FFmpeg
 All Data Structures Namespaces Files Functions Variables Typedefs Enumerations Enumerator Macros Groups Pages
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 "formats.h"
36 #include "hermite.h"
37 #include "internal.h"
38 
39 typedef struct SidechainCompressContext {
40  const AVClass *class;
41 
42  double level_in;
43  double level_sc;
46  double lin_slope;
47  double ratio;
48  double threshold;
49  double makeup;
50  double mix;
51  double thres;
52  double knee;
53  double knee_start;
54  double knee_stop;
58  int link;
59  int detection;
60 
62  int64_t pts;
64 
65 #define OFFSET(x) offsetof(SidechainCompressContext, x)
66 #define A AV_OPT_FLAG_AUDIO_PARAM
67 #define F AV_OPT_FLAG_FILTERING_PARAM
68 
69 static const AVOption options[] = {
70  { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F },
71  { "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0.000976563, 1, A|F },
72  { "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 20, A|F },
73  { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 2000, A|F },
74  { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A|F },
75  { "makeup", "set make up gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 64, A|F },
76  { "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.82843}, 1, 8, A|F },
77  { "link", "set link type", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A|F, "link" },
78  { "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F, "link" },
79  { "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F, "link" },
80  { "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A|F, "detection" },
81  { "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A|F, "detection" },
82  { "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A|F, "detection" },
83  { "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A|F },
84  { "mix", "set mix", OFFSET(mix), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A|F },
85  { NULL }
86 };
87 
88 #define sidechaincompress_options options
89 AVFILTER_DEFINE_CLASS(sidechaincompress);
90 
91 // A fake infinity value (because real infinity may break some hosts)
92 #define FAKE_INFINITY (65536.0 * 65536.0)
93 
94 // Check for infinity (with appropriate-ish tolerance)
95 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
96 
97 static double output_gain(double lin_slope, double ratio, double thres,
98  double knee, double knee_start, double knee_stop,
99  double compressed_knee_stop, int detection)
100 {
101  double slope = log(lin_slope);
102  double gain = 0.0;
103  double delta = 0.0;
104 
105  if (detection)
106  slope *= 0.5;
107 
108  if (IS_FAKE_INFINITY(ratio)) {
109  gain = thres;
110  delta = 0.0;
111  } else {
112  gain = (slope - thres) / ratio + thres;
113  delta = 1.0 / ratio;
114  }
115 
116  if (knee > 1.0 && slope < knee_stop)
117  gain = hermite_interpolation(slope, knee_start, knee_stop,
118  knee_start, compressed_knee_stop,
119  1.0, delta);
120 
121  return exp(gain - slope);
122 }
123 
125 {
126  AVFilterContext *ctx = outlink->src;
128 
129  s->thres = log(s->threshold);
130  s->lin_knee_start = s->threshold / sqrt(s->knee);
132  s->knee_start = log(s->lin_knee_start);
133  s->knee_stop = log(s->threshold * sqrt(s->knee));
134  s->compressed_knee_stop = (s->knee_stop - s->thres) / s->ratio + s->thres;
135 
136  s->attack_coeff = FFMIN(1., 1. / (s->attack * outlink->sample_rate / 4000.));
137  s->release_coeff = FFMIN(1., 1. / (s->release * outlink->sample_rate / 4000.));
138 
139  return 0;
140 }
141 
143  const double *src, double *dst, const double *scsrc, int nb_samples,
144  double level_in, double level_sc,
145  AVFilterLink *inlink, AVFilterLink *sclink)
146 {
147  const double makeup = s->makeup;
148  const double mix = s->mix;
149  int i, c;
150 
151  for (i = 0; i < nb_samples; i++) {
152  double abs_sample, gain = 1.0;
153 
154  abs_sample = fabs(scsrc[0] * level_sc);
155 
156  if (s->link == 1) {
157  for (c = 1; c < sclink->channels; c++)
158  abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
159  } else {
160  for (c = 1; c < sclink->channels; c++)
161  abs_sample += fabs(scsrc[c] * level_sc);
162 
163  abs_sample /= sclink->channels;
164  }
165 
166  if (s->detection)
167  abs_sample *= abs_sample;
168 
169  s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? s->attack_coeff : s->release_coeff);
170 
171  if (s->lin_slope > 0.0 && s->lin_slope > (s->detection ? s->adj_knee_start : s->lin_knee_start))
172  gain = output_gain(s->lin_slope, s->ratio, s->thres, s->knee,
173  s->knee_start, s->knee_stop,
175 
176  for (c = 0; c < inlink->channels; c++)
177  dst[c] = src[c] * level_in * (gain * makeup * mix + (1. - mix));
178 
179  src += inlink->channels;
180  dst += inlink->channels;
181  scsrc += sclink->channels;
182  }
183 }
184 
185 #if CONFIG_SIDECHAINCOMPRESS_FILTER
186 static int filter_frame(AVFilterLink *link, AVFrame *frame)
187 {
188  AVFilterContext *ctx = link->dst;
