FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_afreqshift.c
Date: 2022-11-26 13:19:19
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1 /*
2 * Copyright (c) Paul B Mahol
3 * Copyright (c) Laurent de Soras, 2005
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/channel_layout.h"
23 #include "libavutil/ffmath.h"
24 #include "libavutil/opt.h"
25 #include "avfilter.h"
26 #include "audio.h"
27 #include "formats.h"
28
29 #define MAX_NB_COEFFS 16
30
31 typedef struct AFreqShift {
32 const AVClass *class;
33
34 double shift;
35 double level;
36 int nb_coeffs;
37 int old_nb_coeffs;
38
39 double cd[MAX_NB_COEFFS * 2];
40 float cf[MAX_NB_COEFFS * 2];
41
42 int64_t in_samples;
43
44 AVFrame *i1, *o1;
45 AVFrame *i2, *o2;
46
47 void (*filter_channel)(AVFilterContext *ctx,
48 int channel,
49 AVFrame *in, AVFrame *out);
50 } AFreqShift;
51
52 static const enum AVSampleFormat sample_fmts[] = {
53 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE
54 };
55
56 #define PFILTER(name, type, sin, cos, cc) \
57 static void pfilter_channel_## name(AVFilterContext *ctx, \
58 int ch, \
59 AVFrame *in, AVFrame *out) \
60 { \
61 AFreqShift *s = ctx->priv; \
62 const int nb_samples = in->nb_samples; \
63 const type *src = (const type *)in->extended_data[ch]; \
64 type *dst = (type *)out->extended_data[ch]; \
65 type *i1 = (type *)s->i1->extended_data[ch]; \
66 type *o1 = (type *)s->o1->extended_data[ch]; \
67 type *i2 = (type *)s->i2->extended_data[ch]; \
68 type *o2 = (type *)s->o2->extended_data[ch]; \
69 const type *c = s->cc; \
70 const type level = s->level; \
71 type shift = s->shift * M_PI; \
72 type cos_theta = cos(shift); \
73 type sin_theta = sin(shift); \
74 \
75 for (int n = 0; n < nb_samples; n++) { \
76 type xn1 = src[n], xn2 = src[n]; \
77 type I, Q; \
78 \
79 for (int j = 0; j < s->nb_coeffs; j++) { \
80 I = c[j] * (xn1 + o2[j]) - i2[j]; \
81 i2[j] = i1[j]; \
82 i1[j] = xn1; \
83 o2[j] = o1[j]; \
84 o1[j] = I; \
85 xn1 = I; \
86 } \
87 \
88 for (int j = s->nb_coeffs; j < s->nb_coeffs*2; j++) { \
89 Q = c[j] * (xn2 + o2[j]) - i2[j]; \
90 i2[j] = i1[j]; \
91 i1[j] = xn2; \
92 o2[j] = o1[j]; \
93 o1[j] = Q; \
94 xn2 = Q; \
95 } \
96 Q = o2[s->nb_coeffs * 2 - 1]; \
97 \
98 dst[n] = (I * cos_theta - Q * sin_theta) * level; \
99 } \
100 }
101
102 PFILTER(flt, float, sin, cos, cf)
103 PFILTER(dbl, double, sin, cos, cd)
104
105 #define FFILTER(name, type, sin, cos, fmod, cc) \
106 static void ffilter_channel_## name(AVFilterContext *ctx, \
107 int ch, \
108 AVFrame *in, AVFrame *out) \
109 { \
110 AFreqShift *s = ctx->priv; \
111 const int nb_samples = in->nb_samples; \
112 const type *src = (const type *)in->extended_data[ch]; \
113 type *dst = (type *)out->extended_data[ch]; \
114 type *i1 = (type *)s->i1->extended_data[ch]; \
115 type *o1 = (type *)s->o1->extended_data[ch]; \
116 type *i2 = (type *)s->i2->extended_data[ch]; \
117 type *o2 = (type *)s->o2->extended_data[ch]; \
118 const type *c = s->cc; \
119 const type level = s->level; \
120 type ts = 1. / in->sample_rate; \
121 type shift = s->shift; \
122 int64_t N = s->in_samples; \
123 \
124 for (int n = 0; n < nb_samples; n++) { \
125 type xn1 = src[n], xn2 = src[n]; \
126 type I, Q, theta; \
127 \
128 for (int j = 0; j < s->nb_coeffs; j++) { \
129 I = c[j] * (xn1 + o2[j]) - i2[j]; \
130 i2[j] = i1[j]; \
131 i1[j] = xn1; \
132 o2[j] = o1[j]; \
133 o1[j] = I; \
134 xn1 = I; \
135 } \
136 \
137 for (int j = s->nb_coeffs; j < s->nb_coeffs*2; j++) { \
138 Q = c[j] * (xn2 + o2[j]) - i2[j]; \
139 i2[j] = i1[j]; \
140 i1[j] = xn2; \
141 o2[j] = o1[j]; \
142 o1[j] = Q; \
143 xn2 = Q; \
144 } \
145 Q = o2[s->nb_coeffs * 2 - 1]; \
146 \
