FFmpeg  4.3.8
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 {
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 
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 
168  int nb_channels; /**< number of channels */
169  int sample_rate; /**< sample rate */
170  int planar;
171  AVAudioFifo **fifos; /**< audio fifo for each input */
172  uint8_t *input_state; /**< current state of each input */
173  float *input_scale; /**< mixing scale factor for each input */
174  float *weights; /**< custom weights for every input */
175  float weight_sum; /**< sum of custom weights for every input */
176  float *scale_norm; /**< normalization factor for every input */
177  int64_t next_pts; /**< calculated pts for next output frame */
178  FrameList *frame_list; /**< list of frame info for the first input */
179 } MixContext;
180 
181 #define OFFSET(x) offsetof(MixContext, x)
182 #define A AV_OPT_FLAG_AUDIO_PARAM
183 #define F AV_OPT_FLAG_FILTERING_PARAM
184 #define T AV_OPT_FLAG_RUNTIME_PARAM
185 static const AVOption amix_options[] = {
186  { "inputs", "Number of inputs.",
187  OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, INT16_MAX, A|F },
188  { "duration", "How to determine the end-of-stream.",
189  OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
190  { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, "duration" },
191  { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, "duration" },
192  { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, "duration" },
193  { "dropout_transition", "Transition time, in seconds, for volume "
194  "renormalization when an input stream ends.",
195  OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
196  { "weights", "Set weight for each input.",
197  OFFSET(weights_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, A|F|T },
198  { NULL }
199 };
200 
202 
203 /**
204  * Update the scaling factors to apply to each input during mixing.
205  *
206  * This balances the full volume range between active inputs and handles
207  * volume transitions when EOF is encountered on an input but mixing continues
208  * with the remaining inputs.
209  */
211 {
212  float weight_sum = 0.f;
213  int i;
214 
215  for (i = 0; i < s->nb_inputs; i++)
216  if (s->input_state[i] & INPUT_ON)
217  weight_sum += FFABS(s->weights[i]);
218 
219  for (i = 0; i < s->nb_inputs; i++) {
220  if (s->input_state[i] & INPUT_ON) {
221  if (s->scale_norm[i] > weight_sum / FFABS(s->weights[i])) {
222  s->scale_norm[i] -= ((s->weight_sum / FFABS(s->weights[i])) / s->nb_inputs) *
223  nb_samples / (s->dropout_transition * s->sample_rate);
224  s->scale_norm[i] = FFMAX(s->scale_norm[i], weight_sum / FFABS(s->weights[i]));
225  }
226  }
227  }
228 
229  for (i = 0; i < s->nb_inputs; i++) {
230  if (s->input_state[i] & INPUT_ON)
231  s->input_scale[i] = 1.0f / s->scale_norm[i] * FFSIGN(s->weights[i]);
232  else
233  s->input_scale[i] = 0.0f;
234  }
235 }
236 
237 static int config_output(AVFilterLink *outlink)
238 {
239  AVFilterContext *ctx = outlink->src;
240  MixContext *s = ctx->priv;
241  int i;
242  char buf[64];
243 
244  s->planar = av_sample_fmt_is_planar(outlink->format);
245  s->sample_rate = outlink->sample_rate;
246  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
248 
249  s->frame_list = av_mallocz(sizeof(*s->frame_list));
250  if (!s->frame_list)
251  return AVERROR(ENOMEM);
252 
253  s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
254  if (!s->fifos)
255  return AVERROR(ENOMEM);
256 
257  s->nb_channels = outlink->channels;
258  for (i = 0; i < s->nb_inputs; i++) {
259  s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
260  if (!s->fifos[i])
261  return AVERROR(ENOMEM);
262  }
263 
265  if (!s->input_state)
266  return AVERROR(ENOMEM);
267  memset(s->input_state, INPUT_ON, s->nb_inputs);
268  s->active_inputs = s->nb_inputs;
269 
270  s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
271  s->scale_norm = av_mallocz_array(s->nb_inputs, sizeof(*s->scale_norm));
272  if (!s->input_scale || !s->scale_norm)
273  return AVERROR(ENOMEM);
274  for (i = 0; i < s->nb_inputs; i++)
275  s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
276  calculate_scales(s, 0);
277 
278  av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
279 
280  av_log(ctx, AV_LOG_VERBOSE,
281  "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
282  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
283 
284  return 0;
285 }
286 
287 /**
288  * Read samples from the input FIFOs, mix, and write to the output link.
