FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/asrc_sinc.c
Date: 2024-11-20 23:03:26
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
33 typedef struct SincContext {
34 const AVClass *class;
35
36 int sample_rate, nb_samples;
37 float att, beta, phase, Fc0, Fc1, tbw0, tbw1;
38 int num_taps[2];
39 int round;
40
41 int n, rdft_len;
42 float *coeffs;
43 int64_t pts;
44
45 AVTXContext *tx, *itx;
46 av_tx_fn tx_fn, itx_fn;
47 } SincContext;
48
49 static int activate(AVFilterContext *ctx)
50 {
51 AVFilterLink *outlink = ctx->outputs[0];
52 SincContext *s = ctx->priv;
53 const float *coeffs = s->coeffs;
54 AVFrame *frame = NULL;
55 int nb_samples;
56
57 if (!ff_outlink_frame_wanted(outlink))
58 return FFERROR_NOT_READY;
59
60 nb_samples = FFMIN(s->nb_samples, s->n - s->pts);
61 if (nb_samples <= 0) {
62 ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
63 return 0;
64 }
65
66 if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
67 return AVERROR(ENOMEM);
68
69 memcpy(frame->data[0], coeffs + s->pts, nb_samples * sizeof(float));
70
71 frame->pts = s->pts;
72 s->pts += nb_samples;
73
74 return ff_filter_frame(outlink, frame);
75 }
76
77 static int query_formats(const AVFilterContext *ctx,
78 AVFilterFormatsConfig **cfg_in,
79 AVFilterFormatsConfig **cfg_out)
80 {
81 const SincContext *s = ctx->priv;
82 static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
83 int sample_rates[] = { s->sample_rate, -1 };
84 static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT,
85 AV_SAMPLE_FMT_NONE };
86 int ret = ff_set_common_formats_from_list2(ctx, cfg_in, cfg_out, sample_fmts);
87 if (ret < 0)
88 return ret;
89
90 ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, chlayouts);
91 if (ret < 0)
92 return ret;
93
94 return ff_set_common_samplerates_from_list2(ctx, cfg_in, cfg_out, sample_rates);
95 }
96
97 static float *make_lpf(int num_taps, float Fc, float beta, float rho,
98 float scale, int dc_norm)
99 {
100 int i, m = num_taps - 1;
101 float *h = av_calloc(num_taps, sizeof(*h)), sum = 0;
102 float mult = scale / av_bessel_i0(beta), mult1 = 1.f / (.5f * m + rho);
103
104 if (!h)
105 return NULL;
106
107 av_assert0(Fc >= 0 && Fc <= 1);
108
109 for (i = 0; i <= m / 2; i++) {
110 float z = i - .5f * m, x = z * M_PI, y = z * mult1;
111 h[i] = x ? sinf(Fc * x) / x : Fc;
112 sum += h[i] *= av_bessel_i0(beta * sqrtf(1.f - y * y)) * mult;
113 if (m - i != i) {
114 h[m - i] = h[i];
115 sum += h[i];
116 }
117 }
118
119 for (i = 0; dc_norm && i < num_taps; i++)
120 h[i] *= scale / sum;
121
122 return h;
123 }
124
125 static float kaiser_beta(float att, float tr_bw)
126 {
127 if (att >= 60.f) {
128 static const float coefs[][4] = {
129 {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
130 {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
131 {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
132 {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
133 {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
134 {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
135 {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
136 {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
137 {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
138 {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
139 };
140 float realm = logf(tr_bw / .0005f) / logf(2.f);
141 float const *c0 = coefs[av_clip((int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
142 float const *c1 = coefs[av_clip(1 + (int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
143 float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
144 float b1 = ((c1[0] * att + c1[1]) * att + c1[2]) * att + c1[3];
145
146 return b0 + (b1 - b0) * (realm - (int)realm);
147 }
148 if (att > 50.f)
149 return .1102f * (att - 8.7f);
150 if (att > 20.96f)
151 return .58417f * powf(att - 20.96f, .4f) + .07886f * (att - 20.96f);
152 return 0;
153 }
154
155 static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
156 {
157 *beta = *beta < 0.f ? kaiser_beta(att, tr_bw * .5f / Fc): *beta;
158 att = att < 60.f ? (att - 7.95f) / (2.285f * M_PI * 2.f) :
159 ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022f) * *beta + .06186902f;
160 *num_taps = !*num_taps ? ceilf(att/tr_bw + 1) : *num_taps;
