FFmpeg
asrc_sinc.c
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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"
24 #include "libavutil/opt.h"
25 #include "libavutil/tx.h"
26 
27 #include "audio.h"
28 #include "avfilter.h"
29 #include "filters.h"
30 #include "internal.h"
31 
32 typedef struct SincContext {
33  const AVClass *class;
34 
36  float att, beta, phase, Fc0, Fc1, tbw0, tbw1;
37  int num_taps[2];
38  int round;
39 
40  int n, rdft_len;
41  float *coeffs;
42  int64_t pts;
43 
46 } SincContext;
47 
49 {
50  AVFilterLink *outlink = ctx->outputs[0];
51  SincContext *s = ctx->priv;
52  const float *coeffs = s->coeffs;
53  AVFrame *frame = NULL;
54  int nb_samples;
55 
56  if (!ff_outlink_frame_wanted(outlink))
57  return FFERROR_NOT_READY;
58 
59  nb_samples = FFMIN(s->nb_samples, s->n - s->pts);
60  if (nb_samples <= 0) {
61  ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
62  return 0;
63  }
64 
65  if (!(frame = ff_get_audio_buffer(outlink, nb_samples)))
66  return AVERROR(ENOMEM);
67 
68  memcpy(frame->data[0], coeffs + s->pts, nb_samples * sizeof(float));
69 
70  frame->pts = s->pts;
71  s->pts += nb_samples;
72 
73  return ff_filter_frame(outlink, frame);
74 }
75 
77 {
78  SincContext *s = ctx->priv;
79  static const AVChannelLayout chlayouts[] = { AV_CHANNEL_LAYOUT_MONO, { 0 } };
80  int sample_rates[] = { s->sample_rate, -1 };
81  static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT,
84  if (ret < 0)
85  return ret;
86 
88  if (ret < 0)
89  return ret;
90 
92 }
93 
94 static float bessel_I_0(float x)
95 {
96  float term = 1, sum = 1, last_sum, x2 = x / 2;
97  int i = 1;
98 
99  do {
100  float y = x2 / i++;
101 
102  last_sum = sum;
103  sum += term *= y * y;
104  } while (sum != last_sum);
105 
106  return sum;
107 }
108 
109 static float *make_lpf(int num_taps, float Fc, float beta, float rho,
110  float scale, int dc_norm)
111 {
112  int i, m = num_taps - 1;
113  float *h = av_calloc(num_taps, sizeof(*h)), sum = 0;
114  float mult = scale / bessel_I_0(beta), mult1 = 1.f / (.5f * m + rho);
115 
116  if (!h)
117  return NULL;
118 
119  av_assert0(Fc >= 0 && Fc <= 1);
120 
121  for (i = 0; i <= m / 2; i++) {
122  float z = i - .5f * m, x = z * M_PI, y = z * mult1;
123  h[i] = x ? sinf(Fc * x) / x : Fc;
124  sum += h[i] *= bessel_I_0(beta * sqrtf(1.f - y * y)) * mult;
125  if (m - i != i) {
126  h[m - i] = h[i];
127  sum += h[i];
128  }
129  }
130 
131  for (i = 0; dc_norm && i < num_taps; i++)
132  h[i] *= scale / sum;
133 
134  return h;
135 }
136 
137 static float kaiser_beta(float att, float tr_bw)
138 {
139  if (att >= 60.f) {
140  static const float coefs[][4] = {
141  {-6.784957e-10, 1.02856e-05, 0.1087556, -0.8988365 + .001},
142  {-6.897885e-10, 1.027433e-05, 0.10876, -0.8994658 + .002},
143  {-1.000683e-09, 1.030092e-05, 0.1087677, -0.9007898 + .003},
144  {-3.654474e-10, 1.040631e-05, 0.1087085, -0.8977766 + .006},
145  {8.106988e-09, 6.983091e-06, 0.1091387, -0.9172048 + .015},
146  {9.519571e-09, 7.272678e-06, 0.1090068, -0.9140768 + .025},
147  {-5.626821e-09, 1.342186e-05, 0.1083999, -0.9065452 + .05},
148  {-9.965946e-08, 5.073548e-05, 0.1040967, -0.7672778 + .085},
149  {1.604808e-07, -5.856462e-05, 0.1185998, -1.34824 + .1},
150  {-1.511964e-07, 6.363034e-05, 0.1064627, -0.9876665 + .18},
151  };
152  float realm = logf(tr_bw / .0005f) / logf(2.f);