289  */
290 static int output_frame(AVFilterLink *outlink)
291 {
292  AVFilterContext *ctx = outlink->src;
293  MixContext *s = ctx->priv;
294  AVFrame *out_buf, *in_buf;
295  int nb_samples, ns, i;
296 
297  if (s->input_state[0] & INPUT_ON) {
298  /* first input live: use the corresponding frame size */
299  nb_samples = frame_list_next_frame_size(s->frame_list);
300  for (i = 1; i < s->nb_inputs; i++) {
301  if (s->input_state[i] & INPUT_ON) {
302  ns = av_audio_fifo_size(s->fifos[i]);
303  if (ns < nb_samples) {
304  if (!(s->input_state[i] & INPUT_EOF))
305  /* unclosed input with not enough samples */
306  return 0;
307  /* closed input to drain */
308  nb_samples = ns;
309  }
310  }
311  }
312  } else {
313  /* first input closed: use the available samples */
314  nb_samples = INT_MAX;
315  for (i = 1; i < s->nb_inputs; i++) {
316  if (s->input_state[i] & INPUT_ON) {
317  ns = av_audio_fifo_size(s->fifos[i]);
318  nb_samples = FFMIN(nb_samples, ns);
319  }
320  }
321  if (nb_samples == INT_MAX) {
323  return 0;
324  }
325  }
326 
328  frame_list_remove_samples(s->frame_list, nb_samples);
329 
330  calculate_scales(s, nb_samples);
331 
332  if (nb_samples == 0)
333  return 0;
334 
335  out_buf = ff_get_audio_buffer(outlink, nb_samples);
336  if (!out_buf)
337  return AVERROR(ENOMEM);
338 
339  in_buf = ff_get_audio_buffer(outlink, nb_samples);
340  if (!in_buf) {
341  av_frame_free(&out_buf);
342  return AVERROR(ENOMEM);
343  }
344 
345  for (i = 0; i < s->nb_inputs; i++) {
346  if (s->input_state[i] & INPUT_ON) {
347  int planes, plane_size, p;
348 
349  av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
350  nb_samples);
351 
352  planes = s->planar ? s->nb_channels : 1;
353  plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
354  plane_size = FFALIGN(plane_size, 16);
355 
356  if (out_buf->format == AV_SAMPLE_FMT_FLT ||
357  out_buf->format == AV_SAMPLE_FMT_FLTP) {
358  for (p = 0; p < planes; p++) {
359  s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
360  (float *) in_buf->extended_data[p],
361  s->input_scale[i], plane_size);
362  }
363  } else {
364  for (p = 0; p < planes; p++) {
365  s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
366  (double *) in_buf->extended_data[p],
367  s->input_scale[i], plane_size);
368  }
369  }
370  }
371  }
372  av_frame_free(&in_buf);
373 
374  out_buf->pts = s->next_pts;
375  if (s->next_pts != AV_NOPTS_VALUE)
376  s->next_pts += nb_samples;
377 
378  return ff_filter_frame(outlink, out_buf);
379 }
380 
381 /**
382  * Requests a frame, if needed, from each input link other than the first.
