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af_agate.c
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1 /*
2  * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 /**
22  * @file
23  * Audio (Sidechain) Gate filter
24  */
25 
26 #include "libavutil/audio_fifo.h"
27 #include "libavutil/avassert.h"
29 #include "libavutil/opt.h"
30 #include "avfilter.h"
31 #include "audio.h"
32 #include "formats.h"
33 #include "hermite.h"
34 
35 typedef struct AudioGateContext {
36  const AVClass *class;
37 
38  double level_in;
39  double level_sc;
40  double attack;
41  double release;
42  double threshold;
43  double ratio;
44  double knee;
45  double makeup;
46  double range;
47  int link;
48  int detection;
49 
50  double thres;
51  double knee_start;
52  double lin_knee_stop;
53  double knee_stop;
54  double lin_slope;
55  double attack_coeff;
56  double release_coeff;
57 
59  int64_t pts;
61 
62 #define OFFSET(x) offsetof(AudioGateContext, x)
63 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
64 
65 static const AVOption options[] = {
66  { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
67  { "range", "set max gain reduction", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0.06125}, 0, 1, A },
68  { "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0, 1, A },
69  { "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 9000, A },
70  { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 9000, A },
71  { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A },
72  { "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 64, A },
73  { "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1, 8, A },
74  { "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A, "detection" },
75  { "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "detection" },
76  { "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "detection" },
77  { "link", "set link", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "link" },
78  { "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "link" },
79  { "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "link" },
80  { "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
81  { NULL }
82 };
83 
84 static int agate_config_input(AVFilterLink *inlink)
85 {
86  AVFilterContext *ctx = inlink->dst;
87  AudioGateContext *s = ctx->priv;
88  double lin_threshold = s->threshold;
89  double lin_knee_sqrt = sqrt(s->knee);
90  double lin_knee_start;
91 
92  if (s->detection)
93  lin_threshold *= lin_threshold;
94 
95  s->attack_coeff = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.));
96  s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.));
97  s->lin_knee_stop = lin_threshold * lin_knee_sqrt;
98  lin_knee_start = lin_threshold / lin_knee_sqrt;
99  s->thres = log(lin_threshold);
100  s->knee_start = log(lin_knee_start);
101  s->knee_stop = log(s->lin_knee_stop);
102 
103  return 0;
104 }
105 
106 // A fake infinity value (because real infinity may break some hosts)
107 #define FAKE_INFINITY (65536.0 * 65536.0)
108 
109 // Check for infinity (with appropriate-ish tolerance)
110 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
111 
112 static double output_gain(double lin_slope, double ratio, double thres,
113  double knee, double knee_start, double knee_stop,
114  double lin_knee_stop, double range)
115 {
116  if (lin_slope < lin_knee_stop) {
117  double slope = log(lin_slope);
118  double tratio = ratio;
119  double gain = 0.;
120  double delta = 0.;
121 
122  if (IS_FAKE_INFINITY(ratio))
123  tratio = 1000.;
124  gain = (slope - thres) * tratio + thres;
125  delta = tratio;
126 
127  if (knee > 1. && slope > knee_start) {
128  gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio + thres), knee_stop, delta, 1.);
129  }
130  return FFMAX(range, exp(gain - slope));
131  }
132 
133  return 1.;
134 }
135 
136 static void gate(AudioGateContext *s,
137  const double *src, double *dst, const double *scsrc,
138  int nb_samples, double level_in, double level_sc,
139  AVFilterLink *inlink, AVFilterLink *sclink)
140 {
141  const double makeup = s->makeup;
