FFmpeg
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/float_dsp.h"
38 #include "libavutil/mathematics.h"
39 #include "libavutil/opt.h"
40 #include "libavutil/samplefmt.h"
41 
42 #include "audio.h"
43 #include "avfilter.h"
44 #include "filters.h"
45 #include "formats.h"
46 #include "internal.h"
47 
48 #define INPUT_ON 1 /**< input is active */
49 #define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
50 
51 #define DURATION_LONGEST 0
52 #define DURATION_SHORTEST 1
53 #define DURATION_FIRST 2
54 
55 
56 typedef struct FrameInfo {
58  int64_t pts;
59  struct FrameInfo *next;
60 } FrameInfo;
61 
62 /**
63  * Linked list used to store timestamps and frame sizes of all frames in the
64  * FIFO for the first input.
65  *
66  * This is needed to keep timestamps synchronized for the case where multiple
67  * input frames are pushed to the filter for processing before a frame is
68  * requested by the output link.
69  */
70 typedef struct FrameList {
71  int nb_frames;
75 } FrameList;
76 
77 static void frame_list_clear(FrameList *frame_list)
78 {
79  if (frame_list) {
80  while (frame_list->list) {
81  FrameInfo *info = frame_list->list;
82  frame_list->list = info->next;
83  av_free(info);
84  }
85  frame_list->nb_frames = 0;
86  frame_list->nb_samples = 0;
87  frame_list->end = NULL;
88  }
89 }
90 
91 static int frame_list_next_frame_size(FrameList *frame_list)
92 {
93  if (!frame_list->list)
94  return 0;
95  return frame_list->list->nb_samples;
96 }
97 
98 static int64_t frame_list_next_pts(FrameList *frame_list)
99 {
100  if (!frame_list->list)
101  return AV_NOPTS_VALUE;
102  return frame_list->list->pts;
103 }
104 
105 static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
106 {
107  if (nb_samples >= frame_list->nb_samples) {
108  frame_list_clear(frame_list);
109  } else {
110  int samples = nb_samples;
111  while (samples > 0) {
112  FrameInfo *info = frame_list->list;
113  av_assert0(info);
114  if (info->nb_samples <= samples) {
115  samples -= info->nb_samples;
116  frame_list->list = info->next;
117  if (!frame_list->list)
118  frame_list->end = NULL;
119  frame_list->nb_frames--;
120  frame_list->nb_samples -= info->nb_samples;
121  av_free(info);
122  } else {
123  info->nb_samples -= samples;
124  info->pts += samples;
125  frame_list->nb_samples -= samples;
126  samples = 0;
127  }
128  }
129  }
130 }
131 
132 static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
133 {
134  FrameInfo *info = av_malloc(sizeof(*info));
135  if (!info)
136  return AVERROR(ENOMEM);
137  info->nb_samples = nb_samples;
138  info->pts = pts;
139  info->next = NULL;
140 
141  if (!frame_list->list) {
142  frame_list->list = info;
143  frame_list->end = info;
144  } else {
145  av_assert0(frame_list->end);
146  frame_list->end->next = info;
147  frame_list->end = info;
148  }
149  frame_list->nb_frames++;
150  frame_list->nb_samples += nb_samples;
151 
152  return 0;
153 }
154 
155 /* FIXME: use directly links fifo */
156 
157 typedef struct MixContext {
158  const AVClass *class; /**< class for AVOptions */
160 
161  int nb_inputs; /**< number of inputs */
162  int active_inputs; /**< number of input currently active */
163  int duration_mode; /**< mode for determining duration */
164  float dropout_transition; /**< transition time when an input drops out */
165  char *weights_str; /**< string for custom weights for every input */
166 
167  int nb_channels; /**< number of channels */
168  int sample_rate; /**< sample rate */
169  int planar;
170  AVAudioFifo **fifos; /**< audio fifo for each input */
171  uint8_t *input_state; /**< current state of each input */
172  float *input_scale; /**< mixing scale factor for each input */
173  float *weights; /**< custom weights for every input */
174  float weight_sum; /**< sum of custom weights for every input */
175  float *scale_norm; /**< normalization factor for every input */
176  int64_t next_pts; /**< calculated pts for next output frame */
177  FrameList *frame_list; /**< list of frame info for the first input */
178 } MixContext;
179 
180 #define OFFSET(x) offsetof(MixContext, x)
181 #define A AV_OPT_FLAG_AUDIO_PARAM
182 #define F AV_OPT_FLAG_FILTERING_PARAM
183 static const AVOption amix_options[] = {
184  { "inputs", "Number of inputs.",
185  OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 1024, A|F },
186  { "duration", "How to determine the end-of-stream.",
187  OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
188  { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, "duration" },
189  { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, "duration" },
190  { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, "duration" },
191  { "dropout_transition", "Transition time, in seconds, for volume "
192  "renormalization when an input stream ends.",
193  OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
194  { "weights", "Set weight for each input.",
195  OFFSET(weights_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, A|F },
196  { NULL }
197 };
198 
200 
201 /**
202  * Update the scaling factors to apply to each input during mixing.