161 }
162
163 static float *lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
164 {
165 int n = *num_taps;
166
167 if ((Fc /= Fn) <= 0.f || Fc >= 1.f) {
168 *num_taps = 0;
169 return NULL;
170 }
171
172 att = att ? att : 120.f;
173
174 kaiser_params(att, Fc, (tbw ? tbw / Fn : .05f) * .5f, beta, num_taps);
175
176 if (!n) {
177 n = *num_taps;
178 *num_taps = av_clip(n, 11, 32767);
179 if (round)
180 *num_taps = 1 + 2 * (int)((int)((*num_taps / 2) * Fc + .5f) / Fc + .5f);
181 }
182
183 return make_lpf(*num_taps |= 1, Fc, *beta, 0.f, 1.f, 0);
184 }
185
186 static void invert(float *h, int n)
187 {
188 for (int i = 0; i < n; i++)
189 h[i] = -h[i];
190
191 h[(n - 1) / 2] += 1;
192 }
193
194 #define SQR(a) ((a) * (a))
195
196 static float safe_log(float x)
197 {
198 av_assert0(x >= 0);
199 if (x)
200 return logf(x);
201 return -26;
202 }
203
204 static int fir_to_phase(SincContext *s, float **h, int *len, int *post_len, float phase)
205 {
206 float *pi_wraps, *work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.f;
207 int i, work_len, begin, end, imp_peak = 0, peak = 0, ret;
208 float imp_sum = 0, peak_imp_sum = 0, scale = 1.f;
209 float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
210
211 for (i = *len, work_len = 2 * 2 * 8; i > 1; work_len <<= 1, i >>= 1);
212
213 /* The first part is for work (+2 for (UN)PACK), the latter for pi_wraps. */
214 work = av_calloc((work_len + 2) + (work_len / 2 + 1), sizeof(float));
215 if (!work)
216 return AVERROR(ENOMEM);
217 pi_wraps = &work[work_len + 2];
218
219 memcpy(work, *h, *len * sizeof(*work));
220
221 av_tx_uninit(&s->tx);
222 av_tx_uninit(&s->itx);
223 ret = av_tx_init(&s->tx, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, work_len, &scale, AV_TX_INPLACE);
224 if (ret < 0)
225 goto fail;
226 ret = av_tx_init(&s->itx, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, work_len, &scale, AV_TX_INPLACE);
227 if (ret < 0)
228 goto fail;
229
230 s->tx_fn(s->tx, work, work, sizeof(float)); /* Cepstral: */
231
232 for (i = 0; i <= work_len; i += 2) {
233 float angle = atan2f(work[i + 1], work[i]);
234 float detect = 2 * M_PI;
235 float delta = angle - prev_angle2;
236 float adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
237
238 prev_angle2 = angle;
239 cum_2pi += adjust;
240 angle += cum_2pi;
241 detect = M_PI;
242 delta = angle - prev_angle1;
243 adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
244 prev_angle1 = angle;
245 cum_1pi += fabsf(adjust); /* fabs for when 2pi and 1pi have combined */
246 pi_wraps[i >> 1] = cum_1pi;
247
248 work[i] = safe_log(sqrtf(SQR(work[i]) + SQR(work[i + 1])));
249 work[i + 1] = 0;
250 }
251
252 s->itx_fn(s->itx, work, work, sizeof(AVComplexFloat));
253
254 for (i = 0; i < work_len; i++)
255 work[i] *= 2.f / work_len;
256
257 for (i = 1; i < work_len / 2; i++) { /* Window to reject acausal components */
258 work[i] *= 2;
259 work[i + work_len / 2] = 0;
260 }
261 s->tx_fn(s->tx, work, work, sizeof(float));
262
263 for (i = 2; i < work_len; i += 2) /* Interpolate between linear & min phase */
264 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];
265
266 work[0] = exp(work[0]);
267 work[1] = exp(work[1]);
268 for (i = 2; i < work_len; i += 2) {
269 float x = expf(work[i]);
270
271 work[i ] = x * cosf(work[i + 1]);
272 work[i + 1] = x * sinf(work[i + 1]);
273 }
274
275 s->itx_fn(s->itx, work, work, sizeof(AVComplexFloat));
276 for (i = 0; i < work_len; i++)
277 work[i] *= 2.f / work_len;
278
279 /* Find peak pos. */
280 for (i = 0; i <= (int) (pi_wraps[work_len >> 1] / M_PI + .5f); i++) {
281 imp_sum += work[i];
282 if (fabs(imp_sum) > fabs(peak_imp_sum)) {
283 peak_imp_sum = imp_sum;
284 peak = i;
285 }
286 if (work[i] > work[imp_peak]) /* For debug check only */
287 imp_peak = i;
288 }
289
290 while (peak && fabsf(work[peak - 1]) > fabsf(work[peak]) && (work[peak - 1] * work[peak] > 0)) {
291 peak--;
292 }
293
294 if (!phase1) {
295 begin = 0;
296 } else if (phase1 == 1) {
297 begin = peak - *len / 2;
298 } else {
299 begin = (.997f - (2 - phase1) * .22f) * *len + .5f;
300 end = (.997f + (0 - phase1) * .22f) * *len + .5f;
301 begin = peak - (begin & ~3);
302 end = peak + 1 + ((end + 3) & ~3);
303 *len = end - begin;
