FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/asrc_sinc.c
Date: 2024-04-25 05:10:44
Exec Total Coverage
Lines: 0 214 0.0%
Functions: 0 11 0.0%
Branches: 0 124 0.0%

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1 /*
2 * Copyright (c) 2008-2009 Rob Sykes <robs@users.sourceforge.net>
3 * Copyright (c) 2017 Paul B Mahol
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/avassert.h"
23 #include "libavutil/channel_layout.h"
24 #include "libavutil/mem.h"
25 #include "libavutil/opt.h"
26 #include "libavutil/tx.h"
27
28 #include "audio.h"
29 #include "avfilter.h"
30 #include "filters.h"
31 #include "formats.h"
32 #include "internal.h"
33
34 typedef struct SincContext {
35 const AVClass *class;
36
37 int sample_rate, nb_samples;
38 float att, beta, phase, Fc0, Fc1, tbw0, tbw1;
39 int num_taps[2];
40 int round;
41
42 int n, rdft_len;
43 float *coeffs;
44 int64_t pts;
45
46 AVTXContext *tx, *itx;
47 av_tx_fn tx_fn, itx_fn;
48 } SincContext;
49
50 static int activate(AVFilterContext *ctx)
51 {
52 AVFilterLink *outlink = ctx->outputs[0];
53 SincContext *s = ctx->priv;
54 const float *coeffs = s->coeffs;
55 AVFrame *frame = NULL;
56 int nb_samples;
57
58 if (!ff_outlink_frame_wanted(outlink))
59 return FFERROR_NOT_READY;
60
61 nb_samples = FFMIN(s->nb_samples, s->n - s->pts);
62 if (nb_samples <= 0) {
63 ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
64 return 0;
65 }
66
67 if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
68 return AVERROR(ENOMEM);
69
70 memcpy(frame->data[0], coeffs + s->pts, nb_samples * sizeof(float));
71
72 frame->pts = s->pts;
73 s->pts += nb_samples;
74
75 return ff_filter_frame(outlink, frame);
76 }
77
78 static int query_formats(AVFilterContext *ctx)
79 {
80 SincContext *s = ctx->priv;
81 static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
82 int sample_rates[] = { s->sample_rate, -1 };
83 static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT,
84 AV_SAMPLE_FMT_NONE };
85 int ret = ff_set_common_formats_from_list(ctx, sample_fmts);
86 if (ret < 0)
87 return ret;
88
89 ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts);
90 if (ret < 0)
91 return ret;
92
93 return ff_set_common_samplerates_from_list(ctx, sample_rates);
94 }
95
96 static float *make_lpf(int num_taps, float Fc, float beta, float rho,
97 float scale, int dc_norm)
98 {
99 int i, m = num_taps - 1;
100 float *h = av_calloc(num_taps, sizeof(*h)), sum = 0;
101 float mult = scale / av_bessel_i0(beta), mult1 = 1.f / (.5f * m + rho);
102
103 if (!h)
104 return NULL;
105
106 av_assert0(Fc >= 0 && Fc <= 1);
107
108 for (i = 0; i <= m / 2; i++) {
109 float z = i - .5f * m, x = z * M_PI, y = z * mult1;
110 h[i] = x ? sinf(Fc * x) / x : Fc;
111 sum += h[i] *= av_bessel_i0(beta * sqrtf(1.f - y * y)) * mult;
112 if (m - i != i) {
113 h[m - i] = h[i];
114 sum += h[i];
115 }
116 }
117
118 for (i = 0; dc_norm && i < num_taps; i++)
119 h[i] *= scale / sum;
120
121 return h;
122 }
123
124 static float kaiser_beta(float att, float tr_bw)
125 {
126 if (att >= 60.f) {
127 static const float coefs[][4] = {
128 {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
129 {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
130 {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
131 {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
