/* * 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/opt.h" #include "libavcodec/avfft.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; int winlen, tabsize; AVFrame *in, *out; RDFTContext *rdft, *irdft; } 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) { int i,j; s->rdft = av_rdft_init(wb, DFT_R2C); s->irdft = av_rdft_init(wb, IDFT_C2R); if (!s->rdft || !s->irdft) return AVERROR(ENOMEM); s->aa = 96; s->winlen = (1 << (wb-1))-1; s->tabsize = 1 << wb; s->ires = av_calloc(s->tabsize, sizeof(float)); s->irest = av_calloc(s->tabsize, sizeof(float)); s->fsamples = av_calloc(s->tabsize, sizeof(float)); 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; float *nires; 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; av_rdft_calc(s->rdft, s->irest); nires = s->ires; for (i = 0; i < tabsize; i++) nires[i] = s->irest[i]; } 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 = s->fsamples; int ch, i; AVFrame *out = ff_get_audio_buffer(outlink, s->winlen); float *src, *dst, *ptr; if (!out) { av_frame_free(&in); return AVERROR(ENOMEM); } for (ch = 0; ch < in->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; av_rdft_calc(s->rdft, fsamples); fsamples[0] = ires[0] * fsamples[0]; fsamples[1] = ires[1] * fsamples[1]; for (i = 1; i < s->tabsize / 2; i++) { float re, im; re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1]; im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1]; fsamples[i*2 ] = re; fsamples[i*2+1] = im; } av_rdft_calc(s->irdft, fsamples); for (i = 0; i < s->winlen; i++) dst[i] += fsamples[i] / s->tabsize * 2; for (i = s->winlen; i < s->tabsize; i++) dst[i] = fsamples[i] / s->tabsize * 2; for (i = 0; i < s->winlen; 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 query_formats(AVFilterContext *ctx) { AVFilterFormats *formats; AVFilterChannelLayouts *layouts; static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }; int ret; layouts = ff_all_channel_counts(); if (!layouts) return AVERROR(ENOMEM); ret = ff_set_common_channel_layouts(ctx, layouts); if (ret < 0) return ret; formats = ff_make_format_list(sample_fmts); if ((ret = ff_set_common_formats(ctx, formats)) < 0) return ret; formats = ff_all_samplerates(); return ff_set_common_samplerates(ctx, formats); } 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_rdft_end(s->rdft); av_rdft_end(s->irdft); } static const AVFilterPad superequalizer_inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, }, { NULL } }; static const AVFilterPad superequalizer_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_output, }, { NULL } }; #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); AVFilter ff_af_superequalizer = { .name = "superequalizer", .description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."), .priv_size = sizeof(SuperEqualizerContext), .priv_class = &superequalizer_class, .query_formats = query_formats, .init = init, .activate = activate, .uninit = uninit, .inputs = superequalizer_inputs, .outputs = superequalizer_outputs, };