FFmpeg  4.3
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 "filters.h"
33 #include "formats.h"
34 #include "hermite.h"
35 
36 typedef struct AudioGateContext {
37  const AVClass *class;
38 
39  double level_in;
40  double level_sc;
41  double attack;
42  double release;
43  double threshold;
44  double ratio;
45  double knee;
46  double makeup;
47  double range;
48  int link;
49  int detection;
50  int mode;
51 
52  double thres;
53  double knee_start;
54  double knee_stop;
56  double lin_knee_stop;
57  double lin_slope;
58  double attack_coeff;
59  double release_coeff;
60 
62  int64_t pts;
64 
65 #define OFFSET(x) offsetof(AudioGateContext, x)
66 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
67 
68 static const AVOption options[] = {
69  { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
70  { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "mode" },
71  { "downward",0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
72  { "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
73  { "range", "set max gain reduction", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0.06125}, 0, 1, A },
74  { "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0, 1, A },
75  { "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 9000, A },
76  { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 9000, A },
77  { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A },
78  { "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 64, A },
79  { "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1, 8, A },
80  { "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, A, "detection" },
81  { "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "detection" },
82  { "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "detection" },
83  { "link", "set link", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "link" },
84  { "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "link" },
85  { "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "link" },
86  { "level_sc", "set sidechain gain", OFFSET(level_sc), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
87  { NULL }
88 };
89 
90 static int agate_config_input(AVFilterLink *inlink)
91 {
92  AVFilterContext *ctx = inlink->dst;
93  AudioGateContext *s = ctx->priv;
94  double lin_threshold = s->threshold;
95  double lin_knee_sqrt = sqrt(s->knee);
96 
97  if (s->detection)
98  lin_threshold *= lin_threshold;
99 
100  s->attack_coeff = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.));
101  s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.));
102  s->lin_knee_stop = lin_threshold * lin_knee_sqrt;
103  s->lin_knee_start = lin_threshold / lin_knee_sqrt;
104  s->thres = log(lin_threshold);
105  s->knee_start = log(s->lin_knee_start);
106  s->knee_stop = log(s->lin_knee_stop);
107 
108  return 0;
109 }
110 
111 // A fake infinity value (because real infinity may break some hosts)
112 #define FAKE_INFINITY (65536.0 * 65536.0)
113 
114 // Check for infinity (with appropriate-ish tolerance)
115 #define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0)
116 
117 static double output_gain(double lin_slope, double ratio, double thres,
118  double knee, double knee_start, double knee_stop,
119  double range, int mode)
120 {
121  double slope = log(lin_slope);
122  double tratio = ratio;
123  double gain = 0.;
124  double delta = 0.;
125 
126  if (IS_FAKE_INFINITY(ratio))
127  tratio = 1000.;
128  gain = (slope - thres) * tratio + thres;
129  delta = tratio;
130 
131  if (mode) {
132  if (knee > 1. && slope < knee_stop)
133  gain = hermite_interpolation(slope, knee_stop, knee_start, ((knee_stop - thres) * tratio + thres), knee_start, delta, 1.);
134  } else {
135  if (knee > 1. && slope > knee_start)
136  gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio + thres), knee_stop, delta, 1.);
137  }
138  return FFMAX(range, exp(gain - slope));
139 }
140 
141 static void gate(AudioGateContext *s,
142  const double *src, double *dst, const double *scsrc,
143  int nb_samples, double level_in, double level_sc,
