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->channels * jobnr) / nb_jobs;
109 const int end = (in->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 ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
147 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
query_formats(AVFilterContext * ctx)154 static int query_formats(AVFilterContext *ctx)
155 {
156 AVFilterChannelLayouts *layouts;
157 AVFilterFormats *formats;
158 static const enum AVSampleFormat sample_fmts[] = {
159 AV_SAMPLE_FMT_DBLP,
160 AV_SAMPLE_FMT_NONE
161 };
162 int ret;
163
164 layouts = ff_all_channel_counts();
165 if (!layouts)
166 return AVERROR(ENOMEM);
167 ret = ff_set_common_channel_layouts(ctx, layouts);
168 if (ret < 0)
169 return ret;
170
171 formats = ff_make_format_list(sample_fmts);
172 if (!formats)
173 return AVERROR(ENOMEM);
174 ret = ff_set_common_formats(ctx, formats);
175 if (ret < 0)
176 return ret;
177
178 formats = ff_all_samplerates();
179 if (!formats)
180 return AVERROR(ENOMEM);
181 return ff_set_common_samplerates(ctx, formats);
182 }
183
set_highshelf_rbj(BiquadCoeffs * bq,double freq,double q,double peak,double sr)184 static inline void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
185 {
186 double A = sqrt(peak);
187 double w0 = freq * 2 * M_PI / sr;
188 double alpha = sin(w0) / (2 * q);
189 double cw0 = cos(w0);
190 double tmp = 2 * sqrt(A) * alpha;
191 double b0 = 0, ib0 = 0;
192
193 bq->a0 = A*( (A+1) + (A-1)*cw0 + tmp);
194 bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
195 bq->a2 = A*( (A+1) + (A-1)*cw0 - tmp);
196 b0 = (A+1) - (A-1)*cw0 + tmp;
197 bq->b1 = 2*( (A-1) - (A+1)*cw0);
198 bq->b2 = (A+1) - (A-1)*cw0 - tmp;
199
200 ib0 = 1 / b0;
201 bq->b1 *= ib0;
202 bq->b2 *= ib0;
203 bq->a0 *= ib0;
204 bq->a1 *= ib0;
205 bq->a2 *= ib0;
206 }
207
set_lp_rbj(BiquadCoeffs * bq,double fc,double q,double sr,double gain)208 static inline void set_lp_rbj(BiquadCoeffs *bq, double fc, double q, double sr, double gain)
209 {
210 double omega = 2.0 * M_PI * fc / sr;
211 double sn = sin(omega);
212 double cs = cos(omega);
213 double alpha = sn/(2 * q);
214 double inv = 1.0/(1.0 + alpha);
215
216 bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
217 bq->a1 = bq->a0 + bq->a0;
218 bq->b1 = (-2.0 * cs * inv);
219 bq->b2 = ((1.0 - alpha) * inv);
220 }
221
freq_gain(BiquadCoeffs * c,double freq,double sr)222 static double freq_gain(BiquadCoeffs *c, double freq, double sr)
223 {
224 double zr, zi;
225
226 freq *= 2.0 * M_PI / sr;
227 zr = cos(freq);
228 zi = -sin(freq);
229
230 /* |(a0 + a1*z + a2*z^2)/(1 + b1*z + b2*z^2)| */
231 return hypot(c->a0 + c->a1*zr + c->a2*(zr*zr-zi*zi), c->a1*zi + 2*c->a2*zr*zi) /
232 hypot(1 + c->b1*zr + c->b2*(zr*zr-zi*zi), c->b1*zi + 2*c->b2*zr*zi);
233 }
234
config_input(AVFilterLink * inlink)235 static int config_input(AVFilterLink *inlink)
236 {
237 double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
238 double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
239 AVFilterContext *ctx = inlink->dst;
240 AudioEmphasisContext *s = ctx->priv;
241 BiquadCoeffs coeffs;
242
243 if (!s->w)
244 s->w = ff_get_audio_buffer(inlink, 4);
245 if (!s->w)
246 return AVERROR(ENOMEM);
247
248 switch (s->type) {
249 case 0: //"Columbia"
250 i = 100.;
251 j = 500.;
252 k = 1590.;
253 break;
254 case 1: //"EMI"
255 i = 70.;
256 j = 500.;
257 k = 2500.;
258 break;
259 case 2: //"BSI(78rpm)"
260 i = 50.;
261 j = 353.;
262 k = 3180.;
263 break;
264 case 3: //"RIAA"
265 default:
266 tau1 = 0.003180;
267 tau2 = 0.000318;
268 tau3 = 0.000075;
269 i = 1. / (2. * M_PI * tau1);
270 j = 1. / (2. * M_PI * tau2);
271 k = 1. / (2. * M_PI * tau3);
272 break;
273 case 4: //"CD Mastering"
274 tau1 = 0.000050;
275 tau2 = 0.000015;
276 tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
277 i = 1. / (2. * M_PI * tau1);
278 j = 1. / (2. * M_PI * tau2);
279 k = 1. / (2. * M_PI * tau3);
