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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"
35#include "libavutil/channel_layout.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
55typedef struct FrameInfo {
56 int nb_samples;
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 */
69typedef struct FrameList {
70 int nb_frames;
71 int nb_samples;
72 FrameInfo *list;
73 FrameInfo *end;
74} FrameList;
75
76static 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
90static 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
97static 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
104static 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
131static 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
155typedef struct MixContext {
156 const AVClass *class; /**< class for AVOptions */
157 AVFloatDSPContext *fdsp;
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
178static 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
192AVFILTER_DEFINE_CLASS(amix);
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 */
201static 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);
207 s->scale_norm = FFMAX(s->scale_norm, s->active_inputs);
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
218static 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 };
228 s->next_pts = AV_NOPTS_VALUE;
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
238 s->nb_channels = outlink->channels;
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
245 s->input_state = av_malloc(s->nb_inputs);
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
266static int calc_active_inputs(MixContext *s);
267
268/**
269 * Read samples from the input FIFOs, mix, and write to the output link.
270 */
271static 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
310 s->next_pts = frame_list_next_pts(s->frame_list);
311 frame_list_remove_samples(s->frame_list, nb_samples);
312
313 calculate_scales(s, nb_samples);
314
315 if (nb_samples == 0)
316 return 0;
317
318 out_buf = ff_get_audio_buffer(outlink, nb_samples);
319 if (!out_buf)
320 return AVERROR(ENOMEM);
321
322 in_buf = ff_get_audio_buffer(outlink, nb_samples);
323 if (!in_buf) {
324 av_frame_free(&out_buf);
325 return AVERROR(ENOMEM);
326 }
327
328 for (i = 0; i < s->nb_inputs; i++) {
329 if (s->input_state[i] & INPUT_ON) {
330 int planes, plane_size, p;
331
332 av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
333 nb_samples);
334
335 planes = s->planar ? s->nb_channels : 1;
336 plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
337 plane_size = FFALIGN(plane_size, 16);
338
339 for (p = 0; p < planes; p++) {
340 s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
341 (float *) in_buf->extended_data[p],
342 s->input_scale[i], plane_size);
343 }
344 }
345 }
346 av_frame_free(&in_buf);
347
348 out_buf->pts = s->next_pts;
349 if (s->next_pts != AV_NOPTS_VALUE)
350 s->next_pts += nb_samples;
351
352 return ff_filter_frame(outlink, out_buf);
353}
354
355/**
356 * Requests a frame, if needed, from each input link other than the first.
357 */
358static int request_samples(AVFilterContext *ctx, int min_samples)
359{
360 MixContext *s = ctx->priv;
361 int i, ret;
362
363 av_assert0(s->nb_inputs > 1);
364
365 for (i = 1; i < s->nb_inputs; i++) {
366 ret = 0;
367 if (!(s->input_state[i] & INPUT_ON))
368 continue;
369 if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
370 continue;
371 ret = ff_request_frame(ctx->inputs[i]);
372 if (ret == AVERROR_EOF) {
373 s->input_state[i] |= INPUT_EOF;
374 if (av_audio_fifo_size(s->fifos[i]) == 0) {
375 s->input_state[i] = 0;
376 continue;
377 }
378 } else if (ret < 0)
379 return ret;
380 }
381 return output_frame(ctx->outputs[0]);
382}
383
384/**
385 * Calculates the number of active inputs and determines EOF based on the
386 * duration option.
