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