FFmpeg  4.3.7
af_sidechaincompress.c
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1 /*
2  * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
3  * Copyright (c) 2015 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 /**
23  * @file
24  * Audio (Sidechain) Compressor filter
25  */
26 
27 #include "libavutil/audio_fifo.h"
28 #include "libavutil/avassert.h"
30 #include "libavutil/common.h"
31 #include "libavutil/opt.h"
32 
33 #include "audio.h"
34 #include "avfilter.h"
35 #include "filters.h"
36 #include "formats.h"
37 #include "hermite.h"
38 #include "internal.h"
39 
40 typedef struct SidechainCompressContext {
41  const AVClass *class;
42 
43  double level_in;
44  double level_sc;
47  double lin_slope;
48  double ratio;
49  double threshold;
50  double makeup;
51  double mix;
52  double thres;
53  double knee;
54  double knee_start;
55  double knee_stop;
57  double lin_knee_stop;
59  double adj_knee_stop;
62  int link;
63  int detection;
64  int mode;
65 
67  int64_t pts;
69 
70 #define OFFSET(x) offsetof(SidechainCompressContext, x)
71 #define A AV_OPT_FLAG_AUDIO_PARAM
72 #define F AV_OPT_FLAG_FILTERING_PARAM
73 #define R AV_OPT_FLAG_RUNTIME_PARAM
74 
75 static const AVOption options[] = {
76  { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F|R },
77  { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A|F|R, "mode" },
78  { "downward",0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F|R, "mode" },
79  { "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F|R, "mode" },
80  { "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0.000976563, 1, A|F|R },
81  { "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 20, A|F|R },
82  { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 2000, A|F|R },
83  { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A|F|R },
84  { "makeup", "set make up gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 64, A|F|R },
85  { "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.82843}, 1, 8, A|F|R },
86  { "link", "set link type", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A|F|R, "link" },
87  { "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F|R, "link" },
88  { "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F|R, "link" },
89  { "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A|F|R, "detection" },
90  { "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F|R, "detection" },
91  { "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F|R, "detection" },
92  { "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F|R },
93  { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A|F|R },
94  { NULL }
95 };
96 
97 #define sidechaincompress_options options
98 AVFILTER_DEFINE_CLASS(sidechaincompress);
99 
100 // A fake infinity value (because real infinity may break some hosts)
101 #define FAKE_INFINITY (65536.0 * 65536.0)
102 
103 // Check for infinity (with appropriate-ish tolerance)
104 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
105 
106 static double output_gain(double lin_slope, double ratio, double thres,
107  double knee, double knee_start, double knee_stop,
108  double compressed_knee_start,
109  double compressed_knee_stop,
110  int detection, int mode)
111 {
112  double slope = log(lin_slope);
113  double gain = 0.0;
114  double delta = 0.0;
115 
116  if (detection)
117  slope *= 0.5;
118 
119  if (IS_FAKE_INFINITY(ratio)) {
120  gain = thres;
121  delta = 0.0;
122  } else {
123  gain = (slope - thres) / ratio + thres;
124  delta = 1.0 / ratio;
125  }
126 
127  if (mode) {
128  if (knee > 1.0 && slope > knee_start)
129  gain = hermite_interpolation(slope, knee_stop, knee_start,
130  knee_stop, compressed_knee_start,
131  1.0, delta);
132  } else {
133  if (knee > 1.0 && slope < knee_stop)
134  gain = hermite_interpolation(slope, knee_start, knee_stop,
135  knee_start, compressed_knee_stop,
136  1.0, delta);
137  }
138 
139  return exp(gain - slope);
140 }
141 
143 {
144  AVFilterContext *ctx = outlink->src;
146 
147  s->thres = log(s->threshold);
148  s->lin_knee_start = s->threshold / sqrt(s->knee);
149  s->lin_knee_stop = s->threshold * sqrt(s->knee);
152  s->knee_start = log(s->lin_knee_start);
153  s->knee_stop = log(s->lin_knee_stop);
154  s->compressed_knee_start = (s->knee_start - s->thres) / s->ratio + s->thres;
