FFmpeg
af_adelay.c
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1 /*
2  * Copyright (c) 2013 Paul B Mahol
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 #include "libavutil/avstring.h"
22 #include "libavutil/mem.h"
23 #include "libavutil/opt.h"
24 #include "libavutil/samplefmt.h"
25 #include "avfilter.h"
26 #include "audio.h"
27 #include "filters.h"
28 
29 typedef struct ChanDelay {
31  size_t delay_index;
32  size_t index;
33  unsigned int samples_size;
34  uint8_t *samples;
35 } ChanDelay;
36 
37 typedef struct AudioDelayContext {
38  const AVClass *class;
39  int all;
40  char *delays;
42  int nb_delays;
48  int eof;
49 
51 
52  void (*delay_channel)(ChanDelay *d, int nb_samples,
53  const uint8_t *src, uint8_t *dst);
54  int (*resize_channel_samples)(ChanDelay *d, int64_t new_delay);
56 
57 #define OFFSET(x) offsetof(AudioDelayContext, x)
58 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
59 
60 static const AVOption adelay_options[] = {
61  { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A | AV_OPT_FLAG_RUNTIME_PARAM },
62  { "all", "use last available delay for remained channels", OFFSET(all), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
63  { NULL }
64 };
65 
66 AVFILTER_DEFINE_CLASS(adelay);
67 
68 #define DELAY(name, type, fill) \
69 static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \
70  const uint8_t *ssrc, uint8_t *ddst) \
71 { \
72  const type *src = (type *)ssrc; \
73  type *dst = (type *)ddst; \
74  type *samples = (type *)d->samples; \
75  \
76  while (nb_samples) { \
77  if (d->delay_index < d->delay) { \
78  const int len = FFMIN(nb_samples, d->delay - d->delay_index); \
79  \
80  memcpy(&samples[d->delay_index], src, len * sizeof(type)); \
81  memset(dst, fill, len * sizeof(type)); \
82  d->delay_index += len; \
83  src += len; \
84  dst += len; \
85  nb_samples -= len; \
86  } else { \
87  *dst = samples[d->index]; \
88  samples[d->index] = *src; \
89  nb_samples--; \
90  d->index++; \
91  src++, dst++; \
92  d->index = d->index >= d->delay ? 0 : d->index; \
93  } \
94  } \
95 }
96 
97 DELAY(u8, uint8_t, 0x80)
98 DELAY(s16, int16_t, 0)
99 DELAY(s32, int32_t, 0)
100 DELAY(flt, float, 0)
101 DELAY(dbl, double, 0)
102 
103 #define CHANGE_DELAY(name, type, fill) \
104 static int resize_samples_## name ##p(ChanDelay *d, int64_t new_delay) \
105 { \
106  type *samples; \
107  \
108  if (new_delay == d->delay) { \
109  return 0; \
110  } \
111  \
112  if (new_delay == 0) { \
113  av_freep(&d->samples); \
114  d->samples_size = 0; \
115  d->delay = 0; \
116  d->index = 0; \
117  d->delay_index = 0; \
118  return 0; \
119  } \
120  \
121  samples = (type *) av_fast_realloc(d->samples, &d->samples_size, new_delay * sizeof(type)); \
122  if (!samples) { \
123  return AVERROR(ENOMEM); \
124  } \
125  \
126  if (new_delay < d->delay) { \
127  if (d->index > new_delay) { \
128  d->index -= new_delay; \
129  memmove(samples, &samples[new_delay], d->index * sizeof(type)); \
130  d->delay_index = new_delay; \
131  } else if (d->delay_index > d->index) { \
132  memmove(&samples[d->index], &samples[d->index+(d->delay-new_delay)], \
133  (new_delay - d->index) * sizeof(type)); \
134  d->delay_index -= d->delay - new_delay; \
135  } \
136  } else { \
137  size_t block_size; \
138  if (d->delay_index >= d->delay) { \
139  block_size = (d->delay - d->index) * sizeof(type); \
140  memmove(&samples[d->index+(new_delay - d->delay)], &samples[d->index], block_size); \
141  d->delay_index = new_delay; \
142  } else { \
143  d->delay_index += new_delay - d->delay; \
144  } \
145  block_size = (new_delay - d->delay) * sizeof(type); \
146  memset(&samples[d->index], fill, block_size); \
147  } \
148  d->delay = new_delay; \
149  d->samples = (void *) samples; \
150  return 0; \
151 }
152 
