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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 "formats.h"
45 #include "internal.h"
46 
47 #define INPUT_ON 1 /**< input is active */
48 #define INPUT_EOF 2 /**< input has reached EOF (may still be active) */
49 
50 #define DURATION_LONGEST 0
51 #define DURATION_SHORTEST 1
52 #define DURATION_FIRST 2
53 
54 
55 typedef struct FrameInfo {
57  int64_t pts;
58  struct FrameInfo *next;
59 } FrameInfo;
60 
61 /**
62  * Linked list used to store timestamps and frame sizes of all frames in the
63  * FIFO for the first input.
64  *
65  * This is needed to keep timestamps synchronized for the case where multiple
66  * input frames are pushed to the filter for processing before a frame is
67  * requested by the output link.
68  */
69 typedef struct FrameList {
70  int nb_frames;
74 } FrameList;
75 
76 static void frame_list_clear(FrameList *frame_list)
77 {
78  if (frame_list) {
79  while (frame_list->list) {
80  FrameInfo *info = frame_list->list;
81  frame_list->list = info->next;
82  av_free(info);
83  }
84  frame_list->nb_frames = 0;
85  frame_list->nb_samples = 0;
86  frame_list->end = NULL;
87  }
88 }
89 
90 static int frame_list_next_frame_size(FrameList *frame_list)
91 {
92  if (!frame_list->list)
93  return 0;
94  return frame_list->list->nb_samples;
95 }
96 
97 static int64_t frame_list_next_pts(FrameList *frame_list)
98 {
99  if (!frame_list->list)
100  return AV_NOPTS_VALUE;
101  return frame_list->list->pts;
102 }
103 
104 static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
105 {
106  if (nb_samples >= frame_list->nb_samples) {
107  frame_list_clear(frame_list);
108  } else {
109  int samples = nb_samples;
110  while (samples > 0) {
111  FrameInfo *info = frame_list->list;
112  av_assert0(info);
113  if (info->nb_samples <= samples) {
114  samples -= info->nb_samples;
115  frame_list->list = info->next;
116  if (!frame_list->list)
117  frame_list->end = NULL;
118  frame_list->nb_frames--;
119  frame_list->nb_samples -= info->nb_samples;
120  av_free(info);
121  } else {
122  info->nb_samples -= samples;
123  info->pts += samples;
124  frame_list->nb_samples -= samples;
125  samples = 0;
126  }
127  }
128  }
129 }
130 
131 static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
132 {
133  FrameInfo *info = av_malloc(sizeof(*info));
134  if (!info)
135  return AVERROR(ENOMEM);
136  info->nb_samples = nb_samples;
137  info->pts = pts;
138  info->next = NULL;
139 
140  if (!frame_list->list) {
141  frame_list->list = info;
142  frame_list->end = info;
143  } else {
144  av_assert0(frame_list->end);
145  frame_list->end->next = info;
146  frame_list->end = info;
147  }
148  frame_list->nb_frames++;
149  frame_list->nb_samples += nb_samples;
150 
151  return 0;
152 }
153 
154 
155 typedef struct MixContext {
156  const AVClass *class; /**< class for AVOptions */
158 
159  int nb_inputs; /**< number of inputs */
160  int active_inputs; /**< number of input currently active */
161  int duration_mode; /**< mode for determining duration */
162  float dropout_transition; /**< transition time when an input drops out */
163 
164  int nb_channels; /**< number of channels */
165  int sample_rate; /**< sample rate */
166  int planar;
167  AVAudioFifo **fifos; /**< audio fifo for each input */
168  uint8_t *input_state; /**< current state of each input */
169  float *input_scale; /**< mixing scale factor for each input */
170  float scale_norm; /**< normalization factor for all inputs */
171  int64_t next_pts; /**< calculated pts for next output frame */
172  FrameList *frame_list; /**< list of frame info for the first input */
173 } MixContext;
174 
175 #define OFFSET(x) offsetof(MixContext, x)
176 #define A AV_OPT_FLAG_AUDIO_PARAM
177 #define F AV_OPT_FLAG_FILTERING_PARAM
178 static const AVOption amix_options[] = {
179  { "inputs", "Number of inputs.",
180  OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 32, A|F },
181  { "duration", "How to determine the end-of-stream.",
182  OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" },
183  { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, INT_MIN, INT_MAX, A|F, "duration" },
184  { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A|F, "duration" },
185  { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, INT_MIN, INT_MAX, A|F, "duration" },
186  { "dropout_transition", "Transition time, in seconds, for volume "
187  "renormalization when an input stream ends.",
188  OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
189  { NULL }
190 };
191 
193 
194 /**
195  * Update the scaling factors to apply to each input during mixing.
