FFmpeg
af_amix.c
Go to the documentation of this file.
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 
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 
314  } else {
315  /* first input closed: use the available samples */
316  nb_samples = INT_MAX;
317  for (i = 1; i < s->nb_inputs; i++) {
318  if (s->input_state[i] & INPUT_ON) {
319  ns = av_audio_fifo_size(s->fifos[i]);
320  nb_samples = FFMIN(nb_samples, ns);
321  }
322  }
323  if (nb_samples == INT_MAX) {
325  return 0;
326  }
327  }
328 
329  frame_list_remove_samples(s->frame_list, nb_samples);
330 
331  calculate_scales(s, nb_samples);
332 
333  if (nb_samples == 0)
334  return 0;
335 
336  out_buf = ff_get_audio_buffer(outlink, nb_samples);
337  if (!out_buf)
338  return AVERROR(ENOMEM);
339 
340  in_buf = ff_get_audio_buffer(outlink, nb_samples);
341  if (!in_buf) {
342  av_frame_free(&out_buf);
343  return AVERROR(ENOMEM);
344  }
345 
346  for (i = 0; i < s->nb_inputs; i++) {
347  if (s->input_state[i] & INPUT_ON) {
348  int planes, plane_size, p;
349 
350  av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
351  nb_samples);
352 
353  planes = s->planar ? s->nb_channels : 1;
354  plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
355  plane_size = FFALIGN(plane_size, 16);
356 
357  if (out_buf->format == AV_SAMPLE_FMT_FLT ||
358  out_buf->format == AV_SAMPLE_FMT_FLTP) {
359  for (p = 0; p < planes; p++) {
360  s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
361  (float *) in_buf->extended_data[p],
362  s->input_scale[i], plane_size);
363  }
364  } else {
365  for (p = 0; p < planes; p++) {
366  s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
367  (double *) in_buf->extended_data[p],
368  s->input_scale[i], plane_size);
369  }
370  }
371  }
372  }
373  av_frame_free(&in_buf);
374 
375  out_buf->pts = s->next_pts;
376  if (s->next_pts != AV_NOPTS_VALUE)
377  s->next_pts += nb_samples;
378 
379  return ff_filter_frame(outlink, out_buf);
380 }
381 
382 /**
383  * Requests a frame, if needed, from each input link other than the first.
384  */
385 static int request_samples(AVFilterContext *ctx, int min_samples)
386 {
387  MixContext *s = ctx->priv;
388  int i;
389 
390  av_assert0(s->nb_inputs > 1);
391 
392  for (i = 1; i < s->nb_inputs; i++) {
393  if (!(s->input_state[i] & INPUT_ON) ||
394  (s->input_state[i] & INPUT_EOF))
395  continue;
396  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
397  continue;
399  }
400  return output_frame(ctx->outputs[0]);
401 }
402 
403 /**
404  * Calculates the number of active inputs and determines EOF based on the
405  * duration option.
406  *
407  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
408  */
410 {
411  int i;
412  int active_inputs = 0;
413  for (i = 0; i < s->nb_inputs; i++)
414  active_inputs += !!(s->input_state[i] & INPUT_ON);
415  s->active_inputs = active_inputs;
416 
417  if (!active_inputs ||
418  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
419  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
420  return AVERROR_EOF;
421  return 0;
422 }
423 
425 {
426  AVFilterLink *outlink = ctx->outputs[0];
427  MixContext *s = ctx->priv;
428  AVFrame *buf = NULL;
429  int i, ret;
430 
431  FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
432 
433  for (i = 0; i < s->nb_inputs; i++) {
434  AVFilterLink *inlink = ctx->inputs[i];
435 
436  if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
437  if (i == 0) {
438  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
439  outlink->time_base);
440  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
441  if (ret < 0) {
442  av_frame_free(&buf);
443  return ret;
444  }
445  }
446 
447  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
448  buf->nb_samples);
449  if (ret < 0) {
450  av_frame_free(&buf);
451  return ret;
452  }
453 
454  av_frame_free(&buf);
455 
456  ret = output_frame(outlink);
457  if (ret < 0)
458  return ret;
459  }
460  }
461 
462  for (i = 0; i < s->nb_inputs; i++) {
463  int64_t pts;
464  int status;
465 
466  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
467  if (status == AVERROR_EOF) {
468  if (i == 0) {
469  s->input_state[i] = 0;
470  if (s->nb_inputs == 1) {
471  ff_outlink_set_status(outlink, status, pts);
472  return 0;
473  }
474  } else {
475  s->input_state[i] |= INPUT_EOF;
476  if (av_audio_fifo_size(s->fifos[i]) == 0) {
477  s->input_state[i] = 0;
478  }
479  }
480  }
481  }
482  }
483 
484  if (calc_active_inputs(s)) {
486  return 0;
487  }
488 
489  if (ff_outlink_frame_wanted(outlink)) {
490  int wanted_samples;
491 
492  if (!(s->input_state[0] & INPUT_ON))
493  return request_samples(ctx, 1);
494 
495  if (s->frame_list->nb_frames == 0) {
497  return 0;
498  }
500 
501  wanted_samples = frame_list_next_frame_size(s->frame_list);
502 
503  return request_samples(ctx, wanted_samples);
504  }
505 
506  return 0;
507 }
508 
510 {
511  MixContext *s = ctx->priv;
512  float last_weight = 1.f;
513  char *p;
514  int i;
515 
516  s->weight_sum = 0.f;
517  p = s->weights_str;
518  for (i = 0; i < s->nb_inputs; i++) {
519  last_weight = av_strtod(p, &p);