144  AVFilterLink *inlink, AVFilterLink *sclink)
145 {
146  const double makeup = s->makeup;
147  const double attack_coeff = s->attack_coeff;
148  const double release_coeff = s->release_coeff;
149  int n, c;
150 
151  for (n = 0; n < nb_samples; n++, src += inlink->channels, dst += inlink->channels, scsrc += sclink->channels) {
152  double abs_sample = fabs(scsrc[0] * level_sc), gain = 1.0;
153  int detected;
154 
155  if (s->link == 1) {
156  for (c = 1; c < sclink->channels; c++)
157  abs_sample = FFMAX(fabs(scsrc[c] * level_sc), abs_sample);
158  } else {
159  for (c = 1; c < sclink->channels; c++)
160  abs_sample += fabs(scsrc[c] * level_sc);
161 
162  abs_sample /= sclink->channels;
163  }
164 
165  if (s->detection)
166  abs_sample *= abs_sample;
167 
168  s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? attack_coeff : release_coeff);
169 
170  if (s->mode)
171  detected = s->lin_slope > s->lin_knee_start;
172  else
173  detected = s->lin_slope < s->lin_knee_stop;
174 
175  if (s->lin_slope > 0.0 && detected)
176  gain = output_gain(s->lin_slope, s->ratio, s->thres,
177  s->knee, s->knee_start, s->knee_stop,
178  s->range, s->mode);
179 
180  for (c = 0; c < inlink->channels; c++)
181  dst[c] = src[c] * level_in * gain * makeup;
182  }
183 }
184 
185 #if CONFIG_AGATE_FILTER
186 
187 #define agate_options options
188 AVFILTER_DEFINE_CLASS(agate);
189 
190 static int query_formats(AVFilterContext *ctx)
191 {
194  int ret;
195 
196  if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL)) < 0)
197  return ret;
198  ret = ff_set_common_formats(ctx, formats);
199  if (ret < 0)
200  return ret;
201 
202  layouts = ff_all_channel_counts();
203  if (!layouts)
204  return AVERROR(ENOMEM);
205  ret = ff_set_common_channel_layouts(ctx, layouts);
206  if (ret < 0)
207  return ret;
208 
209  formats = ff_all_samplerates();
210  if (!formats)
211  return AVERROR(ENOMEM);
212 
213  return ff_set_common_samplerates(ctx, formats);
214 }
215 
216 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
217 {
218  const double *src = (const double *)in->data[0];
219  AVFilterContext *ctx = inlink->dst;
220  AVFilterLink *outlink = ctx->outputs[0];
221  AudioGateContext *s = ctx->priv;
222  AVFrame *out;
223  double *dst;
224 
225  if (av_frame_is_writable(in)) {
226  out = in;
227  } else {
228  out = ff_get_audio_buffer(outlink, in->nb_samples);
229  if (!out) {
230  av_frame_free(&in);
231  return AVERROR(ENOMEM);
232  }
234  }
235  dst = (double *)out->data[0];
236 
237  gate(s, src, dst, src, in->nb_samples,
238  s->level_in, s->level_in, inlink, inlink);
239 
240  if (out != in)
241  av_frame_free(&in);
242  return ff_filter_frame(outlink, out);
243 }
244 
245 static const AVFilterPad inputs[] = {
246  {
247  .name = "default",
248  .type = AVMEDIA_TYPE_AUDIO,
249  .filter_frame = filter_frame,
250  .config_props = agate_config_input,
251  },
252  { NULL }
253 };
254 
255 static const AVFilterPad outputs[] = {
256  {
257  .name = "default",
258  .type = AVMEDIA_TYPE_AUDIO,
259  },
260  { NULL }
261 };
262 
264  .name = "agate",
265  .description = NULL_IF_CONFIG_SMALL("Audio gate."),
266  .query_formats = query_formats,
267  .priv_size = sizeof(AudioGateContext),
268  .priv_class = &agate_class,
269  .inputs = inputs,
270  .outputs = outputs,
271 };
272 
273 #endif /* CONFIG_AGATE_FILTER */
274 
275 #if CONFIG_SIDECHAINGATE_FILTER
276 
277 #define sidechaingate_options options
278 AVFILTER_DEFINE_CLASS(sidechaingate);
279 
280 static int activate(AVFilterContext *ctx)
281 {
282  AudioGateContext *s = ctx->priv;
283  AVFrame *out = NULL, *in[2] = { NULL };
284  int ret, i, nb_samples;
285  double *dst;
286 
288  if ((ret = ff_inlink_consume_frame(ctx->inputs[0], &in[0])) > 0) {
289  av_audio_fifo_write(s->fifo[0], (void **)in[0]->extended_data,
290  in[0]->nb_samples);
291  av_frame_free(&in[0]);
292  }
293  if (ret < 0)
294  return ret;
295  if ((ret = ff_inlink_consume_frame(ctx->inputs[1], &in[1])) > 0) {
296  av_audio_fifo_write(s->fifo[1], (void **)in[1]->extended_data,
