1 /*
2 * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
3 *
4 * This file is part of FFmpeg.
5 *
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/opt.h"
22 #include "avfilter.h"
23 #include "internal.h"
24 #include "audio.h"
25
26 typedef struct BiquadCoeffs {
27 double a0, a1, a2, b1, b2;
28 } BiquadCoeffs;
29
30 typedef struct RIAACurve {
31 BiquadCoeffs r1;
32 BiquadCoeffs brickw;
33 int use_brickw;
34 } RIAACurve;
35
36 typedef struct AudioEmphasisContext {
37 const AVClass *class;
38 int mode, type;
39 double level_in, level_out;
40
41 RIAACurve rc;
42
43 AVFrame *w;
44 } AudioEmphasisContext;
45
46 #define OFFSET(x) offsetof(AudioEmphasisContext, x)
47 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
48
49 static const AVOption aemphasis_options[] = {
50 { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
51 { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
52 { "mode", "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "mode" },
53 { "reproduction", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
54 { "production", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
55 { "type", "set filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=4}, 0, 8, FLAGS, "type" },
56 { "col", "Columbia", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
57 { "emi", "EMI", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
58 { "bsi", "BSI (78RPM)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
59 { "riaa", "RIAA", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "type" },
60 { "cd", "Compact Disc (CD)", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, "type" },
61 { "50fm", "50µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, "type" },
62 { "75fm", "75µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, "type" },
63 { "50kf", "50µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, "type" },
64 { "75kf", "75µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, "type" },
65 { NULL }
66 };
67
68 AVFILTER_DEFINE_CLASS(aemphasis);
69
biquad_process(BiquadCoeffs * bq,double * dst,const double * src,int nb_samples,double * w,double level_in,double level_out)70 static inline void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples,
71 double *w, double level_in, double level_out)
72 {
73 const double a0 = bq->a0;
74 const double a1 = bq->a1;
75 const double a2 = bq->a2;
76 const double b1 = bq->b1;
77 const double b2 = bq->b2;
78 double w1 = w[0];
79 double w2 = w[1];
80
81 for (int i = 0; i < nb_samples; i++) {
82 double n = src[i] * level_in;
83 double tmp = n - w1 * b1 - w2 * b2;
84 double out = tmp * a0 + w1 * a1 + w2 * a2;
85
86 w2 = w1;
87 w1 = tmp;
88
89 dst[i] = out * level_out;
90 }
91
92 w[0] = w1;
93 w[1] = w2;
94 }
95
96 typedef struct ThreadData {
97 AVFrame *in, *out;
98 } ThreadData;
99
filter_channels(AVFilterContext * ctx,void * arg,int jobnr,int nb_jobs)100 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
101 {
102 AudioEmphasisContext *s = ctx->priv;
103 const double level_out = s->level_out;
104 const double level_in = s->level_in;
105 ThreadData *td = arg;
106 AVFrame *out = td->out;
107 AVFrame *in = td->in;
108 const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
109 const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
110
111 for (int ch = start; ch < end; ch++) {
112 const double *src = (const double *)in->extended_data[ch];
113 double *w = (double *)s->w->extended_data[ch];
114 double *dst = (double *)out->extended_data[ch];
115
116 if (s->rc.use_brickw) {
117 biquad_process(&s->rc.brickw, dst, src, in->nb_samples, w + 2, level_in, 1.);
118 biquad_process(&s->rc.r1, dst, dst, in->nb_samples, w, 1., level_out);
119 } else {
120 biquad_process(&s->rc.r1, dst, src, in->nb_samples, w, level_in, level_out);
