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af_silenceremove.c
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1 /*
2  * Copyright (c) 2001 Heikki Leinonen
3  * Copyright (c) 2001 Chris Bagwell
4  * Copyright (c) 2003 Donnie Smith
5  * Copyright (c) 2014 Paul B Mahol
6  *
7  * This file is part of FFmpeg.
8  *
9  * FFmpeg is free software; you can redistribute it and/or
10  * modify it under the terms of the GNU Lesser General Public
11  * License as published by the Free Software Foundation; either
12  * version 2.1 of the License, or (at your option) any later version.
13  *
14  * FFmpeg is distributed in the hope that it will be useful,
15  * but WITHOUT ANY WARRANTY; without even the implied warranty of
16  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17  * Lesser General Public License for more details.
18  *
19  * You should have received a copy of the GNU Lesser General Public
20  * License along with FFmpeg; if not, write to the Free Software
21  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22  */
23 
24 #include <float.h> /* DBL_MAX */
25 
26 #include "libavutil/opt.h"
27 #include "libavutil/timestamp.h"
28 #include "audio.h"
29 #include "formats.h"
30 #include "avfilter.h"
31 #include "internal.h"
32 
39 };
40 
41 typedef struct SilenceRemoveContext {
42  const AVClass *class;
43 
45 
47  int64_t start_duration;
49 
51  int64_t stop_duration;
53 
54  double *start_holdoff;
58 
59  double *stop_holdoff;
63 
64  double *window;
65  double *window_current;
66  double *window_end;
68  double rms_sum;
69 
71  int restart;
72  int64_t next_pts;
74 
75 #define OFFSET(x) offsetof(SilenceRemoveContext, x)
76 #define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_AUDIO_PARAM
77 static const AVOption silenceremove_options[] = {
78  { "start_periods", NULL, OFFSET(start_periods), AV_OPT_TYPE_INT, {.i64=0}, 0, 9000, FLAGS },
79  { "start_duration", NULL, OFFSET(start_duration), AV_OPT_TYPE_DURATION, {.i64=0}, 0, 9000, FLAGS },
80  { "start_threshold", NULL, OFFSET(start_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, DBL_MAX, FLAGS },
81  { "stop_periods", NULL, OFFSET(stop_periods), AV_OPT_TYPE_INT, {.i64=0}, -9000, 9000, FLAGS },
82  { "stop_duration", NULL, OFFSET(stop_duration), AV_OPT_TYPE_DURATION, {.i64=0}, 0, 9000, FLAGS },
83  { "stop_threshold", NULL, OFFSET(stop_threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, DBL_MAX, FLAGS },
84  { "leave_silence", NULL, OFFSET(leave_silence), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS },
85  { NULL }
86 };
87 
88 AVFILTER_DEFINE_CLASS(silenceremove);
89 
90 static av_cold int init(AVFilterContext *ctx)
91 {
92  SilenceRemoveContext *s = ctx->priv;
93 
94  if (s->stop_periods < 0) {
95  s->stop_periods = -s->stop_periods;
96  s->restart = 1;
97  }
98 
99  return 0;
100 }
101 
103 {
104  memset(s->window, 0, s->window_size * sizeof(*s->window));
105 
106  s->window_current = s->window;
107  s->window_end = s->window + s->window_size;
108  s->rms_sum = 0;
109 }
110 
111 static int config_input(AVFilterLink *inlink)
112 {
113  AVFilterContext *ctx = inlink->dst;
114  SilenceRemoveContext *s = ctx->priv;
115 
116  s->window_size = (inlink->sample_rate / 50) * inlink->channels;
117  s->window = av_malloc_array(s->window_size, sizeof(*s->window));
118  if (!s->window)
119  return AVERROR(ENOMEM);
120 
121  clear_rms(s);
122 
124  AV_TIME_BASE);
126  AV_TIME_BASE);
127 
129  sizeof(*s->start_holdoff) *
130  inlink->channels);
131  if (!s->start_holdoff)
132  return AVERROR(ENOMEM);
133 
134  s->start_holdoff_offset = 0;
135  s->start_holdoff_end = 0;
136  s->start_found_periods = 0;
137 
139  sizeof(*s->stop_holdoff) *
140  inlink->channels);
141  if (!s->stop_holdoff)
142  return AVERROR(ENOMEM);
143 
144  s->stop_holdoff_offset = 0;
