FFmpeg
af_amix.c
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1 /*
2  * Audio Mix Filter
3  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * Audio Mix Filter
25  *
26  * Mixes audio from multiple sources into a single output. The channel layout,
27  * sample rate, and sample format will be the same for all inputs and the
28  * output.
29  */
30 
31 #include "libavutil/attributes.h"
32 #include "libavutil/audio_fifo.h"
33 #include "libavutil/avassert.h"
34 #include "libavutil/avstring.h"
36 #include "libavutil/common.h"
37 #include "libavutil/float_dsp.h"
38 #include "libavutil/mathematics.h"
39 #include "libavutil/opt.h"
40 #include "libavutil/samplefmt.h"
41 
42 #include "audio.h"
43 #include "avfilter.h"
44 #include "filters.h"
45 #include "formats.h"
46 #include "internal.h"
47 
48 #define INPUT_ON 1 /**< input is active */
49 #define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
50 
51 #define DURATION_LONGEST 0
52 #define DURATION_SHORTEST 1
53 #define DURATION_FIRST 2
54 
55 
56 typedef struct FrameInfo {
58  int64_t pts;
59  struct FrameInfo *next;
60 } FrameInfo;
61 
62 /**
63  * Linked list used to store timestamps and frame sizes of all frames in the
64  * FIFO for the first input.
65  *
66  * This is needed to keep timestamps synchronized for the case where multiple
67  * input frames are pushed to the filter for processing before a frame is
68  * requested by the output link.
69  */
70 typedef struct FrameList {
71  int nb_frames;
75 } FrameList;
76 
77 static void frame_list_clear(FrameList *frame_list)
78 {
79  if (frame_list) {
80  while (frame_list->list) {
81  FrameInfo *info = frame_list->list;
82  frame_list->list = info->next;
83  av_free(info);
84  }
85  frame_list->nb_frames = 0;
86  frame_list->nb_samples = 0;
87  frame_list->end = NULL;
88  }
89 }
90 
91 static int frame_list_next_frame_size(FrameList *frame_list)
92 {
93  if (!frame_list->list)
94  return 0;
95  return frame_list->list->nb_samples;
96 }
97 
98 static int64_t frame_list_next_pts(FrameList *frame_list)
99 {
100  if (!frame_list->list)
101  return AV_NOPTS_VALUE;
102  return frame_list->list->pts;
103 }
104 
105 static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
106 {
107  if (nb_samples >= frame_list->nb_samples) {
108  frame_list_clear(frame_list);
109  } else {
110  int samples = nb_samples;
111  while (samples > 0) {
112  FrameInfo *info = frame_list->list;
113  av_assert0(info);
114  if (info->nb_samples <= samples) {
115  samples -= info->nb_samples;
116  frame_list->list = info->next;
117  if (!frame_list->list)
118  frame_list->end = NULL;
119  frame_list->nb_frames--;
120  frame_list->nb_samples -= info->nb_samples;
121  av_free(info);
122  } else {
123  info->nb_samples -= samples;
124  info->pts += samples;
125  frame_list->nb_samples -= samples;
126  samples = 0;
127  }
128  }
129  }
130 }
131 
132 static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
133 {
134  FrameInfo *info = av_malloc(sizeof(*info));
135  if (!info)
136  return AVERROR(ENOMEM);
137  info->nb_samples = nb_samples;
138  info->pts = pts;
139  info->next = NULL;
140 
141  if (!frame_list->list) {
142  frame_list->list = info;
143  frame_list->end = info;
144  } else {
145  av_assert0(frame_list->end);
146  frame_list->end->next = info;
147  frame_list->end = info;
148  }
149  frame_list->nb_frames++;
150  frame_list->nb_samples += nb_samples;
151 
152  return 0;
153 }
154 
155 /* FIXME: use directly links fifo */
156 
157 typedef struct MixContext {
158  const AVClass *class; /**< class for AVOptions */
160 
161  int nb_inputs; /**< number of inputs */
162  int active_inputs; /**< number of input currently active */
163  int duration_mode; /**< mode for determining duration */
164  float dropout_transition; /**< transition time when an input drops out */
165  char *weights_str; /**< string for custom weights for every input */
166 
167  int nb_channels; /**< number of channels */
168  int sample_rate; /**< sample rate */
169  int planar;
170  AVAudioFifo **fifos; /**< audio fifo for each input */
171  uint8_t *input_state; /**< current state of each input */