383  */
384 static int request_samples(AVFilterContext *ctx, int min_samples)
385 {
386  MixContext *s = ctx->priv;
387  int i;
388 
389  av_assert0(s->nb_inputs > 1);
390 
391  for (i = 1; i < s->nb_inputs; i++) {
392  if (!(s->input_state[i] & INPUT_ON) ||
393  (s->input_state[i] & INPUT_EOF))
394  continue;
395  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
396  continue;
398  }
399  return output_frame(ctx->outputs[0]);
400 }
401 
402 /**
403  * Calculates the number of active inputs and determines EOF based on the
404  * duration option.
405  *
406  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
407  */
409 {
410  int i;
411  int active_inputs = 0;
412  for (i = 0; i < s->nb_inputs; i++)
413  active_inputs += !!(s->input_state[i] & INPUT_ON);
414  s->active_inputs = active_inputs;
415 
416  if (!active_inputs ||
417  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
418  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
419  return AVERROR_EOF;
420  return 0;
421 }
422 
424 {
425  AVFilterLink *outlink = ctx->outputs[0];
426  MixContext *s = ctx->priv;
427  AVFrame *buf = NULL;
428  int i, ret;
429 
430  FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
431 
432  for (i = 0; i < s->nb_inputs; i++) {
433  AVFilterLink *inlink = ctx->inputs[i];
434 
435  if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
436  if (i == 0) {
437  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
438  outlink->time_base);
439  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
440  if (ret < 0) {
441  av_frame_free(&buf);
442  return ret;
443  }
444  }
445 
446  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
447  buf->nb_samples);
448  if (ret < 0) {
449  av_frame_free(&buf);
450  return ret;
451  }
452 
453  av_frame_free(&buf);
454 
455  ret = output_frame(outlink);
456  if (ret < 0)
457  return ret;
458  }
459  }
460 
461  for (i = 0; i < s->nb_inputs; i++) {
462  int64_t pts;
463  int status;
464 
465  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
466  if (status == AVERROR_EOF) {
467  if (i == 0) {
468  s->input_state[i] = 0;
469  if (s->nb_inputs == 1) {
470  ff_outlink_set_status(outlink, status, pts);
471  return 0;
472  }
473  } else {
474  s->input_state[i] |= INPUT_EOF;
475  if (av_audio_fifo_size(s->fifos[i]) == 0) {
476  s->input_state[i] = 0;
477  }
478  }
479  }
480  }
481  }
482 
483  if (calc_active_inputs(s)) {
485  return 0;
486  }
487 
488  if (ff_outlink_frame_wanted(outlink)) {
489  int wanted_samples;
490 
491  if (!(s->input_state[0] & INPUT_ON))
492  return request_samples(ctx, 1);
493 
494  if (s->frame_list->nb_frames == 0) {
496  return 0;
497  }
499 
500  wanted_samples = frame_list_next_frame_size(s->frame_list);
501 
502  return request_samples(ctx, wanted_samples);
503  }
504 
505  return 0;
506 }
507 
509 {
510  MixContext *s = ctx->priv;
511  float last_weight = 1.f;
512  char *p;
513  int i;
514 
515  s->weight_sum = 0.f;
516  p = s->weights_str;
517  for (i = 0; i < s->nb_inputs; i++) {
518  last_weight = av_strtod(p, &p);
519  s->weights[i] = last_weight;
520  s->weight_sum += FFABS(last_weight);
521  if (p && *p) {
522  p++;
523  } else {
524  i++;
525  break;
526  }
527  }
528 
529  for (; i < s->nb_inputs; i++) {
530  s->weights[i] = last_weight;
531  s->weight_sum += FFABS(last_weight);
532  }
533 }
534 
536 {
537  MixContext *s = ctx->priv;
538  int i, ret;
539 
540  for (i = 0; i < s->nb_inputs; i++) {
541  AVFilterPad pad = { 0 };
542 
543  pad.type = AVMEDIA_TYPE_AUDIO;
544  pad.name = av_asprintf("input%d", i);
545  if (!pad.name)
546  return AVERROR(ENOMEM);
547 
548  if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
549  av_freep(&pad.name);
550  return ret;
551  }
552  }
553 
555  if (!s->fdsp)
556  return AVERROR(ENOMEM);
557 
558  s->weights = av_mallocz_array(s->nb_inputs, sizeof(*s->weights));
559  if (!s->weights)
560  return AVERROR(ENOMEM);
561 
562  parse_weights(ctx);
563 
564  return 0;
565 }
566 
568 {
569  int i;
570  MixContext *s = ctx->priv;
571 
572  if (s->fifos) {
573  for (i = 0; i < s->nb_inputs; i++)
574  av_audio_fifo_free(s->fifos[i]);
575  av_freep(&s->fifos);
576  }
578  av_freep(&s->frame_list);
579  av_freep(&s->input_state);
580  av_freep(&s->input_scale);