145  s->stop_holdoff_end = 0;
146  s->stop_found_periods = 0;
147 
148  if (s->start_periods)
149  s->mode = SILENCE_TRIM;
150  else
151  s->mode = SILENCE_COPY;
152 
153  return 0;
154 }
155 
156 static int config_output(AVFilterLink *outlink)
157 {
158  outlink->flags |= FF_LINK_FLAG_REQUEST_LOOP;
159 
160  return 0;
161 }
162 
163 static double compute_rms(SilenceRemoveContext *s, double sample)
164 {
165  double new_sum;
166 
167  new_sum = s->rms_sum;
168  new_sum -= *s->window_current;
169  new_sum += sample * sample;
170 
171  return sqrt(new_sum / s->window_size);
172 }
173 
175 {
176  s->rms_sum -= *s->window_current;
177  *s->window_current = sample * sample;
178  s->rms_sum += *s->window_current;
179 
180  s->window_current++;
181  if (s->window_current >= s->window_end)
182  s->window_current = s->window;
183 }
184 
185 static void flush(AVFrame *out, AVFilterLink *outlink,
186  int *nb_samples_written, int *ret)
187 {
188  if (*nb_samples_written) {
189  out->nb_samples = *nb_samples_written / outlink->channels;
190  *ret = ff_filter_frame(outlink, out);
191  *nb_samples_written = 0;
192  } else {
193  av_frame_free(&out);
194  }
195 }
196 
197 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
198 {
199  AVFilterContext *ctx = inlink->dst;
200  AVFilterLink *outlink = ctx->outputs[0];
201  SilenceRemoveContext *s = ctx->priv;
202  int i, j, threshold, ret = 0;
203  int nbs, nb_samples_read, nb_samples_written;
204  double *obuf, *ibuf = (double *)in->data[0];
205  AVFrame *out;
206 
207  nb_samples_read = nb_samples_written = 0;
208 
209  switch (s->mode) {
210  case SILENCE_TRIM:
211 silence_trim:
212  nbs = in->nb_samples - nb_samples_read / inlink->channels;
213  if (!nbs)
214  break;
215 
216  for (i = 0; i < nbs; i++) {
217  threshold = 0;
218  for (j = 0; j < inlink->channels; j++) {
219  threshold |= compute_rms(s, ibuf[j]) > s->start_threshold;
220  }
221 
222  if (threshold) {
223  for (j = 0; j < inlink->channels; j++) {
224  update_rms(s, *ibuf);
225  s->start_holdoff[s->start_holdoff_end++] = *ibuf++;
226  nb_samples_read++;
227  }
228 
229  if (s->start_holdoff_end >= s->start_duration * inlink->channels) {
230  if (++s->start_found_periods >= s->start_periods) {
232  goto silence_trim_flush;
233  }
234 
235  s->start_holdoff_offset = 0;
236  s->start_holdoff_end = 0;
237  }
238  } else {
239  s->start_holdoff_end = 0;
240 
241  for (j = 0; j < inlink->channels; j++)
242  update_rms(s, ibuf[j]);
243 
244  ibuf += inlink->channels;
245  nb_samples_read += inlink->channels;
246  }
247  }
248  break;
249 
250  case SILENCE_TRIM_FLUSH:
251 silence_trim_flush:
253  nbs -= nbs % inlink->channels;
254  if (!nbs)
255  break;
256 
257  out = ff_get_audio_buffer(inlink, nbs / inlink->channels);
258  if (!out) {
259  av_frame_free(&in);
260  return AVERROR(ENOMEM);
261  }
262 
263  memcpy(out->data[0], &s->start_holdoff[s->start_holdoff_offset],
264  nbs * sizeof(double));
265  s->start_holdoff_offset += nbs;
266 
267  ret = ff_filter_frame(outlink, out);
268 
270  s->start_holdoff_offset = 0;
271  s->start_holdoff_end = 0;
272  s->mode = SILENCE_COPY;
273  goto silence_copy;
274  }
275  break;
276 
277  case SILENCE_COPY:
278 silence_copy:
279  nbs = in->nb_samples - nb_samples_read / inlink->channels;
280  if (!nbs)
281  break;
282 
283  out = ff_get_audio_buffer(inlink, nbs);
284  if (!out) {
285  av_frame_free(&in);
286  return AVERROR(ENOMEM);
287  }
288  obuf = (double *)out->data[0];
289 
290  if (s->stop_periods) {
291  for (i = 0; i < nbs; i++) {
292  threshold = 1;
293  for (j = 0; j < inlink->channels; j++)
294  threshold &= compute_rms(s, ibuf[j]) > s->stop_threshold;
295 
296  if (threshold && s->stop_holdoff_end && !s->leave_silence) {
298  flush(out, outlink, &nb_samples_written, &ret);
299  goto silence_copy_flush;