280 break;
281 case 5: //"50µs FM (Europe)"
282 tau1 = 0.000050;
283 tau2 = tau1 / 20;// not used
284 tau3 = tau1 / 50;//
285 i = 1. / (2. * M_PI * tau1);
286 j = 1. / (2. * M_PI * tau2);
287 k = 1. / (2. * M_PI * tau3);
288 break;
289 case 6: //"75µs FM (US)"
290 tau1 = 0.000075;
291 tau2 = tau1 / 20;// not used
292 tau3 = tau1 / 50;//
293 i = 1. / (2. * M_PI * tau1);
294 j = 1. / (2. * M_PI * tau2);
295 k = 1. / (2. * M_PI * tau3);
296 break;
297 }
298
299 i *= 2 * M_PI;
300 j *= 2 * M_PI;
301 k *= 2 * M_PI;
302
303 t = 1. / sr;
304
305 //swap a1 b1, a2 b2
306 if (s->type == 7 || s->type == 8) {
307 double tau = (s->type == 7 ? 0.000050 : 0.000075);
308 double f = 1.0 / (2 * M_PI * tau);
309 double nyq = sr * 0.5;
310 double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
311 double cfreq = sqrt((gain - 1.0) * f * f); // frequency
312 double q = 1.0;
313
314 if (s->type == 8)
315 q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
316 if (s->type == 7)
317 q = pow((sr / 4750.0) + 19.5, -0.25);
318 if (s->mode == 0)
319 set_highshelf_rbj(&s->rc.r1, cfreq, q, 1. / gain, sr);
320 else
321 set_highshelf_rbj(&s->rc.r1, cfreq, q, gain, sr);
322 s->rc.use_brickw = 0;
323 } else {
324 s->rc.use_brickw = 1;
325 if (s->mode == 0) { // Reproduction
326 g = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
327 a0 = (2.*t+j*t*t)*g;
328 a1 = (2.*j*t*t)*g;
329 a2 = (-2.*t+j*t*t)*g;
330 b1 = (-8.+2.*i*k*t*t)*g;
331 b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
332 } else { // Production
333 g = 1. / (2.*t+j*t*t);
334 a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
335 a1 = (-8.+2.*i*k*t*t)*g;
336 a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
337 b1 = (2.*j*t*t)*g;
338 b2 = (-2.*t+j*t*t)*g;
339 }
340
341 coeffs.a0 = a0;
342 coeffs.a1 = a1;
343 coeffs.a2 = a2;
344 coeffs.b1 = b1;
345 coeffs.b2 = b2;
346
347 // the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
348 // find actual gain
349 // Note: for FM emphasis, use 100 Hz for normalization instead
350 gain1kHz = freq_gain(&coeffs, 1000.0, sr);
351 // divide one filter's x[n-m] coefficients by that value
352 gc = 1.0 / gain1kHz;
353 s->rc.r1.a0 = coeffs.a0 * gc;
354 s->rc.r1.a1 = coeffs.a1 * gc;
355 s->rc.r1.a2 = coeffs.a2 * gc;
356 s->rc.r1.b1 = coeffs.b1;
357 s->rc.r1.b2 = coeffs.b2;
358 }
359
360 cutfreq = FFMIN(0.45 * sr, 21000.);
361 set_lp_rbj(&s->rc.brickw, cutfreq, 0.707, sr, 1.);
362
363 return 0;
364 }
365
process_command(AVFilterContext * ctx,const char * cmd,const char * args,char * res,int res_len,int flags)366 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
367 char *res, int res_len, int flags)
368 {
369 int ret;
370
371 ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
372 if (ret < 0)
373 return ret;
374
375 return config_input(ctx->inputs[0]);
376 }
377
uninit(AVFilterContext * ctx)378 static av_cold void uninit(AVFilterContext *ctx)
379 {
380 AudioEmphasisContext *s = ctx->priv;
381
382 av_frame_free(&s->w);
383 }
384
385 static const AVFilterPad avfilter_af_aemphasis_inputs[] = {
386 {
387 .name = "default",
388 .type = AVMEDIA_TYPE_AUDIO,
389 .config_props = config_input,
390 .filter_frame = filter_frame,
391 },
392 { NULL }
393 };
394
395 static const AVFilterPad avfilter_af_aemphasis_outputs[] = {
396 {
397 .name = "default",
398 .type = AVMEDIA_TYPE_AUDIO,
399 },
400 { NULL }
401 };
402
403 AVFilter ff_af_aemphasis = {
404 .name = "aemphasis",
405 .description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
406 .priv_size = sizeof(AudioEmphasisContext),
407 .priv_class = &aemphasis_class,
408 .uninit = uninit,
409 .query_formats = query_formats,
410 .inputs = avfilter_af_aemphasis_inputs,
411 .outputs = avfilter_af_aemphasis_outputs,
412 .process_command = process_command,
413 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
414 AVFILTER_FLAG_SLICE_THREADS,
415 };
416