FFmpeg  4.4.4
af_amix.c
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1 /*
2  * Audio Mix Filter
3  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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  * Audio Mix Filter
25  *
26  * Mixes audio from multiple sources into a single output. The channel layout,
27  * sample rate, and sample format will be the same for all inputs and the
28  * output.
29  */
30 
31 #include "libavutil/attributes.h"
32 #include "libavutil/audio_fifo.h"
33 #include "libavutil/avassert.h"
34 #include "libavutil/avstring.h"
36 #include "libavutil/common.h"
37 #include "libavutil/eval.h"
38 #include "libavutil/float_dsp.h"
39 #include "libavutil/mathematics.h"
40 #include "libavutil/opt.h"
41 #include "libavutil/samplefmt.h"
42 
43 #include "audio.h"
44 #include "avfilter.h"
45 #include "filters.h"
46 #include "formats.h"
47 #include "internal.h"
48 
49 #define INPUT_ON 1 /**< input is active */
50 #define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
51 
52 #define DURATION_LONGEST 0
53 #define DURATION_SHORTEST 1
54 #define DURATION_FIRST 2
55 
56 
57 typedef struct FrameInfo {
59  int64_t pts;
60  struct FrameInfo *next;
61 } FrameInfo;
62 
63 /**
64  * Linked list used to store timestamps and frame sizes of all frames in the
65  * FIFO for the first input.
66  *
67  * This is needed to keep timestamps synchronized for the case where multiple
68  * input frames are pushed to the filter for processing before a frame is
69  * requested by the output link.
70  */
71 typedef struct FrameList {
72  int nb_frames;
76 } FrameList;
77 
78 static void frame_list_clear(FrameList *frame_list)
79 {
80  if (frame_list) {
81  while (frame_list->list) {
82  FrameInfo *info = frame_list->list;
83  frame_list->list = info->next;
84  av_free(info);
85  }
86  frame_list->nb_frames = 0;
87  frame_list->nb_samples = 0;
88  frame_list->end = NULL;
89  }
90 }
91 
92 static int frame_list_next_frame_size(FrameList *frame_list)
93 {
94  if (!frame_list->list)
95  return 0;
96  return frame_list->list->nb_samples;
97 }
98 
99 static int64_t frame_list_next_pts(FrameList *frame_list)
100 {
101  if (!frame_list->list)
102  return AV_NOPTS_VALUE;
103  return frame_list->list->pts;
104 }
105 
106 static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
107 {
108  if (nb_samples >= frame_list->nb_samples) {
109  frame_list_clear(frame_list);
110  } else {
111  int samples = nb_samples;
112  while (samples > 0) {
113  FrameInfo *info = frame_list->list;
114  av_assert0(info);
115  if (info->nb_samples <= samples) {
116  samples -= info->nb_samples;
117  frame_list->list = info->next;
118  if (!frame_list->list)
119  frame_list->end = NULL;
120  frame_list->nb_frames--;
121  frame_list->nb_samples -= info->nb_samples;
122  av_free(info);
123  } else {
124  info->nb_samples -= samples;
125  info->pts += samples;
126  frame_list->nb_samples -= samples;
127  samples = 0;
128  }
129  }
130  }
131 }
132 
133 static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
134 {
135  FrameInfo *info = av_malloc(sizeof(*info));
136  if (!info)
137  return AVERROR(ENOMEM);
138  info->nb_samples = nb_samples;
139  info->pts = pts;
140  info->next = NULL;
141 
142  if (!frame_list->list) {
143  frame_list->list = info;
144  frame_list->end = info;
145  } else {
146  av_assert0(frame_list->end);
147  frame_list->end->next = info;
148  frame_list->end = info;
149  }
150  frame_list->nb_frames++;
151  frame_list->nb_samples += nb_samples;
152 
153  return 0;
154 }
155 
156 /* FIXME: use directly links fifo */
157 
158 typedef struct MixContext {
159  const AVClass *class; /**< class for AVOptions */
161 
162  int nb_inputs; /**< number of inputs */
163  int active_inputs; /**< number of input currently active */
164  int duration_mode; /**< mode for determining duration */
165  float dropout_transition; /**< transition time when an input drops out */
166  char *weights_str; /**< string for custom weights for every input */
167  int normalize; /**< if inputs are scaled */
168 
169  int nb_channels; /**< number of channels */
170  int sample_rate; /**< sample rate */
171  int planar;
172  AVAudioFifo **fifos; /**< audio fifo for each input */
173  uint8_t *input_state; /**< current state of each input */
174  float *input_scale; /**< mixing scale factor for each input */
