1 /*
2 * Copyright (c) 2002 Naoki Shibata
3 * Copyright (c) 2017 Paul B Mahol
4 *
5 * This file is part of FFmpeg.
6 *
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.
11 *
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.
16 *
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
20 */
21
22 #include "libavutil/opt.h"
23 #include "libavutil/tx.h"
24
25 #include "audio.h"
26 #include "avfilter.h"
27 #include "filters.h"
28 #include "internal.h"
29
30 #define NBANDS 17
31 #define M 15
32
33 typedef struct EqParameter {
34 float lower, upper, gain;
35 } EqParameter;
36
37 typedef struct SuperEqualizerContext {
38 const AVClass *class;
39
40 EqParameter params[NBANDS + 1];
41
42 float gains[NBANDS + 1];
43
44 float fact[M + 1];
45 float aa;
46 float iza;
47 float *ires, *irest;
48 float *fsamples, *fsamples_out;
49 int winlen, tabsize;
50
51 AVFrame *in, *out;
52 AVTXContext *rdft, *irdft;
53 av_tx_fn tx_fn, itx_fn;
54 } SuperEqualizerContext;
55
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
59 };
60
izero(SuperEqualizerContext * s,float x)61 static float izero(SuperEqualizerContext *s, float x)
62 {
63 float ret = 1;
64 int m;
65
66 for (m = 1; m <= M; m++) {
67 float t;
68
69 t = pow(x / 2, m) / s->fact[m];
70 ret += t*t;
71 }
72
73 return ret;
74 }
75
hn_lpf(int n,float f,float fs)76 static float hn_lpf(int n, float f, float fs)
77 {
78 float t = 1 / fs;
79 float omega = 2 * M_PI * f;
80
81 if (n * omega * t == 0)
82 return 2 * f * t;
83 return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
84 }
85
hn_imp(int n)86 static float hn_imp(int n)
87 {
88 return n == 0 ? 1.f : 0.f;
89 }
90
hn(int n,EqParameter * param,float fs)91 static float hn(int n, EqParameter *param, float fs)
92 {
93 float ret, lhn;
94 int i;
95
96 lhn = hn_lpf(n, param[0].upper, fs);
97 ret = param[0].gain*lhn;
98
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);
102 lhn = lhn2;
103 }
104
105 ret += param[i].gain * (hn_imp(n) - lhn);
106
107 return ret;
108 }
109
alpha(float a)110 static float alpha(float a)
111 {
112 if (a <= 21)
113 return 0;
114 if (a <= 50)
115 return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
116 return .1102f * (a - 8.7f);
117 }
118
win(SuperEqualizerContext * s,float n,int N)119 static float win(SuperEqualizerContext *s, float n, int N)
120 {
121 return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
122 }
123
process_param(float * bc,EqParameter * param,float fs)124 static void process_param(float *bc, EqParameter *param, float fs)
125 {
126 int i;
127
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];
132 }
133 }
134
equ_init(SuperEqualizerContext * s,int wb)135 static int equ_init(SuperEqualizerContext *s, int wb)
136 {
137 float scale = 1.f, iscale = 1.f;
138 int i, j, ret;
139
140 ret = av_tx_init(&s->rdft, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, 1 << wb, &scale, 0);
141 if (ret < 0)
142 return ret;
143
144 ret = av_tx_init(&s->irdft, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, 1 << wb, &iscale, 0);
145 if (ret < 0)
146 return ret;
147
148 s->aa = 96;
149 s->winlen = (1 << (wb-1))-1;
150 s->tabsize = 1 << wb;
151
152 s->ires = av_calloc(s->tabsize + 2, sizeof(float));
153 s->irest = av_calloc(s->tabsize, sizeof(float));
154 s->fsamples = av_calloc(s->tabsize, sizeof(float));
155 s->fsamples_out = av_calloc(s->tabsize + 2, sizeof(float));
156 if (!s->ires || !s->irest || !s->fsamples || !s->fsamples_out)
157 return AVERROR(ENOMEM);
158
159 for (i = 0; i <= M; i++) {
160 s->fact[i] = 1;
161 for (j = 1; j <= i; j++)
162 s->fact[i] *= j;
163 }
164
165 s->iza = izero(s, alpha(s->aa));
166
167 return 0;
168 }
169
make_fir(SuperEqualizerContext * s,float * lbc,float * rbc,EqParameter * param,float fs)170 static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
171 {
172 const int winlen = s->winlen;
173 const int tabsize = s->tabsize;
174 int i;
175
176 if (fs <= 0)
177 return;
178
179 process_param(lbc, param, fs);
180 for (i = 0; i < winlen; i++)
181 s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
182 for (; i < tabsize; i++)
183 s->irest[i] = 0;
184
185 s->tx_fn(s->rdft, s->ires, s->irest, sizeof(float));
186 }
187
filter_frame(AVFilterLink * inlink,AVFrame * in)188 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
189 {
190 AVFilterContext *ctx = inlink->dst;
191 SuperEqualizerContext *s = ctx->priv;
192 AVFilterLink *outlink = ctx->outputs[0];
193 const float *ires = s->ires;
194 float *fsamples_out = s->fsamples_out;
195 float *fsamples = s->fsamples;
196 int ch, i;
197
198 AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
199 float *src, *dst, *ptr;
200
201 if (!out) {
202 av_frame_free(&in);
203 return AVERROR(ENOMEM);
204 }
205
206 for (ch = 0; ch < in->ch_layout.nb_channels; ch++) {
207 ptr = (float *)out->extended_data[ch];
208 dst = (float *)s->out->extended_data[ch];