153  float const *c0 = coefs[av_clip((int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
154  float const *c1 = coefs[av_clip(1 + (int)realm, 0, FF_ARRAY_ELEMS(coefs) - 1)];
155  float b0 = ((c0[0] * att + c0[1]) * att + c0[2]) * att + c0[3];
156  float b1 = ((c1[0] * att + c1[1]) * att + c1[2]) * att + c1[3];
157 
158  return b0 + (b1 - b0) * (realm - (int)realm);
159  }
160  if (att > 50.f)
161  return .1102f * (att - 8.7f);
162  if (att > 20.96f)
163  return .58417f * powf(att - 20.96f, .4f) + .07886f * (att - 20.96f);
164  return 0;
165 }
166 
167 static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
168 {
169  *beta = *beta < 0.f ? kaiser_beta(att, tr_bw * .5f / Fc): *beta;
170  att = att < 60.f ? (att - 7.95f) / (2.285f * M_PI * 2.f) :
171  ((.0007528358f-1.577737e-05 * *beta) * *beta + 0.6248022f) * *beta + .06186902f;
172  *num_taps = !*num_taps ? ceilf(att/tr_bw + 1) : *num_taps;
173 }
174 
175 static float *lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
176 {
177  int n = *num_taps;
178 
179  if ((Fc /= Fn) <= 0.f || Fc >= 1.f) {
180  *num_taps = 0;
181  return NULL;
182  }
183 
184  att = att ? att : 120.f;
185 
186  kaiser_params(att, Fc, (tbw ? tbw / Fn : .05f) * .5f, beta, num_taps);
187 
188  if (!n) {
189  n = *num_taps;
190  *num_taps = av_clip(n, 11, 32767);
191  if (round)
192  *num_taps = 1 + 2 * (int)((int)((*num_taps / 2) * Fc + .5f) / Fc + .5f);
193  }
194 
195  return make_lpf(*num_taps |= 1, Fc, *beta, 0.f, 1.f, 0);
196 }
197 
198 static void invert(float *h, int n)
199 {
200  for (int i = 0; i < n; i++)
201  h[i] = -h[i];
202 
203  h[(n - 1) / 2] += 1;
204 }
205 
206 #define SQR(a) ((a) * (a))
207 
208 static float safe_log(float x)
209 {
210  av_assert0(x >= 0);
211  if (x)
212  return logf(x);
213  return -26;
214 }
215 
216 static int fir_to_phase(SincContext *s, float **h, int *len, int *post_len, float phase)
217 {
218  float *pi_wraps, *work, phase1 = (phase > 50.f ? 100.f - phase : phase) / 50.f;
219  int i, work_len, begin, end, imp_peak = 0, peak = 0;
220  float imp_sum = 0, peak_imp_sum = 0, scale = 1.f;
221  float prev_angle2 = 0, cum_2pi = 0, prev_angle1 = 0, cum_1pi = 0;
222 
223  for (i = *len, work_len = 2 * 2 * 8; i > 1; work_len <<= 1, i >>= 1);
224 
225  /* The first part is for work (+2 for (UN)PACK), the latter for pi_wraps. */
226  work = av_calloc((work_len + 2) + (work_len / 2 + 1), sizeof(float));
227  if (!work)
228  return AVERROR(ENOMEM);
229  pi_wraps = &work[work_len + 2];
230 
231  memcpy(work, *h, *len * sizeof(*work));
232 
233  av_tx_uninit(&s->tx);
234  av_tx_uninit(&s->itx);
235  av_tx_init(&s->tx, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, work_len, &scale, AV_TX_INPLACE);
236  av_tx_init(&s->itx, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, work_len, &scale, AV_TX_INPLACE);
237  if (!s->tx || !s->itx) {
238  av_free(work);
239  return AVERROR(ENOMEM);
240  }
241 
242  s->tx_fn(s->tx, work, work, sizeof(float)); /* Cepstral: */
243 
244  for (i = 0; i <= work_len; i += 2) {
245  float angle = atan2f(work[i + 1], work[i]);
246  float detect = 2 * M_PI;
247  float delta = angle - prev_angle2;
248  float adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
249 
250  prev_angle2 = angle;
251  cum_2pi += adjust;
252  angle += cum_2pi;
253  detect = M_PI;