142  const double attack_coeff = s->attack_coeff;
143  const double release_coeff = s->release_coeff;
144  int n, c;
145 
146  for (n = 0; n < nb_samples; n++, src += inlink->channels, dst += inlink->channels, scsrc += sclink->channels) {
147  double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;
148 
149  if (s->link == 1) {
150  for (c = 1; c < sclink->channels; c++)
151  abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
152  } else {
153  for (c = 1; c < sclink->channels; c++)
154  abs_sample += fabs(scsrc[c] * level_sc);
155 
156  abs_sample /= sclink->channels;
157  }
158 
159  if (s->detection)
160  abs_sample *= abs_sample;
161 
162  s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? attack_coeff : release_coeff);
163  if (s->lin_slope > 0.0)
164  gain = output_gain(s->lin_slope, s->ratio, s->thres,
165  s->knee, s->knee_start, s->knee_stop,
166  s->lin_knee_stop, s->range);
167 
168  for (c = 0; c < inlink->channels; c++)
169  dst[c] = src[c] * level_in * gain * makeup;
170  }
171 }
172 
173 #if CONFIG_AGATE_FILTER
174 
175 #define agate_options options
176 AVFILTER_DEFINE_CLASS(agate);
177 
178 static int query_formats(AVFilterContext *ctx)
179 {
182  int ret;
183 
184  if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL)) < 0)
185  return ret;
186  ret = ff_set_common_formats(ctx, formats);
187  if (ret < 0)
188  return ret;
189 
190  layouts = ff_all_channel_counts();
191  if (!layouts)
192  return AVERROR(ENOMEM);
193  ret = ff_set_common_channel_layouts(ctx, layouts);
194  if (ret < 0)
195  return ret;
196 
197  formats = ff_all_samplerates();
198  if (!formats)
199  return AVERROR(ENOMEM);
200 
201  return ff_set_common_samplerates(ctx, formats);
202 }
203 
204 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
205 {
206  const double *src = (const double *)in->data[0];
207  AVFilterContext *ctx = inlink->dst;
208  AVFilterLink *outlink = ctx->outputs[0];
209  AudioGateContext *s = ctx->priv;
210  AVFrame *out;
211  double *dst;
212 
213  if (av_frame_is_writable(in)) {
214  out = in;
215  } else {
216  out = ff_get_audio_buffer(inlink, in->nb_samples);
217  if (!out) {
218  av_frame_free(&in);
219  return AVERROR(ENOMEM);
220  }
222  }
223  dst = (double *)out->data[0];
224 
225  gate(s, src, dst, src, in->nb_samples,
226  s->level_in, s->level_in, inlink, inlink);
227 
228  if (out != in)
229  av_frame_free(&in);
230  return ff_filter_frame(outlink, out);
231 }
232 
233 static const AVFilterPad inputs[] = {
234  {
235  .name = "default",
236  .type = AVMEDIA_TYPE_AUDIO,
237  .filter_frame = filter_frame,
238  .config_props = agate_config_input,
239  },
240  { NULL }
241 };
242 
243 static const AVFilterPad outputs[] = {
244  {
245  .name = "default",
246  .type = AVMEDIA_TYPE_AUDIO,
247  },
248  { NULL }
249 };
250 
251 AVFilter ff_af_agate = {
252  .name = "agate",
253  .description = NULL_IF_CONFIG_SMALL("Audio gate."),
254  .query_formats = query_formats,
255  .priv_size = sizeof(AudioGateContext),
256  .priv_class = &agate_class,
257  .inputs = inputs,
258  .outputs = outputs,
259 };
260 
261 #endif /* CONFIG_AGATE_FILTER */
262 
263 #if CONFIG_SIDECHAINGATE_FILTER
264 
265 #define sidechaingate_options options
266 AVFILTER_DEFINE_CLASS(sidechaingate);
267 
268 static int scfilter_frame(AVFilterLink *link, AVFrame *frame)
269 {
270  AVFilterContext *ctx = link->dst;
271  AudioGateContext *s = ctx->priv;
272  AVFilterLink *outlink = ctx->outputs[0];
273  AVFrame *out, *in[2] = { NULL };
274  double *dst;
275  int nb_samples;
276  int i;
277 
278  for (i = 0; i < 2; i++)
279  if (link == ctx->inputs[i])
280  break;
281  av_assert0(i < 2);
282  av_audio_fifo_write(s->fifo[i], (void **)frame->extended_data,
283  frame->nb_samples);
284  av_frame_free(&frame);
285 
286  nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
287  if (!nb_samples)
288  return 0;
289 
290  out = ff_get_audio_buffer(outlink, nb_samples);
291  if (!out)
292  return AVERROR(ENOMEM);
293  for (i = 0; i < 2; i++) {
294  in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
295  if (!in[i]) {
296  av_frame_free(&in[0]);