196  *
197  * This balances the full volume range between active inputs and handles
198  * volume transitions when EOF is encountered on an input but mixing continues
199  * with the remaining inputs.
200  */
201 static void calculate_scales(MixContext *s, int nb_samples)
202 {
203  int i;
204 
205  if (s->scale_norm > s->active_inputs) {
206  s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate);
208  }
209 
210  for (i = 0; i < s->nb_inputs; i++) {
211  if (s->input_state[i] & INPUT_ON)
212  s->input_scale[i] = 1.0f / s->scale_norm;
213  else
214  s->input_scale[i] = 0.0f;
215  }
216 }
217 
218 static int config_output(AVFilterLink *outlink)
219 {
220  AVFilterContext *ctx = outlink->src;
221  MixContext *s = ctx->priv;
222  int i;
223  char buf[64];
224 
225  s->planar = av_sample_fmt_is_planar(outlink->format);
226  s->sample_rate = outlink->sample_rate;
227  outlink->time_base = (AVRational){ 1, outlink->sample_rate };
229 
230  s->frame_list = av_mallocz(sizeof(*s->frame_list));
231  if (!s->frame_list)
232  return AVERROR(ENOMEM);
233 
234  s->fifos = av_mallocz_array(s->nb_inputs, sizeof(*s->fifos));
235  if (!s->fifos)
236  return AVERROR(ENOMEM);
237 
239  for (i = 0; i < s->nb_inputs; i++) {
240  s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
241  if (!s->fifos[i])
242  return AVERROR(ENOMEM);
243  }
244 
246  if (!s->input_state)
247  return AVERROR(ENOMEM);
248  memset(s->input_state, INPUT_ON, s->nb_inputs);
249  s->active_inputs = s->nb_inputs;
250 
251  s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale));
252  if (!s->input_scale)
253  return AVERROR(ENOMEM);
254  s->scale_norm = s->active_inputs;
255  calculate_scales(s, 0);
256 
257  av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout);
258 
259  av_log(ctx, AV_LOG_VERBOSE,
260  "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
261  av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);
262 
263  return 0;
264 }
265 
266 static int calc_active_inputs(MixContext *s);
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, ret, i;
277 
278  ret = calc_active_inputs(s);
279  if (ret < 0)
280  return ret;
281 
282  if (s->input_state[0] & INPUT_ON) {
283  /* first input live: use the corresponding frame size */
284  nb_samples = frame_list_next_frame_size(s->frame_list);
285  for (i = 1; i < s->nb_inputs; i++) {
286  if (s->input_state[i] & INPUT_ON) {
287  ns = av_audio_fifo_size(s->fifos[i]);
288  if (ns < nb_samples) {
289  if (!(s->input_state[i] & INPUT_EOF))
290  /* unclosed input with not enough samples */
291  return 0;
292  /* closed input to drain */
293  nb_samples = ns;
294  }
295  }
296  }
297  } else {
298  /* first input closed: use the available samples */
299  nb_samples = INT_MAX;
300  for (i = 1; i < s->nb_inputs; i++) {
301  if (s->input_state[i] & INPUT_ON) {
302  ns = av_audio_fifo_size(s->fifos[i]);
303  nb_samples = FFMIN(nb_samples, ns);
304  }
305  }
306  if (nb_samples == INT_MAX)
307  return AVERROR_EOF;
308  }
309 
311  frame_list_remove_samples(s->frame_list, nb_samples);
312 
313  calculate_scales(s, nb_samples);
314 
315  out_buf = ff_get_audio_buffer(outlink, nb_samples);
316  if (!out_buf)
317  return AVERROR(ENOMEM);
318 
319  in_buf = ff_get_audio_buffer(outlink, nb_samples);
320  if (!in_buf) {
321  av_frame_free(&out_buf);
322  return AVERROR(ENOMEM);
323  }
324 
325  for (i = 0; i < s->nb_inputs; i++) {
326  if (s->input_state[i] & INPUT_ON) {
327  int planes, plane_size, p;
328 
329  av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
330  nb_samples);
331 
332  planes = s->planar ? s->nb_channels : 1;
333  plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
