FFmpeg  4.4.5
af_asoftclip.c
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1 /*
2  * Copyright (c) 2019 The FFmpeg Project
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/avassert.h"
23 #include "libavutil/opt.h"
25 #include "avfilter.h"
26 #include "audio.h"
27 #include "formats.h"
28 
30  ASC_HARD = -1,
40 };
41 
42 typedef struct ASoftClipContext {
43  const AVClass *class;
44 
45  int type;
48  double threshold;
49  double output;
50  double param;
51 
54 
56 
57  void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
58  int nb_samples, int channels, int start, int end);
60 
61 #define OFFSET(x) offsetof(ASoftClipContext, x)
62 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
63 #define F AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
64 
65 static const AVOption asoftclip_options[] = {
66  { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, -1, NB_TYPES-1, A, "types" },
67  { "hard", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_HARD}, 0, 0, A, "types" },
68  { "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" },
69  { "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" },
70  { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" },
71  { "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, "types" },
72  { "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" },
73  { "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
74  { "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
75  { "erf", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ERF}, 0, 0, A, "types" },
76  { "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
77  { "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
78  { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
79  { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
80  { NULL }
81 };
82 
84 
86 {
89  static const enum AVSampleFormat sample_fmts[] = {
93  };
94  int ret;
95 
97  if (!formats)
98  return AVERROR(ENOMEM);
100  if (ret < 0)
101  return ret;
102 
104  if (!layouts)
105  return AVERROR(ENOMEM);
106 
108  if (ret < 0)
109  return ret;
110 
113 }
114 
116  void **dptr, const void **sptr,
117  int nb_samples, int channels,
118  int start, int end)
119 {
120  float threshold = s->threshold;
121  float gain = s->output * threshold;
122  float factor = 1.f / threshold;
123  float param = s->param;
124 
125  for (int c = start; c < end; c++) {
126  const float *src = sptr[c];
127  float *dst = dptr[c];
128 
129  switch (s->type) {
130  case ASC_HARD:
131  for (int n = 0; n < nb_samples; n++) {
132  dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
133  dst[n] *= gain;
134  }
135  break;
136  case ASC_TANH:
137  for (int n = 0; n < nb_samples; n++) {
138  dst[n] = tanhf(src[n] * factor * param);
139  dst[n] *= gain;
140  }
141  break;
142  case ASC_ATAN:
143  for (int n = 0; n < nb_samples; n++) {
144  dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
145  dst[n] *= gain;
146  }
147  break;
148  case ASC_CUBIC:
149  for (int n = 0; n < nb_samples; n++) {
150  float sample = src[n] * factor;
151 
152  if (FFABS(sample) >= 1.5f)
153  dst[n] = FFSIGN(sample);
154  else
155  dst[n] = sample - 0.1481f * powf(sample, 3.f);
156  dst[n] *= gain;
157  }
158  break;
159  case ASC_EXP:
160  for (int n = 0; n < nb_samples; n++) {
161  dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
162  dst[n] *= gain;
163  }
164  break;
165  case ASC_ALG:
166  for (int n = 0; n < nb_samples; n++) {
167  float sample = src[n] * factor;
168 
169  dst[n] = sample / (sqrtf(param + sample * sample));
170  dst[n] *= gain;
171  }
172  break;
173  case ASC_QUINTIC:
174  for (int n = 0; n < nb_samples; n++) {
175  float sample = src[n] * factor;
176 
177  if (FFABS(sample) >= 1.25)
178  dst[n] = FFSIGN(sample);
179  else
180  dst[n] = sample - 0.08192f * powf(sample, 5.f);
181  dst[n] *= gain;
182  }
183  break;
184  case ASC_SIN:
185  for (int n = 0; n < nb_samples; n++) {
186  float sample = src[n] * factor;
187 
188  if (FFABS(sample) >= M_PI_2)
189  dst[n] = FFSIGN(sample);
190  else
191  dst[n] = sinf(sample);
192  dst[n] *= gain;
193  }
194  break;
195  case ASC_ERF:
196  for (int n = 0; n < nb_samples; n++) {
197  dst[n] = erff(src[n] * factor);
198  dst[n] *= gain;
199  }
200  break;
201  default:
202  av_assert0(0);
203  }
204  }
205 }
206 
208  void **dptr, const void **sptr,
209  int nb_samples, int channels,
210  int start, int end)
211 {
212  double threshold = s->threshold;
213  double gain = s->output * threshold;
