2 * Copyright (c) 2002 Naoki Shibata
3 * Copyright (c) 2017 Paul B Mahol
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 #include "libavutil/opt.h"
24 #include "libavcodec/avfft.h"
34 typedef struct EqParameter {
35 float lower, upper, gain;
38 typedef struct SuperEqualizerContext {
41 EqParameter params[NBANDS + 1];
43 float gains[NBANDS + 1];
53 RDFTContext *rdft, *irdft;
54 } SuperEqualizerContext;
56 static const float bands[] = {
57 65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
58 1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
61 static float izero(SuperEqualizerContext *s, float x)
66 for (m = 1; m <= M; m++) {
69 t = pow(x / 2, m) / s->fact[m];
76 static float hn_lpf(int n, float f, float fs)
79 float omega = 2 * M_PI * f;
81 if (n * omega * t == 0)
83 return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
86 static float hn_imp(int n)
88 return n == 0 ? 1.f : 0.f;
91 static float hn(int n, EqParameter *param, float fs)
96 lhn = hn_lpf(n, param[0].upper, fs);
97 ret = param[0].gain*lhn;
99 for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
100 float lhn2 = hn_lpf(n, param[i].upper, fs);
101 ret += param[i].gain * (lhn2 - lhn);
105 ret += param[i].gain * (hn_imp(n) - lhn);
110 static float alpha(float a)
115 return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
116 return .1102f * (a - 8.7f);
119 static float win(SuperEqualizerContext *s, float n, int N)
121 return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
124 static void process_param(float *bc, EqParameter *param, float fs)
128 for (i = 0; i <= NBANDS; i++) {
129 param[i].lower = i == 0 ? 0 : bands[i - 1];
130 param[i].upper = i == NBANDS ? fs : bands[i];
131 param[i].gain = bc[i];
135 static int equ_init(SuperEqualizerContext *s, int wb)
139 s->rdft = av_rdft_init(wb, DFT_R2C);
140 s->irdft = av_rdft_init(wb, IDFT_C2R);
141 if (!s->rdft || !s->irdft)
142 return AVERROR(ENOMEM);
145 s->winlen = (1 << (wb-1))-1;
146 s->tabsize = 1 << wb;
148 s->ires = av_calloc(s->tabsize, sizeof(float));
149 s->irest = av_calloc(s->tabsize, sizeof(float));
150 s->fsamples = av_calloc(s->tabsize, sizeof(float));
152 for (i = 0; i <= M; i++) {
154 for (j = 1; j <= i; j++)
158 s->iza = izero(s, alpha(s->aa));
163 static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
165 const int winlen = s->winlen;
166 const int tabsize = s->tabsize;
173 process_param(lbc, param, fs);
174 for (i = 0; i < winlen; i++)
175 s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
176 for (; i < tabsize; i++)
179 av_rdft_calc(s->rdft, s->irest);
181 for (i = 0; i < tabsize; i++)
182 nires[i] = s->irest[i];
185 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
187 AVFilterContext *ctx = inlink->dst;
188 SuperEqualizerContext *s = ctx->priv;
189 AVFilterLink *outlink = ctx->outputs[0];
190 const float *ires = s->ires;
191 float *fsamples = s->fsamples;
194 AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
195 float *src, *dst, *ptr;
199 return AVERROR(ENOMEM);
202 for (ch = 0; ch < in->channels; ch++) {
203 ptr = (float *)out->extended_data[ch];
204 dst = (float *)s->out->extended_data[ch];
205 src = (float *)in->extended_data[ch];
207 for (i = 0; i < in->nb_samples; i++)
208 fsamples[i] = src[i];
209 for (; i < s->tabsize; i++)
212 av_rdft_calc(s->rdft, fsamples);
214 fsamples[0] = ires[0] * fsamples[0];
215 fsamples[1] = ires[1] * fsamples[1];
216 for (i = 1; i < s->tabsize / 2; i++) {
219 re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
220 im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1];
223 fsamples[i*2+1] = im;
226 av_rdft_calc(s->irdft, fsamples);
228 for (i = 0; i < s->winlen; i++)
229 dst[i] += fsamples[i] / s->tabsize * 2;
230 for (i = s->winlen; i < s->tabsize; i++)
231 dst[i] = fsamples[i] / s->tabsize * 2;
232 for (i = 0; i < s->winlen; i++)
234 for (i = 0; i < s->winlen; i++)
235 dst[i] = dst[i+s->winlen];
241 return ff_filter_frame(outlink, out);
244 static int activate(AVFilterContext *ctx)
246 AVFilterLink *inlink = ctx->inputs[0];
247 AVFilterLink *outlink = ctx->outputs[0];
248 SuperEqualizerContext *s = ctx->priv;
252 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
254 ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
258 return filter_frame(inlink, in);
260 FF_FILTER_FORWARD_STATUS(inlink, outlink);
261 FF_FILTER_FORWARD_WANTED(outlink, inlink);
263 return FFERROR_NOT_READY;
266 static av_cold int init(AVFilterContext *ctx)
268 SuperEqualizerContext *s = ctx->priv;
270 return equ_init(s, 14);
273 static int query_formats(AVFilterContext *ctx)
275 AVFilterFormats *formats;
276 AVFilterChannelLayouts *layouts;
277 static const enum AVSampleFormat sample_fmts[] = {
283 layouts = ff_all_channel_counts();
285 return AVERROR(ENOMEM);
286 ret = ff_set_common_channel_layouts(ctx, layouts);
290 formats = ff_make_format_list(sample_fmts);
291 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
294 formats = ff_all_samplerates();
295 return ff_set_common_samplerates(ctx, formats);
298 static int config_input(AVFilterLink *inlink)
300 AVFilterContext *ctx = inlink->dst;
301 SuperEqualizerContext *s = ctx->priv;
303 s->out = ff_get_audio_buffer(inlink, s->tabsize);
305 return AVERROR(ENOMEM);
310 static int config_output(AVFilterLink *outlink)
312 AVFilterContext *ctx = outlink->src;
313 SuperEqualizerContext *s = ctx->priv;
315 make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
320 static av_cold void uninit(AVFilterContext *ctx)
322 SuperEqualizerContext *s = ctx->priv;
324 av_frame_free(&s->out);
327 av_freep(&s->fsamples);
328 av_rdft_end(s->rdft);
329 av_rdft_end(s->irdft);
332 static const AVFilterPad superequalizer_inputs[] = {
335 .type = AVMEDIA_TYPE_AUDIO,
336 .config_props = config_input,
341 static const AVFilterPad superequalizer_outputs[] = {
344 .type = AVMEDIA_TYPE_AUDIO,
345 .config_props = config_output,
350 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
351 #define OFFSET(x) offsetof(SuperEqualizerContext, x)
353 static const AVOption superequalizer_options[] = {
354 { "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
355 { "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
356 { "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
357 { "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
358 { "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
359 { "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
360 { "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
361 { "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
362 { "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
363 { "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
364 { "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
365 { "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
366 { "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
367 { "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
368 { "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
369 { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
370 { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
371 { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
375 AVFILTER_DEFINE_CLASS(superequalizer);
377 const AVFilter ff_af_superequalizer = {
378 .name = "superequalizer",
379 .description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
380 .priv_size = sizeof(SuperEqualizerContext),
381 .priv_class = &superequalizer_class,
382 .query_formats = query_formats,
384 .activate = activate,
386 .inputs = superequalizer_inputs,
387 .outputs = superequalizer_outputs,