254  delta = angle - prev_angle1;
255  adjust = detect * ((delta < -detect * .7f) - (delta > detect * .7f));
256  prev_angle1 = angle;
257  cum_1pi += fabsf(adjust); /* fabs for when 2pi and 1pi have combined */
258  pi_wraps[i >> 1] = cum_1pi;
259 
260  work[i] = safe_log(sqrtf(SQR(work[i]) + SQR(work[i + 1])));
261  work[i + 1] = 0;
262  }
263 
264  s->itx_fn(s->itx, work, work, sizeof(AVComplexFloat));
265 
266  for (i = 0; i < work_len; i++)
267  work[i] *= 2.f / work_len;
268 
269  for (i = 1; i < work_len / 2; i++) { /* Window to reject acausal components */
270  work[i] *= 2;
271  work[i + work_len / 2] = 0;
272  }
273  s->tx_fn(s->tx, work, work, sizeof(float));
274 
275  for (i = 2; i < work_len; i += 2) /* Interpolate between linear & min phase */
276  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];
277 
278  work[0] = exp(work[0]);
279  work[1] = exp(work[1]);
280  for (i = 2; i < work_len; i += 2) {
281  float x = expf(work[i]);
282 
283  work[i ] = x * cosf(work[i + 1]);
284  work[i + 1] = x * sinf(work[i + 1]);
285  }
286 
287  s->itx_fn(s->itx, work, work, sizeof(AVComplexFloat));
288  for (i = 0; i < work_len; i++)
289  work[i] *= 2.f / work_len;
290 
291  /* Find peak pos. */
292  for (i = 0; i <= (int) (pi_wraps[work_len >> 1] / M_PI + .5f); i++) {
293  imp_sum += work[i];
294  if (fabs(imp_sum) > fabs(peak_imp_sum)) {
295  peak_imp_sum = imp_sum;
296  peak = i;
297  }
298  if (work[i] > work[imp_peak]) /* For debug check only */
299  imp_peak = i;
300  }
301 
302  while (peak && fabsf(work[peak - 1]) > fabsf(work[peak]) && (work[peak - 1] * work[peak] > 0)) {
303  peak--;
304  }
305 
306  if (!phase1) {
307  begin = 0;
308  } else if (phase1 == 1) {
309  begin = peak - *len / 2;
310  } else {
311  begin = (.997f - (2 - phase1) * .22f) * *len + .5f;
312  end = (.997f + (0 - phase1) * .22f) * *len + .5f;
313  begin = peak - (begin & ~3);
314  end = peak + 1 + ((end + 3) & ~3);
315  *len = end - begin;
316  *h = av_realloc_f(*h, *len, sizeof(**h));
317  if (!*h) {
318  av_free(work);
319  return AVERROR(ENOMEM);
320  }
321  }
322 
323  for (i = 0; i < *len; i++) {
324  (*h)[i] = work[(begin + (phase > 50.f ? *len - 1 - i : i) + work_len) & (work_len - 1)];
325  }
326  *post_len = phase > 50 ? peak - begin : begin + *len - (peak + 1);
327 
328  av_log(s, AV_LOG_DEBUG, "%d nPI=%g peak-sum@%i=%g (val@%i=%g); len=%i post=%i (%g%%)\n",
329  work_len, pi_wraps[work_len >> 1] / M_PI, peak, peak_imp_sum, imp_peak,
330  work[imp_peak], *len, *post_len, 100.f - 100.f * *post_len / (*len - 1));
331 
332  av_free(work);
333 
334  return 0;
335 }
336 
337 static int config_output(AVFilterLink *outlink)
338 {
339  AVFilterContext *ctx = outlink->src;
340  SincContext *s = ctx->priv;
341  float Fn = s->sample_rate * .5f;
342  float *h[2];
343  int i, n, post_peak, longer;
344 
345  outlink->sample_rate = s->sample_rate;
346  s->pts = 0;
347 
348  if (s->Fc0 >= Fn || s->Fc1 >= Fn) {
350  "filter frequency must be less than %d/2.\n", s->sample_rate);
351  return AVERROR(EINVAL);
352  }
353 
354  h[0] = lpf(Fn, s->Fc0, s->tbw0, &s->num_taps[0], s->att, &s->beta, s->round);
355  h[1] = lpf(Fn, s->Fc1, s->tbw1, &s->num_taps[1], s->att, &s->beta, s->round);
356 
357  if (h[0])
358  invert(h[0], s->num_taps[0]);
359 
360  longer = s->num_taps[1] > s->num_taps[0];
361  n = s->num_taps[longer];
362 
