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/mem.h"
41 #include "libavutil/opt.h"
42 #include "libavutil/samplefmt.h"
43 
44 #include "audio.h"
45 #include "avfilter.h"
46 #include "filters.h"
47 #include "internal.h"
48 
49 #define INPUT_ON 1 /**< input is active */
50 #define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
51 
52 #define DURATION_LONGEST 0
53 #define DURATION_SHORTEST 1
54 #define DURATION_FIRST 2
55 
56 
57 typedef struct FrameInfo {
60  struct FrameInfo *next;
61 } FrameInfo;
62 
63 /**
64  * Linked list used to store timestamps and frame sizes of all frames in the
65  * FIFO for the first input.
66  *
67  * This is needed to keep timestamps synchronized for the case where multiple
68  * input frames are pushed to the filter for processing before a frame is
69  * requested by the output link.
70  */
71 typedef struct FrameList {
72  int nb_frames;
76 } FrameList;
77 
78 static void frame_list_clear(FrameList *frame_list)
79 {
80  if (frame_list) {
81  while (frame_list->list) {
82  FrameInfo *info = frame_list->list;
83  frame_list->list = info->next;
84  av_free(info);
85  }
86  frame_list->nb_frames = 0;
87  frame_list->nb_samples = 0;
88  frame_list->end = NULL;
89  }
90 }
91 
92 static int frame_list_next_frame_size(FrameList *frame_list)
93 {
94  if (!frame_list->list)
95  return 0;
96  return frame_list->list->nb_samples;
97 }
98 
100 {
101  if (!frame_list->list)
102  return AV_NOPTS_VALUE;
103  return frame_list->list->pts;
104 }
105 
106 static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
107 {
108  if (nb_samples >= frame_list->nb_samples) {
109  frame_list_clear(frame_list);
110  } else {
111  int samples = nb_samples;
112  while (samples > 0) {
113  FrameInfo *info = frame_list->list;
114  av_assert0(info);
115  if (info->nb_samples <= samples) {
116  samples -= info->nb_samples;
117  frame_list->list = info->next;
118  if (!frame_list->list)
119  frame_list->end = NULL;
120  frame_list->nb_frames--;
121  frame_list->nb_samples -= info->nb_samples;
122  av_free(info);
123  } else {
124  info->nb_samples -= samples;
125  info->pts += samples;
126  frame_list->nb_samples -= samples;
127  samples = 0;
128  }
129  }
130  }
131 }
132 
133 static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
134 {
135  FrameInfo *info = av_malloc(sizeof(*info));
136  if (!info)
137  return AVERROR(ENOMEM);
138  info->nb_samples = nb_samples;
139  info->pts = pts;
140  info->next = NULL;
141 
142  if (!frame_list->list) {
143  frame_list->list = info;
144  frame_list->end = info;
145  } else {
146  av_assert0(frame_list->end);
147  frame_list->end->next = info;
148  frame_list->end = info;
149  }
150  frame_list->nb_frames++;
151  frame_list->nb_samples += nb_samples;
152 
153  return 0;
154 }
155 
156 /* FIXME: use directly links fifo */
157 
158 typedef struct MixContext {
159  const AVClass *class; /**< class for AVOptions */
161 
162  int nb_inputs; /**< number of inputs */
163  int active_inputs; /**< number of input currently active */
164  int duration_mode; /**< mode for determining duration */
165  float dropout_transition; /**< transition time when an input drops out */
166  char *weights_str; /**< string for custom weights for every input */
167  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, .unit = "duration" },
191  { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, .unit = "duration" },
192  { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, .unit = "duration" },
193  { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, .unit = "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  */
213 static void calculate_scales(MixContext *s, int nb_samples)
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 };
254  s->next_pts = AV_NOPTS_VALUE;
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_calloc(s->nb_inputs, sizeof(*s->fifos));
261  if (!s->fifos)
262  return AVERROR(ENOMEM);
263 
264  s->nb_channels = outlink->ch_layout.nb_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 
271  s->input_state = av_malloc(s->nb_inputs);
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_calloc(s->nb_inputs, sizeof(*s->input_scale));
278  s->scale_norm = av_calloc(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_channel_layout_describe(&outlink->ch_layout, buf, sizeof(buf));
286 
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 
320  s->next_pts = frame_list_next_pts(s->frame_list);
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) {
331  ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
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  out_buf->duration = av_rescale_q(out_buf->nb_samples, av_make_q(1, outlink->sample_rate),
384  outlink->time_base);
385 
386  if (s->next_pts != AV_NOPTS_VALUE)
387  s->next_pts += nb_samples;
388 
389  return ff_filter_frame(outlink, out_buf);
390 }
391 
392 /**
393  * Requests a frame, if needed, from each input link other than the first.