297  in[1]->nb_samples);
298  av_frame_free(&in[1]);
299  }
300  if (ret < 0)
301  return ret;
302 
303  nb_samples = FFMIN(av_audio_fifo_size(s->fifo[0]), av_audio_fifo_size(s->fifo[1]));
304  if (nb_samples) {
305  out = ff_get_audio_buffer(ctx->outputs[0], nb_samples);
306  if (!out)
307  return AVERROR(ENOMEM);
308  for (i = 0; i < 2; i++) {
309  in[i] = ff_get_audio_buffer(ctx->inputs[i], nb_samples);
310  if (!in[i]) {
311  av_frame_free(&in[0]);
312  av_frame_free(&in[1]);
313  av_frame_free(&out);
314  return AVERROR(ENOMEM);
315  }
316  av_audio_fifo_read(s->fifo[i], (void **)in[i]->data, nb_samples);
317  }
318 
319  dst = (double *)out->data[0];
320  out->pts = s->pts;
321  s->pts += av_rescale_q(nb_samples, (AVRational){1, ctx->outputs[0]->sample_rate}, ctx->outputs[0]->time_base);
322 
323  gate(s, (double *)in[0]->data[0], dst,
324  (double *)in[1]->data[0], nb_samples,
325  s->level_in, s->level_sc,
326  ctx->inputs[0], ctx->inputs[1]);
327 
328  av_frame_free(&in[0]);
329  av_frame_free(&in[1]);
330 
331  ret = ff_filter_frame(ctx->outputs[0], out);
332  if (ret < 0)
333  return ret;
334  }
335  FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
336  FF_FILTER_FORWARD_STATUS(ctx->inputs[1], ctx->outputs[0]);
337  if (ff_outlink_frame_wanted(ctx->outputs[0])) {
338  if (!av_audio_fifo_size(s->fifo[0]))
340  if (!av_audio_fifo_size(s->fifo[1]))
342  }
343  return 0;
344 }
345 
346 static int scquery_formats(AVFilterContext *ctx)
347 {
350  static const enum AVSampleFormat sample_fmts[] = {
353  };
354  int ret, i;
355 
356  if (!ctx->inputs[0]->in_channel_layouts ||
358  av_log(ctx, AV_LOG_WARNING,
359  "No channel layout for input 1\n");
360  return AVERROR(EAGAIN);
361  }
362 
363  if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||
364  (ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
365  return ret;
366 
367  for (i = 0; i < 2; i++) {
368  layouts = ff_all_channel_counts();
369  if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
370  return ret;
371  }
372 
373  formats = ff_make_format_list(sample_fmts);
374  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
375  return ret;
376 
377  formats = ff_all_samplerates();
378  return ff_set_common_samplerates(ctx, formats);
379 }
380 
381 static int scconfig_output(AVFilterLink *outlink)
382 {
383  AVFilterContext *ctx = outlink->src;
384  AudioGateContext *s = ctx->priv;
385 
386  if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
387  av_log(ctx, AV_LOG_ERROR,
388  "Inputs must have the same sample rate "
389  "%d for in0 vs %d for in1\n",
390  ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
391  return AVERROR(EINVAL);
392  }
393 
394  outlink->sample_rate = ctx->inputs[0]->sample_rate;
395  outlink->time_base = ctx->inputs[0]->time_base;
396  outlink->channel_layout = ctx->inputs[0]->channel_layout;
397  outlink->channels = ctx->inputs[0]->channels;
398 
399  s->fifo[0] = av_audio_fifo_alloc(ctx->inputs[0]->format, ctx->inputs[0]->channels, 1024);
400  s->fifo[1] = av_audio_fifo_alloc(ctx->inputs[1]->format, ctx->inputs[1]->channels, 1024);
401  if (!s->fifo[0] || !s->fifo[1])
402  return AVERROR(ENOMEM);
403 
404 
405  agate_config_input(ctx->inputs[0]);
406 
407  return 0;
408 }
409 
410 static av_cold void uninit(AVFilterContext *ctx)
411 {
412  AudioGateContext *s = ctx->priv;
413 
414  av_audio_fifo_free(s->fifo[0]);
415  av_audio_fifo_free(s->fifo[1]);
416 }
417 
418 static const AVFilterPad sidechaingate_inputs[] = {
419  {
420  .name = "main",
421  .type = AVMEDIA_TYPE_AUDIO,
422  },{
423  .name = "sidechain",
424  .type = AVMEDIA_TYPE_AUDIO,
425  },
426  { NULL }
427 };
428 
429 static const AVFilterPad sidechaingate_outputs[] = {
430  {
431  .name = "default",
432  .type = AVMEDIA_TYPE_AUDIO,
433  .config_props = scconfig_output,
434  },
435  { NULL }
436 };
437 
439  .name = "sidechaingate",
440  .description = NULL_IF_CONFIG_SMALL("Audio sidechain gate."),
441  .priv_size = sizeof(AudioGateContext),
442  .priv_class = &sidechaingate_class,
443  .query_formats = scquery_formats,