175  float *weights; /**< custom weights for every input */
176  float weight_sum; /**< sum of custom weights for every input */
177  float *scale_norm; /**< normalization factor for every input */
178  int64_t next_pts; /**< calculated pts for next output frame */
179  FrameList *frame_list; /**< list of frame info for the first input */
180 } MixContext;
181 
182 #define OFFSET(x) offsetof(MixContext, x)
183 #define A AV_OPT_FLAG_AUDIO_PARAM
184 #define F AV_OPT_FLAG_FILTERING_PARAM
185 #define T AV_OPT_FLAG_RUNTIME_PARAM
186 static const AVOption amix_options[] = {
187  { "inputs", "Number of inputs.",
188  OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, INT16_MAX, A|F },
189  { "duration", "How to determine the end-of-stream.",
190  OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
191  { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, "duration" },
192  { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, "duration" },
193  { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, "duration" },
194  { "dropout_transition", "Transition time, in seconds, for volume "
195  "renormalization when an input stream ends.",
196  OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
197  { "weights", "Set weight for each input.",
198  OFFSET(weights_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, A|F|T },
199  { "normalize", "Scale inputs",
200  OFFSET(normalize), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A|F|T },
201  { NULL }
202 };
203 
205 
206 /**
207  * Update the scaling factors to apply to each input during mixing.
208  *
209  * This balances the full volume range between active inputs and handles
210  * volume transitions when EOF is encountered on an input but mixing continues
211  * with the remaining inputs.
212  */
213 static void calculate_scales(MixContext *s, int nb_samples)
214 {
215  float weight_sum = 0.f;
216  int i;
217 
218  for (i = 0; i < s->nb_inputs; i++)
219  if (s->input_state[i] & INPUT_ON)
220  weight_sum += FFABS(s->weights[i]);
221 
222  for (i = 0; i < s->nb_inputs; i++) {
223  if (s->input_state[i] & INPUT_ON) {
224  if (s->scale_norm[i] > weight_sum / FFABS(s->weights[i])) {
225  s->scale_norm[i] -= ((s->weight_sum / FFABS(s->weights[i])) / s->nb_inputs) *
226  nb_samples / (s->dropout_transition * s->sample_rate);
227  s->scale_norm[i] = FFMAX(s->scale_norm[i], weight_sum / FFABS(s->weights[i]));
228  }
229  }
230  }
231 
232  for (i = 0; i < s->nb_inputs; i++) {
233  if (s->input_state[i] & INPUT_ON) {
234  if (!s->normalize)
235  s->input_scale[i] = FFABS(s->weights[i]);
236  else
237  s->input_scale[i] = 1.0f / s->scale_norm[i] * FFSIGN(s->weights[i]);
238  } else {
239  s->input_scale[i] = 0.0f;
240  }
241  }
242 }
243 
244 static int config_output(AVFilterLink *outlink)
245 {
246  AVFilterContext *ctx = outlink->src;
247  MixContext *s = ctx->priv;
248  int i;
249  char buf[64];
250 
251  s->planar = av_sample_fmt_is_planar(outlink->format);
252  s->sample_rate = outlink->sample_rate;
253  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
254  s->next_pts = AV_NOPTS_VALUE;
255 
256  s->frame_list = av_mallocz(sizeof(*s->frame_list));
257  if (!s->frame_list)
258  return AVERROR(ENOMEM);
259 
260  s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
261  if (!s->fifos)
262  return AVERROR(ENOMEM);
263 
264  s->nb_channels = outlink->channels;
265  for (i = 0; i < s->nb_inputs; i++) {
266  s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
267  if (!s->fifos[i])
268  return AVERROR(ENOMEM);
269  }
270 
271  s->input_state = av_malloc(s->nb_inputs);
272  if (!s->input_state)
273  return AVERROR(ENOMEM);
274  memset(s->input_state, INPUT_ON, s->nb_inputs);
275  s->active_inputs = s->nb_inputs;
276 
277  s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
278  s->scale_norm = av_mallocz_array(s->nb_inputs, sizeof(*s->scale_norm));
279  if (!s->input_scale || !s->scale_norm)
280  return AVERROR(ENOMEM);
281  for (i = 0; i < s->nb_inputs; i++)
282  s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
283  calculate_scales(s, 0);
284 
285  av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
286 
288  "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
289  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
290 
291  return 0;
292 }
293 
294 /**
295  * Read samples from the input FIFOs, mix, and write to the output link.