121 }
122 }
123
124 return 0;
125 }
126
filter_frame(AVFilterLink * inlink,AVFrame * in)127 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
128 {
129 AVFilterContext *ctx = inlink->dst;
130 AVFilterLink *outlink = ctx->outputs[0];
131 ThreadData td;
132 AVFrame *out;
133
134 if (av_frame_is_writable(in)) {
135 out = in;
136 } else {
137 out = ff_get_audio_buffer(outlink, in->nb_samples);
138 if (!out) {
139 av_frame_free(&in);
140 return AVERROR(ENOMEM);
141 }
142 av_frame_copy_props(out, in);
143 }
144
145 td.in = in; td.out = out;
146 ff_filter_execute(ctx, filter_channels, &td, NULL,
147 FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
148
149 if (in != out)
150 av_frame_free(&in);
151 return ff_filter_frame(outlink, out);
152 }
153
set_highshelf_rbj(BiquadCoeffs * bq,double freq,double q,double peak,double sr)154 static inline void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
155 {
156 double A = sqrt(peak);
157 double w0 = freq * 2 * M_PI / sr;
158 double alpha = sin(w0) / (2 * q);
159 double cw0 = cos(w0);
160 double tmp = 2 * sqrt(A) * alpha;
161 double b0 = 0, ib0 = 0;
162
163 bq->a0 = A*( (A+1) + (A-1)*cw0 + tmp);
164 bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
165 bq->a2 = A*( (A+1) + (A-1)*cw0 - tmp);
166 b0 = (A+1) - (A-1)*cw0 + tmp;
167 bq->b1 = 2*( (A-1) - (A+1)*cw0);
168 bq->b2 = (A+1) - (A-1)*cw0 - tmp;
169
170 ib0 = 1 / b0;
171 bq->b1 *= ib0;
172 bq->b2 *= ib0;
173 bq->a0 *= ib0;
174 bq->a1 *= ib0;
175 bq->a2 *= ib0;
176 }
177
set_lp_rbj(BiquadCoeffs * bq,double fc,double q,double sr,double gain)178 static inline void set_lp_rbj(BiquadCoeffs *bq, double fc, double q, double sr, double gain)
179 {
180 double omega = 2.0 * M_PI * fc / sr;
181 double sn = sin(omega);
182 double cs = cos(omega);
183 double alpha = sn/(2 * q);
184 double inv = 1.0/(1.0 + alpha);
185
186 bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
187 bq->a1 = bq->a0 + bq->a0;
188 bq->b1 = (-2.0 * cs * inv);
189 bq->b2 = ((1.0 - alpha) * inv);
190 }
191
freq_gain(BiquadCoeffs * c,double freq,double sr)192 static double freq_gain(BiquadCoeffs *c, double freq, double sr)
193 {
194 double zr, zi;
195
196 freq *= 2.0 * M_PI / sr;
197 zr = cos(freq);
198 zi = -sin(freq);
199
200 /* |(a0 + a1*z + a2*z^2)/(1 + b1*z + b2*z^2)| */
201 return hypot(c->a0 + c->a1*zr + c->a2*(zr*zr-zi*zi), c->a1*zi + 2*c->a2*zr*zi) /
202 hypot(1 + c->b1*zr + c->b2*(zr*zr-zi*zi), c->b1*zi + 2*c->b2*zr*zi);
203 }
204
config_input(AVFilterLink * inlink)205 static int config_input(AVFilterLink *inlink)
206 {
207 double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
208 double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
209 AVFilterContext *ctx = inlink->dst;
210 AudioEmphasisContext *s = ctx->priv;
211 BiquadCoeffs coeffs;
212
213 if (!s->w)
214 s->w = ff_get_audio_buffer(inlink, 4);
215 if (!s->w)
216 return AVERROR(ENOMEM);
217
218 switch (s->type) {
219 case 0: //"Columbia"
220 i = 100.;
221 j = 500.;
222 k = 1590.;
223 break;
224 case 1: //"EMI"
225 i = 70.;
226 j = 500.;
227 k = 2500.;
228 break;
229 case 2: //"BSI(78rpm)"
230 i = 50.;
231 j = 353.;
232 k = 3180.;
233 break;
234 case 3: //"RIAA"
235 default:
236 tau1 = 0.003180;
237 tau2 = 0.000318;
238 tau3 = 0.000075;
239 i = 1. / (2. * M_PI * tau1);
240 j = 1. / (2. * M_PI * tau2);
241 k = 1. / (2. * M_PI * tau3);
242 break;
243 case 4: //"CD Mastering"
244 tau1 = 0.000050;
245 tau2 = 0.000015;
246 tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
247 i = 1. / (2. * M_PI * tau1);
248 j = 1. / (2. * M_PI * tau2);
249 k = 1. / (2. * M_PI * tau3);
250 break;
251 case 5: //"50µs FM (Europe)"
252 tau1 = 0.000050;
253 tau2 = tau1 / 20;// not used
254 tau3 = tau1 / 50;//