209 src = (float *)in->extended_data[ch];
210
211 for (i = 0; i < in->nb_samples; i++)
212 fsamples[i] = src[i];
213 for (; i < s->tabsize; i++)
214 fsamples[i] = 0;
215
216 s->tx_fn(s->rdft, fsamples_out, fsamples, sizeof(float));
217
218 for (i = 0; i <= s->tabsize / 2; i++) {
219 float re, im;
220
221 re = ires[i*2 ] * fsamples_out[i*2] - ires[i*2+1] * fsamples_out[i*2+1];
222 im = ires[i*2+1] * fsamples_out[i*2] + ires[i*2 ] * fsamples_out[i*2+1];
223
224 fsamples_out[i*2 ] = re;
225 fsamples_out[i*2+1] = im;
226 }
227
228 s->itx_fn(s->irdft, fsamples, fsamples_out, sizeof(float));
229
230 for (i = 0; i < s->winlen; i++)
231 dst[i] += fsamples[i] / s->tabsize;
232 for (i = s->winlen; i < s->tabsize; i++)
233 dst[i] = fsamples[i] / s->tabsize;
234 for (i = 0; i < out->nb_samples; i++)
235 ptr[i] = dst[i];
236 for (i = 0; i < s->winlen; i++)
237 dst[i] = dst[i+s->winlen];
238 }
239
240 out->pts = in->pts;
241 av_frame_free(&in);
242
243 return ff_filter_frame(outlink, out);
244 }
245
activate(AVFilterContext * ctx)246 static int activate(AVFilterContext *ctx)
247 {
248 AVFilterLink *inlink = ctx->inputs[0];
249 AVFilterLink *outlink = ctx->outputs[0];
250 SuperEqualizerContext *s = ctx->priv;
251 AVFrame *in = NULL;
252 int ret;
253
254 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
255
256 ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
257 if (ret < 0)
258 return ret;
259 if (ret > 0)
260 return filter_frame(inlink, in);
261
262 FF_FILTER_FORWARD_STATUS(inlink, outlink);
263 FF_FILTER_FORWARD_WANTED(outlink, inlink);
264
265 return FFERROR_NOT_READY;
266 }
267
init(AVFilterContext * ctx)268 static av_cold int init(AVFilterContext *ctx)
269 {
270 SuperEqualizerContext *s = ctx->priv;
271
272 return equ_init(s, 14);
273 }
274
config_input(AVFilterLink * inlink)275 static int config_input(AVFilterLink *inlink)
276 {
277 AVFilterContext *ctx = inlink->dst;
278 SuperEqualizerContext *s = ctx->priv;
279
280 s->out = ff_get_audio_buffer(inlink, s->tabsize);
281 if (!s->out)
282 return AVERROR(ENOMEM);
283
284 return 0;
285 }
286
config_output(AVFilterLink * outlink)287 static int config_output(AVFilterLink *outlink)
288 {
289 AVFilterContext *ctx = outlink->src;
290 SuperEqualizerContext *s = ctx->priv;
291
292 make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
293
294 return 0;
295 }
296
uninit(AVFilterContext * ctx)297 static av_cold void uninit(AVFilterContext *ctx)
298 {
299 SuperEqualizerContext *s = ctx->priv;
300
301 av_frame_free(&s->out);
302 av_freep(&s->irest);
303 av_freep(&s->ires);
304 av_freep(&s->fsamples);
305 av_freep(&s->fsamples_out);
306 av_tx_uninit(&s->rdft);
307 av_tx_uninit(&s->irdft);
308 }
309
310 static const AVFilterPad superequalizer_inputs[] = {
311 {
312 .name = "default",
313 .type = AVMEDIA_TYPE_AUDIO,
314 .config_props = config_input,
315 },
316 };
317
318 static const AVFilterPad superequalizer_outputs[] = {
319 {
320 .name = "default",
321 .type = AVMEDIA_TYPE_AUDIO,
322 .config_props = config_output,
323 },
324 };
325
326 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
327 #define OFFSET(x) offsetof(SuperEqualizerContext, x)
328
329 static const AVOption superequalizer_options[] = {
330 { "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
331 { "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
332 { "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
333 { "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
334 { "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
335 { "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
336 { "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
337 { "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
338 { "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
339 { "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
340 { "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
341 { "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
342 { "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
343 { "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
344 { "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
345 { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
346 { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
347 { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
348 { NULL }
349 };
350
351 AVFILTER_DEFINE_CLASS(superequalizer);
352
353 const AVFilter ff_af_superequalizer = {
354 .name = "superequalizer",
355 .description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
356 .priv_size = sizeof(SuperEqualizerContext),
357 .priv_class = &superequalizer_class,
358 .init = init,
359 .activate = activate,
360 .uninit = uninit,
361 FILTER_INPUTS(superequalizer_inputs),
362 FILTER_OUTPUTS(superequalizer_outputs),
363 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
364 };
365