172  float *input_scale; /**< mixing scale factor for each input */
173  float *weights; /**< custom weights for every input */
174  float weight_sum; /**< sum of custom weights for every input */
175  float *scale_norm; /**< normalization factor for every input */
176  int64_t next_pts; /**< calculated pts for next output frame */
177  FrameList *frame_list; /**< list of frame info for the first input */
178 } MixContext;
179 
180 #define OFFSET(x) offsetof(MixContext, x)
181 #define A AV_OPT_FLAG_AUDIO_PARAM
182 #define F AV_OPT_FLAG_FILTERING_PARAM
183 static const AVOption amix_options[] = {
184  { "inputs", "Number of inputs.",
185  OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 1024, A|F },
186  { "duration", "How to determine the end-of-stream.",
187  OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
188  { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, 0, 0, A|F, "duration" },
189  { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, "duration" },
190  { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, 0, 0, A|F, "duration" },
191  { "dropout_transition", "Transition time, in seconds, for volume "
192  "renormalization when an input stream ends.",
193  OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
194  { "weights", "Set weight for each input.",
195  OFFSET(weights_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, A|F },
196  { NULL }
197 };
198 
200 
201 /**
202  * Update the scaling factors to apply to each input during mixing.
203  *
204  * This balances the full volume range between active inputs and handles
205  * volume transitions when EOF is encountered on an input but mixing continues
206  * with the remaining inputs.
207  */
208 static void calculate_scales(MixContext *s, int nb_samples)
209 {
210  float weight_sum = 0.f;
211  int i;
212 
213  for (i = 0; i < s->nb_inputs; i++)
214  if (s->input_state[i] & INPUT_ON)
215  weight_sum += s->weights[i];
216 
217  for (i = 0; i < s->nb_inputs; i++) {
218  if (s->input_state[i] & INPUT_ON) {
219  if (s->scale_norm[i] > weight_sum / s->weights[i]) {
220  s->scale_norm[i] -= ((s->weight_sum / s->weights[i]) / s->nb_inputs) *
221  nb_samples / (s->dropout_transition * s->sample_rate);
222  s->scale_norm[i] = FFMAX(s->scale_norm[i], weight_sum / s->weights[i]);
223  }
224  }
225  }
226 
227  for (i = 0; i < s->nb_inputs; i++) {
228  if (s->input_state[i] & INPUT_ON)
229  s->input_scale[i] = 1.0f / s->scale_norm[i];
230  else
231  s->input_scale[i] = 0.0f;
232  }
233 }
234 
235 static int config_output(AVFilterLink *outlink)
236 {
237  AVFilterContext *ctx = outlink->src;
238  MixContext *s = ctx->priv;
239  int i;
240  char buf[64];
241 
242  s->planar = av_sample_fmt_is_planar(outlink->format);
243  s->sample_rate = outlink->sample_rate;
244  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
245  s->next_pts = AV_NOPTS_VALUE;
246 
247  s->frame_list = av_mallocz(sizeof(*s->frame_list));
248  if (!s->frame_list)
249  return AVERROR(ENOMEM);
250 
251  s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
252  if (!s->fifos)
253  return AVERROR(ENOMEM);
254 
255  s->nb_channels = outlink->channels;
256  for (i = 0; i < s->nb_inputs; i++) {
257  s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
258  if (!s->fifos[i])
259  return AVERROR(ENOMEM);
260  }
261 
262  s->input_state = av_malloc(s->nb_inputs);
263  if (!s->input_state)
264  return AVERROR(ENOMEM);
265  memset(s->input_state, INPUT_ON, s->nb_inputs);
266  s->active_inputs = s->nb_inputs;
267 
268  s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
269  s->scale_norm = av_mallocz_array(s->nb_inputs, sizeof(*s->scale_norm));
270  if (!s->input_scale || !s->scale_norm)
271  return AVERROR(ENOMEM);
272  for (i = 0; i < s->nb_inputs; i++)
273  s->scale_norm[i] = s->weight_sum / s->weights[i];
274  calculate_scales(s, 0);
275 
276  av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
277 
279  "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
280  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
281 
282  return 0;
283 }
284 
285 /**
286  * Read samples from the input FIFOs, mix, and write to the output link.