190  AVFilterLink *outlink = ctx->outputs[0];
191  AVFrame *out = NULL, *in[2] = { NULL };
192  double *dst;
193  int nb_samples;
194  int i;
195 
196  for (i = 0; i < 2; i++)
197  if (link == ctx->inputs[i])
198  break;
199  av_assert0(i < 2);
200  av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
201  frame->nb_samples);
202  av_frame_free(&frame);
203 
204  nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
205  if (!nb_samples)
206  return 0;
207 
208  out = ff_get_audio_buffer(outlink, nb_samples);
209  if (!out)
210  return AVERROR(ENOMEM);
211  for (i = 0; i < 2; i++) {
212  in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
213  if (!in[i]) {
214  av_frame_free(&in[0]);
215  av_frame_free(&in[1]);
216  av_frame_free(&out);
217  return AVERROR(ENOMEM);
218  }
219  av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
220  }
221 
222  dst = (double *)out->data[0];
223  out->pts = s->pts;
224  s->pts += nb_samples;
225 
226  compressor(s, (double *)in[0]->data[0], dst,
227  (double *)in[1]->data[0], nb_samples,
228  s->level_in, s->level_sc,
229  ctx->inputs[0], ctx->inputs[1]);
230 
231  av_frame_free(&in[0]);
232  av_frame_free(&in[1]);
233 
234  return ff_filter_frame(outlink, out);
235 }
236 
237 static int request_frame(AVFilterLink *outlink)
238 {
239  AVFilterContext *ctx = outlink->src;
240  SidechainCompressContext *s = ctx->priv;
241  int i;
242 
243  /* get a frame on each input */
244  for (i = 0; i < 2; i++) {
245  AVFilterLink *inlink = ctx->inputs[i];
246  if (!av_audio_fifo_size(s->fifo[i]))
247  return ff_request_frame(inlink);
248  }
249 
250  return 0;
251 }
252 
253 static int query_formats(AVFilterContext *ctx)
254 {
257  static const enum AVSampleFormat sample_fmts[] = {
260  };
261  int ret, i;
262 
263  if (!ctx->inputs[0]->in_channel_layouts ||
265  av_log(ctx, AV_LOG_WARNING,
266  "No channel layout for input 1\n");
267  return AVERROR(EAGAIN);
268  }
269 
270  if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||
271  (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
272  return ret;
273 
274  for (i = 0; i < 2; i++) {
275  layouts = ff_all_channel_counts();
276  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
277  return ret;
278  }
279 
280  formats = ff_make_format_list(sample_fmts);
281  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
282  return ret;
283 
284  formats = ff_all_samplerates();
285  return ff_set_common_samplerates(ctx, formats);
286 }
287 
288 static int config_output(AVFilterLink *outlink)
289 {
290  AVFilterContext *ctx = outlink->src;
291  SidechainCompressContext *s = ctx->priv;
292 
293  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
294  av_log(ctx, AV_LOG_ERROR,
295  "Inputs must have the same sample rate "
296  "%d for in0 vs %d for in1\n",
297  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
298  return AVERROR(EINVAL);
299  }
300 
301  outlink->sample_rate = ctx->inputs[0]->sample_rate;
302  outlink->time_base = ctx->inputs[0]->time_base;
303  outlink->channel_layout = ctx->inputs[0]->channel_layout;
304  outlink->channels = ctx->inputs[0]->channels;
305 
306  s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
307  s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
308  if (!s->fifo[0] || !s->fifo[1])
309  return AVERROR(ENOMEM);
310 
311  compressor_config_output(outlink);
312 
313  return 0;
314 }
315 
316 static av_cold void uninit(AVFilterContext *ctx)
317 {
318  SidechainCompressContext *s = ctx->priv;
319 
320  av_audio_fifo_free(s->fifo[0]);
321  av_audio_fifo_free(s->fifo[1]);
322 }
323 
324 static const AVFilterPad sidechaincompress_inputs[] = {
325  {
326  .name = "main",
327  .type = AVMEDIA_TYPE_AUDIO,
328  .filter_frame = filter_frame,
329  },{
330  .name = "sidechain",
331  .type = AVMEDIA_TYPE_AUDIO,
332  .filter_frame = filter_frame,
333  },
334  { NULL }
335 };
336 
337 static const AVFilterPad sidechaincompress_outputs[] = {
338  {
339  .name = "default",
340  .type = AVMEDIA_TYPE_AUDIO,
341  .config_props = config_output,
342  .request_frame = request_frame,
343  },
344  { NULL }
345 };
346 
347 AVFilter ff_af_sidechaincompress = {
348  .name = "sidechaincompress",
349  .description = NULL_IF_CONFIG_SMALL("Sidechain compressor."),
350  .priv_size = sizeof(SidechainCompressContext),
351  .priv_class = &sidechaincompress_class,
353  .uninit = uninit,
354  .inputs = sidechaincompress_inputs,
355  .outputs = sidechaincompress_outputs,
356 };
357 #endif /* CONFIG_SIDECHAINCOMPRESS_FILTER */
358 
359 #if CONFIG_ACOMPRESSOR_FILTER
360 static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *in)
361 {
362  const double *src = (const double *)in->data[0];
363  AVFilterContext *ctx = inlink->dst;
364  SidechainCompressContext *s = ctx->priv;
365  AVFilterLink *outlink = ctx->outputs[0];
366  AVFrame *out;
367  double *dst;
368 
369  if (av_frame_is_writable(in)) {
370  out = in;
371  } else {