581  av_freep(&s->scale_norm);
582  av_freep(&s->weights);
583  av_freep(&s->fdsp);
584 
585  for (i = 0; i < ctx->nb_inputs; i++)
586  av_freep(&ctx->input_pads[i].name);
587 }
588 
590 {
591  static const enum AVSampleFormat sample_fmts[] = {
595  };
596  int ret;
597 
598  if ((ret = ff_set_common_formats(ctx, ff_make_format_list(sample_fmts))) < 0 ||
599  (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
600  return ret;
601 
603 }
604 
605 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
606  char *res, int res_len, int flags)
607 {
608  MixContext *s = ctx->priv;
609  int ret;
610 
611  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
612  if (ret < 0)
613  return ret;
614 
615  parse_weights(ctx);
616  for (int i = 0; i < s->nb_inputs; i++)
617  s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
618  calculate_scales(s, 0);
619 
620  return 0;
621 }
622 
624  {
625  .name = "default",
626  .type = AVMEDIA_TYPE_AUDIO,
627  .config_props = config_output,
628  },
629  { NULL }
630 };
631 
633  .name = "amix",
634  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
635  .priv_size = sizeof(MixContext),
636  .priv_class = &amix_class,
637  .init = init,
638  .uninit = uninit,
639  .activate = activate,
641  .inputs = NULL,
642  .outputs = avfilter_af_amix_outputs,
645 };
float, planar
Definition: samplefmt.h:69
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link&#39;s FIFO and update the link&#39;s stats.
Definition: avfilter.c:1476
#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:586
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
#define DURATION_LONGEST
Definition: af_amix.c:52
AVOption.
Definition: opt.h:246
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
Main libavfilter public API header.
#define A
Definition: af_amix.c:182
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:567
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
Definition: avfilter.h:105
double, planar
Definition: samplefmt.h:70
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
static int frame_list_next_frame_size(FrameList *frame_list)
Definition: af_amix.c:92
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
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1602
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:172
Macro definitions for various function/variable attributes.
Linked list used to store timestamps and frame sizes of all frames in the FIFO for the first input...
Definition: af_amix.c:71
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:300
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
void(* vector_fmac_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float and add to destination vector.
Definition: float_dsp.h:54
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1075
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:423
uint8_t
#define av_cold
Definition: attributes.h:88
#define av_malloc(s)
#define T
Definition: af_amix.c:184
static int64_t frame_list_next_pts(FrameList *frame_list)
Definition: af_amix.c:99
AVOptions.
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:384
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:393
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:623
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:408
int sample_rate
sample rate
Definition: af_amix.c:169
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:589
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
float dropout_transition
transition time when an input drops out
Definition: af_amix.c:165
FrameList * frame_list
list of frame info for the first input
Definition: af_amix.c:178
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
float * input_scale
mixing scale factor for each input
Definition: af_amix.c:173
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
int nb_samples
Definition: af_amix.c:73
A filter pad used for either input or output.
Definition: internal.h:54
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
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:1431
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:171
void(* vector_dmac_scalar)(double *dst, const double *src, double mul, int len)
Multiply a vector of doubles by a scalar double and add to destination vector.
