FFmpeg  4.2.1
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 
74 static const AVOption options[] = {
75  { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F },
76  { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A|F, "mode" },
77  { "downward",0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F, "mode" },
78  { "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F, "mode" },
79  { "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0.000976563, 1, A|F },
80  { "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 20, A|F },
81  { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 2000, A|F },
82  { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A|F },
83  { "makeup", "set make up gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 64, A|F },
84  { "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.82843}, 1, 8, A|F },
85  { "link", "set link type", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A|F, "link" },
86  { "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F, "link" },
87  { "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F, "link" },
88  { "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A|F, "detection" },
89  { "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F, "detection" },
90  { "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F, "detection" },
91  { "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F },
92  { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A|F },
93  { NULL }
94 };
95 
96 #define sidechaincompress_options options
97 AVFILTER_DEFINE_CLASS(sidechaincompress);
98 
99 // A fake infinity value (because real infinity may break some hosts)
100 #define FAKE_INFINITY (65536.0 * 65536.0)
101 
102 // Check for infinity (with appropriate-ish tolerance)
103 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
104 
105 static double output_gain(double lin_slope, double ratio, double thres,
106  double knee, double knee_start, double knee_stop,
107  double compressed_knee_start,
108  double compressed_knee_stop,
109  int detection, int mode)
110 {
111  double slope = log(lin_slope);
112  double gain = 0.0;
113  double delta = 0.0;
114 
115  if (detection)
116  slope *= 0.5;
117 
118  if (IS_FAKE_INFINITY(ratio)) {
119  gain = thres;
120  delta = 0.0;
121  } else {
122  gain = (slope - thres) / ratio + thres;
123  delta = 1.0 / ratio;
124  }
125 
126  if (mode) {
127  if (knee > 1.0 && slope > knee_start)
128  gain = hermite_interpolation(slope, knee_stop, knee_start,
129  knee_stop, compressed_knee_start,
130  1.0, delta);
131  } else {
132  if (knee > 1.0 && slope < knee_stop)
133  gain = hermite_interpolation(slope, knee_start, knee_stop,
134  knee_start, compressed_knee_stop,
135  1.0, delta);
136  }
137 
138  return exp(gain - slope);
139 }
140 
142 {
143  AVFilterContext *ctx = outlink->src;
145 
146  s->thres = log(s->threshold);
147  s->lin_knee_start = s->threshold / sqrt(s->knee);
148  s->lin_knee_stop = s->threshold * sqrt(s->knee);
151  s->knee_start = log(s->lin_knee_start);
152  s->knee_stop = log(s->lin_knee_stop);
153  s->compressed_knee_start = (s->knee_start - s->thres) / s->ratio + s->thres;
154  s->compressed_knee_stop = (s->knee_stop - s->thres) / s->ratio + s->thres;
155 
156  s->attack_coeff = FFMIN(1., 1. / (s->attack * outlink->sample_rate / 4000.));
157  s->release_coeff = FFMIN(1., 1. / (s->release * outlink->sample_rate / 4000.));
158 
159  return 0;
160 }
161 
163  const double *src, double *dst, const double *scsrc, int nb_samples,
164  double level_in, double level_sc,
165  AVFilterLink *inlink, AVFilterLink *sclink)
166 {
167  const double makeup = s->makeup;
168  const double mix = s->mix;
169  int i, c;
170 
171  for (i = 0; i < nb_samples; i++) {
172  double abs_sample, gain = 1.0;
173  double detector;
174  int detected;
175 
176  abs_sample = fabs(scsrc[0] * level_sc);
177 
178  if (s->link == 1) {
179  for (c = 1; c < sclink->channels; c++)
180  abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
181  } else {
182  for (c = 1; c < sclink->channels; c++)
183  abs_sample += fabs(scsrc[c] * level_sc);
184 
185  abs_sample /= sclink->channels;
186  }
187 
188  if (s->detection)
189  abs_sample *= abs_sample;
190 
191  s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? s->attack_coeff : s->release_coeff);
192 
193  if (s->mode) {
194  detector = (s->detection ? s->adj_knee_stop : s->lin_knee_stop);
195  detected = s->lin_slope < detector;
196  } else {