147 theta = 2. * M_PI * fmod(shift * (N + n) * ts, 1.); \
148 dst[n] = (I * cos(theta) - Q * sin(theta)) * level; \
149 } \
150 }
151
152 FFILTER(flt, float, sinf, cosf, fmodf, cf)
153 FFILTER(dbl, double, sin, cos, fmod, cd)
154
155 static void compute_transition_param(double *K, double *Q, double transition)
156 {
157 double kksqrt, e, e2, e4, k, q;
158
159 k = tan((1. - transition * 2.) * M_PI / 4.);
160 k *= k;
161 kksqrt = pow(1 - k * k, 0.25);
162 e = 0.5 * (1. - kksqrt) / (1. + kksqrt);
163 e2 = e * e;
164 e4 = e2 * e2;
165 q = e * (1. + e4 * (2. + e4 * (15. + 150. * e4)));
166
167 *Q = q;
168 *K = k;
169 }
170
171 static double ipowp(double x, int64_t n)
172 {
173 double z = 1.;
174
175 while (n != 0) {
176 if (n & 1)
177 z *= x;
178 n >>= 1;
179 x *= x;
180 }
181
182 return z;
183 }
184
185 static double compute_acc_num(double q, int order, int c)
186 {
187 int64_t i = 0;
188 int j = 1;
189 double acc = 0.;
190 double q_ii1;
191
192 do {
193 q_ii1 = ipowp(q, i * (i + 1));
194 q_ii1 *= sin((i * 2 + 1) * c * M_PI / order) * j;
195 acc += q_ii1;
196
197 j = -j;
198 i++;
199 } while (fabs(q_ii1) > 1e-100);
200
201 return acc;
202 }
203
204 static double compute_acc_den(double q, int order, int c)
205 {
206 int64_t i = 1;
207 int j = -1;
208 double acc = 0.;
209 double q_i2;
210
211 do {
212 q_i2 = ipowp(q, i * i);
213 q_i2 *= cos(i * 2 * c * M_PI / order) * j;
214 acc += q_i2;
215
216 j = -j;
217 i++;
218 } while (fabs(q_i2) > 1e-100);
219
220 return acc;
221 }
222
223 static double compute_coef(int index, double k, double q, int order)
224 {
225 const int c = index + 1;
226 const double num = compute_acc_num(q, order, c) * pow(q, 0.25);
227 const double den = compute_acc_den(q, order, c) + 0.5;
228 const double ww = num / den;
229 const double wwsq = ww * ww;
230
231 const double x = sqrt((1 - wwsq * k) * (1 - wwsq / k)) / (1 + wwsq);
232 const double coef = (1 - x) / (1 + x);
233
234 return coef;
235 }
236
237 static void compute_coefs(double *coef_arrd, float *coef_arrf, int nbr_coefs, double transition)
238 {
239 const int order = nbr_coefs * 2 + 1;
240 double k, q;
241
242 compute_transition_param(&k, &q, transition);
243
244 for (int n = 0; n < nbr_coefs; n++) {
245 const int idx = (n / 2) + (n & 1) * nbr_coefs / 2;
246
247 coef_arrd[idx] = compute_coef(n, k, q, order);
248 coef_arrf[idx] = coef_arrd[idx];
249 }
250 }
251
252 static int config_input(AVFilterLink *inlink)
253 {
254 AVFilterContext *ctx = inlink->dst;
255 AFreqShift *s = ctx->priv;
256
257 if (s->old_nb_coeffs != s->nb_coeffs)
258 compute_coefs(s->cd, s->cf, s->nb_coeffs * 2, 2. * 20. / inlink->sample_rate);
259 s->old_nb_coeffs = s->nb_coeffs;
260
261 s->i1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
262 s->o1 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
263 s->i2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
264 s->o2 = ff_get_audio_buffer(inlink, MAX_NB_COEFFS * 2);
265 if (!s->i1 || !s->o1 || !s->i2 || !s->o2)
266 return AVERROR(ENOMEM);
267
268 if (inlink->format == AV_SAMPLE_FMT_DBLP) {
269 if (!strcmp(ctx->filter->name, "afreqshift"))
270 s->filter_channel = ffilter_channel_dbl;
271 else
272 s->filter_channel = pfilter_channel_dbl;
273 } else {
274 if (!strcmp(ctx->filter->name, "afreqshift"))
275 s->filter_channel = ffilter_channel_flt;
276 else
277 s->filter_channel = pfilter_channel_flt;
278 }
279
280 return 0;
281 }
282
283 typedef struct ThreadData {
284 AVFrame *in, *out;
285 } ThreadData;
286
287 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
288 {
289 AFreqShift *s = ctx->priv;
290 ThreadData *td = arg;
291 AVFrame *out = td->out;
292 AVFrame *in = td->in;
293 const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