155  s->compressed_knee_stop = (s->knee_stop - s->thres) / s->ratio + s->thres;
156 
157  s->attack_coeff = FFMIN(1., 1. / (s->attack * outlink->sample_rate / 4000.));
158  s->release_coeff = FFMIN(1., 1. / (s->release * outlink->sample_rate / 4000.));
159 
160  return 0;
161 }
162 
164  const double *src, double *dst, const double *scsrc, int nb_samples,
165  double level_in, double level_sc,
166  AVFilterLink *inlink, AVFilterLink *sclink)
167 {
168  const double makeup = s->makeup;
169  const double mix = s->mix;
170  int i, c;
171 
172  for (i = 0; i < nb_samples; i++) {
173  double abs_sample, gain = 1.0;
174  double detector;
175  int detected;
176 
177  abs_sample = fabs(scsrc[0] * level_sc);
178 
179  if (s->link == 1) {
180  for (c = 1; c < sclink->channels; c++)
181  abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
182  } else {
183  for (c = 1; c < sclink->channels; c++)
184  abs_sample += fabs(scsrc[c] * level_sc);
185 
186  abs_sample /= sclink->channels;
187  }
188 
189  if (s->detection)
190  abs_sample *= abs_sample;
191 
192  s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? s->attack_coeff : s->release_coeff);
193 
194  if (s->mode) {
195  detector = (s->detection ? s->adj_knee_stop : s->lin_knee_stop);
196  detected = s->lin_slope < detector;
197  } else {
198  detector = (s->detection ? s->adj_knee_start : s->lin_knee_start);
199  detected = s->lin_slope > detector;
200  }
201 
202  if (s->lin_slope > 0.0 && detected)
203  gain = output_gain(s->lin_slope, s->ratio, s->thres, s->knee,
204  s->knee_start, s->knee_stop,
207  s->detection, s->mode);
208 
209  for (c = 0; c < inlink->channels; c++)
210  dst[c] = src[c] * level_in * (gain * makeup * mix + (1. - mix));
211 
212  src += inlink->channels;
213  dst += inlink->channels;
214  scsrc += sclink->channels;
215  }
216 }
217 
218 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
219  char *res, int res_len, int flags)
220 {
221  int ret;
222 
223  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
224  if (ret < 0)
225  return ret;
226 
228 
229  return 0;
230 }
231 
232 #if CONFIG_SIDECHAINCOMPRESS_FILTER
233 static int activate(AVFilterContext *ctx)
234 {
236  AVFrame *out = NULL, *in[2] = { NULL };
237  int ret, i, nb_samples;
238  double *dst;
239 
241  if ((ret = ff_inlink_consume_frame(ctx->inputs[0], &in[0])) > 0) {
242  av_audio_fifo_write(s->fifo[0], (void **)in[0]->extended_data,
243  in[0]->nb_samples);
244  av_frame_free(&in[0]);
245  }
246  if (ret < 0)
247  return ret;
248  if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &in[1])) > 0) {
249  av_audio_fifo_write(s->fifo[1], (void **)in[1]->extended_data,
250  in[1]->nb_samples);
251  av_frame_free(&in[1]);
252  }
253  if (ret < 0)
254  return ret;
255 
256  nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
257  if (nb_samples) {
258  out = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
259  if (!out)
260  return AVERROR(ENOMEM);
261  for (i = 0; i < 2; i++) {
262  in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
263  if (!in[i]) {
264  av_frame_free(&in[0]);
265  av_frame_free(&in[1]);
266  av_frame_free(&out);
267  return AVERROR(ENOMEM);
268  }
269  av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
270  }
271 
272  dst = (double *)out->data[0];
273  out->pts = s->pts;
274  s->pts += av_rescale_q(nb_samples, (AVRational){1, ctx->outputs[0]->sample_rate}, ctx->outputs[0]->time_base);
275 
276  compressor(s, (double *)in[0]->data[0], dst,
277  (double *)in[1]->data[0], nb_samples,
278  s->level_in, s->level_sc,
279  ctx->inputs[0], ctx->inputs[1]);
280 
281  av_frame_free(&in[0]);
282  av_frame_free(&in[1]);
283 
284  ret = ff_filter_frame(ctx->outputs[0], out);
285  if (ret < 0)
286  return ret;
287  }
288  FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
289  FF_FILTER_FORWARD_STATUS(ctx->inputs[1], ctx->outputs[0]);
290  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
291  if (!av_audio_fifo_size(s->fifo[0]))
293  if (!av_audio_fifo_size(s->fifo[1]))
295  }
296  return 0;
297 }
298 
299 static int query_formats(AVFilterContext *ctx)
300 {
303  static const enum AVSampleFormat sample_fmts[] = {
306  };
307  int ret, i;
308 
309  if (!ctx->inputs[0]->in_channel_layouts ||
311  av_log(ctx, AV_LOG_WARNING,
312  "No channel layout for input 1\n");
313  return AVERROR(EAGAIN);
314  }
315 