132 {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
133 {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
134 {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
135 {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
136 {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
137 {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
138 };
139 float realm = logf(tr_bw / .0005f) / logf(2.f);
140 float const *c0 = coefs[av_clip((int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
141 float const *c1 = coefs[av_clip(1 + (int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
142 float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
143 float b1 = ((c1[0] * att + c1[1]) * att + c1[2]) * att + c1[3];
144
145 return b0 + (b1 - b0) * (realm - (int)realm);
146 }
147 if (att > 50.f)
148 return .1102f * (att - 8.7f);
149 if (att > 20.96f)
150 return .58417f * powf(att - 20.96f, .4f) + .07886f * (att - 20.96f);
151 return 0;
152 }
153
154 static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
155 {
156 *beta = *beta < 0.f ? kaiser_beta(att, tr_bw * .5f / Fc): *beta;
157 att = att < 60.f ? (att - 7.95f) / (2.285f * M_PI * 2.f) :
158 ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022f) * *beta + .06186902f;
159 *num_taps = !*num_taps ? ceilf(att/tr_bw + 1) : *num_taps;
160 }
161
162 static float *lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
163 {
164 int n = *num_taps;
165
166 if ((Fc /= Fn) <= 0.f || Fc >= 1.f) {
167 *num_taps = 0;
168 return NULL;
169 }
170
171 att = att ? att : 120.f;
172
173 kaiser_params(att, Fc, (tbw ? tbw / Fn : .05f) * .5f, beta, num_taps);
174
175 if (!n) {
176 n = *num_taps;
177 *num_taps = av_clip(n, 11, 32767);
178 if (round)
179 *num_taps = 1 + 2 * (int)((int)((*num_taps / 2) * Fc + .5f) / Fc + .5f);
180 }
181
182 return make_lpf(*num_taps |= 1, Fc, *beta, 0.f, 1.f, 0);
183 }
184
185 static void invert(float *h, int n)
186 {
187 for (int i = 0; i < n; i++)
188 h[i] = -h[i];
189
190 h[(n - 1) / 2] += 1;
191 }
192
193 #define SQR(a) ((a) * (a))
194
195 static float safe_log(float x)
196 {
197 av_assert0(x >= 0);
198 if (x)
199 return logf(x);
200 return -26;
201 }
202
203 static int fir_to_phase(SincContext *s, float **h, int *len, int *post_len, float phase)
204 {
205 float *pi_wraps, *work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.f;
206 int i, work_len, begin, end, imp_peak = 0, peak = 0, ret;
207 float imp_sum = 0, peak_imp_sum = 0, scale = 1.f;
208 float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
209
210 for (i = *len, work_len = 2 * 2 * 8; i > 1; work_len <<= 1, i >>= 1);
211
212 /* The first part is for work (+2 for (UN)PACK), the latter for pi_wraps. */
213 work = av_calloc((work_len + 2) + (work_len / 2 + 1), sizeof(float));
214 if (!work)
215 return AVERROR(ENOMEM);
216 pi_wraps = &work[work_len + 2];
217
218 memcpy(work, *h, *len * sizeof(*work));
219
220 av_tx_uninit(&s->tx);
221 av_tx_uninit(&s->itx);
222 ret = av_tx_init(&s->tx, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, work_len, &scale, AV_TX_INPLACE);
223 if (ret < 0)
224 goto fail;
225 ret = av_tx_init(&s->itx, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, work_len, &scale, AV_TX_INPLACE);
226 if (ret < 0)
227 goto fail;
228