300  } else if (threshold) {
301  for (j = 0; j < inlink->channels; j++) {
302  update_rms(s, *ibuf);
303  *obuf++ = *ibuf++;
304  nb_samples_read++;
305  nb_samples_written++;
306  }
307  } else if (!threshold) {
308  for (j = 0; j < inlink->channels; j++) {
309  update_rms(s, *ibuf);
310  if (s->leave_silence) {
311  *obuf++ = *ibuf;
312  nb_samples_written++;
313  }
314 
315  s->stop_holdoff[s->stop_holdoff_end++] = *ibuf++;
316  nb_samples_read++;
317  }
318 
319  if (s->stop_holdoff_end >= s->stop_duration * inlink->channels) {
320  if (++s->stop_found_periods >= s->stop_periods) {
321  s->stop_holdoff_offset = 0;
322  s->stop_holdoff_end = 0;
323 
324  if (!s->restart) {
325  s->mode = SILENCE_STOP;
326  flush(out, outlink, &nb_samples_written, &ret);
327  goto silence_stop;
328  } else {
329  s->stop_found_periods = 0;
330  s->start_found_periods = 0;
331  s->start_holdoff_offset = 0;
332  s->start_holdoff_end = 0;
333  clear_rms(s);
334  s->mode = SILENCE_TRIM;
335  flush(out, outlink, &nb_samples_written, &ret);
336  goto silence_trim;
337  }
338  }
340  flush(out, outlink, &nb_samples_written, &ret);
341  goto silence_copy_flush;
342  }
343  }
344  }
345  flush(out, outlink, &nb_samples_written, &ret);
346  } else {
347  memcpy(obuf, ibuf, sizeof(double) * nbs * inlink->channels);
348  ret = ff_filter_frame(outlink, out);
349  }
350  break;
351 
352  case SILENCE_COPY_FLUSH:
353 silence_copy_flush:
355  nbs -= nbs % inlink->channels;
356  if (!nbs)
357  break;
358 
359  out = ff_get_audio_buffer(inlink, nbs / inlink->channels);
360  if (!out) {
361  av_frame_free(&in);
362  return AVERROR(ENOMEM);
363  }
364 
365  memcpy(out->data[0], &s->stop_holdoff[s->stop_holdoff_offset],
366  nbs * sizeof(double));
367  s->stop_holdoff_offset += nbs;
368 
369  ret = ff_filter_frame(outlink, out);
370 
371  if (s->stop_holdoff_offset == s->stop_holdoff_end) {
372  s->stop_holdoff_offset = 0;
373  s->stop_holdoff_end = 0;
374  s->mode = SILENCE_COPY;
375  goto silence_copy;
376  }
377  break;
378  case SILENCE_STOP:
379 silence_stop:
380  break;
381  }
382 
383  av_frame_free(&in);
384 
385  return ret;
386 }
387 
388 static int request_frame(AVFilterLink *outlink)
389 {
390  AVFilterContext *ctx = outlink->src;
391  SilenceRemoveContext *s = ctx->priv;
392  int ret;
393 
394  ret = ff_request_frame(ctx->inputs[0]);
395  if (ret == AVERROR_EOF && (s->mode == SILENCE_COPY_FLUSH ||
396  s->mode == SILENCE_COPY)) {
397  int nbs = s->stop_holdoff_end - s->stop_holdoff_offset;
398  if (nbs) {
399  AVFrame *frame;
400 
401  frame = ff_get_audio_buffer(outlink, nbs / outlink->channels);
402  if (!frame)
403  return AVERROR(ENOMEM);
404 
405  memcpy(frame->data[0], &s->stop_holdoff[s->stop_holdoff_offset],
406  nbs * sizeof(double));
407  ret = ff_filter_frame(ctx->inputs[0], frame);
408  }
409  s->mode = SILENCE_STOP;
410  }
411  return ret;
412 }
413 
415 {
418  static const enum AVSampleFormat sample_fmts[] = {
420  };
421  int ret;
422 
423  layouts = ff_all_channel_layouts();
424  if (!layouts)
425  return AVERROR(ENOMEM);
426  ret = ff_set_common_channel_layouts(ctx, layouts);
427  if (ret < 0)
428  return ret;
429 
430  formats = ff_make_format_list(sample_fmts);
431  if (!formats)
432  return AVERROR(ENOMEM);
433  ret = ff_set_common_formats(ctx, formats);
434  if (ret < 0)
435  return ret;
436 
437  formats = ff_all_samplerates();
438  if (!formats)
439  return AVERROR(ENOMEM);
440  return ff_set_common_samplerates(ctx, formats);
441 }
442 
443 static av_cold void uninit(AVFilterContext *ctx)
444 {
445  SilenceRemoveContext *s = ctx->priv;
446 
447  av_freep(&s->start_holdoff);
448  av_freep(&s->stop_holdoff);
449  av_freep(&s->window);
450 }
451 
453  {
454  .name = "default",
455  .type = AVMEDIA_TYPE_AUDIO,
456  .config_props = config_input,
457  .filter_frame = filter_frame,