296  */
297 static int output_frame(AVFilterLink *outlink)
298 {
299  AVFilterContext *ctx = outlink->src;
300  MixContext *s = ctx->priv;
301  AVFrame *out_buf, *in_buf;
302  int nb_samples, ns, i;
303 
304  if (s->input_state[0] & INPUT_ON) {
305  /* first input live: use the corresponding frame size */
306  nb_samples = frame_list_next_frame_size(s->frame_list);
307  for (i = 1; i < s->nb_inputs; i++) {
308  if (s->input_state[i] & INPUT_ON) {
309  ns = av_audio_fifo_size(s->fifos[i]);
310  if (ns < nb_samples) {
311  if (!(s->input_state[i] & INPUT_EOF))
312  /* unclosed input with not enough samples */
313  return 0;
314  /* closed input to drain */
315  nb_samples = ns;
316  }
317  }
318  }
319 
320  s->next_pts = frame_list_next_pts(s->frame_list);
321  } else {
322  /* first input closed: use the available samples */
323  nb_samples = INT_MAX;
324  for (i = 1; i < s->nb_inputs; i++) {
325  if (s->input_state[i] & INPUT_ON) {
326  ns = av_audio_fifo_size(s->fifos[i]);
327  nb_samples = FFMIN(nb_samples, ns);
328  }
329  }
330  if (nb_samples == INT_MAX) {
331  ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
332  return 0;
333  }
334  }
335 
336  frame_list_remove_samples(s->frame_list, nb_samples);
337 
338  calculate_scales(s, nb_samples);
339 
340  if (nb_samples == 0)
341  return 0;
342 
343  out_buf = ff_get_audio_buffer(outlink, nb_samples);
344  if (!out_buf)
345  return AVERROR(ENOMEM);
346 
347  in_buf = ff_get_audio_buffer(outlink, nb_samples);
348  if (!in_buf) {
349  av_frame_free(&out_buf);
350  return AVERROR(ENOMEM);
351  }
352 
353  for (i = 0; i < s->nb_inputs; i++) {
354  if (s->input_state[i] & INPUT_ON) {
355  int planes, plane_size, p;
356 
357  av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
358  nb_samples);
359 
360  planes = s->planar ? s->nb_channels : 1;
361  plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
362  plane_size = FFALIGN(plane_size, 16);
363 
364  if (out_buf->format == AV_SAMPLE_FMT_FLT ||
365  out_buf->format == AV_SAMPLE_FMT_FLTP) {
366  for (p = 0; p < planes; p++) {
367  s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
368  (float *) in_buf->extended_data[p],
369  s->input_scale[i], plane_size);
370  }
371  } else {
372  for (p = 0; p < planes; p++) {
373  s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
374  (double *) in_buf->extended_data[p],
375  s->input_scale[i], plane_size);
376  }
377  }
378  }
379  }
380  av_frame_free(&in_buf);
381 
382  out_buf->pts = s->next_pts;
383  if (s->next_pts != AV_NOPTS_VALUE)
384  s->next_pts += nb_samples;
385 
386  return ff_filter_frame(outlink, out_buf);
387 }
388 
389 /**
390  * Requests a frame, if needed, from each input link other than the first.
