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/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  */
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 };
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 
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 
287  av_log(ctx, AV_LOG_VERBOSE,
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 
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) {
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;
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 
438  FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
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)) {
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) {
504  return 0;
505  }
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 
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 
570  parse_weights(ctx);
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  }
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 
606  if ((ret = ff_set_common_formats(ctx, ff_make_format_list(sample_fmts))) < 0 ||
607  (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
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 
623  parse_weights(ctx);
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,
650  .outputs = avfilter_af_amix_outputs,
653 };
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:1492
#define NULL
Definition: coverity.c:32
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:318
#define DURATION_LONGEST
Definition: af_amix.c:52
AVOption.
Definition: opt.h:248
#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:183
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:575
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:106
double, planar
Definition: samplefmt.h:70
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
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:569
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 int frame_list_next_frame_size(FrameList *frame_list)
Definition: af_amix.c:92
static const struct @322 planes[]
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:1618
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:287
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:349
#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:1094
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:431
uint8_t
#define av_cold
Definition: attributes.h:88
#define av_malloc(s)
#define T
Definition: af_amix.c:185
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:392
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:411
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:631
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
int sample_rate
sample rate
Definition: af_amix.c:170
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:597
#define AVERROR_EOF
End of file.
Definition: error.h:55
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:210
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:179
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:174
int normalize
if inputs are scaled
Definition: af_amix.c:167
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:1447
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:172
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:348
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:588
#define OFFSET(x)
Definition: af_amix.c:182
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
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:204
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:117
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
void * priv
private data for use by the filter
Definition: avfilter.h:356
simple assert() macros that are a bit more flexible than ISO C assert().
static void parse_weights(AVFilterContext *ctx)
Definition: af_amix.c:516
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:103
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
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
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
unsigned nb_inputs
number of input pads
Definition: avfilter.h:350
#define FFMIN(a, b)
Definition: common.h:105
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:171
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:176
int nb_channels
number of channels
Definition: af_amix.c:169
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:391
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t * input_state
current state of each input
Definition: af_amix.c:173
float * scale_norm
normalization factor for every input
Definition: af_amix.c:177
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:145
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:149
static int output_frame(AVFilterLink *outlink)
Read samples from the input FIFOs, mix, and write to the output link.
Definition: af_amix.c:297
#define INPUT_ON
input is active
Definition: af_amix.c:49
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:353
int64_t next_pts
calculated pts for next output frame
Definition: af_amix.c:178
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:422
#define DURATION_SHORTEST
Definition: af_amix.c:53
#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
#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: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:213
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:613
int nb_frames
Definition: af_amix.c:72
static const AVOption amix_options[]
Definition: af_amix.c:186
float * weights
custom weights for every input
Definition: af_amix.c:175
#define av_free(p)
Audio FIFO Buffer.
#define F
Definition: af_amix.c:184
#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:341
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:941
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:543
#define av_freep(p)
AVFilter ff_af_amix
Definition: af_amix.c:640
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:244
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:437
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:576
int i
Definition: input.c:407
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)
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:240
AVFloatDSPContext * fdsp
Definition: af_amix.c:160