363  if (h[0] && h[1]) {
364  for (i = 0; i < s->num_taps[!longer]; i++)
365  h[longer][i + (n - s->num_taps[!longer]) / 2] += h[!longer][i];
366 
367  if (s->Fc0 < s->Fc1)
368  invert(h[longer], n);
369 
370  av_free(h[!longer]);
371  }
372 
373  if (s->phase != 50.f) {
374  int ret = fir_to_phase(s, &h[longer], &n, &post_peak, s->phase);
375  if (ret < 0)
376  return ret;
377  } else {
378  post_peak = n >> 1;
379  }
380 
381  s->n = 1 << (av_log2(n) + 1);
382  s->rdft_len = 1 << av_log2(n);
383  s->coeffs = av_calloc(s->n, sizeof(*s->coeffs));
384  if (!s->coeffs)
385  return AVERROR(ENOMEM);
386 
387  for (i = 0; i < n; i++)
388  s->coeffs[i] = h[longer][i];
389  av_free(h[longer]);
390 
391  av_tx_uninit(&s->tx);
392  av_tx_uninit(&s->itx);
393 
394  return 0;
395 }
396 
398 {
399  SincContext *s = ctx->priv;
400 
401  av_freep(&s->coeffs);
402  av_tx_uninit(&s->tx);
403  av_tx_uninit(&s->itx);
404 }
405 
406 static const AVFilterPad sinc_outputs[] = {
407  {
408  .name = "default",
409  .type = AVMEDIA_TYPE_AUDIO,
410  .config_props = config_output,
411  },
412 };
413 
414 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
415 #define OFFSET(x) offsetof(SincContext, x)
416 
417 static const AVOption sinc_options[] = {
418  { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
419  { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF },
420  { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
421  { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF },
422  { "hp", "set high-pass filter frequency", OFFSET(Fc0), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
423  { "lp", "set low-pass filter frequency", OFFSET(Fc1), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, INT_MAX, AF },
424  { "phase", "set filter phase response", OFFSET(phase), AV_OPT_TYPE_FLOAT, {.dbl=50}, 0, 100, AF },
425  { "beta", "set kaiser window beta", OFFSET(beta), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 256, AF },
426  { "att", "set stop-band attenuation", OFFSET(att), AV_OPT_TYPE_FLOAT, {.dbl=120}, 40, 180, AF },
427  { "round", "enable rounding", OFFSET(round), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
428  { "hptaps", "set number of taps for high-pass filter", OFFSET(num_taps[0]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
429  { "lptaps", "set number of taps for low-pass filter", OFFSET(num_taps[1]), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF },
430  { NULL }
431 };
432 
434 
436  .name = "sinc",
437  .description = NULL_IF_CONFIG_SMALL("Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR coefficients."),
438  .priv_size = sizeof(SincContext),
439  .priv_class = &sinc_class,
440  .uninit = uninit,
441  .activate = activate,
442  .inputs = NULL,
445 };
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:100
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#define av_clip
Definition: common.h:95
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Definition: asrc_sinc.c:32
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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int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1009
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static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:947
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#define AVERROR_EOF
End of file.