203  *
204  * This balances the full volume range between active inputs and handles
205  * volume transitions when EOF is encountered on an input but mixing continues
206  * with the remaining inputs.
207  */
209 {
210  float weight_sum = 0.f;
211  int i;
212 
213  for (i = 0; i < s->nb_inputs; i++)
214  if (s->input_state[i] & INPUT_ON)
215  weight_sum += FFABS(s->weights[i]);
216 
217  for (i = 0; i < s->nb_inputs; i++) {
218  if (s->input_state[i] & INPUT_ON) {
219  if (s->scale_norm[i] > weight_sum / FFABS(s->weights[i])) {
220  s->scale_norm[i] -= ((s->weight_sum / FFABS(s->weights[i])) / s->nb_inputs) *
221  nb_samples / (s->dropout_transition * s->sample_rate);
222  s->scale_norm[i] = FFMAX(s->scale_norm[i], weight_sum / FFABS(s->weights[i]));
223  }
224  }
225  }
226 
227  for (i = 0; i < s->nb_inputs; i++) {
228  if (s->input_state[i] & INPUT_ON)
229  s->input_scale[i] = 1.0f / s->scale_norm[i] * FFSIGN(s->weights[i]);
230  else
231  s->input_scale[i] = 0.0f;
232  }
233 }
234 
235 static int config_output(AVFilterLink *outlink)
236 {
237  AVFilterContext *ctx = outlink->src;
238  MixContext *s = ctx->priv;
239  int i;
240  char buf[64];
241 
242  s->planar = av_sample_fmt_is_planar(outlink->format);
243  s->sample_rate = outlink->sample_rate;
244  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
246 
247  s->frame_list = av_mallocz(sizeof(*s->frame_list));
248  if (!s->frame_list)
249  return AVERROR(ENOMEM);
250 
251  s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
252  if (!s->fifos)
253  return AVERROR(ENOMEM);
254 
255  s->nb_channels = outlink->channels;
256  for (i = 0; i < s->nb_inputs; i++) {
257  s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
258  if (!s->fifos[i])
259  return AVERROR(ENOMEM);
260  }
261 
263  if (!s->input_state)
264  return AVERROR(ENOMEM);
265  memset(s->input_state, INPUT_ON, s->nb_inputs);
266  s->active_inputs = s->nb_inputs;
267 
268  s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
269  s->scale_norm = av_mallocz_array(s->nb_inputs, sizeof(*s->scale_norm));
270  if (!s->input_scale || !s->scale_norm)
271  return AVERROR(ENOMEM);
272  for (i = 0; i < s->nb_inputs; i++)
273  s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
274  calculate_scales(s, 0);
275 
276  av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
277 
278  av_log(ctx, AV_LOG_VERBOSE,
279  "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
280  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
281 
282  return 0;
283 }
284 
285 /**
286  * Read samples from the input FIFOs, mix, and write to the output link.