255 i = 1. / (2. * M_PI * tau1);
256 j = 1. / (2. * M_PI * tau2);
257 k = 1. / (2. * M_PI * tau3);
258 break;
259 case 6: //"75µs FM (US)"
260 tau1 = 0.000075;
261 tau2 = tau1 / 20;// not used
262 tau3 = tau1 / 50;//
263 i = 1. / (2. * M_PI * tau1);
264 j = 1. / (2. * M_PI * tau2);
265 k = 1. / (2. * M_PI * tau3);
266 break;
267 }
268
269 i *= 2 * M_PI;
270 j *= 2 * M_PI;
271 k *= 2 * M_PI;
272
273 t = 1. / sr;
274
275 //swap a1 b1, a2 b2
276 if (s->type == 7 || s->type == 8) {
277 double tau = (s->type == 7 ? 0.000050 : 0.000075);
278 double f = 1.0 / (2 * M_PI * tau);
279 double nyq = sr * 0.5;
280 double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
281 double cfreq = sqrt((gain - 1.0) * f * f); // frequency
282 double q = 1.0;
283
284 if (s->type == 8)
285 q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
286 if (s->type == 7)
287 q = pow((sr / 4750.0) + 19.5, -0.25);
288 if (s->mode == 0)
289 set_highshelf_rbj(&s->rc.r1, cfreq, q, 1. / gain, sr);
290 else
291 set_highshelf_rbj(&s->rc.r1, cfreq, q, gain, sr);
292 s->rc.use_brickw = 0;
293 } else {
294 s->rc.use_brickw = 1;
295 if (s->mode == 0) { // Reproduction
296 g = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
297 a0 = (2.*t+j*t*t)*g;
298 a1 = (2.*j*t*t)*g;
299 a2 = (-2.*t+j*t*t)*g;
300 b1 = (-8.+2.*i*k*t*t)*g;
301 b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
302 } else { // Production
303 g = 1. / (2.*t+j*t*t);
304 a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
305 a1 = (-8.+2.*i*k*t*t)*g;
306 a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
307 b1 = (2.*j*t*t)*g;
308 b2 = (-2.*t+j*t*t)*g;
309 }
310
311 coeffs.a0 = a0;
312 coeffs.a1 = a1;
313 coeffs.a2 = a2;
314 coeffs.b1 = b1;
315 coeffs.b2 = b2;
316
317 // the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
318 // find actual gain
319 // Note: for FM emphasis, use 100 Hz for normalization instead
320 gain1kHz = freq_gain(&coeffs, 1000.0, sr);
321 // divide one filter's x[n-m] coefficients by that value
322 gc = 1.0 / gain1kHz;
323 s->rc.r1.a0 = coeffs.a0 * gc;
324 s->rc.r1.a1 = coeffs.a1 * gc;
325 s->rc.r1.a2 = coeffs.a2 * gc;
326 s->rc.r1.b1 = coeffs.b1;
327 s->rc.r1.b2 = coeffs.b2;
328 }
329
330 cutfreq = FFMIN(0.45 * sr, 21000.);
331 set_lp_rbj(&s->rc.brickw, cutfreq, 0.707, sr, 1.);
332
333 return 0;
334 }
335
process_command(AVFilterContext * ctx,const char * cmd,const char * args,char * res,int res_len,int flags)336 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
337 char *res, int res_len, int flags)
338 {
339 int ret;
340
341 ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
342 if (ret < 0)
343 return ret;
344
345 return config_input(ctx->inputs[0]);
346 }
347
uninit(AVFilterContext * ctx)348 static av_cold void uninit(AVFilterContext *ctx)
349 {
350 AudioEmphasisContext *s = ctx->priv;
351
352 av_frame_free(&s->w);
353 }
354
355 static const AVFilterPad avfilter_af_aemphasis_inputs[] = {
356 {
357 .name = "default",
358 .type = AVMEDIA_TYPE_AUDIO,
359 .config_props = config_input,
360 .filter_frame = filter_frame,
361 },
362 };
363
364 static const AVFilterPad avfilter_af_aemphasis_outputs[] = {
365 {
366 .name = "default",
367 .type = AVMEDIA_TYPE_AUDIO,
368 },
369 };
370
371 const AVFilter ff_af_aemphasis = {
372 .name = "aemphasis",
373 .description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
374 .priv_size = sizeof(AudioEmphasisContext),
375 .priv_class = &aemphasis_class,
376 .uninit = uninit,
377 FILTER_INPUTS(avfilter_af_aemphasis_inputs),
378 FILTER_OUTPUTS(avfilter_af_aemphasis_outputs),
379 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
380 .process_command = process_command,
381 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
382 AVFILTER_FLAG_SLICE_THREADS,
383 };
384