
|
||||
|
||||
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
|
||||
|
||||
Loading…
Reference in New Issue
Block a user