287  */
288 static int output_frame(AVFilterLink *outlink)
289 {
290  AVFilterContext *ctx = outlink->src;
291  MixContext *s = ctx->priv;
292  AVFrame *out_buf, *in_buf;
293  int nb_samples, ns, i;
294 
295  if (s->input_state[0] & INPUT_ON) {
296  /* first input live: use the corresponding frame size */
297  nb_samples = frame_list_next_frame_size(s->frame_list);
298  for (i = 1; i < s->nb_inputs; i++) {
299  if (s->input_state[i] & INPUT_ON) {
300  ns = av_audio_fifo_size(s->fifos[i]);
301  if (ns < nb_samples) {
302  if (!(s->input_state[i] & INPUT_EOF))
303  /* unclosed input with not enough samples */
304  return 0;
305  /* closed input to drain */
306  nb_samples = ns;
307  }
308  }
309  }
310  } else {
311  /* first input closed: use the available samples */
312  nb_samples = INT_MAX;
313  for (i = 1; i < s->nb_inputs; i++) {
314  if (s->input_state[i] & INPUT_ON) {
315  ns = av_audio_fifo_size(s->fifos[i]);
316  nb_samples = FFMIN(nb_samples, ns);
317  }
318  }
319  if (nb_samples == INT_MAX) {
320  ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
321  return 0;
322  }
323  }
324 
325  s->next_pts = frame_list_next_pts(s->frame_list);
326  frame_list_remove_samples(s->frame_list, nb_samples);
327 
328  calculate_scales(s, nb_samples);
329 
330  if (nb_samples == 0)
331  return 0;
332 
333  out_buf = ff_get_audio_buffer(outlink, nb_samples);
334  if (!out_buf)
335  return AVERROR(ENOMEM);
336 
337  in_buf = ff_get_audio_buffer(outlink, nb_samples);
338  if (!in_buf) {
339  av_frame_free(&out_buf);
340  return AVERROR(ENOMEM);
341  }
342 
343  for (i = 0; i < s->nb_inputs; i++) {
344  if (s->input_state[i] & INPUT_ON) {
345  int planes, plane_size, p;
346 
347  av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
348  nb_samples);
349 
350  planes = s->planar ? s->nb_channels : 1;
351  plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
352  plane_size = FFALIGN(plane_size, 16);
353 
354  if (out_buf->format == AV_SAMPLE_FMT_FLT ||
355  out_buf->format == AV_SAMPLE_FMT_FLTP) {
356  for (p = 0; p < planes; p++) {
357  s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
358  (float *) in_buf->extended_data[p],
359  s->input_scale[i], plane_size);
360  }
361  } else {
362  for (p = 0; p < planes; p++) {
363  s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
364  (double *) in_buf->extended_data[p],
365  s->input_scale[i], plane_size);
366  }
367  }
368  }
369  }
370  av_frame_free(&in_buf);
371 
372  out_buf->pts = s->next_pts;
373  if (s->next_pts != AV_NOPTS_VALUE)
374  s->next_pts += nb_samples;
375 
376  return ff_filter_frame(outlink, out_buf);
377 }
378 
379 /**
380  * Requests a frame, if needed, from each input link other than the first.