294 const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
295
296 for (int ch = start; ch < end; ch++)
297 s->filter_channel(ctx, ch, in, out);
298
299 return 0;
300 }
301
302 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
303 {
304 AVFilterContext *ctx = inlink->dst;
305 AVFilterLink *outlink = ctx->outputs[0];
306 AFreqShift *s = ctx->priv;
307 AVFrame *out;
308 ThreadData td;
309
310 if (s->old_nb_coeffs != s->nb_coeffs)
311 compute_coefs(s->cd, s->cf, s->nb_coeffs * 2, 2. * 20. / inlink->sample_rate);
312 s->old_nb_coeffs = s->nb_coeffs;
313
314 if (av_frame_is_writable(in)) {
315 out = in;
316 } else {
317 out = ff_get_audio_buffer(outlink, in->nb_samples);
318 if (!out) {
319 av_frame_free(&in);
320 return AVERROR(ENOMEM);
321 }
322 av_frame_copy_props(out, in);
323 }
324
325 td.in = in; td.out = out;
326 ff_filter_execute(ctx, filter_channels, &td, NULL,
327 FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
328
329 s->in_samples += in->nb_samples;
330
331 if (out != in)
332 av_frame_free(&in);
333 return ff_filter_frame(outlink, out);
334 }
335
336 static av_cold void uninit(AVFilterContext *ctx)
337 {
338 AFreqShift *s = ctx->priv;
339
340 av_frame_free(&s->i1);
341 av_frame_free(&s->o1);
342 av_frame_free(&s->i2);
343 av_frame_free(&s->o2);
344 }
345
346 #define OFFSET(x) offsetof(AFreqShift, x)
347 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
348
349 static const AVOption afreqshift_options[] = {
350 { "shift", "set frequency shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -INT_MAX, INT_MAX, FLAGS },
351 { "level", "set output level", OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS },
352 { "order", "set filter order", OFFSET(nb_coeffs),AV_OPT_TYPE_INT, {.i64=8}, 1, MAX_NB_COEFFS, FLAGS },
353 { NULL }
354 };
355
356 AVFILTER_DEFINE_CLASS(afreqshift);
357
358 static const AVFilterPad inputs[] = {
359 {
360 .name = "default",
361 .type = AVMEDIA_TYPE_AUDIO,
362 .filter_frame = filter_frame,
363 .config_props = config_input,
364 },
365 };
366
367 static const AVFilterPad outputs[] = {
368 {
369 .name = "default",
370 .type = AVMEDIA_TYPE_AUDIO,
371 },
372 };
373
374 const AVFilter ff_af_afreqshift = {
375 .name = "afreqshift",
376 .description = NULL_IF_CONFIG_SMALL("Apply frequency shifting to input audio."),
377 .priv_size = sizeof(AFreqShift),
378 .priv_class = &afreqshift_class,
379 .uninit = uninit,
380 FILTER_INPUTS(inputs),
381 FILTER_OUTPUTS(outputs),
382 FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
383 .process_command = ff_filter_process_command,
384 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
385 AVFILTER_FLAG_SLICE_THREADS,
386 };
387
388 static const AVOption aphaseshift_options[] = {
389 { "shift", "set phase shift", OFFSET(shift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1.0, 1.0, FLAGS },
390 { "level", "set output level",OFFSET(level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0, 1.0, FLAGS },
391 { "order", "set filter order",OFFSET(nb_coeffs), AV_OPT_TYPE_INT,{.i64=8}, 1, MAX_NB_COEFFS, FLAGS },
392 { NULL }
393 };
394
395 AVFILTER_DEFINE_CLASS(aphaseshift);
396
397 const AVFilter ff_af_aphaseshift = {
398 .name = "aphaseshift",
399 .description = NULL_IF_CONFIG_SMALL("Apply phase shifting to input audio."),
400 .priv_size = sizeof(AFreqShift),
401 .priv_class = &aphaseshift_class,
402 .uninit = uninit,
403 FILTER_INPUTS(inputs),
404 FILTER_OUTPUTS(outputs),
405 FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
406 .process_command = ff_filter_process_command,
407 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
408 AVFILTER_FLAG_SLICE_THREADS,
409 };
410