FFmpeg
af_sidechaincompress.c
Go to the documentation of this file.
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:1492
#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:581
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:308
AVOption.
Definition: opt.h:248
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#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
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:1618
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:465
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1091
#define av_cold
Definition: attributes.h:88
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:401
#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
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 AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
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:600
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:203
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:153
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:885
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
#define s(width, name)
Definition: cbs_vp9.c:257
#define R
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: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)
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
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)\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
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
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:425
#define flags(name, subs,...)
Definition: cbs_av1.c:560
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:322
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:723
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:440
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:374
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:588
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:659
int i
Definition: input.c:406