Definition: error.h:57
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Definition: filter_design.txt:204
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Definition: tx_priv.h:225
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#define atan2f(y, x)
Definition: libm.h:45
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int ff_set_common_samplerates_from_list(AVFilterContext *ctx, const int *samplerates)
Equivalent to ff_set_common_samplerates(ctx, ff_make_format_list(samplerates))
Definition: formats.c:733
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AVTXContext * tx
Definition: asrc_sinc.c:44
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This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
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static float kaiser_beta(float att, float tr_bw)
Definition: asrc_sinc.c:137
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av_tx_fn tx_fn
Definition: asrc_sinc.c:45
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AVTXContext * itx
Definition: asrc_sinc.c:44
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AVOption.
Definition: opt.h:251
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#define FILTER_QUERY_FUNC(func)
Definition: internal.h:167
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#define expf(x)
Definition: libm.h:283
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#define AF
Definition: asrc_sinc.c:414
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Definition: tx.h:27
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const char * name
Filter name.
Definition: avfilter.h:175
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static const uint64_t c1
Definition: murmur3.c:51
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int sample_rate
Definition: asrc_sinc.c:35
SincContext::att
float att
Definition: asrc_sinc.c:36
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int nb_samples
Definition: asrc_sinc.c:35
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float * coeffs
Definition: asrc_sinc.c:41
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static __device__ float ceilf(float a)
Definition: cuda_runtime.h:175
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Definition: ffmpeg_filter.c:156
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av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
Definition: tx.c:883
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static float * lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
Definition: asrc_sinc.c:175
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static double b1(void *priv, double x, double y)
Definition: vf_xfade.c:1771
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#define cosf(x)
Definition: libm.h:78
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float tbw1
Definition: asrc_sinc.c:36
SincContext::Fc1
float Fc1
Definition: asrc_sinc.c:36
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int64_t pts
Definition: asrc_sinc.c:42
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static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1389
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static float * make_lpf(int num_taps, float Fc, float beta, float rho, float scale, int dc_norm)
Definition: asrc_sinc.c:109
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static __device__ float fabsf(float a)
Definition: cuda_runtime.h:181
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#define OFFSET(x)
Definition: asrc_sinc.c:415
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A filter pad used for either input or output.
Definition: internal.h:49
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static int16_t mult(Float11 *f1, Float11 *f2)
Definition: g726.c:60
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SincContext::rdft_len
int rdft_len
Definition: asrc_sinc.c:40
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#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
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#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen.c:29
av_cold
#define av_cold
Definition: attributes.h:90
av_tx_fn
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
Definition: tx.h:127
ff_outlink_set_status
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
Definition: filters.h:189
s
#define s(width, name)
Definition: cbs_vp9.c:256
adjust
static int adjust(int x, int size)
Definition: mobiclip.c:515
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
SincContext::tbw0
float tbw0
Definition: asrc_sinc.c:36
ff_set_common_formats_from_list
int ff_set_common_formats_from_list(AVFilterContext *ctx, const int *fmts)
Equivalent to ff_set_common_formats(ctx, ff_make_format_list(fmts))
Definition: formats.c:755
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
filters.h
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:201
ctx
AVFormatContext * ctx
Definition: movenc.c:48
ff_set_common_channel_layouts_from_list
int ff_set_common_channel_layouts_from_list(AVFilterContext *ctx, const AVChannelLayout *fmts)
Equivalent to ff_set_common_channel_layouts(ctx, ff_make_channel_layout_list(fmts))
Definition: formats.c:715
SincContext::n
int n
Definition: asrc_sinc.c:40
ff_asrc_sinc
const AVFilter ff_asrc_sinc
Definition: asrc_sinc.c:435
SincContext::phase
float phase
Definition: asrc_sinc.c:36
av_realloc_f
#define av_realloc_f(p, o, n)
Definition: tableprint_vlc.h:32
fir_to_phase
static int fir_to_phase(SincContext *s, float **h, int *len, int *post_len, float phase)
Definition: asrc_sinc.c:216
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
fabs
static __device__ float fabs(float a)
Definition: cuda_runtime.h:182
NULL
#define NULL
Definition: coverity.c:32
sinc_outputs
static const AVFilterPad sinc_outputs[]
Definition: asrc_sinc.c:406
SincContext::round
int round
Definition: asrc_sinc.c:38
AV_TX_INPLACE
@ AV_TX_INPLACE
Allows for in-place transformations, where input == output.