287  */
288 static int output_frame(AVFilterLink *outlink)
289 {
290  AVFilterContext *ctx = outlink->src;
291  MixContext *s = ctx->priv;
292  AVFrame *out_buf, *in_buf;
293  int nb_samples, ns, i;
294 
295  if (s->input_state[0] & INPUT_ON) {
296  /* first input live: use the corresponding frame size */
297  nb_samples = frame_list_next_frame_size(s->frame_list);
298  for (i = 1; i < s->nb_inputs; i++) {
299  if (s->input_state[i] & INPUT_ON) {
300  ns = av_audio_fifo_size(s->fifos[i]);
301  if (ns < nb_samples) {
302  if (!(s->input_state[i] & INPUT_EOF))
303  /* unclosed input with not enough samples */
304  return 0;
305  /* closed input to drain */
306  nb_samples = ns;
307  }
308  }
309  }
310  } else {
311  /* first input closed: use the available samples */
312  nb_samples = INT_MAX;
313  for (i = 1; i < s->nb_inputs; i++) {
314  if (s->input_state[i] & INPUT_ON) {
315  ns = av_audio_fifo_size(s->fifos[i]);
316  nb_samples = FFMIN(nb_samples, ns);
317  }
318  }
319  if (nb_samples == INT_MAX) {
321  return 0;
322  }
323  }
324 
326  frame_list_remove_samples(s->frame_list, nb_samples);
327 
328  calculate_scales(s, nb_samples);
329 
330  if (nb_samples == 0)
331  return 0;
332 
333  out_buf = ff_get_audio_buffer(outlink, nb_samples);
334  if (!out_buf)
335  return AVERROR(ENOMEM);
336 
337  in_buf = ff_get_audio_buffer(outlink, nb_samples);
338  if (!in_buf) {
339  av_frame_free(&out_buf);
340  return AVERROR(ENOMEM);
341  }
342 
343  for (i = 0; i < s->nb_inputs; i++) {
344  if (s->input_state[i] & INPUT_ON) {
345  int planes, plane_size, p;
346 
347  av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
348  nb_samples);
349 
350  planes = s->planar ? s->nb_channels : 1;
351  plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
352  plane_size = FFALIGN(plane_size, 16);
353 
354  if (out_buf->format == AV_SAMPLE_FMT_FLT ||
355  out_buf->format == AV_SAMPLE_FMT_FLTP) {
356  for (p = 0; p < planes; p++) {
357  s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
358  (float *) in_buf->extended_data[p],
359  s->input_scale[i], plane_size);
360  }
361  } else {
362  for (p = 0; p < planes; p++) {
363  s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
364  (double *) in_buf->extended_data[p],
365  s->input_scale[i], plane_size);
366  }
367  }
368  }
369  }
370  av_frame_free(&in_buf);
371 
372  out_buf->pts = s->next_pts;
373  if (s->next_pts != AV_NOPTS_VALUE)
374  s->next_pts += nb_samples;
375 
376  return ff_filter_frame(outlink, out_buf);
377 }
378 
379 /**
380  * Requests a frame, if needed, from each input link other than the first.
381  */
382 static int request_samples(AVFilterContext *ctx, int min_samples)
383 {
384  MixContext *s = ctx->priv;
385  int i;
386 
387  av_assert0(s->nb_inputs > 1);
388 
389  for (i = 1; i < s->nb_inputs; i++) {
390  if (!(s->input_state[i] & INPUT_ON) ||
391  (s->input_state[i] & INPUT_EOF))
392  continue;
393  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
394  continue;
396  }
397  return output_frame(ctx->outputs[0]);
398 }
399 
400 /**
401  * Calculates the number of active inputs and determines EOF based on the
402  * duration option.