Definition: float_dsp.h:70
AVFilterPad * input_pads
array of input pads
Definition: avfilter.h:345
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
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:605
#define OFFSET(x)
Definition: af_amix.c:181
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
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
int64_t pts
Definition: af_amix.c:59
#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 NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:188
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:869
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().
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
static void parse_weights(AVFilterContext *ctx)
Definition: af_amix.c:508
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
#define FFMAX(a, b)
Definition: common.h:94
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
int active_inputs
number of input currently active
Definition: af_amix.c:163
static const struct @315 planes[]
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
audio channel layout utility functions
unsigned nb_inputs
number of input pads
Definition: avfilter.h:347
#define FFMIN(a, b)
Definition: common.h:96
struct FrameInfo * next
Definition: af_amix.c:60
int nb_samples
Definition: af_amix.c:58
#define FFSIGN(a)
Definition: common.h:73
AVFormatContext * ctx
Definition: movenc.c:48
static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
Definition: af_amix.c:106
int planar
Definition: af_amix.c:170
int duration_mode
mode for determining duration
Definition: af_amix.c:164
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define s(width, name)
Definition: cbs_vp9.c:257
float weight_sum
sum of custom weights for every input
Definition: af_amix.c:175
int nb_channels
number of channels
Definition: af_amix.c:168
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
Definition: af_amix.c:133
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.
double av_strtod(const char *numstr, char **tail)
Parse the string in numstr and return its value as a double.
Definition: eval.c:106
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:373
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t * input_state
current state of each input
Definition: af_amix.c:172
long long int64_t
Definition: coverity.c:34
float * scale_norm
normalization factor for every input
Definition: af_amix.c:176
char * weights_str
string for custom weights for every input
Definition: af_amix.c:166
FrameInfo * list
Definition: af_amix.c:74
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
const char * name
Filter name.
Definition: avfilter.h:148
static int output_frame(AVFilterLink *outlink)
Read samples from the input FIFOs, mix, and write to the output link.
Definition: af_amix.c:290
#define INPUT_ON
input is active
Definition: af_amix.c:49
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
int64_t next_pts
calculated pts for next output frame
Definition: af_amix.c:177
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:439
#define DURATION_SHORTEST
Definition: af_amix.c:53
#define flags(name, subs,...)
Definition: cbs_av1.c:576
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
#define ns(max_value, name, subs,...)
Definition: cbs_av1.c:697
#define INPUT_EOF
input has reached EOF (may still be active)
Definition: af_amix.c:50
static void calculate_scales(MixContext *s, int nb_samples)
Update the scaling factors to apply to each input during mixing.
Definition: af_amix.c:210
FrameInfo * end
Definition: af_amix.c:75
common internal and external API header
AVFILTER_DEFINE_CLASS(amix)
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_amix.c:605
int nb_frames
Definition: af_amix.c:72
static const AVOption amix_options[]
Definition: af_amix.c:185
float * weights
custom weights for every input
Definition: af_amix.c:174
#define av_free(p)
Audio FIFO Buffer.
#define F
Definition: af_amix.c:183
#define DURATION_FIRST
Definition: af_amix.c:54
int nb_inputs
number of inputs
Definition: af_amix.c:162
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:731
static av_cold int init(AVFilterContext *ctx)
Definition: af_amix.c:535
#define av_freep(p)
AVFilter ff_af_amix
Definition: af_amix.c:632
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:237
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:454
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:347
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:593
static void frame_list_clear(FrameList *frame_list)
Definition: af_amix.c:78
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
simple arithmetic expression evaluator
void * av_mallocz_array(size_t nmemb, size_t size)
Allocate a memory block for an array with av_mallocz().
Definition: mem.c:190
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
Definition: internal.h:266
AVFloatDSPContext * fdsp
Definition: af_amix.c:160