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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 
166  int nb_channels; /**< number of channels */
167  int sample_rate; /**< sample rate */
168  int planar;
169  AVAudioFifo **fifos; /**< audio fifo for each input */
170  uint8_t *input_state; /**< current state of each input */
171  float *input_scale; /**< mixing scale factor for each input */
172  float scale_norm; /**< normalization factor for all inputs */
173  int64_t next_pts; /**< calculated pts for next output frame */
174  FrameList *frame_list; /**< list of frame info for the first input */
175 } MixContext;
176 
177 #define OFFSET(x) offsetof(MixContext, x)
178 #define A AV_OPT_FLAG_AUDIO_PARAM
179 #define F AV_OPT_FLAG_FILTERING_PARAM
180 static const AVOption amix_options[] = {
181  { "inputs", "Number of inputs.",
182  OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 1024, A|F },
183  { "duration", "How to determine the end-of-stream.",
184  OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
185  { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, "duration" },
186  { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, "duration" },
187  { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, "duration" },
188  { "dropout_transition", "Transition time, in seconds, for volume "
189  "renormalization when an input stream ends.",
190  OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
191  { NULL }
192 };
193 
195 
196 /**
197  * Update the scaling factors to apply to each input during mixing.
198  *
199  * This balances the full volume range between active inputs and handles
200  * volume transitions when EOF is encountered on an input but mixing continues
201  * with the remaining inputs.
202  */
203 static void calculate_scales(MixContext *s, int nb_samples)
204 {
205  int i;
206 
207  if (s->scale_norm > s->active_inputs) {
208  s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);
210  }
211 
212  for (i = 0; i < s->nb_inputs; i++) {
213  if (s->input_state[i] & INPUT_ON)
214  s->input_scale[i] = 1.0f / s->scale_norm;
215  else
216  s->input_scale[i] = 0.0f;
217  }
218 }
219 
220 static int config_output(AVFilterLink *outlink)
221 {
222  AVFilterContext *ctx = outlink->src;
223  MixContext *s = ctx->priv;
224  int i;
225  char buf[64];
226 
227  s->planar = av_sample_fmt_is_planar(outlink->format);
228  s->sample_rate = outlink->sample_rate;
229  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
231 
232  s->frame_list = av_mallocz(sizeof(*s->frame_list));
233  if (!s->frame_list)
234  return AVERROR(ENOMEM);
235 
236  s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
237  if (!s->fifos)
238  return AVERROR(ENOMEM);
239 
240  s->nb_channels = outlink->channels;
241  for (i = 0; i < s->nb_inputs; i++) {
242  s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
243  if (!s->fifos[i])
244  return AVERROR(ENOMEM);
245  }
246 
248  if (!s->input_state)
249  return AVERROR(ENOMEM);
250  memset(s->input_state, INPUT_ON, s->nb_inputs);
251  s->active_inputs = s->nb_inputs;
252 
253  s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
254  if (!s->input_scale)
255  return AVERROR(ENOMEM);
256  s->scale_norm = s->active_inputs;
257  calculate_scales(s, 0);
258 
259  av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
260 
261  av_log(ctx, AV_LOG_VERBOSE,
262  "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
263  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
264 
265  return 0;
266 }
267 
268 /**
269  * Read samples from the input FIFOs, mix, and write to the output link.