297  av_frame_free(&in[1]);
298  av_frame_free(&out);
299  return AVERROR(ENOMEM);
300  }
301  av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
302  }
303 
304  dst = (double *)out->data[0];
305  out->pts = s->pts;
306  s->pts += nb_samples;
307 
308  gate(s, (double *)in[0]->data[0], dst,
309  (double *)in[1]->data[0], nb_samples,
310  s->level_in, s->level_sc,
311  ctx->inputs[0], ctx->inputs[1]);
312 
313  av_frame_free(&in[0]);
314  av_frame_free(&in[1]);
315 
316  return ff_filter_frame(outlink, out);
317 }
318 
319 static int screquest_frame(AVFilterLink *outlink)
320 {
321  AVFilterContext *ctx = outlink->src;
322  AudioGateContext *s = ctx->priv;
323  int i;
324 
325  /* get a frame on each input */
326  for (i = 0; i < 2; i++) {
327  AVFilterLink *inlink = ctx->inputs[i];
328  if (!av_audio_fifo_size(s->fifo[i]))
329  return ff_request_frame(inlink);
330  }
331 
332  return 0;
333 }
334 
335 static int scquery_formats(AVFilterContext *ctx)
336 {
338  AVFilterChannelLayouts *layouts = NULL;
339  static const enum AVSampleFormat sample_fmts[] = {
342  };
343  int ret, i;
344 
345  if (!ctx->inputs[0]->in_channel_layouts ||
347  av_log(ctx, AV_LOG_WARNING,
348  "No channel layout for input 1\n");
349  return AVERROR(EAGAIN);
350  }
351 
352  if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||
353  (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
354  return ret;
355 
356  for (i = 0; i < 2; i++) {
357  layouts = ff_all_channel_counts();
358  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
359  return ret;
360  }
361 
362  formats = ff_make_format_list(sample_fmts);
363  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
364  return ret;
365 
366  formats = ff_all_samplerates();
367  return ff_set_common_samplerates(ctx, formats);
368 }
369 
370 static int scconfig_output(AVFilterLink *outlink)
371 {
372  AVFilterContext *ctx = outlink->src;
373  AudioGateContext *s = ctx->priv;
374 
375  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
376  av_log(ctx, AV_LOG_ERROR,
377  "Inputs must have the same sample rate "
378  "%d for in0 vs %d for in1\n",
379  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
380  return AVERROR(EINVAL);
381  }
382 
383  outlink->sample_rate = ctx->inputs[0]->sample_rate;
384  outlink->time_base = ctx->inputs[0]->time_base;
385  outlink->channel_layout = ctx->inputs[0]->channel_layout;
386  outlink->channels = ctx->inputs[0]->channels;
387 
388  s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
389  s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
390  if (!s->fifo[0] || !s->fifo[1])
391  return AVERROR(ENOMEM);
392 
393 
394  agate_config_input(ctx->inputs[0]);
395 
396  return 0;
397 }
398 
399 static av_cold void uninit(AVFilterContext *ctx)
400 {
401  AudioGateContext *s = ctx->priv;
402 
403  av_audio_fifo_free(s->fifo[0]);
404  av_audio_fifo_free(s->fifo[1]);
405 }
406 
407 static const AVFilterPad sidechaingate_inputs[] = {
408  {
409  .name = "main",
410  .type = AVMEDIA_TYPE_AUDIO,
411  .filter_frame = scfilter_frame,
412  },{
413  .name = "sidechain",
414  .type = AVMEDIA_TYPE_AUDIO,
415  .filter_frame = scfilter_frame,
416  },
417  { NULL }
418 };
419 
420 static const AVFilterPad sidechaingate_outputs[] = {
421  {
422  .name = "default",
423  .type = AVMEDIA_TYPE_AUDIO,
424  .config_props = scconfig_output,
425  .request_frame = screquest_frame,
426  },
427  { NULL }
428 };
429 
430 AVFilter ff_af_sidechaingate = {
431  .name = "sidechaingate",
432  .description = NULL_IF_CONFIG_SMALL("Audio sidechain gate."),
433  .priv_size = sizeof(AudioGateContext),
434  .priv_class = &sidechaingate_class,
435  .query_formats = scquery_formats,
436  .uninit = uninit,
437  .inputs = sidechaingate_inputs,
438  .outputs = sidechaingate_outputs,
439 };
440 #endif /* CONFIG_SIDECHAINGATE_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:631
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:60
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:182
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
double lin_slope
Definition: af_agate.c:54
AVOption.