372  out = ff_get_audio_buffer(inlink, in->nb_samples);
373  if (!out) {
374  av_frame_free(&in);
375  return AVERROR(ENOMEM);
376  }
377  av_frame_copy_props(out, in);
378  }
379  dst = (double *)out->data[0];
380 
381  compressor(s, src, dst, src, in->nb_samples,
382  s->level_in, s->level_in,
383  inlink, inlink);
384 
385  if (out != in)
386  av_frame_free(&in);
387  return ff_filter_frame(outlink, out);
388 }
389 
390 static int acompressor_query_formats(AVFilterContext *ctx)
391 {
394  static const enum AVSampleFormat sample_fmts[] = {
397  };
398  int ret;
399 
400  layouts = ff_all_channel_counts();
401  if (!layouts)
402  return AVERROR(ENOMEM);
403  ret = ff_set_common_channel_layouts(ctx, layouts);
404  if (ret < 0)
405  return ret;
406 
407  formats = ff_make_format_list(sample_fmts);
408  if (!formats)
409  return AVERROR(ENOMEM);
410  ret = ff_set_common_formats(ctx, formats);
411  if (ret < 0)
412  return ret;
413 
414  formats = ff_all_samplerates();
415  if (!formats)
416  return AVERROR(ENOMEM);
417  return ff_set_common_samplerates(ctx, formats);
418 }
419 
420 #define acompressor_options options
421 AVFILTER_DEFINE_CLASS(acompressor);
422 
423 static const AVFilterPad acompressor_inputs[] = {
424  {
425  .name = "default",
426  .type = AVMEDIA_TYPE_AUDIO,
427  .filter_frame = acompressor_filter_frame,
428  },
429  { NULL }
430 };
431 
432 static const AVFilterPad acompressor_outputs[] = {
433  {
434  .name = "default",
435  .type = AVMEDIA_TYPE_AUDIO,
436  .config_props = compressor_config_output,
437  },
438  { NULL }
439 };
440 
441 AVFilter ff_af_acompressor = {
442  .name = "acompressor",
443  .description = NULL_IF_CONFIG_SMALL("Audio compressor."),
444  .priv_size = sizeof(SidechainCompressContext),
445  .priv_class = &acompressor_class,
446  .query_formats = acompressor_query_formats,
447  .inputs = acompressor_inputs,
448  .outputs = acompressor_outputs,
449 };
450 #endif /* CONFIG_ACOMPRESSOR_FILTER */
#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
const char * s
Definition: avisynth_c.h:768
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:184
AVOption.
Definition: opt.h:245
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
static double output_gain(double lin_slope, double ratio, double thres, double knee, double knee_start, double knee_stop, double compressed_knee_stop, int detection)
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
Main libavfilter public API header.
#define F
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
static enum AVSampleFormat formats[]
Definition: avresample.c:163
static int compressor_config_output(AVFilterLink *outlink)
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:59
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:315
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
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:1189
#define av_cold
Definition: attributes.h:82
static av_cold int uninit(AVCodecContext *avctx)
Definition: crystalhd.c:336
float delta
AVOptions.
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:268
static AVFrame * frame
#define A
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:53
#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:64
#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:158
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
void * priv
private data for use by the filter
Definition: avfilter.h:322
uint64_t * channel_layouts
list of channel layouts
Definition: formats.h:86
static int config_output(AVFilterLink *outlink)
Definition: af_aecho.c:231
#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:64
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 int request_frame(AVFilterLink *outlink)
Definition: aeval.c:274
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
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: aeval.c:413
AVFormatContext * ctx
Definition: movenc.c:48
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
#define src
Definition: vp9dsp.c:530
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)
#define IS_FAKE_INFINITY(value)
static int mix(int c0, int c1)
Definition: 4xm.c:707
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
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:529
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_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:319
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
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:198
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:282
int nb_channel_layouts
number of channel layouts
Definition: formats.h:87
static double c[64]
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:307
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:369
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
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:231
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
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:589