270  */
271 static int output_frame(AVFilterLink *outlink)
272 {
273  AVFilterContext *ctx = outlink->src;
274  MixContext *s = ctx->priv;
275  AVFrame *out_buf, *in_buf;
276  int nb_samples, ns, i;
277 
278  if (s->input_state[0] & INPUT_ON) {
279  /* first input live: use the corresponding frame size */
280  nb_samples = frame_list_next_frame_size(s->frame_list);
281  for (i = 1; i < s->nb_inputs; i++) {
282  if (s->input_state[i] & INPUT_ON) {
283  ns = av_audio_fifo_size(s->fifos[i]);
284  if (ns < nb_samples) {
285  if (!(s->input_state[i] & INPUT_EOF))
286  /* unclosed input with not enough samples */
287  return 0;
288  /* closed input to drain */
289  nb_samples = ns;
290  }
291  }
292  }
293  } else {
294  /* first input closed: use the available samples */
295  nb_samples = INT_MAX;
296  for (i = 1; i < s->nb_inputs; i++) {
297  if (s->input_state[i] & INPUT_ON) {
298  ns = av_audio_fifo_size(s->fifos[i]);
299  nb_samples = FFMIN(nb_samples, ns);
300  }
301  }
302  if (nb_samples == INT_MAX) {
304  return 0;
305  }
306  }
307 
309  frame_list_remove_samples(s->frame_list, nb_samples);
310 
311  calculate_scales(s, nb_samples);
312 
313  if (nb_samples == 0)
314  return 0;
315 
316  out_buf = ff_get_audio_buffer(outlink, nb_samples);
317  if (!out_buf)
318  return AVERROR(ENOMEM);
319 
320  in_buf = ff_get_audio_buffer(outlink, nb_samples);
321  if (!in_buf) {
322  av_frame_free(&out_buf);
323  return AVERROR(ENOMEM);
324  }
325 
326  for (i = 0; i < s->nb_inputs; i++) {
327  if (s->input_state[i] & INPUT_ON) {
328  int planes, plane_size, p;
329 
330  av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
331  nb_samples);
332 
333  planes = s->planar ? s->nb_channels : 1;
334  plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
335  plane_size = FFALIGN(plane_size, 16);
336 
337  if (out_buf->format == AV_SAMPLE_FMT_FLT ||
338  out_buf->format == AV_SAMPLE_FMT_FLTP) {
339  for (p = 0; p < planes; p++) {
340  s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
341  (float *) in_buf->extended_data[p],
342  s->input_scale[i], plane_size);
343  }
344  } else {
345  for (p = 0; p < planes; p++) {
346  s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
347  (double *) in_buf->extended_data[p],
348  s->input_scale[i], plane_size);
349  }
350  }
351  }
352  }
353  av_frame_free(&in_buf);
354 
355  out_buf->pts = s->next_pts;
356  if (s->next_pts != AV_NOPTS_VALUE)
357  s->next_pts += nb_samples;
358 
359  return ff_filter_frame(outlink, out_buf);
360 }
361 
362 /**
363  * Requests a frame, if needed, from each input link other than the first.
364  */
365 static int request_samples(AVFilterContext *ctx, int min_samples)
366 {
367  MixContext *s = ctx->priv;
368  int i;
369 
370  av_assert0(s->nb_inputs > 1);
371 
372  for (i = 1; i < s->nb_inputs; i++) {
373  if (!(s->input_state[i] & INPUT_ON) ||
374  (s->input_state[i] & INPUT_EOF))
375  continue;
376  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
377  continue;
379  }
380  return output_frame(ctx->outputs[0]);
381 }
382 
383 /**
384  * Calculates the number of active inputs and determines EOF based on the
385  * duration option.
386  *
387  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
388  */
390 {
391  int i;
392  int active_inputs = 0;
393  for (i = 0; i < s->nb_inputs; i++)
394  active_inputs += !!(s->input_state[i] & INPUT_ON);
395  s->active_inputs = active_inputs;
396 
397  if (!active_inputs ||
398  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
399  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
400  return AVERROR_EOF;
401  return 0;
402 }
403 
405 {
406  AVFilterLink *outlink = ctx->outputs[0];
407  MixContext *s = ctx->priv;
408  AVFrame *buf = NULL;
409  int i, ret;
410 
411  for (i = 0; i < s->nb_inputs; i++) {
412  AVFilterLink *inlink = ctx->inputs[i];
413 
414  if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
415  if (i == 0) {
416  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
417  outlink->time_base);
418  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
419  if (ret < 0) {
420  av_frame_free(&buf);
421  return ret;
422  }
423  }
424 
425  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
426  buf->nb_samples);
427  if (ret < 0) {
428  av_frame_free(&buf);
429  return ret;
430  }
431 
432  av_frame_free(&buf);
433 
434  ret = output_frame(outlink);
435  if (ret < 0)
436  return ret;
437  }
438  }
439 
440  for (i = 0; i < s->nb_inputs; i++) {
441  int64_t pts;
442  int status;
443 
444  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
445  if (status == AVERROR_EOF) {
446  if (i == 0) {
447  s->input_state[i] = 0;
448  if (s->nb_inputs == 1) {
449  ff_outlink_set_status(outlink, status, pts);
450  return 0;
451  }
452  } else {
453  s->input_state[i] |= INPUT_EOF;
454  if (av_audio_fifo_size(s->fifos[i]) == 0) {
455  s->input_state[i] = 0;
456  }
457  }
458  }
459  }
460  }
461 
462  if (calc_active_inputs(s)) {
464  return 0;
465  }
466 
467  if (ff_outlink_frame_wanted(outlink)) {
468  int wanted_samples;
469 
470  if (!(s->input_state[0] & INPUT_ON))
471  return request_samples(ctx, 1);
472 
473  if (s->frame_list->nb_frames == 0) {
475  return 0;
476  }
478 
479  wanted_samples = frame_list_next_frame_size(s->frame_list);
480 
481  return request_samples(ctx, wanted_samples);
482  }
483 
484  return 0;
485 }
486 
488 {
489  MixContext *s = ctx->priv;
490  int i, ret;
491 
492  for (i = 0; i < s->nb_inputs; i++) {
493  char name[32];
494  AVFilterPad pad = { 0 };
495 
496  snprintf(name, sizeof(name), "input%d", i);
497  pad.type = AVMEDIA_TYPE_AUDIO;
498  pad.name = av_strdup(name);
499  if (!pad.name)
500  return AVERROR(ENOMEM);
501 
502  if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
503  av_freep(&pad.name);
504  return ret;
505  }
506  }
507 
509  if (!s->fdsp)
510  return AVERROR(ENOMEM);
511 
512  return 0;
513 }
514 
516 {
517  int i;
518  MixContext *s = ctx->priv;
519 
520  if (s->fifos) {