214  double factor = 1. / threshold;
215  double param = s->param;
216 
217  for (int c = start; c < end; c++) {
218  const double *src = sptr[c];
219  double *dst = dptr[c];
220 
221  switch (s->type) {
222  case ASC_HARD:
223  for (int n = 0; n < nb_samples; n++) {
224  dst[n] = av_clipd(src[n] * factor, -1., 1.);
225  dst[n] *= gain;
226  }
227  break;
228  case ASC_TANH:
229  for (int n = 0; n < nb_samples; n++) {
230  dst[n] = tanh(src[n] * factor * param);
231  dst[n] *= gain;
232  }
233  break;
234  case ASC_ATAN:
235  for (int n = 0; n < nb_samples; n++) {
236  dst[n] = 2. / M_PI * atan(src[n] * factor * param);
237  dst[n] *= gain;
238  }
239  break;
240  case ASC_CUBIC:
241  for (int n = 0; n < nb_samples; n++) {
242  double sample = src[n] * factor;
243 
244  if (FFABS(sample) >= 1.5)
245  dst[n] = FFSIGN(sample);
246  else
247  dst[n] = sample - 0.1481 * pow(sample, 3.);
248  dst[n] *= gain;
249  }
250  break;
251  case ASC_EXP:
252  for (int n = 0; n < nb_samples; n++) {
253  dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
254  dst[n] *= gain;
255  }
256  break;
257  case ASC_ALG:
258  for (int n = 0; n < nb_samples; n++) {
259  double sample = src[n] * factor;
260 
261  dst[n] = sample / (sqrt(param + sample * sample));
262  dst[n] *= gain;
263  }
264  break;
265  case ASC_QUINTIC:
266  for (int n = 0; n < nb_samples; n++) {
267  double sample = src[n] * factor;
268 
269  if (FFABS(sample) >= 1.25)
270  dst[n] = FFSIGN(sample);
271  else
272  dst[n] = sample - 0.08192 * pow(sample, 5.);
273  dst[n] *= gain;
274  }
275  break;
276  case ASC_SIN:
277  for (int n = 0; n < nb_samples; n++) {
278  double sample = src[n] * factor;
279 
280  if (FFABS(sample) >= M_PI_2)
281  dst[n] = FFSIGN(sample);
282  else
283  dst[n] = sin(sample);
284  dst[n] *= gain;
285  }
286  break;
287  case ASC_ERF:
288  for (int n = 0; n < nb_samples; n++) {
289  dst[n] = erf(src[n] * factor);
290  dst[n] *= gain;
291  }
292  break;
293  default:
294  av_assert0(0);
295  }
296  }
297 }
298 
299 static int config_input(AVFilterLink *inlink)
300 {
301  AVFilterContext *ctx = inlink->dst;
302  ASoftClipContext *s = ctx->priv;
303  int ret;
304 
305  switch (inlink->format) {
306  case AV_SAMPLE_FMT_FLT:
307  case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
308  case AV_SAMPLE_FMT_DBL:
309  case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
310  default: av_assert0(0);
311  }
312 
313  if (s->oversample <= 1)
314  return 0;
315 
316  s->up_ctx = swr_alloc();
317  s->down_ctx = swr_alloc();
318  if (!s->up_ctx || !s->down_ctx)
319  return AVERROR(ENOMEM);
320 
321  av_opt_set_int(s->up_ctx, "in_channel_layout", inlink->channel_layout, 0);
322  av_opt_set_int(s->up_ctx, "in_sample_rate", inlink->sample_rate, 0);
323  av_opt_set_sample_fmt(s->up_ctx, "in_sample_fmt", inlink->format, 0);
324 
325  av_opt_set_int(s->up_ctx, "out_channel_layout", inlink->channel_layout, 0);
326  av_opt_set_int(s->up_ctx, "out_sample_rate", inlink->sample_rate * s->oversample, 0);
327  av_opt_set_sample_fmt(s->up_ctx, "out_sample_fmt", inlink->format, 0);
328 
329  av_opt_set_int(s->down_ctx, "in_channel_layout", inlink->channel_layout, 0);
330  av_opt_set_int(s->down_ctx, "in_sample_rate", inlink->sample_rate * s->oversample, 0);
331  av_opt_set_sample_fmt(s->down_ctx, "in_sample_fmt", inlink->format, 0);
332 
333  av_opt_set_int(s->down_ctx, "out_channel_layout", inlink->channel_layout, 0);
334  av_opt_set_int(s->down_ctx, "out_sample_rate", inlink->sample_rate, 0);
335  av_opt_set_sample_fmt(s->down_ctx, "out_sample_fmt", inlink->format, 0);
336 
337  ret = swr_init(s->up_ctx);
338  if (ret < 0)
339  return ret;
340 
341  ret = swr_init(s->down_ctx);
342  if (ret < 0)
343  return ret;
344 
345  return 0;
346 }
347 
348 typedef struct ThreadData {
349  AVFrame *in, *out;
351  int channels;
352 } ThreadData;
353 
354 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
355 {
356  ASoftClipContext *s = ctx->priv;
357  ThreadData *td = arg;
358  AVFrame *out = td->out;
359  AVFrame *in = td->in;
360  const int channels = td->channels;
361  const int nb_samples = td->nb_samples;
362  const int start = (channels * jobnr) / nb_jobs;
363  const int end = (channels * (jobnr+1)) / nb_jobs;
364 
365  s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
366  nb_samples, channels, start, end);
367 
368  return 0;
369 }
370 
371 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
372 {
373  AVFilterContext *ctx = inlink->dst;