381  */
382 static int request_samples(AVFilterContext *ctx, int min_samples)
383 {
384  MixContext *s = ctx->priv;
385  int i;
386 
387  av_assert0(s->nb_inputs > 1);
388 
389  for (i = 1; i < s->nb_inputs; i++) {
390  if (!(s->input_state[i] & INPUT_ON) ||
391  (s->input_state[i] & INPUT_EOF))
392  continue;
393  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
394  continue;
395  ff_inlink_request_frame(ctx->inputs[i]);
396  }
397  return output_frame(ctx->outputs[0]);
398 }
399 
400 /**
401  * Calculates the number of active inputs and determines EOF based on the
402  * duration option.
403  *
404  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
405  */
407 {
408  int i;
409  int active_inputs = 0;
410  for (i = 0; i < s->nb_inputs; i++)
411  active_inputs += !!(s->input_state[i] & INPUT_ON);
412  s->active_inputs = active_inputs;
413 
414  if (!active_inputs ||
415  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
416  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
417  return AVERROR_EOF;
418  return 0;
419 }
420 
422 {
423  AVFilterLink *outlink = ctx->outputs[0];
424  MixContext *s = ctx->priv;
425  AVFrame *buf = NULL;
426  int i, ret;
427 
429 
430  for (i = 0; i < s->nb_inputs; i++) {
431  AVFilterLink *inlink = ctx->inputs[i];
432 
433  if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
434  if (i == 0) {
435  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
436  outlink->time_base);
437  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
438  if (ret < 0) {
439  av_frame_free(&buf);
440  return ret;
441  }
442  }
443 
444  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
445  buf->nb_samples);
446  if (ret < 0) {
447  av_frame_free(&buf);
448  return ret;
449  }
450 
451  av_frame_free(&buf);
452 
453  ret = output_frame(outlink);
454  if (ret < 0)
455  return ret;
456  }
457  }
458 
459  for (i = 0; i < s->nb_inputs; i++) {
460  int64_t pts;
461  int status;
462 
463  if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
464  if (status == AVERROR_EOF) {
465  if (i == 0) {
466  s->input_state[i] = 0;
467  if (s->nb_inputs == 1) {
468  ff_outlink_set_status(outlink, status, pts);
469  return 0;
470  }
471  } else {
472  s->input_state[i] |= INPUT_EOF;
473  if (av_audio_fifo_size(s->fifos[i]) == 0) {
474  s->input_state[i] = 0;
475  }
476  }
477  }
478  }
479  }
480 
481  if (calc_active_inputs(s)) {
482  ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
483  return 0;
484  }
485 
486  if (ff_outlink_frame_wanted(outlink)) {
487  int wanted_samples;
488 
489  if (!(s->input_state[0] & INPUT_ON))
490  return request_samples(ctx, 1);
491 
492  if (s->frame_list->nb_frames == 0) {
493  ff_inlink_request_frame(ctx->inputs[0]);
494  return 0;
495  }
496  av_assert0(s->frame_list->nb_frames > 0);
497 
498  wanted_samples = frame_list_next_frame_size(s->frame_list);
499 
500  return request_samples(ctx, wanted_samples);
501  }
502 
503  return 0;
504 }
505 
507 {
508  MixContext *s = ctx->priv;
509  char *p, *arg, *saveptr = NULL;
510  float last_weight = 1.f;
511  int i, ret;
512 
513  for (i = 0; i < s->nb_inputs; i++) {
514  AVFilterPad pad = { 0 };
515 
516  pad.type = AVMEDIA_TYPE_AUDIO;
517  pad.name = av_asprintf("input%d", i);
518  if (!pad.name)
519  return AVERROR(ENOMEM);
520 
521  if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
522  av_freep(&pad.name);
523  return ret;
524  }
525  }
526 
527  s->fdsp = avpriv_float_dsp_alloc(0);
528  if (!s->fdsp)
529  return AVERROR(ENOMEM);
530 
531  s->weights = av_mallocz_array(s->nb_inputs, sizeof(*s->weights));