153 CHANGE_DELAY(u8, uint8_t, 0x80)
154 CHANGE_DELAY(s16, int16_t, 0)
155 CHANGE_DELAY(s32, int32_t, 0)
156 CHANGE_DELAY(flt, float, 0)
157 CHANGE_DELAY(dbl, double, 0)
158 
159 static int parse_delays(char *p, char **saveptr, int64_t *result, AVFilterContext *ctx, int sample_rate) {
160  float delay, div;
161  int ret;
162  char *arg;
163  char type = 0;
164 
165  if (!(arg = av_strtok(p, "|", saveptr)))
166  return 1;
167 
168  ret = av_sscanf(arg, "%"SCNd64"%c", result, &type);
169  if (ret != 2 || type != 'S') {
170  div = type == 's' ? 1.0 : 1000.0;
171  if (av_sscanf(arg, "%f", &delay) != 1) {
172  av_log(ctx, AV_LOG_ERROR, "Invalid syntax for delay.\n");
173  return AVERROR(EINVAL);
174  }
175  *result = delay * sample_rate / div;
176  }
177 
178  if (*result < 0) {
179  av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");
180  return AVERROR(EINVAL);
181  }
182  return 0;
183 }
184 
186 {
187  AVFilterContext *ctx = inlink->dst;
188  AudioDelayContext *s = ctx->priv;
189  char *p, *saveptr = NULL;
190  int i;
191 
192  s->next_pts = AV_NOPTS_VALUE;
193  s->chandelay = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->chandelay));
194  if (!s->chandelay)
195  return AVERROR(ENOMEM);
196  s->nb_delays = inlink->ch_layout.nb_channels;
197  s->block_align = av_get_bytes_per_sample(inlink->format);
198 
199  p = s->delays;
200  for (i = 0; i < s->nb_delays; i++) {
201  ChanDelay *d = &s->chandelay[i];
202  int ret;
203 
204  ret = parse_delays(p, &saveptr, &d->delay, ctx, inlink->sample_rate);
205  if (ret == 1)
206  break;
207  else if (ret < 0)
208  return ret;
209  p = NULL;
210  }
211 
212  if (s->all && i) {
213  for (int j = i; j < s->nb_delays; j++)
214  s->chandelay[j].delay = s->chandelay[i-1].delay;
215  }
216 
217  s->padding = s->chandelay[0].delay;
218  for (i = 1; i < s->nb_delays; i++) {
219  ChanDelay *d = &s->chandelay[i];
220 
221  s->padding = FFMIN(s->padding, d->delay);
222  }
223 
224  if (s->padding) {
225  for (i = 0; i < s->nb_delays; i++) {
226  ChanDelay *d = &s->chandelay[i];
227 
228  d->delay -= s->padding;
229  }
230 
231  s->offset = av_rescale_q(s->padding,
232  av_make_q(1, inlink->sample_rate),
233  inlink->time_base);
234  }
235 
236  for (i = 0; i < s->nb_delays; i++) {
237  ChanDelay *d = &s->chandelay[i];
238 
239  if (!d->delay)
240  continue;
241 
242  if (d->delay > SIZE_MAX) {
243  av_log(ctx, AV_LOG_ERROR, "Requested delay is too big.\n");
244  return AVERROR(EINVAL);
245  }
246 
247  d->samples = av_malloc_array(d->delay, s->block_align);
248  if (!d->samples)
249  return AVERROR(ENOMEM);
250  d->samples_size = d->delay * s->block_align;
251 
252  s->max_delay = FFMAX(s->max_delay, d->delay);
253  }
254 
255  switch (inlink->format) {
256  case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ;
257  s->resize_channel_samples = resize_samples_u8p; break;
258  case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p;
259  s->resize_channel_samples = resize_samples_s16p; break;
260  case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p;
261  s->resize_channel_samples = resize_samples_s32p; break;
262  case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp;
263  s->resize_channel_samples = resize_samples_fltp; break;
264  case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp;
265  s->resize_channel_samples = resize_samples_dblp; break;
266  }
267 
268  return 0;
269 }
270 
271 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
272  char *res, int res_len, int flags)
273 {
274  int ret = AVERROR(ENOSYS);
275  AVFilterLink *inlink = ctx->inputs[0];
276  AudioDelayContext *s = ctx->priv;
277 
278  if (!strcmp(cmd, "delays")) {
279  int64_t delay;
280  char *p, *saveptr = NULL;
281  int64_t all_delay = -1;
282  int64_t max_delay = 0;
