FFmpeg
af_aemphasis.c
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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, .unit = "mode" },
53  { "reproduction", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "mode" },
54  { "production", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "mode" },
55  { "type", "set filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=4}, 0, 8, FLAGS, .unit = "type" },
56  { "col", "Columbia", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "type" },
57  { "emi", "EMI", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "type" },
58  { "bsi", "BSI (78RPM)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, .unit = "type" },
59  { "riaa", "RIAA", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, .unit = "type" },
60  { "cd", "Compact Disc (CD)", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, .unit = "type" },
61  { "50fm", "50µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, .unit = "type" },
62  { "75fm", "75µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, .unit = "type" },
63  { "50kf", "50µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, .unit = "type" },
64  { "75kf", "75µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, .unit = "type" },
65  { NULL }
66 };
67 
68 AVFILTER_DEFINE_CLASS(aemphasis);
69 
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 
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 
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 
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 
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 
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 
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 
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 
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 const AVFilter ff_af_aemphasis = {
365  .name = "aemphasis",
366  .description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
367  .priv_size = sizeof(AudioEmphasisContext),
368  .priv_class = &aemphasis_class,
369  .uninit = uninit,
370  FILTER_INPUTS(avfilter_af_aemphasis_inputs),
371  FILTER_OUTPUTS(ff_audio_default_filterpad),
372  FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
373  .process_command = process_command,
374  .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
375  AVFILTER_FLAG_SLICE_THREADS,
376 };
A
#define A(x)
Definition: vpx_arith.h:28
biquad_process
static void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples, double *w, double level_in, double level_out)
Definition: af_aemphasis.c:70
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:97
td
#define td
Definition: regdef.h:70
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
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FILE * out
Definition: movenc.c:55
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static double freq_gain(BiquadCoeffs *c, double freq, double sr)
Definition: af_aemphasis.c:192
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1015
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#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
Definition: internal.h:175
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Definition: filter_design.txt:212
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#define OFFSET(x)
Definition: af_aemphasis.c:46
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void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:160
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This structure describes decoded (raw) audio or video data.
Definition: frame.h:375
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Definition: opt.h:346
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Definition: avfilter.h:170
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int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:313
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Definition: af_aemphasis.c:27
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Definition: af_aemphasis.c:27
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int type
Definition: af_aemphasis.c:38
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int mode
Definition: af_aemphasis.c:38
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Definition: af_aemphasis.c:27
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AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:776
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Definition: writing_filters.txt:86
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A filter pad used for either input or output.
Definition: internal.h:33
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Definition: regdef.h:47
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Definition: attributes.h:90
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static const AVOption aemphasis_options[]
Definition: af_aemphasis.c:49
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@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:237
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@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
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static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_aemphasis.c:336
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static const AVFilterPad avfilter_af_aemphasis_inputs[]
Definition: af_aemphasis.c:355
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Definition: movenc.c:49
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#define FILTER_INPUTS(array)
Definition: internal.h:182
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Definition: filter_design.txt:179
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Definition: coverity.c:32
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Copy only "metadata" fields from src to dst.
Definition: frame.c:709
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Definition: af_aemphasis.c:43
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An AVFilterPad array whose only entry has name "default" and is of type AVMEDIA_TYPE_AUDIO.
Definition: audio.c:33
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
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Definition: af_aemphasis.c:100
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Definition: af_aemphasis.c:39
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Definition: af_aemphasis.c:178
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Definition: af_crystalizer.c:121
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Definition: af_aemphasis.c:33
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Definition: libm.h:366
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Definition: frame.c:645
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Generic processing of user supplied commands that are set in the same way as the filter options.
Definition: avfilter.c:887
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Definition: af_aemphasis.c:36
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Definition: regdef.h:46
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#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
Definition: avfilter.h:147
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:455
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
set_highshelf_rbj
static void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
Definition: af_aemphasis.c:154
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:436
a2
#define a2
Definition: regdef.h:48
ff_filter_get_nb_threads
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:827
AudioEmphasisContext::rc
RIAACurve rc
Definition: af_aemphasis.c:41
ThreadData
Used for passing data between threads.
Definition: dsddec.c:71
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:39
FLAGS
#define FLAGS
Definition: af_aemphasis.c:47
BiquadCoeffs::b1
double b1
Definition: af_aemphasis.c:27
AVFilter
Filter definition.
Definition: avfilter.h:166
ret
ret
Definition: filter_design.txt:187
mode
mode
Definition: ebur128.h:83
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:235
avfilter.h
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:67
RIAACurve::brickw
BiquadCoeffs brickw
Definition: af_aemphasis.c:32
AVFilterContext
An instance of a filter.
Definition: avfilter.h:407
AVFILTER_FLAG_SLICE_THREADS
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
audio.h
RIAACurve::r1
BiquadCoeffs r1
Definition: af_aemphasis.c:31
alpha
static const int16_t alpha[]
Definition: ilbcdata.h:55
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:183
src
INIT_CLIP pixel * src
Definition: h264pred_template.c:418
RIAACurve
Definition: af_aemphasis.c:30
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_aemphasis.c:205
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_aemphasis.c:348
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:474
b0
static double b0(void *priv, double x, double y)
Definition: vf_xfade.c:2034
ff_af_aemphasis
const AVFilter ff_af_aemphasis
Definition: af_aemphasis.c:364
AudioEmphasisContext::level_out
double level_out
Definition: af_aemphasis.c:39
ff_filter_execute
static av_always_inline int ff_filter_execute(AVFilterContext *ctx, avfilter_action_func *func, void *arg, int *ret, int nb_jobs)
Definition: internal.h:134
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:244
BiquadCoeffs::a0
double a0
Definition: af_aemphasis.c:27
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