374  ASoftClipContext *s = ctx->priv;
375  AVFilterLink *outlink = ctx->outputs[0];
376  int ret, nb_samples, channels;
377  ThreadData td;
378  AVFrame *out;
379 
380  if (av_frame_is_writable(in)) {
381  out = in;
382  } else {
383  out = ff_get_audio_buffer(outlink, in->nb_samples);
384  if (!out) {
385  av_frame_free(&in);
386  return AVERROR(ENOMEM);
387  }
389  }
390 
391  if (av_sample_fmt_is_planar(in->format)) {
392  nb_samples = in->nb_samples;
393  channels = in->channels;
394  } else {
395  nb_samples = in->channels * in->nb_samples;
396  channels = 1;
397  }
398 
399  if (s->oversample > 1) {
400  s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
401  if (!s->frame) {
402  ret = AVERROR(ENOMEM);
403  goto fail;
404  }
405 
406  ret = swr_convert(s->up_ctx, (uint8_t**)s->frame->extended_data, in->nb_samples * s->oversample,
407  (const uint8_t **)in->extended_data, in->nb_samples);
408  if (ret < 0)
409  goto fail;
410 
411  td.in = s->frame;
412  td.out = s->frame;
413  td.nb_samples = av_sample_fmt_is_planar(in->format) ? ret : ret * in->channels;
414  td.channels = channels;
417 
418  ret = swr_convert(s->down_ctx, (uint8_t**)out->extended_data, out->nb_samples,
419  (const uint8_t **)s->frame->extended_data, ret);
420  if (ret < 0)
421  goto fail;
422 
423  if (out->pts)
424  out->pts -= s->delay;
425  s->delay += in->nb_samples - ret;
426  out->nb_samples = ret;
427 
428  av_frame_free(&s->frame);
429  } else {
430  td.in = in;
431  td.out = out;
432  td.nb_samples = nb_samples;
433  td.channels = channels;
436  }
437 
438  if (out != in)
439  av_frame_free(&in);
440 
441  return ff_filter_frame(outlink, out);
442 fail:
443  if (out != in)
444  av_frame_free(&out);
445  av_frame_free(&in);
446  av_frame_free(&s->frame);
447 
448  return ret;
449 }
450 
452 {
453  ASoftClipContext *s = ctx->priv;
454 
455  swr_free(&s->up_ctx);
456  swr_free(&s->down_ctx);
457 }
458 
459 static const AVFilterPad inputs[] = {
460  {
461  .name = "default",
462  .type = AVMEDIA_TYPE_AUDIO,
463  .filter_frame = filter_frame,
464  .config_props = config_input,
465  },
466  { NULL }
467 };
468 
469 static const AVFilterPad outputs[] = {
470  {
471  .name = "default",
472  .type = AVMEDIA_TYPE_AUDIO,
473  },
474  { NULL }
475 };
476 
478  .name = "asoftclip",
479  .description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
480  .query_formats = query_formats,
481  .priv_size = sizeof(ASoftClipContext),
482  .priv_class = &asoftclip_class,
483  .inputs = inputs,
484  .outputs = outputs,
485  .uninit = uninit,
489 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_acrusher.c:336
static const AVOption asoftclip_options[]
Definition: af_asoftclip.c:65
static void filter_dbl(ASoftClipContext *s, void **dptr, const void **sptr, int nb_samples, int channels, int start, int end)
Definition: af_asoftclip.c:207
#define F
Definition: af_asoftclip.c:63
static int query_formats(AVFilterContext *ctx)
Definition: af_asoftclip.c:85
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_asoftclip.c:354
static int config_input(AVFilterLink *inlink)
Definition: af_asoftclip.c:299
ASoftClipTypes
Definition: af_asoftclip.c:29
@ ASC_HARD
Definition: af_asoftclip.c:30
@ ASC_ALG
Definition: af_asoftclip.c:35
@ ASC_SIN
Definition: af_asoftclip.c:37
@ ASC_TANH
Definition: af_asoftclip.c:31
@ ASC_ATAN
Definition: af_asoftclip.c:32
@ ASC_ERF
Definition: af_asoftclip.c:38
@ ASC_CUBIC
Definition: af_asoftclip.c:33
@ ASC_EXP
Definition: af_asoftclip.c:34
@ ASC_QUINTIC
Definition: af_asoftclip.c:36
@ NB_TYPES
Definition: af_asoftclip.c:39
static const AVFilterPad inputs[]
Definition: af_asoftclip.c:459
static const AVFilterPad outputs[]
Definition: af_asoftclip.c:469
static void filter_flt(ASoftClipContext *s, void **dptr, const void **sptr, int nb_samples, int channels, int start, int end)
Definition: af_asoftclip.c:115
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_asoftclip.c:371
#define A
Definition: af_asoftclip.c:62
AVFILTER_DEFINE_CLASS(asoftclip)
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_asoftclip.c:451
AVFilter ff_af_asoftclip
Definition: af_asoftclip.c:477
#define OFFSET(x)
Definition: af_asoftclip.c:61
channels
Definition: aptx.h:33
#define av_cold
Definition: attributes.h:88
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1096
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Definition: avfilter.c:882
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:802
Main libavfilter public API header.