283  char *args_cpy = av_strdup(args);
284  if (args_cpy == NULL) {
285  return AVERROR(ENOMEM);
286  }
287 
288  ret = 0;
289  p = args_cpy;
290 
291  if (!strncmp(args, "all:", 4)) {
292  p = &args_cpy[4];
293  ret = parse_delays(p, &saveptr, &all_delay, ctx, inlink->sample_rate);
294  if (ret == 1)
295  ret = AVERROR(EINVAL);
296  else if (ret == 0)
297  delay = all_delay;
298  }
299 
300  if (!ret) {
301  for (int i = 0; i < s->nb_delays; i++) {
302  ChanDelay *d = &s->chandelay[i];
303 
304  if (all_delay < 0) {
305  ret = parse_delays(p, &saveptr, &delay, ctx, inlink->sample_rate);
306  if (ret != 0) {
307  ret = 0;
308  break;
309  }
310  p = NULL;
311  }
312 
313  ret = s->resize_channel_samples(d, delay);
314  if (ret)
315  break;
316  max_delay = FFMAX(max_delay, d->delay);
317  }
318  s->max_delay = FFMAX(s->max_delay, max_delay);
319  }
320  av_freep(&args_cpy);
321  }
322  return ret;
323 }
324 
326 {
327  AVFilterContext *ctx = inlink->dst;
328  AVFilterLink *outlink = ctx->outputs[0];
329  AudioDelayContext *s = ctx->priv;
330  AVFrame *out_frame;
331  int i;
332 
333  if (ctx->is_disabled || !s->delays) {
334  s->input = NULL;
335  return ff_filter_frame(outlink, frame);
336  }
337 
338  s->next_pts = av_rescale_q(frame->pts, inlink->time_base, outlink->time_base);
339 
340  out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);
341  if (!out_frame) {
342  s->input = NULL;
344  return AVERROR(ENOMEM);
345  }
346  av_frame_copy_props(out_frame, frame);
347 
348  for (i = 0; i < s->nb_delays; i++) {
349  ChanDelay *d = &s->chandelay[i];
350  const uint8_t *src = frame->extended_data[i];
351  uint8_t *dst = out_frame->extended_data[i];
352 
353  if (!d->delay)
354  memcpy(dst, src, frame->nb_samples * s->block_align);
355  else
356  s->delay_channel(d, frame->nb_samples, src, dst);
357  }
358 
359  out_frame->pts = s->next_pts + s->offset;
360  out_frame->duration = av_rescale_q(out_frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
361  s->next_pts += out_frame->duration;
363  s->input = NULL;
364  return ff_filter_frame(outlink, out_frame);
365 }
366 
368 {
369  AVFilterLink *inlink = ctx->inputs[0];
370  AVFilterLink *outlink = ctx->outputs[0];
371  AudioDelayContext *s = ctx->priv;
372  AVFrame *frame = NULL;
373  int ret, status;
374  int64_t pts;
375 
377 
378  if (!s->input) {
379  ret = ff_inlink_consume_frame(inlink, &s->input);
380  if (ret < 0)
381  return ret;
382  }
383 
385  if (status == AVERROR_EOF)
386  s->eof = 1;
387  }
388 
389  if (s->next_pts == AV_NOPTS_VALUE && pts != AV_NOPTS_VALUE)
390  s->next_pts = av_rescale_q(pts, inlink->time_base, outlink->time_base);
391 
392  if (s->padding) {
393  int nb_samples = FFMIN(s->padding, 2048);
394 
395  frame = ff_get_audio_buffer(outlink, nb_samples);
396  if (!frame)
397  return AVERROR(ENOMEM);
398  s->padding -= nb_samples;
399 
400  av_samples_set_silence(frame->extended_data, 0,
401  frame->nb_samples,
402  outlink->ch_layout.nb_channels,
403  frame->format);
404 
405  frame->duration = av_rescale_q(frame->nb_samples,
406  (AVRational){1, outlink->sample_rate},
407  outlink->time_base);
408  frame->pts = s->next_pts;
409  s->next_pts += frame->duration;
410 
411  return ff_filter_frame(outlink, frame);
412  }
413 
414  if (s->input)
415  return filter_frame(inlink, s->input);
416 
417  if (s->eof && s->max_delay) {
418  int nb_samples = FFMIN(s->max_delay, 2048);
419 
420  frame = ff_get_audio_buffer(outlink, nb_samples);
421  if (!frame)
422  return AVERROR(ENOMEM);
423  s->max_delay -= nb_samples;
424 
425  av_samples_set_silence(frame->extended_data, 0,
426  frame->nb_samples,
427  outlink->ch_layout.nb_channels,
428  frame->format);
429 
430  frame->duration = av_rescale_q(frame->nb_samples,
431  (AVRational){1, outlink->sample_rate},