394  */
395 static int request_samples(AVFilterContext *ctx, int min_samples)
396 {
397  MixContext *s = ctx->priv;
398  int i;
399 
400  av_assert0(s->nb_inputs > 1);
401  if (min_samples == 1 && s->duration_mode == DURATION_FIRST)
402  min_samples = av_audio_fifo_size(s->fifos[0]);
403 
404  for (i = 1; i < s->nb_inputs; i++) {
405  if (!(s->input_state[i] & INPUT_ON) ||
406  (s->input_state[i] & INPUT_EOF))
407  continue;
408  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
409  continue;
410  ff_inlink_request_frame(ctx->inputs[i]);
411  return 0;
412  }
413  return output_frame(ctx->outputs[0]);
414 }
415 
416 /**
417  * Calculates the number of active inputs and determines EOF based on the
418  * duration option.
419  *
420  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
421  */
423 {
424  int i;
425  int active_inputs = 0;
426  for (i = 0; i < s->nb_inputs; i++)
427  active_inputs += !!(s->input_state[i] & INPUT_ON);
428  s->active_inputs = active_inputs;
429 
430  if (!active_inputs ||
431  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
432  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
433  return AVERROR_EOF;
434  return 0;
435 }
436 
438 {
439  AVFilterLink *outlink = ctx->outputs[0];
440  MixContext *s = ctx->priv;
441  AVFrame *buf = NULL;
442  int i, ret;
443 
445 
446  for (i = 0; i < s->nb_inputs; i++) {
447  AVFilterLink *inlink = ctx->inputs[i];
448 
449  if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
450  if (i == 0) {
451  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
452  outlink->time_base);
453  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
454  if (ret < 0) {
455  av_frame_free(&buf);
456  return ret;
457  }
458  }
459 
460  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
461  buf->nb_samples);
462  if (ret < 0) {
463  av_frame_free(&buf);
464  return ret;
465  }
466 
467  av_frame_free(&buf);
468 
469  ret = output_frame(outlink);
470  if (ret < 0)
471  return ret;
472  }
473  }
474 
475  for (i = 0; i < s->nb_inputs; i++) {
476  int64_t pts;
477  int status;
478 
479  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
480  if (status == AVERROR_EOF) {
481  s->input_state[i] |= INPUT_EOF;
482  if (av_audio_fifo_size(s->fifos[i]) == 0) {
483  s->input_state[i] &= ~INPUT_ON;
484  if (s->nb_inputs == 1) {
485  ff_outlink_set_status(outlink, status, pts);
486  return 0;
487  }
488  }
489  }
490  }
491  }
492 
493  if (calc_active_inputs(s)) {
494  ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
495  return 0;
496  }
497 
498  if (ff_outlink_frame_wanted(outlink)) {
499  int wanted_samples;
500 
501  if (!(s->input_state[0] & INPUT_ON))
502  return request_samples(ctx, 1);
503 
504  if (s->frame_list->nb_frames == 0) {
505  ff_inlink_request_frame(ctx->inputs[0]);
506  return 0;
507  }
508  av_assert0(s->frame_list->nb_frames > 0);
509 
510  wanted_samples = frame_list_next_frame_size(s->frame_list);
511 
512  return request_samples(ctx, wanted_samples);
513  }
514 
515  return 0;
516 }
517 
519 {
520  MixContext *s = ctx->priv;
521  float last_weight = 1.f;
522  char *p;
523  int i;
524 
525  s->weight_sum = 0.f;
526  p = s->weights_str;
527  for (i = 0; i < s->nb_inputs; i++) {
528  last_weight = av_strtod(p, &p);
529  s->weights[i] = last_weight;
530  s->weight_sum += FFABS(last_weight);
531  if (p && *p) {
532  p++;
533  } else {
534  i++;
535  break;
536  }
537  }
538 
539  for (; i < s->nb_inputs; i++) {
540  s->weights[i] = last_weight;
541  s->weight_sum += FFABS(last_weight);
542  }
543 }
544 
546 {
547  MixContext *s = ctx->priv;
548  int i, ret;
549 
550  for (i = 0; i < s->nb_inputs; i++) {
551  AVFilterPad pad = { 0 };
552 
553  pad.type = AVMEDIA_TYPE_AUDIO;
554  pad.name = av_asprintf("input%d", i);
555  if (!pad.name)
556  return AVERROR(ENOMEM);
557 
558  if ((ret = ff_append_inpad_free_name(ctx, &pad)) < 0)
559  return ret;
560  }
561 
562  s->fdsp = avpriv_float_dsp_alloc(0);
563  if (!s->fdsp)
564  return AVERROR(ENOMEM);
565 
566  s->weights = av_calloc(s->nb_inputs, sizeof(*s->weights));
567  if (!s->weights)
568  return AVERROR(ENOMEM);
569 
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  }
585  frame_list_clear(s->frame_list);
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 
594 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
595  char *res, int res_len, int flags)
596 {
597  MixContext *s = ctx->priv;
598  int ret;
599 
600  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
