FFmpeg  4.0
af_alimiter.c
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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  * Lookahead limiter filter
25  */
26 
27 #include "libavutil/avassert.h"
29 #include "libavutil/common.h"
30 #include "libavutil/opt.h"
31 
32 #include "audio.h"
33 #include "avfilter.h"
34 #include "formats.h"
35 #include "internal.h"
36 
37 typedef struct AudioLimiterContext {
38  const AVClass *class;
39 
40  double limit;
41  double attack;
42  double release;
43  double att;
44  double level_in;
45  double level_out;
48  double asc;
49  int asc_c;
50  int asc_pos;
51  double asc_coeff;
52 
53  double *buffer;
55  int pos;
56  int *nextpos;
57  double *nextdelta;
58 
59  double delta;
60  int nextiter;
61  int nextlen;
64 
65 #define OFFSET(x) offsetof(AudioLimiterContext, x)
66 #define A AV_OPT_FLAG_AUDIO_PARAM
67 #define F AV_OPT_FLAG_FILTERING_PARAM
68 
69 static const AVOption alimiter_options[] = {
70  { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
71  { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1},.015625, 64, A|F },
72  { "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, A|F },
73  { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, A|F },
74  { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, A|F },
75  { "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A|F },
76  { "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A|F },
77  { "level", "auto level", OFFSET(auto_level), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A|F },
78  { NULL }
79 };
80 
81 AVFILTER_DEFINE_CLASS(alimiter);
82 
84 {
85  AudioLimiterContext *s = ctx->priv;
86 
87  s->attack /= 1000.;
88  s->release /= 1000.;
89  s->att = 1.;
90  s->asc_pos = -1;
91  s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
92 
93  return 0;
94 }
95 
96 static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
97  double peak, double limit, double patt, int asc)
98 {
99  double rdelta = (1.0 - patt) / (sample_rate * release);
100 
101  if (asc && s->auto_release && s->asc_c > 0) {
102  double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
103 
104  if (a_att > patt) {
105  double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
106 
107  if (delta < rdelta)
108  rdelta = delta;
109  }
110  }
111 
112  return rdelta;
113 }
114 
115 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
116 {
117  AVFilterContext *ctx = inlink->dst;
118  AudioLimiterContext *s = ctx->priv;
119  AVFilterLink *outlink = ctx->outputs[0];
120  const double *src = (const double *)in->data[0];
121  const int channels = inlink->channels;
122  const int buffer_size = s->buffer_size;
123  double *dst, *buffer = s->buffer;
124  const double release = s->release;
125  const double limit = s->limit;
126  double *nextdelta = s->nextdelta;
127  double level = s->auto_level ? 1 / limit : 1;
128  const double level_out = s->level_out;
129  const double level_in = s->level_in;
130  int *nextpos = s->nextpos;
131  AVFrame *out;
132  double *buf;
133  int n, c, i;
134 
135  if (av_frame_is_writable(in)) {
136  out = in;
137  } else {
138  out = ff_get_audio_buffer(outlink, in->nb_samples);
139  if (!out) {
140  av_frame_free(&in);
141  return AVERROR(ENOMEM);
142  }
144  }
145  dst = (double *)out->data[0];
146 
147  for (n = 0; n < in->nb_samples; n++) {
148  double peak = 0;
149 
150  for (c = 0; c < channels; c++) {
151  double sample = src[c] * level_in;
152 
153  buffer[s->pos + c] = sample;
154  peak = FFMAX(peak, fabs(sample));
155  }
156 
157  if (s->auto_release && peak > limit) {
158  s->asc += peak;
159  s->asc_c++;
160  }
161 
162  if (peak > limit) {
163  double patt = FFMIN(limit / peak, 1.);
164  double rdelta = get_rdelta(s, release, inlink->sample_rate,
165  peak, limit, patt, 0);
166  double delta = (limit / peak - s->att) / buffer_size * channels;
167  int found = 0;
168 
169  if (delta < s->delta) {
170  s->delta = delta;
171  nextpos[0] = s->pos;
172  nextpos[1] = -1;
173  nextdelta[0] = rdelta;