458  },
459  { NULL }
460 };
461 
463  {
464  .name = "default",
465  .type = AVMEDIA_TYPE_AUDIO,
466  .config_props = config_output,
467  .request_frame = request_frame,
468  },
469  { NULL }
470 };
471 
473  .name = "silenceremove",
474  .description = NULL_IF_CONFIG_SMALL("Remove silence."),
475  .priv_size = sizeof(SilenceRemoveContext),
476  .priv_class = &silenceremove_class,
477  .init = init,
478  .uninit = uninit,
480  .inputs = silenceremove_inputs,
481  .outputs = silenceremove_outputs,
482 };
static void flush(AVFrame *out, AVFilterLink *outlink, int *nb_samples_written, int *ret)
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
#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:523
const char * s
Definition: avisynth_c.h:631
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
AVOption.
Definition: opt.h:255
static const AVFilterPad silenceremove_outputs[]
static const AVFilterPad outputs[]
Definition: af_ashowinfo.c:248
Main libavfilter public API header.
static int config_output(AVFilterLink *outlink)
AVFilter ff_af_silenceremove
static enum AVSampleFormat formats[]
#define sample
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
#define FLAGS
const char * name
Pad name.
Definition: internal.h:69
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:641
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1158
#define av_cold
Definition: attributes.h:74
AVOptions.
timestamp utils, mostly useful for debugging/logging purposes
SilenceMode
static AVFrame * frame
static av_cold int init(AVFilterContext *ctx)
#define AVERROR_EOF
End of file.
Definition: error.h:55
A filter pad used for either input or output.
Definition: internal.h:63
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:542
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:74
#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:148
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:175
void * priv
private data for use by the filter
Definition: avfilter.h:654
#define FFMAX(a, b)
Definition: common.h:79
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:127
#define AV_TIME_BASE
Internal time base represented as integer.
Definition: avutil.h:246
static const AVFilterPad silenceremove_inputs[]
static void clear_rms(SilenceRemoveContext *s)
static int request_frame(AVFilterLink *outlink)
AVFILTER_DEFINE_CLASS(silenceremove)
enum SilenceMode mode
Frame requests may need to loop in order to be fulfilled.
Definition: internal.h:374
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout (w...
Definition: formats.c:385
A list of supported channel layouts.
Definition: formats.h:85
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:59
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
static av_cold void uninit(AVFilterContext *ctx)
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:470
static void update_rms(SilenceRemoveContext *s, double sample)
static const AVFilterPad inputs[]
Definition: af_ashowinfo.c:239
static double compute_rms(SilenceRemoveContext *s, double sample)
const char * name
Filter name.
Definition: avfilter.h:474
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:648
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:379
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:182
if(ret< 0)
Definition: vf_mcdeint.c:280
static const AVOption silenceremove_options[]
A list of supported formats for one end of a filter link.
Definition: formats.h:64
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-> out
An instance of a filter.
Definition: avfilter.h:633
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#define av_freep(p)
#define av_malloc_array(a, b)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:343
internal API functions
#define OFFSET(x)
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:530