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/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 
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 
48 } SincContext;
49 
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 
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,
86  if (ret < 0)
87  return ret;
88 
90  if (ret < 0)
91  return ret;
92 
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) {
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 
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 
422 
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,
433 };
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
av_clip
#define av_clip
Definition: common.h:99
SincContext
Definition: asrc_sinc.c:34
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
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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static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:948
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
FFERROR_NOT_READY
return FFERROR_NOT_READY
Definition: filter_design.txt:204
av_bessel_i0
double av_bessel_i0(double x)
0th order modified bessel function of the first kind.
Definition: mathematics.c:257
AVTXContext
Definition: tx_priv.h:235
atan2f
#define atan2f(y, x)
Definition: libm.h:45
ff_set_common_samplerates_from_list
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:816
SincContext::tx
AVTXContext * tx
Definition: asrc_sinc.c:46
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:374
kaiser_beta
static float kaiser_beta(float att, float tr_bw)
Definition: asrc_sinc.c:124
SincContext::tx_fn
av_tx_fn tx_fn
Definition: asrc_sinc.c:47
SincContext::itx
AVTXContext * itx
Definition: asrc_sinc.c:46
AVOption
AVOption.
Definition: opt.h:357
FILTER_QUERY_FUNC
#define FILTER_QUERY_FUNC(func)
Definition: internal.h:159
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#define expf(x)
Definition: libm.h:283
AF
#define AF
Definition: asrc_sinc.c:402
AVComplexFloat
Definition: tx.h:27
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:170
c1
static const uint64_t c1
Definition: murmur3.c:52
SincContext::sample_rate
int sample_rate
Definition: asrc_sinc.c:37
SincContext::att
float att
Definition: asrc_sinc.c:38
SincContext::nb_samples
int nb_samples
Definition: asrc_sinc.c:37
SincContext::coeffs
float * coeffs
Definition: asrc_sinc.c:43
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static __device__ float ceilf(float a)
Definition: cuda_runtime.h:175
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sample_rate
Definition: ffmpeg_filter.c:424
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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:903
formats.h
lpf
static float * lpf(float Fn, float Fc, float tbw, int *num_taps, float att, float *beta, int round)
Definition: asrc_sinc.c:162
b1
static double b1(void *priv, double x, double y)
Definition: vf_xfade.c:2035
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#define cosf(x)
Definition: libm.h:78
fail
#define fail()
Definition: checkasm.h:182
SincContext::tbw1
float tbw1
Definition: asrc_sinc.c:38
SincContext::Fc1
float Fc1
Definition: asrc_sinc.c:38
SincContext::pts
int64_t pts
Definition: asrc_sinc.c:44
make_lpf
static float * make_lpf(int num_taps, float Fc, float beta, float rho, float scale, int dc_norm)
Definition: asrc_sinc.c:96
fabsf
static __device__ float fabsf(float a)
Definition: cuda_runtime.h:181
OFFSET
#define OFFSET(x)
Definition: asrc_sinc.c:403
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:33
mult
static int16_t mult(Float11 *f1, Float11 *f2)
Definition: g726.c:60
avassert.h
SincContext::rdft_len
int rdft_len
Definition: asrc_sinc.c:42
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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
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#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:151
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:198
adjust
static int adjust(int x, int size)
Definition: mobiclip.c:513
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
SincContext::tbw0
float tbw0
Definition: asrc_sinc.c:38
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:874
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:40
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:49
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:798
SincContext::n
int n
Definition: asrc_sinc.c:42
ff_asrc_sinc
const AVFilter ff_asrc_sinc
Definition: asrc_sinc.c:423
SincContext::phase
float phase
Definition: asrc_sinc.c:38
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:203
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:394
SincContext::round
int round
Definition: asrc_sinc.c:40
AV_TX_INPLACE
@ AV_TX_INPLACE
Allows for in-place transformations, where input == output.
Definition: tx.h:161
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:50
sqrtf
static __device__ float sqrtf(float a)
Definition: cuda_runtime.h:184
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: asrc_sinc.c:385
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:73
kaiser_params
static void kaiser_params(float att, float Fc, float tr_bw, float *beta, int *num_taps)
Definition: asrc_sinc.c:154
SQR
#define SQR(a)
Definition: asrc_sinc.c:193
f
f
Definition: af_crystalizer.c:121
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:94
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:303
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
SincContext::beta
float beta
Definition: asrc_sinc.c:38
M_PI
#define M_PI
Definition: mathematics.h:67
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:295
sample_rates
sample_rates
Definition: ffmpeg_filter.c:424
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:248
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
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:185
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:39
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
AVFilter
Filter definition.
Definition: avfilter.h:166
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:325
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:245
avfilter.h
SincContext::Fc0
float Fc0
Definition: asrc_sinc.c:38
AVFilterContext
An instance of a filter.
Definition: avfilter.h:407
sinc_options
static const AVOption sinc_options[]
Definition: asrc_sinc.c:405
mem.h
audio.h
SincContext::num_taps
int num_taps[2]
Definition: asrc_sinc.c:39
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:378
SincContext::itx_fn
av_tx_fn itx_fn
Definition: asrc_sinc.c:47
av_free
#define av_free(p)
Definition: tableprint_vlc.h:33
scale
static void scale(int *out, const int *in, const int w, const int h, const int shift)
Definition: intra.c:291
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:261
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:183
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
safe_log
static float safe_log(float x)
Definition: asrc_sinc.c:195
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: asrc_sinc.c:78
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:2034
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:424
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
tx.h