532  if (!s->weights)
533  return AVERROR(ENOMEM);
534 
535  p = s->weights_str;
536  for (i = 0; i < s->nb_inputs; i++) {
537  if (!(arg = av_strtok(p, " ", &saveptr)))
538  break;
539 
540  p = NULL;
541  sscanf(arg, "%f", &last_weight);
542  s->weights[i] = last_weight;
543  s->weight_sum += last_weight;
544  }
545 
546  for (; i < s->nb_inputs; i++) {
547  s->weights[i] = last_weight;
548  s->weight_sum += last_weight;
549  }
550 
551  return 0;
552 }
553 
555 {
556  int i;
557  MixContext *s = ctx->priv;
558 
559  if (s->fifos) {
560  for (i = 0; i < s->nb_inputs; i++)
561  av_audio_fifo_free(s->fifos[i]);
562  av_freep(&s->fifos);
563  }
564  frame_list_clear(s->frame_list);
565  av_freep(&s->frame_list);
566  av_freep(&s->input_state);
567  av_freep(&s->input_scale);
568  av_freep(&s->scale_norm);
569  av_freep(&s->weights);
570  av_freep(&s->fdsp);
571 
572  for (i = 0; i < ctx->nb_inputs; i++)
573  av_freep(&ctx->input_pads[i].name);
574 }
575 
577 {
580  int ret;
581 
583  if (!layouts) {
584  ret = AVERROR(ENOMEM);
585  goto fail;
586  }
587 
588  if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
592  (ret = ff_set_common_formats (ctx, formats)) < 0 ||
595  goto fail;
596  return 0;
597 fail:
598  if (layouts)
599  av_freep(&layouts->channel_layouts);
600  av_freep(&layouts);
601  return ret;
602 }
603 
605  {
606  .name = "default",
607  .type = AVMEDIA_TYPE_AUDIO,
608  .config_props = config_output,
609  },
610  { NULL }
611 };
612 
614  .name = "amix",
615  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
616  .priv_size = sizeof(MixContext),
617  .priv_class = &amix_class,
618  .init = init,
619  .uninit = uninit,
620  .activate = activate,
622  .inputs = NULL,
625 };
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
formats
formats
Definition: signature.h: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:86
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
FrameList::end
FrameInfo * end
Definition: af_amix.c:74
status
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
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:71
DURATION_LONGEST
#define DURATION_LONGEST
Definition: af_amix.c:51
DURATION_FIRST
#define DURATION_FIRST
Definition: af_amix.c:53
ff_set_common_channel_layouts
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
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
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:113
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
ff_all_channel_counts
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:410
av_get_channel_layout_string
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.
Definition: channel_layout.c:211
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
MixContext::fdsp
AVFloatDSPContext * fdsp
Definition: af_amix.c:159
AVOption
AVOption.
Definition: opt.h:246
MixContext
Definition: af_amix.c:157
av_mallocz_array
void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.c:191
AV_LOG_VERBOSE
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
mathematics.h
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
FrameList::nb_samples
int nb_samples
Definition: af_amix.c:72
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
INPUT_ON
#define INPUT_ON
input is active
Definition: af_amix.c:48
formats.h
MixContext::input_scale
float * input_scale
mixing scale factor for each input
Definition: af_amix.c:172
ff_insert_inpad
static int ff_insert_inpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new input pad for the filter.