197  detector = (s->detection ? s->adj_knee_start : s->lin_knee_start);
198  detected = s->lin_slope > detector;
199  }
200 
201  if (s->lin_slope > 0.0 && detected)
202  gain = output_gain(s->lin_slope, s->ratio, s->thres, s->knee,
203  s->knee_start, s->knee_stop,
206  s->detection, s->mode);
207 
208  for (c = 0; c < inlink->channels; c++)
209  dst[c] = src[c] * level_in * (gain * makeup * mix + (1. - mix));
210 
211  src += inlink->channels;
212  dst += inlink->channels;
213  scsrc += sclink->channels;
214  }
215 }
216 
217 #if CONFIG_SIDECHAINCOMPRESS_FILTER
218 static int activate(AVFilterContext *ctx)
219 {
221  AVFrame *out = NULL, *in[2] = { NULL };
222  int ret, i, nb_samples;
223  double *dst;
224 
226  if ((ret = ff_inlink_consume_frame(ctx->inputs[0], &in[0])) > 0) {
227  av_audio_fifo_write(s->fifo[0], (void **)in[0]->extended_data,
228  in[0]->nb_samples);
229  av_frame_free(&in[0]);
230  }
231  if (ret < 0)
232  return ret;
233  if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &in[1])) > 0) {
234  av_audio_fifo_write(s->fifo[1], (void **)in[1]->extended_data,
235  in[1]->nb_samples);
236  av_frame_free(&in[1]);
237  }
238  if (ret < 0)
239  return ret;
240 
241  nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
242  if (nb_samples) {
243  out = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
244  if (!out)
245  return AVERROR(ENOMEM);
246  for (i = 0; i < 2; i++) {
247  in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
248  if (!in[i]) {
249  av_frame_free(&in[0]);
250  av_frame_free(&in[1]);
251  av_frame_free(&out);
252  return AVERROR(ENOMEM);
253  }
254  av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
255  }
256 
257  dst = (double *)out->data[0];
258  out->pts = s->pts;
259  s->pts += nb_samples;
260 
261  compressor(s, (double *)in[0]->data[0], dst,
262  (double *)in[1]->data[0], nb_samples,
263  s->level_in, s->level_sc,
264  ctx->inputs[0], ctx->inputs[1]);
265 
266  av_frame_free(&in[0]);
267  av_frame_free(&in[1]);
268 
269  ret = ff_filter_frame(ctx->outputs[0], out);
270  if (ret < 0)
271  return ret;
272  }
273  FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
274  FF_FILTER_FORWARD_STATUS(ctx->inputs[1], ctx->outputs[0]);
275  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
276  if (!av_audio_fifo_size(s->fifo[0]))
278  if (!av_audio_fifo_size(s->fifo[1]))
280  }
281  return 0;
282 }
283 
284 static int query_formats(AVFilterContext *ctx)
285 {
288  static const enum AVSampleFormat sample_fmts[] = {
291  };
292  int ret, i;
293 
294  if (!ctx->inputs[0]->in_channel_layouts ||
296  av_log(ctx, AV_LOG_WARNING,
297  "No channel layout for input 1\n");
298  return AVERROR(EAGAIN);
299  }
300 
301  if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||
302  (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
303  return ret;
304 
305  for (i = 0; i < 2; i++) {
306  layouts = ff_all_channel_counts();
307  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
308  return ret;
309  }
310 
311  formats = ff_make_format_list(sample_fmts);
312  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
313  return ret;
314 
315  formats = ff_all_samplerates();
316  return ff_set_common_samplerates(ctx, formats);
317 }
318 
319 static int config_output(AVFilterLink *outlink)
320 {
321  AVFilterContext *ctx = outlink->src;
323 
324  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
325  av_log(ctx, AV_LOG_ERROR,
326  "Inputs must have the same sample rate "
327  "%d for in0 vs %d for in1\n",
328  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
329  return AVERROR(EINVAL);
330  }
331 
332  outlink->sample_rate = ctx->inputs[0]->sample_rate;
333  outlink->time_base = ctx->inputs[0]->time_base;
334  outlink->channel_layout = ctx->inputs[0]->channel_layout;
335  outlink->channels = ctx->inputs[0]->channels;
336 
337  s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
338  s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
339  if (!s->fifo[0] || !s->fifo[1])
340  return AVERROR(ENOMEM);
341 
342  compressor_config_output(outlink);
343 
344  return 0;
345 }
346 
347 static av_cold void uninit(AVFilterContext *ctx)
348 {
350 
351  av_audio_fifo_free(s->fifo[0]);
352  av_audio_fifo_free(s->fifo[1]);
353 }
354 
355 static const AVFilterPad sidechaincompress_inputs[] = {
356  {
357  .name = "main",
358  .type = AVMEDIA_TYPE_AUDIO,
359  },{
360  .name = "sidechain",
361  .type = AVMEDIA_TYPE_AUDIO,
362  },
363  { NULL }
364 };
365 