316  if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||
317  (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
318  return ret;
319 
320  for (i = 0; i < 2; i++) {
321  layouts = ff_all_channel_counts();
322  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
323  return ret;
324  }
325 
326  formats = ff_make_format_list(sample_fmts);
327  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
328  return ret;
329 
330  formats = ff_all_samplerates();
331  return ff_set_common_samplerates(ctx, formats);
332 }
333 
334 static int config_output(AVFilterLink *outlink)
335 {
336  AVFilterContext *ctx = outlink->src;
338 
339  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
340  av_log(ctx, AV_LOG_ERROR,
341  "Inputs must have the same sample rate "
342  "%d for in0 vs %d for in1\n",
343  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
344  return AVERROR(EINVAL);
345  }
346 
347  outlink->sample_rate = ctx->inputs[0]->sample_rate;
348  outlink->time_base = ctx->inputs[0]->time_base;
349  outlink->channel_layout = ctx->inputs[0]->channel_layout;
350  outlink->channels = ctx->inputs[0]->channels;
351 
352  s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
353  s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
354  if (!s->fifo[0] || !s->fifo[1])
355  return AVERROR(ENOMEM);
356 
357  compressor_config_output(outlink);
358 
359  return 0;
360 }
361 
362 static av_cold void uninit(AVFilterContext *ctx)
363 {
365 
366  av_audio_fifo_free(s->fifo[0]);
367  av_audio_fifo_free(s->fifo[1]);
368 }
369 
370 static const AVFilterPad sidechaincompress_inputs[] = {
371  {
372  .name = "main",
373  .type = AVMEDIA_TYPE_AUDIO,
374  },{
375  .name = "sidechain",
376  .type = AVMEDIA_TYPE_AUDIO,
377  },
378  { NULL }
379 };
380 
381 static const AVFilterPad sidechaincompress_outputs[] = {
382  {
383  .name = "default",
384  .type = AVMEDIA_TYPE_AUDIO,
385  .config_props = config_output,
386  },
387  { NULL }
388 };
389 
391  .name = "sidechaincompress",
392  .description = NULL_IF_CONFIG_SMALL("Sidechain compressor."),
393  .priv_size = sizeof(SidechainCompressContext),
394  .priv_class = &sidechaincompress_class,
396  .activate = activate,
397  .uninit = uninit,
398  .inputs = sidechaincompress_inputs,
399  .outputs = sidechaincompress_outputs,
401 };
402 #endif /* CONFIG_SIDECHAINCOMPRESS_FILTER */
403 
404 #if CONFIG_ACOMPRESSOR_FILTER
405 static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *in)
406 {
407  const double *src = (const double *)in->data[0];
408  AVFilterContext *ctx = inlink->dst;
410  AVFilterLink *outlink = ctx->outputs[0];
411  AVFrame *out;
412  double *dst;
413 
414  if (av_frame_is_writable(in)) {
415  out = in;
416  } else {
417  out = ff_get_audio_buffer(outlink, in->nb_samples);
418  if (!out) {
419  av_frame_free(&in);
420  return AVERROR(ENOMEM);
421  }
423  }
424  dst = (double *)out->data[0];
425 
426  compressor(s, src, dst, src, in->nb_samples,
427  s->level_in, s->level_in,
428  inlink, inlink);
429 
430  if (out != in)
431  av_frame_free(&in);
432  return ff_filter_frame(outlink, out);
433 }
434 
435 static int acompressor_query_formats(AVFilterContext *ctx)
436 {
439  static const enum AVSampleFormat sample_fmts[] = {
442  };
443  int ret;
444 
445  layouts = ff_all_channel_counts();
446  if (!layouts)
447  return AVERROR(ENOMEM);
448  ret = ff_set_common_channel_layouts(ctx, layouts);
449  if (ret < 0)
450  return ret;
451 
452  formats = ff_make_format_list(sample_fmts);
453  if (!formats)
454  return AVERROR(ENOMEM);
455  ret = ff_set_common_formats(ctx, formats);
456  if (ret < 0)
457  return ret;
458 
459  formats = ff_all_samplerates();
460  if (!formats)
461  return AVERROR(ENOMEM);
462  return ff_set_common_samplerates(ctx, formats);
463 }
464 
465 #define acompressor_options options
466 AVFILTER_DEFINE_CLASS(acompressor);
467 
468 static const AVFilterPad acompressor_inputs[] = {
469  {
470  .name = "default",
471  .type = AVMEDIA_TYPE_AUDIO,
472  .filter_frame = acompressor_filter_frame,
473  },
474  { NULL }
475 };
476 
477 static const AVFilterPad acompressor_outputs[] = {
478  {
479  .name = "default",
480  .type = AVMEDIA_TYPE_AUDIO,
481  .config_props = compressor_config_output,
482  },
483  { NULL }
484 };
485 
487  .name = "acompressor",
488  .description = NULL_IF_CONFIG_SMALL("Audio compressor."),