432  outlink->time_base);
433  frame->pts = s->next_pts;
434  s->next_pts += frame->duration;
435  return filter_frame(inlink, frame);
436  }
437 
438  if (s->eof && s->max_delay == 0) {
439  ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
440  return 0;
441  }
442 
443  if (!s->eof)
445 
446  return FFERROR_NOT_READY;
447 }
448 
450 {
451  AudioDelayContext *s = ctx->priv;
452 
453  if (s->chandelay) {
454  for (int i = 0; i < s->nb_delays; i++)
455  av_freep(&s->chandelay[i].samples);
456  }
457  av_freep(&s->chandelay);
458 }
459 
460 static const AVFilterPad adelay_inputs[] = {
461  {
462  .name = "default",
463  .type = AVMEDIA_TYPE_AUDIO,
464  .config_props = config_input,
465  },
466 };
467 
469  .name = "adelay",
470  .description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),
471  .priv_size = sizeof(AudioDelayContext),
472  .priv_class = &adelay_class,
473  .activate = activate,
474  .uninit = uninit,
480  .process_command = process_command,
481 };
AudioDelayContext::next_pts
int64_t next_pts
Definition: af_adelay.c:47
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:98
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
AudioDelayContext::resize_channel_samples
int(* resize_channel_samples)(ChanDelay *d, int64_t new_delay)
Definition: af_adelay.c:54
AVERROR
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
opt.h
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1061
AVFrame::duration
int64_t duration
Duration of the frame, in the same units as pts.
Definition: frame.h:795
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
FFERROR_NOT_READY
return FFERROR_NOT_READY
Definition: filter_design.txt:204
int64_t
long long int64_t
Definition: coverity.c:34
inlink
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
Definition: filter_design.txt:212
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:162
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: filters.h:262
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:501
av_samples_set_silence
int av_samples_set_silence(uint8_t *const *audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)
Fill an audio buffer with silence.
Definition: samplefmt.c:246
AVOption
AVOption.
Definition: opt.h:429
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(adelay)
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:65
ChanDelay
Definition: af_adelay.c:29
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:205
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:327
ChanDelay::delay
int64_t delay
Definition: af_adelay.c:30
FF_FILTER_FORWARD_STATUS_BACK
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
Definition: filters.h:434
ff_af_adelay
const AVFilter ff_af_adelay
Definition: af_adelay.c:468
OFFSET
#define OFFSET(x)
Definition: af_adelay.c:57
ff_inlink_consume_frame
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
Definition: avfilter.c:1490
AudioDelayContext
Definition: af_adelay.c:37
samplefmt.h
type
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type
Definition: writing_filters.txt:86
pts
static int64_t pts
Definition: transcode_aac.c:644
AVFilterPad
A filter pad used for either input or output.
Definition: filters.h:38
activate
static int activate(AVFilterContext *ctx)
Definition: af_adelay.c:367
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_adelay.c:185
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:209
AudioDelayContext::delay_channel
void(* delay_channel)(ChanDelay *d, int nb_samples, const uint8_t *src, uint8_t *dst)
Definition: af_adelay.c:52
av_cold
#define av_cold
Definition: attributes.h:90
FILTER_SAMPLEFMTS
#define FILTER_SAMPLEFMTS(...)
Definition: filters.h:250
ff_outlink_set_status
static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)
Set the status field of a link from the source filter.