366 static const AVFilterPad sidechaincompress_outputs[] = {
367  {
368  .name = "default",
369  .type = AVMEDIA_TYPE_AUDIO,
370  .config_props = config_output,
371  },
372  { NULL }
373 };
374 
376  .name = "sidechaincompress",
377  .description = NULL_IF_CONFIG_SMALL("Sidechain compressor."),
378  .priv_size = sizeof(SidechainCompressContext),
379  .priv_class = &sidechaincompress_class,
381  .activate = activate,
382  .uninit = uninit,
383  .inputs = sidechaincompress_inputs,
384  .outputs = sidechaincompress_outputs,
385 };
386 #endif /* CONFIG_SIDECHAINCOMPRESS_FILTER */
387 
388 #if CONFIG_ACOMPRESSOR_FILTER
389 static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *in)
390 {
391  const double *src = (const double *)in->data[0];
392  AVFilterContext *ctx = inlink->dst;
394  AVFilterLink *outlink = ctx->outputs[0];
395  AVFrame *out;
396  double *dst;
397 
398  if (av_frame_is_writable(in)) {
399  out = in;
400  } else {
401  out = ff_get_audio_buffer(outlink, in->nb_samples);
402  if (!out) {
403  av_frame_free(&in);
404  return AVERROR(ENOMEM);
405  }
407  }
408  dst = (double *)out->data[0];
409 
410  compressor(s, src, dst, src, in->nb_samples,
411  s->level_in, s->level_in,
412  inlink, inlink);
413 
414  if (out != in)
415  av_frame_free(&in);
416  return ff_filter_frame(outlink, out);
417 }
418 
419 static int acompressor_query_formats(AVFilterContext *ctx)
420 {
423  static const enum AVSampleFormat sample_fmts[] = {
426  };
427  int ret;
428 
429  layouts = ff_all_channel_counts();
430  if (!layouts)
431  return AVERROR(ENOMEM);
432  ret = ff_set_common_channel_layouts(ctx, layouts);
433  if (ret < 0)
434  return ret;
435 
436  formats = ff_make_format_list(sample_fmts);
437  if (!formats)
438  return AVERROR(ENOMEM);
439  ret = ff_set_common_formats(ctx, formats);
440  if (ret < 0)
441  return ret;
442 
443  formats = ff_all_samplerates();
444  if (!formats)
445  return AVERROR(ENOMEM);
446  return ff_set_common_samplerates(ctx, formats);
447 }
448 
449 #define acompressor_options options
450 AVFILTER_DEFINE_CLASS(acompressor);
451 
452 static const AVFilterPad acompressor_inputs[] = {
453  {
454  .name = "default",
455  .type = AVMEDIA_TYPE_AUDIO,
456  .filter_frame = acompressor_filter_frame,
457  },
458  { NULL }
459 };
460 
461 static const AVFilterPad acompressor_outputs[] = {
462  {
463  .name = "default",
464  .type = AVMEDIA_TYPE_AUDIO,
465  .config_props = compressor_config_output,
466  },
467  { NULL }
468 };
469 
471  .name = "acompressor",
472  .description = NULL_IF_CONFIG_SMALL("Audio compressor."),
473  .priv_size = sizeof(SidechainCompressContext),
474  .priv_class = &acompressor_class,
475  .query_formats = acompressor_query_formats,
476  .inputs = acompressor_inputs,
477  .outputs = acompressor_outputs,
478 };
479 #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:1481
#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:549
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:295
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
#define src
Definition: vp8dsp.c:254
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:1607
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:283
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:435
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define av_cold
Definition: attributes.h:82
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:388
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
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#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:568
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:343
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:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
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:232
#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:609
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
if(ret< 0)
Definition: vf_mcdeint.c:279
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:594
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)
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
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:395
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
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
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:701
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:410
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:361
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654