391  */
392 static int request_samples(AVFilterContext *ctx, int min_samples)
393 {
394  MixContext *s = ctx->priv;
395  int i;
396 
397  av_assert0(s->nb_inputs > 1);
398 
399  for (i = 1; i < s->nb_inputs; i++) {
400  if (!(s->input_state[i] & INPUT_ON) ||
401  (s->input_state[i] & INPUT_EOF))
402  continue;
403  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
404  continue;
405  ff_inlink_request_frame(ctx->inputs[i]);
406  }
407  return output_frame(ctx->outputs[0]);
408 }
409 
410 /**
411  * Calculates the number of active inputs and determines EOF based on the
412  * duration option.
413  *
414  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
415  */
417 {
418  int i;
419  int active_inputs = 0;
420  for (i = 0; i < s->nb_inputs; i++)
421  active_inputs += !!(s->input_state[i] & INPUT_ON);
422  s->active_inputs = active_inputs;
423 
424  if (!active_inputs ||
425  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
426  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
427  return AVERROR_EOF;
428  return 0;
429 }
430 
432 {
433  AVFilterLink *outlink = ctx->outputs[0];
434  MixContext *s = ctx->priv;
435  AVFrame *buf = NULL;
436  int i, ret;
437 
439 
440  for (i = 0; i < s->nb_inputs; i++) {
441  AVFilterLink *inlink = ctx->inputs[i];
442 
443  if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
444  if (i == 0) {
445  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
446  outlink->time_base);
447  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
448  if (ret < 0) {
449  av_frame_free(&buf);
450  return ret;
451  }
452  }
453 
454  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
455  buf->nb_samples);
456  if (ret < 0) {
457  av_frame_free(&buf);
458  return ret;
459  }
460 
461  av_frame_free(&buf);
462 
463  ret = output_frame(outlink);
464  if (ret < 0)
465  return ret;
466  }
467  }
468 
469  for (i = 0; i < s->nb_inputs; i++) {
470  int64_t pts;
471  int status;
472 
473  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
474  if (status == AVERROR_EOF) {
475  if (i == 0) {
476  s->input_state[i] = 0;
477  if (s->nb_inputs == 1) {
478  ff_outlink_set_status(outlink, status, pts);
479  return 0;
480  }
481  } else {
482  s->input_state[i] |= INPUT_EOF;
483  if (av_audio_fifo_size(s->fifos[i]) == 0) {
484  s->input_state[i] = 0;
485  }
486  }
487  }
488  }
489  }
490 
491  if (calc_active_inputs(s)) {
492  ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
493  return 0;
494  }
495 
496  if (ff_outlink_frame_wanted(outlink)) {
497  int wanted_samples;
498 
499  if (!(s->input_state[0] & INPUT_ON))
500  return request_samples(ctx, 1);
501 
502  if (s->frame_list->nb_frames == 0) {
503  ff_inlink_request_frame(ctx->inputs[0]);
504  return 0;
505  }
506  av_assert0(s->frame_list->nb_frames > 0);
507 
508  wanted_samples = frame_list_next_frame_size(s->frame_list);
509 
510  return request_samples(ctx, wanted_samples);
511  }
512 
513  return 0;
514 }
515 
517 {
518  MixContext *s = ctx->priv;
519  float last_weight = 1.f;
520  char *p;
521  int i;
522 
523  s->weight_sum = 0.f;
524  p = s->weights_str;
525  for (i = 0; i < s->nb_inputs; i++) {
526  last_weight = av_strtod(p, &p);
527  s->weights[i] = last_weight;
528  s->weight_sum += FFABS(last_weight);
529  if (p && *p) {
530  p++;
531  } else {
532  i++;
533  break;
534  }
535  }
536 
537  for (; i < s->nb_inputs; i++) {
538  s->weights[i] = last_weight;
539  s->weight_sum += FFABS(last_weight);
540  }
541 }
542 
544 {
545  MixContext *s = ctx->priv;
546  int i, ret;
547 
548  for (i = 0; i < s->nb_inputs; i++) {
549  AVFilterPad pad = { 0 };
550 
551  pad.type = AVMEDIA_TYPE_AUDIO;
552  pad.name = av_asprintf("input%d", i);
553  if (!pad.name)
554  return AVERROR(ENOMEM);
555 
556  if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