520  s->weights[i] = last_weight;
521  s->weight_sum += FFABS(last_weight);
522  if (p && *p) {
523  p++;
524  } else {
525  i++;
526  break;
527  }
528  }
529 
530  for (; i < s->nb_inputs; i++) {
531  s->weights[i] = last_weight;
532  s->weight_sum += FFABS(last_weight);
533  }
534 }
535 
537 {
538  MixContext *s = ctx->priv;
539  int i, ret;
540 
541  for (i = 0; i < s->nb_inputs; i++) {
542  AVFilterPad pad = { 0 };
543 
544  pad.type = AVMEDIA_TYPE_AUDIO;
545  pad.name = av_asprintf("input%d", i);
546  if (!pad.name)
547  return AVERROR(ENOMEM);
548 
549  if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
550  av_freep(&pad.name);
551  return ret;
552  }
553  }
554 
556  if (!s->fdsp)
557  return AVERROR(ENOMEM);
558 
559  s->weights = av_mallocz_array(s->nb_inputs, sizeof(*s->weights));
560  if (!s->weights)
561  return AVERROR(ENOMEM);
562 
563  parse_weights(ctx);
564 
565  return 0;
566 }
567 
569 {
570  int i;
571  MixContext *s = ctx->priv;
572 
573  if (s->fifos) {
574  for (i = 0; i < s->nb_inputs; i++)
575  av_audio_fifo_free(s->fifos[i]);
576  av_freep(&s->fifos);
577  }
579  av_freep(&s->frame_list);
580  av_freep(&s->input_state);
581  av_freep(&s->input_scale);
582  av_freep(&s->scale_norm);
583  av_freep(&s->weights);
584  av_freep(&s->fdsp);
585 
586  for (i = 0; i < ctx->nb_inputs; i++)
587  av_freep(&ctx->input_pads[i].name);
588 }
589 
591 {
592  static const enum AVSampleFormat sample_fmts[] = {
596  };
597  int ret;
598 
599  if ((ret = ff_set_common_formats(ctx, ff_make_format_list(sample_fmts))) < 0 ||
600  (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
601  return ret;
602 
604 }
605 
606 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
607  char *res, int res_len, int flags)
608 {
609  MixContext *s = ctx->priv;
610  int ret;
611 
612  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
613  if (ret < 0)
614  return ret;
615 
616  parse_weights(ctx);
617  for (int i = 0; i < s->nb_inputs; i++)
618  s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
619  calculate_scales(s, 0);
620 
621  return 0;
622 }
623 
625  {
626  .name = "default",
627  .type = AVMEDIA_TYPE_AUDIO,
628  .config_props = config_output,
629  },
630  { NULL }
631 };
632 
634  .name = "amix",
635  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
636  .priv_size = sizeof(MixContext),
637  .priv_class = &amix_class,
638  .init = init,
639  .uninit = uninit,
640  .activate = activate,
642  .inputs = NULL,
643  .outputs = avfilter_af_amix_outputs,
646 };
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:1489
#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:308
#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:182
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:568
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 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:1615
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:1091
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:424
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:385
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:401
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:624
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:409
int sample_rate
sample rate
Definition: af_amix.c:169
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:590
#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:178
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: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:1444
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:348
static const struct @323 planes[]
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: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
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:153
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:885
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:509
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
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:350
#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 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:381
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t * input_state
current state of each input
Definition: af_amix.c:172
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
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: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:290
#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:177
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:560
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:681
#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:606
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:341
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:845
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:536
#define av_freep(p)
AVFilter ff_af_amix
Definition: af_amix.c:633
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:237
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:355
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:374
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