FFmpeg
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:1494
#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:550
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
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
#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
FF_FILTER_FORWARD_STATUS(inlink, outlink)
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:1620
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:434
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1093
#define av_cold
Definition: attributes.h:82
static av_cold int uninit(AVCodecContext *avctx)
Definition: crystalhd.c:279
float delta
AVOptions.
filter_frame For filters that do not use the activate() callback
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
#define A
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
#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:569
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:342
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
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
#define s(width, name)
Definition: cbs_vp9.c:257
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)
if(ret)
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)
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
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
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
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
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
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
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
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
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
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:409
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:557
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654