174  s->nextlen = 1;
175  s->nextiter= 0;
176  } else {
177  for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
178  int j = i % buffer_size;
179  double ppeak, pdelta;
180 
181  ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
182  fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
183  pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
184  if (pdelta < nextdelta[j]) {
185  nextdelta[j] = pdelta;
186  found = 1;
187  break;
188  }
189  }
190  if (found) {
191  s->nextlen = i - s->nextiter + 1;
192  nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
193  nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
194  nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
195  s->nextlen++;
196  }
197  }
198  }
199 
200  buf = &s->buffer[(s->pos + channels) % buffer_size];
201  peak = 0;
202  for (c = 0; c < channels; c++) {
203  double sample = buf[c];
204 
205  peak = FFMAX(peak, fabs(sample));
206  }
207 
208  if (s->pos == s->asc_pos && !s->asc_changed)
209  s->asc_pos = -1;
210 
211  if (s->auto_release && s->asc_pos == -1 && peak > limit) {
212  s->asc -= peak;
213  s->asc_c--;
214  }
215 
216  s->att += s->delta;
217 
218  for (c = 0; c < channels; c++)
219  dst[c] = buf[c] * s->att;
220 
221  if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
222  if (s->auto_release) {
223  s->delta = get_rdelta(s, release, inlink->sample_rate,
224  peak, limit, s->att, 1);
225  if (s->nextlen > 1) {
226  int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
227  double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
228  fabs(buffer[pnextpos]) :
229  fabs(buffer[pnextpos + 1]);
230  double pdelta = (limit / ppeak - s->att) /
231  (((buffer_size + pnextpos -
232  ((s->pos + channels) % buffer_size)) %
233  buffer_size) / channels);
234  if (pdelta < s->delta)
235  s->delta = pdelta;
236  }
237  } else {
238  s->delta = nextdelta[s->nextiter];
239  s->att = limit / peak;
240  }
241 
242  s->nextlen -= 1;
243  nextpos[s->nextiter] = -1;
244  s->nextiter = (s->nextiter + 1) % buffer_size;
245  }
246 
247  if (s->att > 1.) {
248  s->att = 1.;
249  s->delta = 0.;
250  s->nextiter = 0;
251  s->nextlen = 0;
252  nextpos[0] = -1;
253  }
254 
255  if (s->att <= 0.) {
256  s->att = 0.0000000000001;
257  s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
258  }
259 
260  if (s->att != 1. && (1. - s->att) < 0.0000000000001)
261  s->att = 1.;
262 
263  if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
264  s->delta = 0.;
265 
266  for (c = 0; c < channels; c++)
267  dst[c] = av_clipd(dst[c], -limit, limit) * level * level_out;
268 
269  s->pos = (s->pos + channels) % buffer_size;
270  src += channels;
271  dst += channels;
272  }
273 
274  if (in != out)
275  av_frame_free(&in);
276 
277  return ff_filter_frame(outlink, out);
278 }
279 
281 {
284  static const enum AVSampleFormat sample_fmts[] = {
287  };
288  int ret;
289 
290  layouts = ff_all_channel_counts();
291  if (!layouts)
292  return AVERROR(ENOMEM);
293  ret = ff_set_common_channel_layouts(ctx, layouts);
294  if (ret < 0)
295  return ret;
296 
297  formats = ff_make_format_list(sample_fmts);
298  if (!formats)
299  return AVERROR(ENOMEM);
300  ret = ff_set_common_formats(ctx, formats);
301  if (ret < 0)
302  return ret;
303 
304  formats = ff_all_samplerates();
305  if (!formats)
306  return AVERROR(ENOMEM);
307  return ff_set_common_samplerates(ctx, formats);
308 }
309 
310 static int config_input(AVFilterLink *inlink)
311 {
312  AVFilterContext *ctx = inlink->dst;
313  AudioLimiterContext *s = ctx->priv;
314  int obuffer_size;
315 
316  obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
317  if (obuffer_size < inlink->channels)
318  return AVERROR(EINVAL);
319 
320  s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
321  s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
322  s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
323  if (!s->buffer || !s->nextdelta || !s->nextpos)
324  return AVERROR(ENOMEM);