Definition: filters.h:424
s
#define s(width, name)
Definition: cbs_vp9.c:198
AudioDelayContext::delays
char * delays
Definition: af_adelay.c:40
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_strtok
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok().
Definition: avstring.c:178
filters.h
adelay_options
static const AVOption adelay_options[]
Definition: af_adelay.c:60
ctx
AVFormatContext * ctx
Definition: movenc.c:49
AudioDelayContext::nb_delays
int nb_delays
Definition: af_adelay.c:42
AudioDelayContext::padding
int64_t padding
Definition: af_adelay.c:44
av_rescale_q
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
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: filters.h:263
arg
const char * arg
Definition: jacosubdec.c:67
av_sscanf
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
Definition: avsscanf.c:961
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:75
result
and forward the result(frame or status change) to the corresponding input. If nothing is possible
NULL
#define NULL
Definition: coverity.c:32
av_frame_copy_props
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:725
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
parse_delays
static int parse_delays(char *p, char **saveptr, int64_t *result, AVFilterContext *ctx, int sample_rate)
Definition: af_adelay.c:159
A
#define A
Definition: af_adelay.c:58
ff_audio_default_filterpad
const AVFilterPad ff_audio_default_filterpad[1]
An AVFilterPad array whose only entry has name "default" and is of type AVMEDIA_TYPE_AUDIO.
Definition: audio.c:34
ff_inlink_acknowledge_status
int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)
Test and acknowledge the change of status on the link.
Definition: avfilter.c:1437
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:94
AV_SAMPLE_FMT_U8P
@ AV_SAMPLE_FMT_U8P
unsigned 8 bits, planar
Definition: samplefmt.h:63
DELAY
#define DELAY(name, type, fill)
Definition: af_adelay.c:68
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition: dsp.h:83
AudioDelayContext::block_align
int block_align
Definition: af_adelay.c:43
av_make_q
static AVRational av_make_q(int num, int den)
Create an AVRational.
Definition: rational.h:71
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
Definition: af_adelay.c:325
AudioDelayContext::input
AVFrame * input
Definition: af_adelay.c:50
FF_FILTER_FORWARD_WANTED
FF_FILTER_FORWARD_WANTED(outlink, inlink)
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:64
ChanDelay::index
size_t index
Definition: af_adelay.c:32
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:469
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
av_get_bytes_per_sample
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:108
AudioDelayContext::chandelay
ChanDelay * chandelay
Definition: af_adelay.c:41
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:450
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:31
ChanDelay::samples
uint8_t * samples
Definition: af_adelay.c:34
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AVFilterPad::name
const char * name
Pad name.
Definition: filters.h:44
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
CHANGE_DELAY
#define CHANGE_DELAY(name, type, fill)
Definition: af_adelay.c:103
AVFilter
Filter definition.
Definition: avfilter.h:201
ret
ret
Definition: filter_design.txt:187
frame
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 the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
status
ov_status_e status
Definition: dnn_backend_openvino.c:100
adelay_inputs
static const AVFilterPad adelay_inputs[]
Definition: af_adelay.c:460
AV_OPT_FLAG_RUNTIME_PARAM
#define AV_OPT_FLAG_RUNTIME_PARAM
A generic parameter which can be set by the user at runtime.
Definition: opt.h:377
avfilter.h
process_command
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_adelay.c:271
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:67
AudioDelayContext::eof
int eof
Definition: af_adelay.c:48
AVFilterContext
An instance of a filter.
Definition: avfilter.h:457
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:272
mem.h
audio.h
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Underlying C type is int.
Definition: opt.h:327
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
AudioDelayContext::all
int all
Definition: af_adelay.c:39
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_adelay.c:449
int32_t
int32_t
Definition: audioconvert.c:56
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
Definition: avfilter.h:190
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:482
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
ChanDelay::delay_index
size_t delay_index
Definition: af_adelay.c:31
ChanDelay::samples_size
unsigned int samples_size
Definition: af_adelay.c:33
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Underlying C type is a uint8_t* that is either NULL or points to a C string allocated with the av_mal...
Definition: opt.h:276
AudioDelayContext::max_delay
int64_t max_delay
Definition: af_adelay.c:45
src
#define src
Definition: vp8dsp.c:248
AudioDelayContext::offset
int64_t offset
Definition: af_adelay.c:46