Definition: opt.h:245
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
Main libavfilter public API header.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
static int agate_config_input(AVFilterLink *inlink)
Definition: af_agate.c:84
static enum AVSampleFormat formats[]
Definition: avresample.c:163
static void gate(AudioGateContext *s, const double *src, double *dst, const double *scsrc, int nb_samples, double level_in, double level_sc, AVFilterLink *inlink, AVFilterLink *sclink)
Definition: af_agate.c:136
double level_in
Definition: af_agate.c:38
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
double release
Definition: af_agate.c:41
double ratio
Definition: af_agate.c:43
const char * name
Pad name.
Definition: internal.h:59
#define IS_FAKE_INFINITY(value)
Definition: af_agate.c:110
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:313
#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:1180
#define av_cold
Definition: attributes.h:82
static av_cold int uninit(AVCodecContext *avctx)
Definition: crystalhd.c:337
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
double knee_start
Definition: af_agate.c:51
double makeup
Definition: af_agate.c:45
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:53
#define OFFSET(x)
Definition: af_agate.c:62
#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:65
#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:153
#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:320
uint64_t * channel_layouts
list of channel layouts
Definition: formats.h:86
simple assert() macros that are a bit more flexible than ISO C assert().
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:337
#define FFMAX(a, b)
Definition: common.h:94
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:229
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
double level_sc
Definition: af_agate.c:39
#define FFMIN(a, b)
Definition: common.h:96
double lin_knee_stop
Definition: af_agate.c:52
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: aeval.c:413
AVFormatContext * ctx
Definition: movenc.c:48
int n
Definition: avisynth_c.h:547
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
#define src
Definition: vp9dsp.c:530
double attack_coeff
Definition: af_agate.c:55
A list of supported channel layouts.
Definition: formats.h:85
int64_t pts
Definition: af_agate.c:59
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
static const AVOption options[]
Definition: af_agate.c:65
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:520
double release_coeff
Definition: af_agate.c:56
double thres
Definition: af_agate.c:50
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
double threshold
Definition: af_agate.c:42
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:142
const char * name
Filter name.
Definition: avfilter.h:146
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:317
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:113
double knee_stop
Definition: af_agate.c:53
AVAudioFifo * fifo[2]
Definition: af_agate.c:58
static int query_formats(AVFilterContext *ctx)
Definition: aeval.c:244
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.
#define AVFILTER_DEFINE_CLASS(fname)
Definition: internal.h:339
double range
Definition: af_agate.c:46
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:305
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
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
static double output_gain(double lin_slope, double ratio, double thres, double knee, double knee_start, double knee_stop, double lin_knee_stop, double range)
Definition: af_agate.c:112
#define A
Definition: af_agate.c:63
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:580
double attack
Definition: af_agate.c:40