325 
326  memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
327  s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
328  s->buffer_size -= s->buffer_size % inlink->channels;
329 
330  if (s->buffer_size <= 0) {
331  av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
332  return AVERROR(EINVAL);
333  }
334 
335  return 0;
336 }
337 
339 {
340  AudioLimiterContext *s = ctx->priv;
341 
342  av_freep(&s->buffer);
343  av_freep(&s->nextdelta);
344  av_freep(&s->nextpos);
345 }
346 
347 static const AVFilterPad alimiter_inputs[] = {
348  {
349  .name = "main",
350  .type = AVMEDIA_TYPE_AUDIO,
351  .filter_frame = filter_frame,
352  .config_props = config_input,
353  },
354  { NULL }
355 };
356 
357 static const AVFilterPad alimiter_outputs[] = {
358  {
359  .name = "default",
360  .type = AVMEDIA_TYPE_AUDIO,
361  },
362  { NULL }
363 };
364 
366  .name = "alimiter",
367  .description = NULL_IF_CONFIG_SMALL("Audio lookahead limiter."),
368  .priv_size = sizeof(AudioLimiterContext),
369  .priv_class = &alimiter_class,
370  .init = init,
371  .uninit = uninit,
373  .inputs = alimiter_inputs,
374  .outputs = alimiter_outputs,
375 };
#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
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
AVOption.
Definition: opt.h:246
#define OFFSET(x)
Definition: af_alimiter.c:65
Main libavfilter public API header.
static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate, double peak, double limit, double patt, int asc)
Definition: af_alimiter.c:96
channels
Definition: aptx.c:30
#define src
Definition: vp8dsp.c:254
#define sample
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
const char * name
Pad name.
Definition: internal.h:60
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define av_cold
Definition: attributes.h:82
AVOptions.
AVFILTER_DEFINE_CLASS(alimiter)
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
#define A
Definition: af_alimiter.c:66
#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
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: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
simple assert() macros that are a bit more flexible than ISO C assert().
static av_cold int init(AVFilterContext *ctx)
Definition: af_alimiter.c:83
#define FFMAX(a, b)
Definition: common.h:94
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:96
#define F
Definition: af_alimiter.c:67
AVFormatContext * ctx
Definition: movenc.c:48
int n
Definition: avisynth_c.h:684
static int config_input(AVFilterLink *inlink)
Definition: af_alimiter.c:310
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
if(ret< 0)
Definition: vf_mcdeint.c:279
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
sample_rate
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:592
static const int8_t patt[4]
Definition: vf_noise.c:67
void * buf
Definition: avisynth_c.h:690
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
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
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_alimiter.c:338
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:395
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_alimiter.c:115
static const AVOption alimiter_options[]
Definition: af_alimiter.c:69
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:232
uint8_t level
Definition: svq3.c:207
AVFilter ff_af_alimiter
Definition: af_alimiter.c:365
common internal and external API header
static double c[64]
static const AVFilterPad alimiter_inputs[]
Definition: af_alimiter.c:347
static const AVFilterPad alimiter_outputs[]
Definition: af_alimiter.c:357
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
#define av_freep(p)
#define av_malloc_array(a, b)
formats
Definition: signature.h:48
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
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
for(j=16;j >0;--j)
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:652
static int query_formats(AVFilterContext *ctx)
Definition: af_alimiter.c:280