334  plane_size = FFALIGN(plane_size, 16);
335 
336  for (p = 0; p < planes; p++) {
337  s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
338  (float *) in_buf->extended_data[p],
339  s->input_scale[i], plane_size);
340  }
341  }
342  }
343  av_frame_free(&in_buf);
344 
345  out_buf->pts = s->next_pts;
346  if (s->next_pts != AV_NOPTS_VALUE)
347  s->next_pts += nb_samples;
348 
349  return ff_filter_frame(outlink, out_buf);
350 }
351 
352 /**
353  * Requests a frame, if needed, from each input link other than the first.
354  */
355 static int request_samples(AVFilterContext *ctx, int min_samples)
356 {
357  MixContext *s = ctx->priv;
358  int i, ret;
359 
360  av_assert0(s->nb_inputs > 1);
361 
362  for (i = 1; i < s->nb_inputs; i++) {
363  ret = 0;
364  if (!(s->input_state[i] & INPUT_ON))
365  continue;
366  if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
367  continue;
368  ret = ff_request_frame(ctx->inputs[i]);
369  if (ret == AVERROR_EOF) {
370  s->input_state[i] |= INPUT_EOF;
371  if (av_audio_fifo_size(s->fifos[i]) == 0) {
372  s->input_state[i] = 0;
373  continue;
374  }
375  } else if (ret < 0)
376  return ret;
377  }
378  return output_frame(ctx->outputs[0]);
379 }
380 
381 /**
382  * Calculates the number of active inputs and determines EOF based on the
383  * duration option.
384  *
385  * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
386  */
388 {
389  int i;
390  int active_inputs = 0;
391  for (i = 0; i < s->nb_inputs; i++)
392  active_inputs += !!(s->input_state[i] & INPUT_ON);
393  s->active_inputs = active_inputs;
394 
395  if (!active_inputs ||
396  (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
397  (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
398  return AVERROR_EOF;
399  return 0;
400 }
401 
402 static int request_frame(AVFilterLink *outlink)
403 {
404  AVFilterContext *ctx = outlink->src;
405  MixContext *s = ctx->priv;
406  int ret;
407  int wanted_samples;
408 
409  ret = calc_active_inputs(s);
410  if (ret < 0)
411  return ret;
412 
413  if (!(s->input_state[0] & INPUT_ON))
414  return request_samples(ctx, 1);
415 
416  if (s->frame_list->nb_frames == 0) {
417  ret = ff_request_frame(ctx->inputs[0]);
418  if (ret == AVERROR_EOF) {
419  s->input_state[0] = 0;
420  if (s->nb_inputs == 1)
421  return AVERROR_EOF;
422  return output_frame(ctx->outputs[0]);
423  }
424  return ret;
425  }
427 
428  wanted_samples = frame_list_next_frame_size(s->frame_list);
429 
430  return request_samples(ctx, wanted_samples);
431 }
432 
433 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
434 {
435  AVFilterContext *ctx = inlink->dst;
436  MixContext *s = ctx->priv;
437  AVFilterLink *outlink = ctx->outputs[0];
438  int i, ret = 0;
439 
440  for (i = 0; i < ctx->nb_inputs; i++)
441  if (ctx->inputs[i] == inlink)
442  break;
443  if (i >= ctx->nb_inputs) {
444  av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
445  ret = AVERROR(EINVAL);
446  goto fail;
447  }
448 
449  if (i == 0) {
450  int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
451  outlink->time_base);
452  ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
453  if (ret < 0)
454  goto fail;
455  }
456 
457  ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
458  buf->nb_samples);
459 
460  av_frame_free(&buf);
461  return output_frame(outlink);
462 
463 fail:
464  av_frame_free(&buf);
465 
466  return ret;
467 }
468 
470 {
471  MixContext *s = ctx->priv;
472  int i;
473 
474  for (i = 0; i < s->nb_inputs; i++) {
475  char name[32];
476  AVFilterPad pad = { 0 };
477 
478  snprintf(name, sizeof(name), "input%d", i);
479  pad.type = AVMEDIA_TYPE_AUDIO;