229 s->tx_fn(s->tx, work, work, sizeof(float)); /* Cepstral: */
230
231 for (i = 0; i <= work_len; i += 2) {
232 float angle = atan2f(work[i + 1], work[i]);
233 float detect = 2 * M_PI;
234 float delta = angle - prev_angle2;
235 float adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
236
237 prev_angle2 = angle;
238 cum_2pi += adjust;
239 angle += cum_2pi;
240 detect = M_PI;
241 delta = angle - prev_angle1;
242 adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
243 prev_angle1 = angle;
244 cum_1pi += fabsf(adjust); /* fabs for when 2pi and 1pi have combined */
245 pi_wraps[i >> 1] = cum_1pi;
246
247 work[i] = safe_log(sqrtf(SQR(work[i]) + SQR(work[i + 1])));
248 work[i + 1] = 0;
249 }
250
251 s->itx_fn(s->itx, work, work, sizeof(AVComplexFloat));
252
253 for (i = 0; i < work_len; i++)
254 work[i] *= 2.f / work_len;
255
256 for (i = 1; i < work_len / 2; i++) { /* Window to reject acausal components */
257 work[i] *= 2;
258 work[i + work_len / 2] = 0;
259 }
260 s->tx_fn(s->tx, work, work, sizeof(float));
261
262 for (i = 2; i < work_len; i += 2) /* Interpolate between linear & min phase */
263 work[i + 1] = phase1 * i / work_len * pi_wraps[work_len >> 1] + (1 - phase1) * (work[i + 1] + pi_wraps[i >> 1]) - pi_wraps[i >> 1];
264
265 work[0] = exp(work[0]);
266 work[1] = exp(work[1]);
267 for (i = 2; i < work_len; i += 2) {
268 float x = expf(work[i]);
269
270 work[i ] = x * cosf(work[i + 1]);
271 work[i + 1] = x * sinf(work[i + 1]);
272 }
273
274 s->itx_fn(s->itx, work, work, sizeof(AVComplexFloat));
275 for (i = 0; i < work_len; i++)
276 work[i] *= 2.f / work_len;
277
278 /* Find peak pos. */
279 for (i = 0; i <= (int) (pi_wraps[work_len >> 1] / M_PI + .5f); i++) {
280 imp_sum += work[i];
281 if (fabs(imp_sum) > fabs(peak_imp_sum)) {
282 peak_imp_sum = imp_sum;
283 peak = i;
284 }
285 if (work[i] > work[imp_peak]) /* For debug check only */
286 imp_peak = i;
287 }
288
289 while (peak && fabsf(work[peak - 1]) > fabsf(work[peak]) && (work[peak - 1] * work[peak] > 0)) {
290 peak--;
291 }
292
293 if (!phase1) {
294 begin = 0;
295 } else if (phase1 == 1) {
296 begin = peak - *len / 2;
297 } else {
298 begin = (.997f - (2 - phase1) * .22f) * *len + .5f;
299 end = (.997f + (0 - phase1) * .22f) * *len + .5f;
300 begin = peak - (begin & ~3);
301 end = peak + 1 + ((end + 3) & ~3);
302 *len = end - begin;
303 *h = av_realloc_f(*h, *len, sizeof(**h));
304 if (!*h) {
305 av_free(work);
306 return AVERROR(ENOMEM);
307 }
308 }
309
310 for (i = 0; i < *len; i++) {
311 (*h)[i] = work[(begin + (phase > 50.f ? *len - 1 - i : i) + work_len) & (work_len - 1)];
312 }
313 *post_len = phase > 50 ? peak - begin : begin + *len - (peak + 1);
314
315 av_log(s, AV_LOG_DEBUG, "%d nPI=%g peak-sum@%i=%g (val@%i=%g); len=%i post=%i (%g%%)\n",
316 work_len, pi_wraps[work_len >> 1] / M_PI, peak, peak_imp_sum, imp_peak,
317 work[imp_peak], *len, *post_len, 100.f - 100.f * *post_len / (*len - 1));
318
319 fail:
320 av_free(work);
321
322 return ret;
323 }
324
325 static int config_output(AVFilterLink *outlink)
326 {
327 AVFilterContext *ctx = outlink->src;
328 SincContext *s = ctx->priv;
329 float Fn = s->sample_rate * .5f;
330 float *h[2];
331 int i, n, post_peak, longer;
332
333 outlink->sample_rate = s->sample_rate;
334 s->pts = 0;
335