#define flags(name, subs,...)
Definition: cbs_av1.c:572
#define s(width, name)
Definition: cbs_vp9.c:257
audio channel layout utility functions
#define fail()
Definition: checkasm.h:133
#define FFMIN(a, b)
Definition: common.h:105
#define av_clipd
Definition: common.h:173
#define av_clipf
Definition: common.h:170
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define FFSIGN(a)
Definition: common.h:73
#define NULL
Definition: coverity.c:32
long long int64_t
Definition: coverity.c:34
int8_t exp
Definition: eval.c:72
#define sample
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:436
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:587
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:286
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:568
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_INT
Definition: opt.h:225
@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:227
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
Definition: avfilter.h:126
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
#define AVERROR(e)
Definition: error.h:43
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:658
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
av_cold void swr_free(SwrContext **ss)
Free the given SwrContext and set the pointer to NULL.
Definition: swresample.c:137
av_cold int swr_init(struct SwrContext *s)
Initialize context after user parameters have been set.
Definition: swresample.c:152
av_cold struct SwrContext * swr_alloc(void)
Allocate SwrContext.
Definition: options.c:149
int av_opt_set_int(void *obj, const char *name, int64_t val, int search_flags)
Definition: opt.c:586
int av_opt_set_sample_fmt(void *obj, const char *name, enum AVSampleFormat fmt, int search_flags)
Definition: opt.c:704
cl_device_type type
const char * arg
Definition: jacosubdec.c:66
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
static double erf(double z)
erf function Algorithm taken from the Boost project, source: http://www.boost.org/doc/libs/1_46_1/boo...
Definition: libm.h:121
#define atanf(x)
Definition: libm.h:40
#define sinf(x)
Definition: libm.h:419
#define expf(x)
Definition: libm.h:283
#define powf(x, y)
Definition: libm.h:50
#define M_PI_2
Definition: mathematics.h:55
#define M_PI
Definition: mathematics.h:52
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVOptions.
#define td
Definition: regdef.h:70
typedef void(RENAME(mix_any_func_type))
formats
Definition: signature.h:48
AVFrame * frame
Definition: af_asoftclip.c:55
SwrContext * down_ctx
Definition: af_asoftclip.c:53
void(* filter)(struct ASoftClipContext *s, void **dst, const void **src, int nb_samples, int channels, int start, int end)
Definition: af_asoftclip.c:57
SwrContext * up_ctx
Definition: af_asoftclip.c:52
Describe the class of an AVClass context structure.
Definition: log.h:67
A list of supported channel layouts.
Definition: formats.h:86
An instance of a filter.
Definition: avfilter.h:341
A list of supported formats for one end of a filter link.
Definition: formats.h:65
A filter pad used for either input or output.
Definition: internal.h:54
const char * name
Pad name.
Definition: internal.h:60
Filter definition.
Definition: avfilter.h:145
const char * name
Filter name.
Definition: avfilter.h:149
AVFormatInternal * internal
An opaque field for libavformat internal usage.
Definition: avformat.h:1699
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
AVOption.
Definition: opt.h:248
The libswresample context.
Used for passing data between threads.
Definition: dsddec.c:67
AVFrame * out
Definition: af_adeclick.c:502
AVFrame * in
Definition: af_adenorm.c:223
int attribute_align_arg swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, const uint8_t *in_arg[SWR_CH_MAX], int in_count)
Definition: swresample.c:714
libswresample public header
#define src
Definition: vp8dsp.c:255
FILE * out
Definition: movenc.c:54
AVFormatContext * ctx
Definition: movenc.c:48
static const int factor[16]
Definition: vf_pp7.c:77
static double c[64]