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 
33 typedef struct SincContext {
34  const AVClass *class;
35 
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;
44 
47 } SincContext;
48 
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,
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) {
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 
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 
423 
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,
434 };
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:98
av_clip
#define av_clip
Definition: common.h:100
SincContext
Definition: asrc_sinc.c:33
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:1061
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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
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Definition: tx_priv.h:235
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#define atan2f(y, x)
Definition: libm.h:45
int64_t
long long int64_t
Definition: coverity.c:34
SincContext::tx
AVTXContext * tx
Definition: asrc_sinc.c:45
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static const int sample_rates[]
Definition: dcaenc.h:34
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
kaiser_beta
static float kaiser_beta(float att, float tr_bw)
Definition: asrc_sinc.c:125
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av_tx_fn tx_fn
Definition: asrc_sinc.c:46
SincContext::itx
AVTXContext * itx
Definition: asrc_sinc.c:45
AVOption
AVOption.
Definition: opt.h:429
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#define expf(x)
Definition: libm.h:283
ff_set_common_channel_layouts_from_list2
int ff_set_common_channel_layouts_from_list2(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out, const AVChannelLayout *fmts)
Definition: formats.c:920
AF
#define AF
Definition: asrc_sinc.c:403
AVComplexFloat
Definition: tx.h:27
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:205
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static const uint64_t c1
Definition: murmur3.c:52
SincContext::sample_rate
int sample_rate
Definition: asrc_sinc.c:36
SincContext::att
float att
Definition: asrc_sinc.c:37
SincContext::nb_samples
int nb_samples
Definition: asrc_sinc.c:36
SincContext::coeffs
float * coeffs
Definition: asrc_sinc.c:42
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static __device__ float ceilf(float a)
Definition: cuda_runtime.h:175
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static int query_formats(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out)
Definition: asrc_sinc.c:77
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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:163
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static double b1(void *priv, double x, double y)
Definition: vf_xfade.c:2034
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#define cosf(x)
Definition: libm.h:78
fail
#define fail()
Definition: checkasm.h:193
SincContext::tbw1
float tbw1
Definition: asrc_sinc.c:37
SincContext::Fc1
float Fc1
Definition: asrc_sinc.c:37
SincContext::pts
int64_t pts
Definition: asrc_sinc.c:43
make_lpf
static float * make_lpf(int num_taps, float Fc, float beta, float rho, float scale, int dc_norm)
Definition: asrc_sinc.c:97
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static __device__ float fabsf(float a)
Definition: cuda_runtime.h:181
OFFSET
#define OFFSET(x)
Definition: asrc_sinc.c:404
AVFilterPad
A filter pad used for either input or output.
Definition: filters.h:38
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:41
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:209
FF_ARRAY_ELEMS
#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: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:424
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:37
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:40
filters.h
ff_set_common_samplerates_from_list2
int ff_set_common_samplerates_from_list2(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out, const int *samplerates)
Definition: formats.c:944
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:230
ctx
AVFormatContext * ctx
Definition: movenc.c:49
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: filters.h:263
SincContext::n
int n
Definition: asrc_sinc.c:41
ff_asrc_sinc
const AVFilter ff_asrc_sinc
Definition: asrc_sinc.c:424
SincContext::phase
float phase
Definition: asrc_sinc.c:37
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:204
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:75
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:395
SincContext::round
int round
Definition: asrc_sinc.c:39
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:49
sqrtf
static __device__ float sqrtf(float a)
Definition: cuda_runtime.h:184
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: asrc_sinc.c:386
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:155
AVFilterFormatsConfig
Lists of formats / etc.
Definition: avfilter.h:111
SQR
#define SQR(a)
Definition: asrc_sinc.c:194
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: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:318
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
SincContext::beta
float beta
Definition: asrc_sinc.c:37
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
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Underlying C type is float.
Definition: opt.h:271
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:186
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
delta
float delta
Definition: vorbis_enc_data.h:430
FILTER_QUERY_FUNC2
#define FILTER_QUERY_FUNC2(func)
Definition: filters.h:239
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: filters.h:44
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
AVFilter
Filter definition.
Definition: avfilter.h:201
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:326
AV_TX_FLOAT_RDFT
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
Definition: tx.h:90
ff_set_common_formats_from_list2
int ff_set_common_formats_from_list2(const AVFilterContext *ctx, AVFilterFormatsConfig **cfg_in, AVFilterFormatsConfig **cfg_out, const int *fmts)
Definition: formats.c:1016
channel_layout.h
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Underlying C type is int.
Definition: opt.h:259
avfilter.h
SincContext::Fc0
float Fc0
Definition: asrc_sinc.c:37
AVFilterContext
An instance of a filter.
Definition: avfilter.h:457
sinc_options
static const AVOption sinc_options[]
Definition: asrc_sinc.c:406
mem.h
audio.h
SincContext::num_taps
int num_taps[2]
Definition: asrc_sinc.c:38
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:393
SincContext::itx_fn
av_tx_fn itx_fn
Definition: asrc_sinc.c:46
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
Underlying C type is int.
Definition: opt.h:327
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
safe_log
static float safe_log(float x)
Definition: asrc_sinc.c:196
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:2033
h
h
Definition: vp9dsp_template.c:2070
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)
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