601  if (ret < 0)
602  return ret;
603 
605  for (int i = 0; i < s->nb_inputs; i++)
606  s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
607  calculate_scales(s, 0);
608 
609  return 0;
610 }
611 
613  {
614  .name = "default",
615  .type = AVMEDIA_TYPE_AUDIO,
616  .config_props = config_output,
617  },
618 };
619 
621  .name = "amix",
622  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
623  .priv_size = sizeof(MixContext),
624  .priv_class = &amix_class,
625  .init = init,
626  .uninit = uninit,
627  .activate = activate,
628  .inputs = NULL,
632  .process_command = process_command,
634 };
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:48
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:97
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
FrameList::end
FrameInfo * end
Definition: af_amix.c:75
AVERROR
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
opt.h
FrameList::nb_frames
int nb_frames
Definition: af_amix.c:72
DURATION_LONGEST
#define DURATION_LONGEST
Definition: af_amix.c:52
DURATION_FIRST
#define DURATION_FIRST
Definition: af_amix.c:54
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1015
AVFrame::duration
int64_t duration
Duration of the frame, in the same units as pts.
Definition: frame.h:780
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
av_audio_fifo_write
int av_audio_fifo_write(AVAudioFifo *af, void *const *data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:119
int64_t
long long int64_t
Definition: coverity.c:34
inlink
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
Definition: filter_design.txt:212
av_asprintf
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:115
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:160
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:374
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:486
MixContext::fdsp
AVFloatDSPContext * fdsp
Definition: af_amix.c:160
AVOption
AVOption.
Definition: opt.h:357
MixContext
Definition: af_amix.c:158
AV_LOG_VERBOSE
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:196
ff_af_amix
const AVFilter ff_af_amix
Definition: af_amix.c:620
mathematics.h
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:170
planes
static const struct @438 planes[]
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:321
FrameList::nb_samples
int nb_samples
Definition: af_amix.c:73
process_command
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_amix.c:594
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:30
INPUT_ON
#define INPUT_ON
input is active
Definition: af_amix.c:49
MixContext::input_scale
float * input_scale
mixing scale factor for each input
Definition: af_amix.c:174
ff_inlink_consume_frame
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:1442
INPUT_EOF
#define INPUT_EOF
input has reached EOF (may still be active)
Definition: af_amix.c:50
MixContext::sample_rate
int sample_rate
sample rate
Definition: af_amix.c:170
FF_FILTER_FORWARD_STATUS_BACK_ALL
#define FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, filter)
Forward the status on an output link to all input links.
Definition: filters.h:212
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:37
MixContext::frame_list
FrameList * frame_list
list of frame info for the first input
Definition: af_amix.c:179
FFSIGN
#define FFSIGN(a)
Definition: common.h:75
samplefmt.h
MixContext::normalize
int normalize
if inputs are scaled
Definition: af_amix.c:167
A
#define A
Definition: af_amix.c:183
pts
static int64_t pts
Definition: transcode_aac.c:644
AVFILTER_FLAG_DYNAMIC_INPUTS
#define AVFILTER_FLAG_DYNAMIC_INPUTS
The number of the filter inputs is not determined just by AVFilter.inputs.
Definition: avfilter.h:106
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:33
avfilter_af_amix_outputs
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:612
avassert.h
amix_options
static const AVOption amix_options[]
Definition: af_amix.c:186
av_cold
#define av_cold
Definition: attributes.h:90
calculate_scales
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
av_channel_layout_describe
int av_channel_layout_describe(const AVChannelLayout *channel_layout, char *buf, size_t buf_size)
Get a human-readable string describing the channel layout properties.
Definition: channel_layout.c:648
ff_outlink_set_status
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
ff_inlink_request_frame
void ff_inlink_request_frame(AVFilterLink *link)
Mark that a frame is wanted on the link.
