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"
33 typedef struct EqParameter {
34 float lower, upper, gain;
37 typedef struct SuperEqualizerContext {
40 EqParameter params[NBANDS + 1];
42 float gains[NBANDS + 1];
52 RDFTContext *rdft, *irdft;
53 } SuperEqualizerContext;
55 static const float bands[] = {
56 65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
57 1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
60 static float izero(SuperEqualizerContext *s, float x)
65 for (m = 1; m <= M; m++) {
68 t = pow(x / 2, m) / s->fact[m];
75 static float hn_lpf(int n, float f, float fs)
78 float omega = 2 * M_PI * f;
80 if (n * omega * t == 0)
82 return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
85 static float hn_imp(int n)
87 return n == 0 ? 1.f : 0.f;
90 static float hn(int n, EqParameter *param, float fs)
95 lhn = hn_lpf(n, param[0].upper, fs);
96 ret = param[0].gain*lhn;
98 for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
99 float lhn2 = hn_lpf(n, param[i].upper, fs);
100 ret += param[i].gain * (lhn2 - lhn);
104 ret += param[i].gain * (hn_imp(n) - lhn);
109 static float alpha(float a)
114 return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
115 return .1102f * (a - 8.7f);
118 static float win(SuperEqualizerContext *s, float n, int N)
120 return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
123 static void process_param(float *bc, EqParameter *param, float fs)
127 for (i = 0; i <= NBANDS; i++) {
128 param[i].lower = i == 0 ? 0 : bands[i - 1];
129 param[i].upper = i == NBANDS ? fs : bands[i];
130 param[i].gain = bc[i];
134 static int equ_init(SuperEqualizerContext *s, int wb)
138 s->rdft = av_rdft_init(wb, DFT_R2C);
139 s->irdft = av_rdft_init(wb, IDFT_C2R);
140 if (!s->rdft || !s->irdft)
141 return AVERROR(ENOMEM);
144 s->winlen = (1 << (wb-1))-1;
145 s->tabsize = 1 << wb;
147 s->ires = av_calloc(s->tabsize, sizeof(float));
148 s->irest = av_calloc(s->tabsize, sizeof(float));
149 s->fsamples = av_calloc(s->tabsize, sizeof(float));
151 for (i = 0; i <= M; i++) {
153 for (j = 1; j <= i; j++)
157 s->iza = izero(s, alpha(s->aa));
162 static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
164 const int winlen = s->winlen;
165 const int tabsize = s->tabsize;
172 process_param(lbc, param, fs);
173 for (i = 0; i < winlen; i++)
174 s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
175 for (; i < tabsize; i++)
178 av_rdft_calc(s->rdft, s->irest);
180 for (i = 0; i < tabsize; i++)
181 nires[i] = s->irest[i];
184 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
186 AVFilterContext *ctx = inlink->dst;
187 SuperEqualizerContext *s = ctx->priv;
188 AVFilterLink *outlink = ctx->outputs[0];
189 const float *ires = s->ires;
190 float *fsamples = s->fsamples;
193 AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
194 float *src, *dst, *ptr;
198 return AVERROR(ENOMEM);
201 for (ch = 0; ch < in->channels; ch++) {
202 ptr = (float *)out->extended_data[ch];
203 dst = (float *)s->out->extended_data[ch];
204 src = (float *)in->extended_data[ch];
206 for (i = 0; i < s->winlen; i++)
207 fsamples[i] = src[i];
208 for (; i < s->tabsize; i++)
211 av_rdft_calc(s->rdft, fsamples);
213 fsamples[0] = ires[0] * fsamples[0];
214 fsamples[1] = ires[1] * fsamples[1];
215 for (i = 1; i < s->tabsize / 2; i++) {
218 re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
219 im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1];
222 fsamples[i*2+1] = im;
225 av_rdft_calc(s->irdft, fsamples);
227 for (i = 0; i < s->winlen; i++)
228 dst[i] += fsamples[i] / s->tabsize * 2;
229 for (i = s->winlen; i < s->tabsize; i++)
230 dst[i] = fsamples[i] / s->tabsize * 2;
231 for (i = 0; i < s->winlen; i++)
233 for (i = 0; i < s->winlen; i++)
234 dst[i] = dst[i+s->winlen];
240 return ff_filter_frame(outlink, out);
243 static av_cold int init(AVFilterContext *ctx)
245 SuperEqualizerContext *s = ctx->priv;
247 return equ_init(s, 14);
250 static int query_formats(AVFilterContext *ctx)
252 AVFilterFormats *formats;
253 AVFilterChannelLayouts *layouts;
254 static const enum AVSampleFormat sample_fmts[] = {
260 layouts = ff_all_channel_counts();
262 return AVERROR(ENOMEM);
263 ret = ff_set_common_channel_layouts(ctx, layouts);
267 formats = ff_make_format_list(sample_fmts);
268 if ((ret = ff_set_common_formats(ctx, formats)) < 0)
271 formats = ff_all_samplerates();
272 return ff_set_common_samplerates(ctx, formats);
275 static int config_input(AVFilterLink *inlink)
277 AVFilterContext *ctx = inlink->dst;
278 SuperEqualizerContext *s = ctx->priv;
280 inlink->partial_buf_size =
281 inlink->min_samples =
282 inlink->max_samples = s->winlen;
284 s->out = ff_get_audio_buffer(inlink, s->tabsize);
286 return AVERROR(ENOMEM);
291 static int config_output(AVFilterLink *outlink)
293 AVFilterContext *ctx = outlink->src;
294 SuperEqualizerContext *s = ctx->priv;
296 make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
301 static av_cold void uninit(AVFilterContext *ctx)
303 SuperEqualizerContext *s = ctx->priv;
305 av_frame_free(&s->out);
308 av_freep(&s->fsamples);
309 av_rdft_end(s->rdft);
310 av_rdft_end(s->irdft);
313 static const AVFilterPad superequalizer_inputs[] = {
316 .type = AVMEDIA_TYPE_AUDIO,
317 .filter_frame = filter_frame,
318 .config_props = config_input,
323 static const AVFilterPad superequalizer_outputs[] = {
326 .type = AVMEDIA_TYPE_AUDIO,
327 .config_props = config_output,
332 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
333 #define OFFSET(x) offsetof(SuperEqualizerContext, x)
335 static const AVOption superequalizer_options[] = {
336 { "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
337 { "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
338 { "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
339 { "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
340 { "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
341 { "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
342 { "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
343 { "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
344 { "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
345 { "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
346 { "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
347 { "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
348 { "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
349 { "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
350 { "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
351 { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
352 { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
353 { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
357 AVFILTER_DEFINE_CLASS(superequalizer);
359 AVFilter ff_af_superequalizer = {
360 .name = "superequalizer",
361 .description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
362 .priv_size = sizeof(SuperEqualizerContext),
363 .priv_class = &superequalizer_class,
364 .query_formats = query_formats,
367 .inputs = superequalizer_inputs,
368 .outputs = superequalizer_outputs,