304 *h = av_realloc_f(*h, *len, sizeof(**h));
305 if (!*h) {
306 av_free(work);
307 return AVERROR(ENOMEM);
308 }
309 }
310
311 for (i = 0; i < *len; i++) {
312 (*h)[i] = work[(begin + (phase > 50.f ? *len - 1 - i : i) + work_len) & (work_len - 1)];
313 }
314 *post_len = phase > 50 ? peak - begin : begin + *len - (peak + 1);
315
316 av_log(s, AV_LOG_DEBUG, "%d nPI=%g peak-sum@%i=%g (val@%i=%g); len=%i post=%i (%g%%)\n",
317 work_len, pi_wraps[work_len >> 1] / M_PI, peak, peak_imp_sum, imp_peak,
318 work[imp_peak], *len, *post_len, 100.f - 100.f * *post_len / (*len - 1));
319
320 fail:
321 av_free(work);
322
323 return ret;
324 }
325
326 static int config_output(AVFilterLink *outlink)
327 {
328 AVFilterContext *ctx = outlink->src;
329 SincContext *s = ctx->priv;
330 float Fn = s->sample_rate * .5f;
331 float *h[2];
332 int i, n, post_peak, longer;
333
334 outlink->sample_rate = s->sample_rate;
335 s->pts = 0;
336
337 if (s->Fc0 >= Fn || s->Fc1 >= Fn) {
338 av_log(ctx, AV_LOG_ERROR,
339 "filter frequency must be less than %d/2.\n", s->sample_rate);
340 return AVERROR(EINVAL);
341 }
342
343 h[0] = lpf(Fn, s->Fc0, s->tbw0, &s->num_taps[0], s->att, &s->beta, s->round);
344 h[1] = lpf(Fn, s->Fc1, s->tbw1, &s->num_taps[1], s->att, &s->beta, s->round);
345
346 if (h[0])
347 invert(h[0], s->num_taps[0]);
348
349 longer = s->num_taps[1] > s->num_taps[0];
350 n = s->num_taps[longer];
351
352 if (h[0] && h[1]) {
353 for (i = 0; i < s->num_taps[!longer]; i++)
354 h[longer][i + (n - s->num_taps[!longer]) / 2] += h[!longer][i];
355
356 if (s->Fc0 < s->Fc1)
357 invert(h[longer], n);
358
359 av_free(h[!longer]);
360 }
361
362 if (s->phase != 50.f) {
363 int ret = fir_to_phase(s, &h[longer], &n, &post_peak, s->phase);
364 if (ret < 0)
365 return ret;
366 } else {
367 post_peak = n >> 1;
368 }
369
370 s->n = 1 << (av_log2(n) + 1);
371 s->rdft_len = 1 << av_log2(n);
372 s->coeffs = av_calloc(s->n, sizeof(*s->coeffs));
373 if (!s->coeffs)
374 return AVERROR(ENOMEM);
375
376 for (i = 0; i < n; i++)
377 s->coeffs[i] = h[longer][i];
378 av_free(h[longer]);
379
380 av_tx_uninit(&s->tx);
381 av_tx_uninit(&s->itx);
382
383 return 0;
384 }
385
386 static av_cold void uninit(AVFilterContext *ctx)
387 {
388 SincContext *s = ctx->priv;
389
390 av_freep(&s->coeffs);
391 av_tx_uninit(&s->tx);
392 av_tx_uninit(&s->itx);
393 }
394
395 static const AVFilterPad sinc_outputs[] = {
396 {
397 .name = "default",
398 .type = AVMEDIA_TYPE_AUDIO,
399 .config_props = config_output,
400 },
401 };
402
403 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
404 #define OFFSET(x) offsetof(SincContext, x)
405
406 static const AVOption sinc_options[] = {
407 { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
408 { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
409 { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
410 { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
411 { "hp", "set high-pass filter frequency", OFFSET(Fc0), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
412 { "lp", "set low-pass filter frequency", OFFSET(Fc1), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
413 { "phase", "set filter phase response", OFFSET(phase), AV_OPT_TYPE_FLOAT, {.dbl=50}, 0, 100, AF },
414 { "beta", "set kaiser window beta", OFFSET(beta), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 256, AF },
415 { "att", "set stop-band attenuation", OFFSET(att), AV_OPT_TYPE_FLOAT, {.dbl=120}, 40, 180, AF },
416 { "round", "enable rounding", OFFSET(round), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
417 { "hptaps", "set number of taps for high-pass filter", OFFSET(num_taps[0]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
418 { "lptaps", "set number of taps for low-pass filter", OFFSET(num_taps[1]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
419 { NULL }
420 };
421
422 AVFILTER_DEFINE_CLASS(sinc);
423
424 const AVFilter ff_asrc_sinc = {
425 .name = "sinc",
426 .description = NULL_IF_CONFIG_SMALL("Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
427 .priv_size = sizeof(SincContext),
428 .priv_class = &sinc_class,
429 .uninit = uninit,
430 .activate = activate,
431 .inputs = NULL,
432 FILTER_OUTPUTS(sinc_outputs),
433 FILTER_QUERY_FUNC2(query_formats),
434 };
435