| /* |
| * 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, |
| }; |
