FFmpeg
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 
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 
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 += nb_samples;
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:1494
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:550
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
double lin_slope
Definition: af_agate.c:57
AVOption.
Definition: opt.h:246
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
Main libavfilter public API header.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
FF_FILTER_FORWARD_STATUS(inlink, outlink)
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
#define src
Definition: vp8dsp.c:254
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:1620
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:434
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1093
#define av_cold
Definition: attributes.h:82
static av_cold int uninit(AVCodecContext *avctx)
Definition: crystalhd.c:279
float delta
AVOptions.
double lin_knee_start
Definition: af_agate.c:55
filter_frame For filters that do not use the activate() callback
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
double knee_start
Definition: af_agate.c:53
double makeup
Definition: af_agate.c:46
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function.If this function returns true
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
#define OFFSET(x)
Definition: af_agate.c:65
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:569
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:342
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
uint64_t * channel_layouts
list of channel layouts
Definition: formats.h:86
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
#define s(width, name)
Definition: cbs_vp9.c:257
int n
Definition: avisynth_c.h:760
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
if(ret)
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:594
double release_coeff
Definition: af_agate.c:59
double thres
Definition: af_agate.c:52
double threshold
Definition: af_agate.c:43
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31))))#define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac){}void ff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map){AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);return NULL;}return ac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;}int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){int use_generic=1;int len=in->nb_samples;int p;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
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
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
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:309
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
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
Audio FIFO Buffer.
#define AVFILTER_DEFINE_CLASS(fname)
Definition: internal.h:334
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:701
FILE * out
Definition: movenc.c:54
formats
Definition: signature.h:48
AVFilter ff_af_agate
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:409
#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:361
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:557
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
double attack
Definition: af_agate.c:41