Definition: avfilter.c:1568
s
#define s(width, name)
Definition: cbs_vp9.c:198
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
info
MIPS optimizations info
Definition: mips.txt:2
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:40
av_sample_fmt_is_planar
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:114
filters.h
ctx
AVFormatContext * ctx
Definition: movenc.c:49
av_rescale_q
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
MixContext::active_inputs
int active_inputs
number of input currently active
Definition: af_amix.c:163
av_get_sample_fmt_name
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:51
MixContext::planar
int planar
Definition: af_amix.c:171
MixContext::duration_mode
int duration_mode
mode for determining duration
Definition: af_amix.c:164
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:74
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
MixContext::nb_channels
int nb_channels
number of channels
Definition: af_amix.c:169
MixContext::dropout_transition
float dropout_transition
transition time when an input drops out
Definition: af_amix.c:165
NULL
#define NULL
Definition: coverity.c:32
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
av_audio_fifo_alloc
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:62
ff_append_inpad_free_name
int ff_append_inpad_free_name(AVFilterContext *f, AVFilterPad *p)
Definition: avfilter.c:132
MixContext::fifos
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:172
inputs
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
Definition: filter_design.txt:243
FrameList
Linked list used to store timestamps and frame sizes of all frames in the FIFO for the first input.
Definition: af_amix.c:71
MixContext::input_state
uint8_t * input_state
current state of each input
Definition: af_amix.c:173
ff_inlink_acknowledge_status
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:1389
FrameInfo
Definition: af_amix.c:57
frame_list_add_frame
static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
Definition: af_amix.c:133
float_dsp.h
eval.h
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:244
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:94
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:575
av_audio_fifo_read
int av_audio_fifo_read(AVAudioFifo *af, void *const *data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:175
av_make_q
static AVRational av_make_q(int num, int den)
Create an AVRational.
Definition: rational.h:71
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
MixContext::scale_norm
float * scale_norm
normalization factor for every input
Definition: af_amix.c:177
MixContext::weights_str
char * weights_str
string for custom weights for every input
Definition: af_amix.c:166
AVFloatDSPContext
Definition: float_dsp.h:24
output_frame
static int output_frame(AVFilterLink *outlink)
Read samples from the input FIFOs, mix, and write to the output link.
Definition: af_amix.c:297
AVFrame::format
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
Definition: frame.h:461
activate
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:437
ff_filter_process_command
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:887
frame_list_next_pts
static int64_t frame_list_next_pts(FrameList *frame_list)
Definition: af_amix.c:99
attributes.h
ns
#define ns(max_value, name, subs,...)
Definition: cbs_av1.c:608
av_audio_fifo_size
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:222
MixContext::weight_sum
float weight_sum
sum of custom weights for every input
Definition: af_amix.c:176
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:248
MixContext::next_pts
int64_t next_pts
calculated pts for next output frame
Definition: af_amix.c:178
normalize
Definition: normalize.py:1
FrameInfo::nb_samples
int nb_samples
Definition: af_amix.c:58
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:454
FrameInfo::next
struct FrameInfo * next
Definition: af_amix.c:60
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:435
common.h
frame_list_next_frame_size
static int frame_list_next_frame_size(FrameList *frame_list)
Definition: af_amix.c:92
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:256
audio_fifo.h
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:39
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
AVFilter
Filter definition.
Definition: avfilter.h:166
ret
ret
Definition: filter_design.txt:187
AVFilterPad::type
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:44
av_strtod
double av_strtod(const char *numstr, char **tail)
Parse the string in numstr and return its value as a double.
Definition: eval.c:107
MixContext::weights
float * weights
custom weights for every input
Definition: af_amix.c:175
status
ov_status_e status
Definition: dnn_backend_openvino.c:101
channel_layout.h
calc_active_inputs
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:422
F
#define F
Definition: af_amix.c:184
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:245
avfilter.h
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(amix)
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:67
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
AVFilterContext
An instance of a filter.
Definition: avfilter.h:407
T
#define T
Definition: af_amix.c:185
mem.h
audio.h
request_samples
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:395
DURATION_SHORTEST
#define DURATION_SHORTEST
Definition: af_amix.c:53
av_free
#define av_free(p)
Definition: tableprint_vlc.h:33
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
OFFSET
#define OFFSET(x)
Definition: af_amix.c:182
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:261
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:183
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:146
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_amix.c:545
frame_list_clear
static void frame_list_clear(FrameList *frame_list)
Definition: af_amix.c:78
parse_weights
static void parse_weights(AVFilterContext *ctx)
Definition: af_amix.c:518
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:474
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
ff_outlink_frame_wanted
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
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:61
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Definition: opt.h:249
FrameInfo::pts
int64_t pts
Definition: af_amix.c:59
FrameList::list
FrameInfo * list
Definition: af_amix.c:74
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:254
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
frame_list_remove_samples
static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
Definition: af_amix.c:106
MixContext::nb_inputs
int nb_inputs
number of inputs
Definition: af_amix.c:162
FILTER_SAMPLEFMTS
#define FILTER_SAMPLEFMTS(...)
Definition: internal.h:170