Definition: internal.h:277
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:1481
INPUT_EOF
#define INPUT_EOF
input has reached EOF (may still be active)
Definition: af_amix.c:49
MixContext::sample_rate
int sample_rate
sample rate
Definition: af_amix.c:168
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:34
fail
#define fail()
Definition: checkasm.h:120
MixContext::frame_list
FrameList * frame_list
list of frame info for the first input
Definition: af_amix.c:177
samplefmt.h
A
#define A
Definition: af_amix.c:181
pts
static int64_t pts
Definition: transcode_aac.c:647
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:105
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
avfilter_af_amix_outputs
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:604
avassert.h
amix_options
static const AVOption amix_options[]
Definition: af_amix.c:183
buf
void * buf
Definition: avisynth_c.h:766
av_cold
#define av_cold
Definition: attributes.h:84
ff_set_common_formats
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
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:208
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:1607
s
#define s(width, name)
Definition: cbs_vp9.c:257
av_audio_fifo_write
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
info
MIPS optimizations info
Definition: mips.txt:2
av_strtok
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok().
Definition: avstring.c:184
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
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:112
outputs
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
filters.h
ctx
AVFormatContext * ctx
Definition: movenc.c:48
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:162
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:49
MixContext::planar
int planar
Definition: af_amix.c:169
arg
const char * arg
Definition: jacosubdec.c:66
MixContext::duration_mode
int duration_mode
mode for determining duration
Definition: af_amix.c:163
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
MixContext::nb_channels
int nb_channels
number of channels
Definition: af_amix.c:167
MixContext::dropout_transition
float dropout_transition
transition time when an input drops out
Definition: af_amix.c:164
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:59
ff_add_format
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:337
MixContext::fifos
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:170
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:70
MixContext::input_state
uint8_t * input_state
current state of each input
Definition: af_amix.c:171
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:1436
FrameInfo
Definition: af_amix.c:56
frame_list_add_frame
static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
Definition: af_amix.c:132
float_dsp.h
planes
static const struct @314 planes[]
ff_af_amix
AVFilter ff_af_amix
Definition: af_amix.c:613
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:235
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:188
FFMAX
#define FFMAX(a, b)
Definition: common.h:94
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:554
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:175
MixContext::weights_str
char * weights_str
string for custom weights for every input
Definition: af_amix.c:165
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:288
AVFrame::format
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...
Definition: frame.h:368
activate
static int activate(AVFilterContext *ctx)
Definition: af_amix.c:421
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
frame_list_next_pts
static int64_t frame_list_next_pts(FrameList *frame_list)
Definition: af_amix.c:98
attributes.h
ns
#define ns(max_value, name, subs,...)
Definition: cbs_av1.c:686
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:228
MixContext::weight_sum
float weight_sum
sum of custom weights for every input
Definition: af_amix.c:174
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:226
MixContext::next_pts
int64_t next_pts
calculated pts for next output frame
Definition: af_amix.c:176
av_audio_fifo_read
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:181
FrameInfo::nb_samples
int nb_samples
Definition: af_amix.c:57
FrameInfo::next
struct FrameInfo * next
Definition: af_amix.c:59
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
common.h
frame_list_next_frame_size
static int frame_list_next_frame_size(FrameList *frame_list)
Definition: af_amix.c:91
uint8_t
uint8_t
Definition: audio_convert.c:194
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:236
audio_fifo.h
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
AVFilter
Filter definition.
Definition: avfilter.h:144
ret
ret
Definition: filter_design.txt:187
AVFilterPad::type
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
MixContext::weights
float * weights
custom weights for every input
Definition: af_amix.c:173
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
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:406
F
#define F
Definition: af_amix.c:182
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:223
avfilter.h
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(amix)
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
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:338
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:382
DURATION_SHORTEST
#define DURATION_SHORTEST
Definition: af_amix.c:52
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:48
OFFSET
#define OFFSET(x)
Definition: af_amix.c:180
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_amix.c:506
frame_list_clear
static void frame_list_clear(FrameList *frame_list)
Definition: af_amix.c:77
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:565
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:576
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
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:64
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Definition: opt.h:227
FrameInfo::pts
int64_t pts
Definition: af_amix.c:58
FrameList::list
FrameInfo * list
Definition: af_amix.c:73
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:232
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
frame_list_remove_samples
static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
Definition: af_amix.c:105
MixContext::nb_inputs
int nb_inputs
number of inputs
Definition: af_amix.c:161