557  av_freep(&pad.name);
558  return ret;
559  }
560  }
561 
562  s->fdsp = avpriv_float_dsp_alloc(0);
563  if (!s->fdsp)
564  return AVERROR(ENOMEM);
565 
566  s->weights = av_mallocz_array(s->nb_inputs, sizeof(*s->weights));
567  if (!s->weights)
568  return AVERROR(ENOMEM);
569 
571 
572  return 0;
573 }
574 
576 {
577  int i;
578  MixContext *s = ctx->priv;
579 
580  if (s->fifos) {
581  for (i = 0; i < s->nb_inputs; i++)
582  av_audio_fifo_free(s->fifos[i]);
583  av_freep(&s->fifos);
584  }
585  frame_list_clear(s->frame_list);
586  av_freep(&s->frame_list);
587  av_freep(&s->input_state);
588  av_freep(&s->input_scale);
589  av_freep(&s->scale_norm);
590  av_freep(&s->weights);
591  av_freep(&s->fdsp);
592 
593  for (i = 0; i < ctx->nb_inputs; i++)
594  av_freep(&ctx->input_pads[i].name);
595 }
596 
598 {
599  static const enum AVSampleFormat sample_fmts[] = {
603  };
604  int ret;
605 
608  return ret;
609 
611 }
612 
613 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
614  char *res, int res_len, int flags)
615 {
616  MixContext *s = ctx->priv;
617  int ret;
618 
619  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
620  if (ret < 0)
621  return ret;
622 
624  for (int i = 0; i < s->nb_inputs; i++)
625  s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
626  calculate_scales(s, 0);
627 
628  return 0;
629 }
630 
632  {
633  .name = "default",
634  .type = AVMEDIA_TYPE_AUDIO,
635  .config_props = config_output,
636  },
637  { NULL }
638 };
639 
641  .name = "amix",
642  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
643  .priv_size = sizeof(MixContext),
644  .priv_class = &amix_class,
645  .init = init,
646  .uninit = uninit,
647  .activate = activate,
649  .inputs = NULL,
653 };
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
#define T
Definition: af_amix.c:185
static int output_frame(AVFilterLink *outlink)
Read samples from the input FIFOs, mix, and write to the output link.
Definition: af_amix.c:297
static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
Definition: af_amix.c:133
#define F
Definition: af_amix.c:184
static void calculate_scales(MixContext *s, int nb_samples)
Update the scaling factors to apply to each input during mixing.
Definition: af_amix.c:213
static int64_t frame_list_next_pts(FrameList *frame_list)
Definition: af_amix.c:99
static void parse_weights(AVFilterContext *ctx)
Definition: af_amix.c:516
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:597
#define DURATION_SHORTEST
Definition: af_amix.c:53
#define DURATION_FIRST
Definition: af_amix.c:54
#define A
Definition: af_amix.c:183
static int calc_active_inputs(MixContext *s)
Calculates the number of active inputs and determines EOF based on the duration option.
Definition: af_amix.c:416
AVFilter ff_af_amix
Definition: af_amix.c:640
#define DURATION_LONGEST
Definition: af_amix.c:52
static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
Definition: af_amix.c:106
#define INPUT_EOF
input has reached EOF (may still be active)
Definition: af_amix.c:50
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_amix.c:613
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:431
static av_cold int init(AVFilterContext *ctx)
Definition: af_amix.c:543
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:575
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:631
static int frame_list_next_frame_size(FrameList *frame_list)
Definition: af_amix.c:92
static void frame_list_clear(FrameList *frame_list)
Definition: af_amix.c:78
#define OFFSET(x)
Definition: af_amix.c:182
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:244
AVFILTER_DEFINE_CLASS(amix)
static int request_samples(AVFilterContext *ctx, int min_samples)
Requests a frame, if needed, from each input link other than the first.