Definition: tx.h:137
work
must be printed separately If there s no standard function for printing the type you the WRITE_1D_FUNC_ARGV macro is a very quick way to create one See libavcodec dv_tablegen c for an example The h file This file should the initialization functions should not do and instead of the variable declarations the generated *_tables h file should be included Since that will be generated in the build the path must be i e not Makefile changes To make the automatic table creation work
Definition: tablegen.txt:66
activate
static int activate(AVFilterContext *ctx)
Definition: asrc_sinc.c:48
sqrtf
static __device__ float sqrtf(float a)
Definition: cuda_runtime.h:184
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: asrc_sinc.c:397
sinf
#define sinf(x)
Definition: libm.h:419
inputs
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
Definition: filter_design.txt:243
exp
int8_t exp
Definition: eval.c:72
kaiser_params
static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
Definition: asrc_sinc.c:167
SQR
#define SQR(a)
Definition: asrc_sinc.c:206
f
f
Definition: af_crystalizer.c:122
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:115
powf
#define powf(x, y)
Definition: libm.h:50
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:301
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
SincContext::beta
float beta
Definition: asrc_sinc.c:36
M_PI
#define M_PI
Definition: mathematics.h:52
av_tx_uninit
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
Definition: tx.c:294
sample_rates
sample_rates
Definition: ffmpeg_filter.c:156
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:228
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
round
static av_always_inline av_const double round(double x)
Definition: libm.h:444
invert
static void invert(float *h, int n)
Definition: asrc_sinc.c:198
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
delta
float delta
Definition: vorbis_enc_data.h:430
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
len
int len
Definition: vorbis_enc_data.h:426
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:55
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:272
AVFilter
Filter definition.
Definition: avfilter.h:171
ret
ret
Definition: filter_design.txt:187
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
config_output
static int config_output(AVFilterLink *outlink)
Definition: asrc_sinc.c:337
AV_TX_FLOAT_RDFT
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
Definition: tx.h:90
channel_layout.h
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:225
avfilter.h
SincContext::Fc0
float Fc0
Definition: asrc_sinc.c:36
AVFilterContext
An instance of a filter.
Definition: avfilter.h:415
sinc_options
static const AVOption sinc_options[]
Definition: asrc_sinc.c:417
audio.h
SincContext::num_taps
int num_taps[2]
Definition: asrc_sinc.c:37
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:368
SincContext::itx_fn
av_tx_fn itx_fn
Definition: asrc_sinc.c:45
av_free
#define av_free(p)
Definition: tableprint_vlc.h:33
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:244
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:191
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
safe_log
static float safe_log(float x)
Definition: asrc_sinc.c:208
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: asrc_sinc.c:76
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
b0
static double b0(void *priv, double x, double y)
Definition: vf_xfade.c:1770
h
h
Definition: vp9dsp_template.c:2038
ff_outlink_frame_wanted
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function. If this function returns true
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(sinc)
int
int
Definition: ffmpeg_filter.c:156
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
bessel_I_0
static float bessel_I_0(float x)
Definition: asrc_sinc.c:94
tx.h