480  pad.name = av_strdup(name);
481  if (!pad.name)
482  return AVERROR(ENOMEM);
484 
485  ff_insert_inpad(ctx, i, &pad);
486  }
487 
489  if (!s->fdsp)
490  return AVERROR(ENOMEM);
491 
492  return 0;
493 }
494 
496 {
497  int i;
498  MixContext *s = ctx->priv;
499 
500  if (s->fifos) {
501  for (i = 0; i < s->nb_inputs; i++)
502  av_audio_fifo_free(s->fifos[i]);
503  av_freep(&s->fifos);
504  }
506  av_freep(&s->frame_list);
507  av_freep(&s->input_state);
508  av_freep(&s->input_scale);
509  av_freep(&s->fdsp);
510 
511  for (i = 0; i < ctx->nb_inputs; i++)
512  av_freep(&ctx->input_pads[i].name);
513 }
514 
516 {
519  int ret;
520 
521  layouts = ff_all_channel_layouts();
522  if (!layouts) {
523  ret = AVERROR(ENOMEM);
524  goto fail;
525  }
526 
527  if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
528  (ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP)) < 0 ||
529  (ret = ff_set_common_formats (ctx, formats)) < 0 ||
530  (ret = ff_set_common_channel_layouts(ctx, layouts)) < 0 ||
531  (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
532  goto fail;
533  return 0;
534 fail:
535  if (layouts)
536  av_freep(&layouts->channel_layouts);
537  av_freep(&layouts);
538  return ret;
539 }
540 
542  {
543  .name = "default",
544  .type = AVMEDIA_TYPE_AUDIO,
545  .config_props = config_output,
546  .request_frame = request_frame
547  },
548  { NULL }
549 };
550 
552  .name = "amix",
553  .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
554  .priv_size = sizeof(MixContext),
555  .priv_class = &amix_class,
556  .init = init,
557  .uninit = uninit,
559  .inputs = NULL,
560  .outputs = avfilter_af_amix_outputs,
562 };
float, planar
Definition: samplefmt.h:69
#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:631
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:60
int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
Read data from an AVAudioFifo.
Definition: audio_fifo.c:182
This structure describes decoded (raw) audio or video data.
Definition: frame.h:180
#define DURATION_LONGEST
Definition: af_amix.c:50
AVOption.
Definition: opt.h:245
AVFormatContext * ctx
Definition: movenc-test.c:48
Main libavfilter public API header.
#define A
Definition: af_amix.c:176
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_amix.c:495
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:103
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:64
static enum AVSampleFormat formats[]
static int frame_list_next_frame_size(FrameList *frame_list)
Definition: af_amix.c:90
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:69
int av_get_channel_layout_nb_channels(uint64_t channel_layout)
Return the number of channels in the channel layout.
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:59
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:313
#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:1180
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:97
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:355
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:264
static const AVFilterPad avfilter_af_amix_outputs[]
Definition: af_amix.c:541
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:387
int sample_rate
sample rate
Definition: af_amix.c:165
static int query_formats(AVFilterContext *ctx)
Definition: af_amix.c:515
#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:162
FrameList * frame_list
list of frame info for the first input
Definition: af_amix.c:172
#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:169
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:110
int nb_samples
Definition: af_amix.c:71
A filter pad used for either input or output.