Definition: af_amix.c:392
static const AVOption amix_options[]
Definition: af_amix.c:186
#define INPUT_ON
input is active
Definition: af_amix.c:49
Macro definitions for various function/variable attributes.
#define av_cold
Definition: attributes.h:88
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
uint8_t
Audio FIFO Buffer.
simple assert() macros that are a bit more flexible than ISO C assert().
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
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:1449
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1096
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Definition: avfilter.c:882
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:1494
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1620
Main libavfilter public API header.
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
#define flags(name, subs,...)
Definition: cbs_av1.c:561
#define ns(max_value, name, subs,...)
Definition: cbs_av1.c:682
#define s(width, name)
Definition: cbs_vp9.c:257
audio channel layout utility functions
common internal and external API header
#define FFMIN(a, b)
Definition: common.h:105
#define FFMAX(a, b)
Definition: common.h:103
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define FFSIGN(a)
Definition: common.h:73
#define NULL
Definition: coverity.c:32
double av_strtod(const char *numstr, char **tail)
Parse the string in numstr and return its value as a double.
Definition: eval.c:106
simple arithmetic expression evaluator
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:189
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:436
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:587
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:286
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:568
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_INT
Definition: opt.h:225
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:228
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
@ AV_OPT_TYPE_STRING
Definition: opt.h:229
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
Definition: avfilter.h:106
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
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_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:210
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
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
Definition: mem.c:190
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
const char * av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
Return the name of sample_fmt, or NULL if sample_fmt is not recognized.
Definition: samplefmt.c:49
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
int i
Definition: input.c:407
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
Definition: internal.h:240
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
static const struct @322 planes[]
#define FFALIGN(x, a)
Definition: macros.h:48
AVOptions.
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
Describe the class of an AVClass context structure.
Definition: log.h:67
An instance of a filter.
Definition: avfilter.h:341
A filter pad used for either input or output.
Definition: internal.h:54
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
const char * name
Pad name.
Definition: internal.h:60
Filter definition.
Definition: avfilter.h:145
const char * name
Filter name.
Definition: avfilter.h:149
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:411
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
Definition: frame.h:391
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
AVOption.
Definition: opt.h:248
Rational number (pair of numerator and denominator).
Definition: rational.h:58
int64_t pts
Definition: af_amix.c:59
int nb_samples
Definition: af_amix.c:58
struct FrameInfo * next
Definition: af_amix.c:60
Linked list used to store timestamps and frame sizes of all frames in the FIFO for the first input.
Definition: af_amix.c:71
int nb_samples
Definition: af_amix.c:73
FrameInfo * list
Definition: af_amix.c:74
FrameInfo * end
Definition: af_amix.c:75
int nb_frames
Definition: af_amix.c:72
int nb_inputs
number of inputs
Definition: af_amix.c:162
int64_t next_pts
calculated pts for next output frame
Definition: af_amix.c:178
int active_inputs
number of input currently active
Definition: af_amix.c:163
int nb_channels
number of channels
Definition: af_amix.c:169
int normalize
if inputs are scaled
Definition: af_amix.c:167
uint8_t * input_state
current state of each input
Definition: af_amix.c:173
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:172
int planar
Definition: af_amix.c:171
float * scale_norm
normalization factor for every input
Definition: af_amix.c:177
int duration_mode
mode for determining duration
Definition: af_amix.c:164
char * weights_str
string for custom weights for every input
Definition: af_amix.c:166
float * weights
custom weights for every input
Definition: af_amix.c:175
AVFloatDSPContext * fdsp
Definition: af_amix.c:160
float * input_scale
mixing scale factor for each input
Definition: af_amix.c:174
FrameList * frame_list
list of frame info for the first input
Definition: af_amix.c:179
int sample_rate
sample rate
Definition: af_amix.c:170
float weight_sum
sum of custom weights for every input
Definition: af_amix.c:176
float dropout_transition
transition time when an input drops out
Definition: af_amix.c:165
#define av_free(p)
#define av_freep(p)
#define av_malloc(s)
#define av_log(a,...)
AVFormatContext * ctx
Definition: movenc.c:48
static int64_t pts