521  for (i = 0; i < s->nb_inputs; i++)
522  av_audio_fifo_free(s->fifos[i]);
523  av_freep(&s->fifos);
524  }
526  av_freep(&s->frame_list);
527  av_freep(&s->input_state);
528  av_freep(&s->input_scale);
529  av_freep(&s->fdsp);
530 
531  for (i = 0; i < ctx->nb_inputs; i++)
532  av_freep(&ctx->input_pads[i].name);
533 }
534 
536 {
539  int ret;
540 
541  layouts = ff_all_channel_counts();
542  if (!layouts) {
543  ret = AVERROR(ENOMEM);
544  goto fail;
545  }
546 
547  if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
548  (ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP)) < 0 ||
549  (ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
550  (ret = ff_add_format(&formats, AV_SAMPLE_FMT_DBLP)) < 0 ||
551  (ret = ff_set_common_formats (ctx, formats)) < 0 ||
552  (ret = ff_set_common_channel_layouts(ctx, layouts)) < 0 ||
553  (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
554  goto fail;
555  return 0;
556 fail:
557  if (layouts)
558  av_freep(&layouts->channel_layouts);
559  av_freep(&layouts);
560  return ret;
561 }
562 
564  {
565  .name = "default",
566  .type = AVMEDIA_TYPE_AUDIO,
567  .config_props = config_output,
568  },
569  { NULL }
570 };
571 
573  .name = "amix",
574  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
575  .priv_size = sizeof(MixContext),
576  .priv_class = &amix_class,
577  .init = init,
578  .uninit = uninit,
579  .activate = activate,
581  .inputs = NULL,
582  .outputs = avfilter_af_amix_outputs,
584 };
float, planar
Definition: samplefmt.h:69
int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)
Take a frame from the link's FIFO and update the link's stats.
Definition: avfilter.c:1542
#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:549
const char * s
Definition: avisynth_c.h:768
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:201
#define DURATION_LONGEST
Definition: af_amix.c:51
AVOption.
Definition: opt.h:246
Main libavfilter public API header.
#define A
Definition: af_amix.c:178
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:515
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:222
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:169
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1663
static int ff_outlink_frame_wanted(AVFilterLink *link)
Test if a frame is wanted on an output link.
Definition: filters.h:152
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
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
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1151
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:404
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:365
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:294
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:563
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:389
int sample_rate
sample rate
Definition: af_amix.c:167
static int flags
Definition: log.c:57
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
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:535
#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:174
#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:171
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:1507
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:169
AVFilterPad * input_pads
array of input pads
Definition: avfilter.h:345
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:568
#define OFFSET(x)
Definition: af_amix.c:177
float scale_norm
normalization factor for all inputs
Definition: af_amix.c:172
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:127
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
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:163
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:179
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
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:337
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
static void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.h:229
#define fail()
Definition: checkasm.h:109
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
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
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
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:168
int duration_mode
mode for determining duration
Definition: af_amix.c:163
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:389
int nb_channels
number of channels
Definition: af_amix.c:166
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:274
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:379
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:237
uint8_t * input_state
current state of each input
Definition: af_amix.c:170
void * buf
Definition: avisynth_c.h:690
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
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:271
#define INPUT_ON
input is active
Definition: af_amix.c:48
#define snprintf
Definition: snprintf.h:34
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:173
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
#define DURATION_SHORTEST
Definition: af_amix.c:52
static int64_t pts
Global timestamp for the audio frames.
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
#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:203
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:180
#define av_free(p)
Audio FIFO Buffer.
#define F
Definition: af_amix.c:179
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
static av_cold int init(AVFilterContext *ctx)
Definition: af_amix.c:487
#define av_freep(p)
formats
Definition: signature.h:48
AVFilter ff_af_amix
Definition: af_amix.c:572
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:220
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
static void frame_list_clear(FrameList *frame_list)
Definition: af_amix.c:77
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
const char * name
Definition: opengl_enc.c:103
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