444  .activate = activate,
445  .uninit = uninit,
446  .inputs = sidechaingate_inputs,
447  .outputs = sidechaingate_outputs,
448 };
449 #endif /* CONFIG_SIDECHAINGATE_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:1476
#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:300
if(ret< 0)
Definition: vf_mcdeint.c:279
double lin_slope
Definition: af_agate.c:57
AVOption.
Definition: opt.h:246
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
Main libavfilter public API header.
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:90
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:141
double level_in
Definition: af_agate.c:39
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1602
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
double release
Definition: af_agate.c:42
double ratio
Definition: af_agate.c:44
const char * name
Pad name.
Definition: internal.h:60
#define IS_FAKE_INFINITY(value)
Definition: af_agate.c:115
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:1075
#define av_cold
Definition: attributes.h:88
static av_cold int uninit(AVCodecContext *avctx)
Definition: crystalhd.c:279
float delta
AVOptions.
double lin_knee_start
Definition: af_agate.c:55
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:393
double knee_start
Definition: af_agate.c:53
const char data[16]
Definition: mxf.c:91
double makeup
Definition: af_agate.c:46
#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 OFFSET(x)
Definition: af_agate.c:65
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
#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: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
#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:203
#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
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:336
#define FFMAX(a, b)
Definition: common.h:94
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
double level_sc
Definition: af_agate.c:40
#define FFMIN(a, b)
Definition: common.h:96
AVFilter ff_af_sidechaingate
double lin_knee_stop
Definition: af_agate.c:56
AVFormatContext * ctx
Definition: movenc.c:48
static int activate(AVFilterContext *ctx)
Definition: af_adeclick.c:622
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
double attack_coeff
Definition: af_agate.c:58
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
int nb_samples
number of samples currently in the FIFO
Definition: audio_fifo.c:37
int64_t pts
Definition: af_agate.c:62
static const AVOption options[]
Definition: af_agate.c:68
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
double release_coeff
Definition: af_agate.c:59
double thres
Definition: af_agate.c:52
double threshold
Definition: af_agate.c:43
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static double output_gain(double lin_slope, double ratio, double thres, double knee, double knee_start, double knee_stop, double range, int mode)
Definition: af_agate.c:117
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
#define FF_FILTER_FORWARD_STATUS(inlink, outlink)
Acknowledge the status on an input link and forward it to an output link.
Definition: filters.h:226
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:425
static int filter_frame(DBEContext *s, AVFrame *frame)
Definition: dolby_e.c:565
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:314
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
double knee_stop
Definition: af_agate.c:54
AVAudioFifo * fifo[2]
Definition: af_agate.c:61
static int query_formats(AVFilterContext *ctx)
Definition: aeval.c:244
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:314
double range
Definition: af_agate.c:47
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:731
FILE * out
Definition: movenc.c:54
formats
Definition: signature.h:48
AVFilter ff_af_agate
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:440
#define A
Definition: af_agate.c:66
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:366
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
double attack
Definition: af_agate.c:41