403  *
404  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
405  */
407 {
408  int i;
409  int active_inputs = 0;
410  for (i = 0; i < s->nb_inputs; i++)
411  active_inputs += !!(s->input_state[i] & INPUT_ON);
412  s->active_inputs = active_inputs;
413 
414  if (!active_inputs ||
415  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
416  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
417  return AVERROR_EOF;
418  return 0;
419 }
420 
422 {
423  AVFilterLink *outlink = ctx->outputs[0];
424  MixContext *s = ctx->priv;
425  AVFrame *buf = NULL;
426  int i, ret;
427 
428  FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
429 
430  for (i = 0; i < s->nb_inputs; i++) {
431  AVFilterLink *inlink = ctx->inputs[i];
432 
433  if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
434  if (i == 0) {
435  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
436  outlink->time_base);
437  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
438  if (ret < 0) {
439  av_frame_free(&buf);
440  return ret;
441  }
442  }
443 
444  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
445  buf->nb_samples);
446  if (ret < 0) {
447  av_frame_free(&buf);
448  return ret;
449  }
450 
451  av_frame_free(&buf);
452 
453  ret = output_frame(outlink);
454  if (ret < 0)
455  return ret;
456  }
457  }
458 
459  for (i = 0; i < s->nb_inputs; i++) {
460  int64_t pts;
461  int status;
462 
463  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
464  if (status == AVERROR_EOF) {
465  if (i == 0) {
466  s->input_state[i] = 0;
467  if (s->nb_inputs == 1) {
468  ff_outlink_set_status(outlink, status, pts);
469  return 0;
470  }
471  } else {
472  s->input_state[i] |= INPUT_EOF;
473  if (av_audio_fifo_size(s->fifos[i]) == 0) {
474  s->input_state[i] = 0;
475  }
476  }
477  }
478  }
479  }
480 
481  if (calc_active_inputs(s)) {
483  return 0;
484  }
485 
486  if (ff_outlink_frame_wanted(outlink)) {
487  int wanted_samples;
488 
489  if (!(s->input_state[0] & INPUT_ON))
490  return request_samples(ctx, 1);
491 
492  if (s->frame_list->nb_frames == 0) {
494  return 0;
495  }
497 
498  wanted_samples = frame_list_next_frame_size(s->frame_list);
499 
500  return request_samples(ctx, wanted_samples);
501  }
502 
503  return 0;
504 }
505 
507 {
508  MixContext *s = ctx->priv;
509  char *p, *arg, *saveptr = NULL;
510  float last_weight = 1.f;
511  int i, ret;
512 
513  for (i = 0; i < s->nb_inputs; i++) {
514  AVFilterPad pad = { 0 };
515 
516  pad.type = AVMEDIA_TYPE_AUDIO;
517  pad.name = av_asprintf("input%d", i);
518  if (!pad.name)
519  return AVERROR(ENOMEM);
520 
521  if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
522  av_freep(&pad.name);
523  return ret;
524  }
525  }
526 
528  if (!s->fdsp)
529  return AVERROR(ENOMEM);
530 
531  s->weights = av_mallocz_array(s->nb_inputs, sizeof(*s->weights));
532  if (!s->weights)
533  return AVERROR(ENOMEM);
534 
535  p = s->weights_str;
536  for (i = 0; i < s->nb_inputs; i++) {
537  if (!(arg = av_strtok(p, " ", &saveptr)))
538  break;
539 
540  p = NULL;
541  sscanf(arg, "%f", &last_weight);
542  s->weights[i] = last_weight;
543  s->weight_sum += FFABS(last_weight);
544  }
545 
546  for (; i < s->nb_inputs; i++) {
547  s->weights[i] = last_weight;
548  s->weight_sum += FFABS(last_weight);
549  }
550 
551  return 0;
552 }
553 
555 {
556  int i;
557  MixContext *s = ctx->priv;
558 
559  if (s->fifos) {
560  for (i = 0; i < s->nb_inputs; i++)
561  av_audio_fifo_free(s->fifos[i]);
562  av_freep(&s->fifos);
563  }
565  av_freep(&s->frame_list);
566  av_freep(&s->input_state);
567  av_freep(&s->input_scale);
568  av_freep(&s->scale_norm);
569  av_freep(&s->weights);
570  av_freep(&s->fdsp);
571 
572  for (i = 0; i < ctx->nb_inputs; i++)
573  av_freep(&ctx->input_pads[i].name);
574 }
575 
577 {
580  int ret;
581 
582  layouts = ff_all_channel_counts();
583  if (!layouts) {
584  ret = AVERROR(ENOMEM);
585  goto fail;
586  }
587 
588  if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
589  (ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP)) < 0 ||
590  (ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
591  (ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBLP)) < 0 ||
592  (ret = ff_set_common_formats (ctx, formats)) < 0 ||
593  (ret = ff_set_common_channel_layouts(ctx, layouts)) < 0 ||
594  (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
595  goto fail;
596  return 0;
597 fail:
598  if (layouts)
599  av_freep(&layouts->channel_layouts);
600  av_freep(&layouts);
601  return ret;
602 }
603 
605  {
606  .name = "default",
607  .type = AVMEDIA_TYPE_AUDIO,
608  .config_props = config_output,
609  },
610  { NULL }
611 };
612 
614  .name = "amix",
615  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
616  .priv_size = sizeof(MixContext),
617  .priv_class = &amix_class,
618  .init = init,
619  .uninit = uninit,
620  .activate = activate,
622  .inputs = NULL,
623  .outputs = avfilter_af_amix_outputs,
625 };
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:1494
#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:550
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:295
#define DURATION_LONGEST
Definition: af_amix.c:51
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:181
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:554
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
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
static int frame_list_next_frame_size(FrameList *frame_list)
Definition: af_amix.c:91