Definition: internal.h:53
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
AVAudioFifo ** fifos
audio fifo for each input
Definition: af_amix.c:167
AVFilterPad * input_pads
array of input pads
Definition: avfilter.h:312
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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:175
float scale_norm
normalization factor for all inputs
Definition: af_amix.c:170
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:65
int64_t pts
Definition: af_amix.c:57
#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:153
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
void * priv
private data for use by the filter
Definition: avfilter.h:320
int(* filter_frame)(AVFilterLink *link, AVFrame *frame)
Filtering callback.
Definition: internal.h:92
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:47
#define FFMAX(a, b)
Definition: common.h:94
#define fail()
Definition: checkasm.h:81
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
int active_inputs
number of input currently active
Definition: af_amix.c:160
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:229
audio channel layout utility functions
unsigned nb_inputs
number of input pads
Definition: avfilter.h:314
#define FFMIN(a, b)
Definition: common.h:96
struct FrameInfo * next
Definition: af_amix.c:58
int nb_samples
Definition: af_amix.c:56
static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
Definition: af_amix.c:104
int planar
Definition: af_amix.c:166
int duration_mode
mode for determining duration
Definition: af_amix.c:161
static const AVFilterPad outputs[]
Definition: af_afftfilt.c:386
int nb_channels
number of channels
Definition: af_amix.c:164
AVFilterChannelLayouts * ff_all_channel_layouts(void)
Construct an empty AVFilterChannelLayouts/AVFilterFormats struct – representing any channel layout (w...
Definition: formats.c:401
static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
Definition: af_amix.c:131
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.
static const AVFilterPad inputs[]
Definition: af_afftfilt.c:376
char * av_strdup(const char *s)
Duplicate the string s.
Definition: mem.c:267
uint8_t * input_state
current state of each input
Definition: af_amix.c:168
void * buf
Definition: avisynth_c.h:553
FrameInfo * list
Definition: af_amix.c:72
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:142
rational number numerator/denominator
Definition: rational.h:43
const char * name
Filter name.
Definition: avfilter.h:146
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:47
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:119
#define snprintf
Definition: snprintf.h:34
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:317
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
int64_t next_pts
calculated pts for next output frame
Definition: af_amix.c:171
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
Definition: af_amix.c:433
#define DURATION_SHORTEST
Definition: af_amix.c:51
static int64_t pts
Global timestamp for the audio frames.
static int flags
Definition: cpu.c:47
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:113
#define INPUT_EOF
input has reached EOF (may still be active)
Definition: af_amix.c:48
static void calculate_scales(MixContext *s, int nb_samples)
Update the scaling factors to apply to each input during mixing.
Definition: af_amix.c:201
FrameInfo * end
Definition: af_amix.c:73
common internal and external API header
AVFILTER_DEFINE_CLASS(amix)
static int request_frame(AVFilterLink *outlink)
Definition: af_amix.c:402
int nb_frames
Definition: af_amix.c:70
static const AVOption amix_options[]
Definition: af_amix.c:178
#define av_free(p)
Audio FIFO Buffer.
#define F
Definition: af_amix.c:177
A list of supported formats for one end of a filter link.
Definition: formats.h:64
#define DURATION_FIRST
Definition: af_amix.c:52
int nb_inputs
number of inputs
Definition: af_amix.c:159
An instance of a filter.
Definition: avfilter.h:305
static void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.h:229
static av_cold int init(AVFilterContext *ctx)
Definition: af_amix.c:469
#define av_freep(p)
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:369
AVFilter ff_af_amix
Definition: af_amix.c:551
static int config_output(AVFilterLink *outlink)
Definition: af_amix.c:218
internal API functions
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:227
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:237
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:252
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:76
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:240
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:283
AVFloatDSPContext * fdsp
Definition: af_amix.c:157