336 if (s->Fc0 >= Fn || s->Fc1 >= Fn) {
337 av_log(ctx, AV_LOG_ERROR,
338 "filter frequency must be less than %d/2.\n", s->sample_rate);
339 return AVERROR(EINVAL);
340 }
341
342 h[0] = lpf(Fn, s->Fc0, s->tbw0, &s->num_taps[0], s->att, &s->beta, s->round);
343 h[1] = lpf(Fn, s->Fc1, s->tbw1, &s->num_taps[1], s->att, &s->beta, s->round);
344
345 if (h[0])
346 invert(h[0], s->num_taps[0]);
347
348 longer = s->num_taps[1] > s->num_taps[0];
349 n = s->num_taps[longer];
350
351 if (h[0] && h[1]) {
352 for (i = 0; i < s->num_taps[!longer]; i++)
353 h[longer][i + (n - s->num_taps[!longer]) / 2] += h[!longer][i];
354
355 if (s->Fc0 < s->Fc1)
356 invert(h[longer], n);
357
358 av_free(h[!longer]);
359 }
360
361 if (s->phase != 50.f) {
362 int ret = fir_to_phase(s, &h[longer], &n, &post_peak, s->phase);
363 if (ret < 0)
364 return ret;
365 } else {
366 post_peak = n >> 1;
367 }
368
369 s->n = 1 << (av_log2(n) + 1);
370 s->rdft_len = 1 << av_log2(n);
371 s->coeffs = av_calloc(s->n, sizeof(*s->coeffs));
372 if (!s->coeffs)
373 return AVERROR(ENOMEM);
374
375 for (i = 0; i < n; i++)
376 s->coeffs[i] = h[longer][i];
377 av_free(h[longer]);
378
379 av_tx_uninit(&s->tx);
380 av_tx_uninit(&s->itx);
381
382 return 0;
383 }
384
385 static av_cold void uninit(AVFilterContext *ctx)
386 {
387 SincContext *s = ctx->priv;
388
389 av_freep(&s->coeffs);
390 av_tx_uninit(&s->tx);
391 av_tx_uninit(&s->itx);
392 }
393
394 static const AVFilterPad sinc_outputs[] = {
395 {
396 .name = "default",
397 .type = AVMEDIA_TYPE_AUDIO,
398 .config_props = config_output,
399 },
400 };
401
402 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
403 #define OFFSET(x) offsetof(SincContext, x)
404
405 static const AVOption sinc_options[] = {
406 { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
407 { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
408 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
409 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
410 { "hp", "set high-pass filter frequency", OFFSET(Fc0), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
411 { "lp", "set low-pass filter frequency", OFFSET(Fc1), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
412 { "phase", "set filter phase response", OFFSET(phase), AV_OPT_TYPE_FLOAT, {.dbl=50}, 0, 100, AF },
413 { "beta", "set kaiser window beta", OFFSET(beta), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 256, AF },
414 { "att", "set stop-band attenuation", OFFSET(att), AV_OPT_TYPE_FLOAT, {.dbl=120}, 40, 180, AF },
415 { "round", "enable rounding", OFFSET(round), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
416 { "hptaps", "set number of taps for high-pass filter", OFFSET(num_taps[0]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
417 { "lptaps", "set number of taps for low-pass filter", OFFSET(num_taps[1]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
418 { NULL }
419 };
420
421 AVFILTER_DEFINE_CLASS(sinc);
422
423 const AVFilter ff_asrc_sinc = {
424 .name = "sinc",
425 .description = NULL_IF_CONFIG_SMALL("Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
426 .priv_size = sizeof(SincContext),
427 .priv_class = &sinc_class,
428 .uninit = uninit,
429 .activate = activate,
430 .inputs = NULL,
431 FILTER_OUTPUTS(sinc_outputs),
432 FILTER_QUERY_FUNC(query_formats),
433 };
434