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:1620
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:70
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:1093
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:421
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
static int64_t frame_list_next_pts(FrameList *frame_list)
Definition: af_amix.c:98
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:382
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:604
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:406
int sample_rate
sample rate
Definition: af_amix.c:168
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:576
#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:164
FrameList * frame_list
list of frame info for the first input
Definition: af_amix.c:177
the definition of that something depends on the semantic of the filter The callback must examine the status of the filter s links and proceed accordingly The status of output links is stored in the status_in and status_out fields and tested by the ff_outlink_frame_wanted() function.If this function returns true
#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:172
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:72
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:1449
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:170
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:259
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:569
#define OFFSET(x)
Definition: af_amix.c:180
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:58
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
uint64_t * channel_layouts
list of channel layouts
Definition: formats.h:86
const char * arg
Definition: jacosubdec.c:66
simple assert() macros that are a bit more flexible than ISO C assert().
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:336
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
#define fail()
Definition: checkasm.h:122
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:162
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
MIPS optimizations info
Definition: mips.txt:2
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:59
int nb_samples
Definition: af_amix.c:57
#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:105
int planar
Definition: af_amix.c:169
int duration_mode
mode for determining duration
Definition: af_amix.c:163
#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:174
int nb_channels
number of channels
Definition: af_amix.c:167
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:132
A list of supported channel layouts.
Definition: formats.h:85
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.
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:368
uint8_t * input_state
current state of each input
Definition: af_amix.c:171
float * scale_norm
normalization factor for every input
Definition: af_amix.c:175
char * weights_str
string for custom weights for every input
Definition: af_amix.c:165
void * buf
Definition: avisynth_c.h:766
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 inputs
FrameInfo * list
Definition: af_amix.c:73
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
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
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:288
#define INPUT_ON
input is active
Definition: af_amix.c:48
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
int64_t next_pts
calculated pts for next output frame
Definition: af_amix.c:176
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
#define DURATION_SHORTEST
Definition: af_amix.c:52
#define flags(name, subs,...)
Definition: cbs_av1.c:561
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
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
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:184
#define ns(max_value, name, subs,...)
Definition: cbs_av1.c:682
#define INPUT_EOF
input has reached EOF (may still be active)
Definition: af_amix.c:49
static void calculate_scales(MixContext *s, int nb_samples)
Update the scaling factors to apply to each input during mixing.
Definition: af_amix.c:208
FrameInfo * end
Definition: af_amix.c:74
common internal and external API header
AVFILTER_DEFINE_CLASS(amix)
int nb_frames
Definition: af_amix.c:71
static const AVOption amix_options[]
Definition: af_amix.c:183
float * weights
custom weights for every input
Definition: af_amix.c:173
#define av_free(p)
static const struct @317 planes[]
Audio FIFO Buffer.
#define F
Definition: af_amix.c:182
A list of supported formats for one end of a filter link.
Definition: formats.h:64
#define DURATION_FIRST
Definition: af_amix.c:53
int nb_inputs
number of inputs
Definition: af_amix.c:161
An instance of a filter.
Definition: avfilter.h:338
Filter the word “frame” indicates either a video frame or a group of audio samples
static av_cold int init(AVFilterContext *ctx)
Definition: af_amix.c:506
#define av_freep(p)
formats
Definition: signature.h:48
AVFilter ff_af_amix
Definition: af_amix.c:613
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:235
internal API functions
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
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:409
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:557
static void frame_list_clear(FrameList *frame_list)
Definition: af_amix.c:77
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.c:191
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
Definition: internal.h:277
AVFloatDSPContext * fdsp
Definition: af_amix.c:159