489  .priv_size = sizeof(SidechainCompressContext),
490  .priv_class = &acompressor_class,
491  .query_formats = acompressor_query_formats,
492  .inputs = acompressor_inputs,
493  .outputs = acompressor_outputs,
495 };
496 #endif /* CONFIG_ACOMPRESSOR_FILTER */
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link&#39;s FIFO and update the link&#39;s stats.
Definition: avfilter.c:1476
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:586
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
AVOption.
Definition: opt.h:246
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
Main libavfilter public API header.
#define F
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
AVFilter ff_af_sidechaincompress
static int compressor_config_output(AVFilterLink *outlink)
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1602
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:300
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:479
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1075
#define av_cold
Definition: attributes.h:88
static av_cold int uninit(AVCodecContext *avctx)
Definition: crystalhd.c:279
float delta
AVOptions.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:393
const char data[16]
Definition: mxf.c:91
#define A
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
#define src
Definition: vp8dsp.c:254
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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:605
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:356
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
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:188
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:869
void * priv
private data for use by the filter
Definition: avfilter.h:353
uint64_t * channel_layouts
list of channel layouts
Definition: formats.h:86
static int config_output(AVFilterLink *outlink)
Definition: af_aecho.c:234
#define OFFSET(x)
simple assert() macros that are a bit more flexible than ISO C assert().
#define FFMAX(a, b)
Definition: common.h:94
AVFILTER_DEFINE_CLASS(sidechaincompress)
int8_t exp
Definition: eval.c:72
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
audio channel layout utility functions
static double hermite_interpolation(double x, double x0, double x1, double p0, double p1, double m0, double m1)
Definition: hermite.h:22
#define FFMIN(a, b)
Definition: common.h:96
AVFormatContext * ctx
Definition: movenc.c:48
static int activate(AVFilterContext *ctx)
Definition: af_adeclick.c:622
#define s(width, name)
Definition: cbs_vp9.c:257
#define R
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
static void compressor(SidechainCompressContext *s, const double *src, double *dst, const double *scsrc, int nb_samples, double level_in, double level_sc, AVFilterLink *inlink, AVFilterLink *sclink)
int nb_samples
number of samples currently in the FIFO
Definition: audio_fifo.c:37
#define IS_FAKE_INFINITY(value)
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:595
static double output_gain(double lin_slope, double ratio, double thres, double knee, double knee_start, double knee_stop, double compressed_knee_start, double compressed_knee_stop, int detection, int mode)
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
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)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
Definition: filters.h:226
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:439
#define flags(name, subs,...)
Definition: cbs_av1.c:565
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:314
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
static const AVOption options[]
static int query_formats(AVFilterContext *ctx)
Definition: aeval.c:244
common internal and external API header
if(ret< 0)
Definition: vf_mcdeint.c:279
int nb_channel_layouts
number of channel layouts
Definition: formats.h:87
static double c[64]
AVFilter ff_af_acompressor
Audio FIFO Buffer.
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:731
FILE * out
Definition: movenc.c:54
formats
Definition: signature.h:48
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:454
mode
Use these values in ebur128_init (or&#39;ed).
Definition: ebur128.h:83
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:593
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:659