387 *
388 * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
389 */
390static int calc_active_inputs(MixContext *s)
391{
392 int i;
393 int active_inputs = 0;
394 for (i = 0; i < s->nb_inputs; i++)
395 active_inputs += !!(s->input_state[i] & INPUT_ON);
396 s->active_inputs = active_inputs;
397
398 if (!active_inputs ||
399 (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
400 (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
401 return AVERROR_EOF;
402 return 0;
403}
404
405static int request_frame(AVFilterLink *outlink)
406{
407 AVFilterContext *ctx = outlink->src;
408 MixContext *s = ctx->priv;
409 int ret;
410 int wanted_samples;
411
412 ret = calc_active_inputs(s);
413 if (ret < 0)
414 return ret;
415
416 if (!(s->input_state[0] & INPUT_ON))
417 return request_samples(ctx, 1);
418
419 if (s->frame_list->nb_frames == 0) {
420 ret = ff_request_frame(ctx->inputs[0]);
421 if (ret == AVERROR_EOF) {
422 s->input_state[0] = 0;
423 if (s->nb_inputs == 1)
424 return AVERROR_EOF;
425 return output_frame(ctx->outputs[0]);
426 }
427 return ret;
428 }
429 av_assert0(s->frame_list->nb_frames > 0);
430
431 wanted_samples = frame_list_next_frame_size(s->frame_list);
432
433 return request_samples(ctx, wanted_samples);
434}
435
436static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
437{
438 AVFilterContext *ctx = inlink->dst;
439 MixContext *s = ctx->priv;
440 AVFilterLink *outlink = ctx->outputs[0];
441 int i, ret = 0;
442
443 for (i = 0; i < ctx->nb_inputs; i++)
444 if (ctx->inputs[i] == inlink)
445 break;
446 if (i >= ctx->nb_inputs) {
447 av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
448 ret = AVERROR(EINVAL);
449 goto fail;
450 }
451
452 if (i == 0) {
453 int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
454 outlink->time_base);
455 ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
456 if (ret < 0)
457 goto fail;
458 }
459
460 ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
461 buf->nb_samples);
462
463 av_frame_free(&buf);
464 return output_frame(outlink);
465
466fail:
467 av_frame_free(&buf);
468
469 return ret;
470}
471
472static av_cold int init(AVFilterContext *ctx)
473{
474 MixContext *s = ctx->priv;
475 int i;
476
477 for (i = 0; i < s->nb_inputs; i++) {
478 char name[32];
479 AVFilterPad pad = { 0 };
480
481 snprintf(name, sizeof(name), "input%d", i);
482 pad.type = AVMEDIA_TYPE_AUDIO;
483 pad.name = av_strdup(name);
484 if (!pad.name)
485 return AVERROR(ENOMEM);
486 pad.filter_frame = filter_frame;
487
488 ff_insert_inpad(ctx, i, &pad);
489 }
490
491 s->fdsp = avpriv_float_dsp_alloc(0);
492 if (!s->fdsp)
493 return AVERROR(ENOMEM);
494
495 return 0;
496}
497
498static av_cold void uninit(AVFilterContext *ctx)
499{
500 int i;
501 MixContext *s = ctx->priv;
502
503 if (s->fifos) {
504 for (i = 0; i < s->nb_inputs; i++)
505 av_audio_fifo_free(s->fifos[i]);
506 av_freep(&s->fifos);
507 }
508 frame_list_clear(s->frame_list);
509 av_freep(&s->frame_list);
510 av_freep(&s->input_state);
511 av_freep(&s->input_scale);
512 av_freep(&s->fdsp);
513
514 for (i = 0; i < ctx->nb_inputs; i++)
515 av_freep(&ctx->input_pads[i].name);
516}
517
518static int query_formats(AVFilterContext *ctx)
519{
520 AVFilterFormats *formats = NULL;
521 AVFilterChannelLayouts *layouts;
522 int ret;
523
524 layouts = ff_all_channel_counts();
525 if (!layouts) {
526 ret = AVERROR(ENOMEM);
527 goto fail;
528 }
529
530 if ((ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT )) < 0 ||
531 (ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP)) < 0 ||
532 (ret = ff_set_common_formats (ctx, formats)) < 0 ||
533 (ret = ff_set_common_channel_layouts(ctx, layouts)) < 0 ||
534 (ret = ff_set_common_samplerates(ctx, ff_all_samplerates())) < 0)
535 goto fail;
536 return 0;
537fail:
538 if (layouts)
539 av_freep(&layouts->channel_layouts);
540 av_freep(&layouts);
541 return ret;
542}
543
544static const AVFilterPad avfilter_af_amix_outputs[] = {
545 {
546 .name = "default",
547 .type = AVMEDIA_TYPE_AUDIO,
548 .config_props = config_output,
549 .request_frame = request_frame
550 },
551 { NULL }
552};
553
554AVFilter ff_af_amix = {
555 .name = "amix",
556 .description = NULL_IF_CONFIG_SMALL("Audio mixing."),
557 .priv_size = sizeof(MixContext),
558 .priv_class = &amix_class,
559 .init = init,
560 .uninit = uninit,
561 .query_formats = query_formats,
562 .inputs = NULL,
563 .outputs = avfilter_af_amix_outputs,
564 .flags = AVFILTER_FLAG_DYNAMIC_INPUTS,
565};
566
generated by cgit at 2024-05-17 02:01:36 (GMT)