aboutsummaryrefslogtreecommitdiffstats
path: root/libavfilter/af_amix.c
blob: bc97200926123c077001db5f95ce94cf3b71d6ed (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
/*
 * Audio Mix Filter
 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file
 * Audio Mix Filter
 *
 * Mixes audio from multiple sources into a single output. The channel layout,
 * sample rate, and sample format will be the same for all inputs and the
 * output.
 */

#include "libavutil/attributes.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/common.h"
#include "libavutil/eval.h"
#include "libavutil/float_dsp.h"
#include "libavutil/mathematics.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"

#include "audio.h"
#include "avfilter.h"
#include "filters.h"

#define INPUT_ON       1    /**< input is active */
#define INPUT_EOF      2    /**< input has reached EOF (may still be active) */

#define DURATION_LONGEST  0
#define DURATION_SHORTEST 1
#define DURATION_FIRST    2


typedef struct FrameInfo {
    int nb_samples;
    int64_t pts;
    struct FrameInfo *next;
} FrameInfo;

/**
 * Linked list used to store timestamps and frame sizes of all frames in the
 * FIFO for the first input.
 *
 * This is needed to keep timestamps synchronized for the case where multiple
 * input frames are pushed to the filter for processing before a frame is
 * requested by the output link.
 */
typedef struct FrameList {
    int nb_frames;
    int nb_samples;
    FrameInfo *list;
    FrameInfo *end;
} FrameList;

static void frame_list_clear(FrameList *frame_list)
{
    if (frame_list) {
        while (frame_list->list) {
            FrameInfo *info = frame_list->list;
            frame_list->list = info->next;
            av_free(info);
        }
        frame_list->nb_frames  = 0;
        frame_list->nb_samples = 0;
        frame_list->end        = NULL;
    }
}

static int frame_list_next_frame_size(FrameList *frame_list)
{
    if (!frame_list->list)
        return 0;
    return frame_list->list->nb_samples;
}

static int64_t frame_list_next_pts(FrameList *frame_list)
{
    if (!frame_list->list)
        return AV_NOPTS_VALUE;
    return frame_list->list->pts;
}

static void frame_list_remove_samples(FrameList *frame_list, int nb_samples)
{
    if (nb_samples >= frame_list->nb_samples) {
        frame_list_clear(frame_list);
    } else {
        int samples = nb_samples;
        while (samples > 0) {
            FrameInfo *info = frame_list->list;
            av_assert0(info);
            if (info->nb_samples <= samples) {
                samples -= info->nb_samples;
                frame_list->list = info->next;
                if (!frame_list->list)
                    frame_list->end = NULL;
                frame_list->nb_frames--;
                frame_list->nb_samples -= info->nb_samples;
                av_free(info);
            } else {
                info->nb_samples       -= samples;
                info->pts              += samples;
                frame_list->nb_samples -= samples;
                samples = 0;
            }
        }
    }
}

static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts)
{
    FrameInfo *info = av_malloc(sizeof(*info));
    if (!info)
        return AVERROR(ENOMEM);
    info->nb_samples = nb_samples;
    info->pts        = pts;
    info->next       = NULL;

    if (!frame_list->list) {
        frame_list->list = info;
        frame_list->end  = info;
    } else {
        av_assert0(frame_list->end);
        frame_list->end->next = info;
        frame_list->end       = info;
    }
    frame_list->nb_frames++;
    frame_list->nb_samples += nb_samples;

    return 0;
}

/* FIXME: use directly links fifo */

typedef struct MixContext {
    const AVClass *class;       /**< class for AVOptions */
    AVFloatDSPContext *fdsp;

    int nb_inputs;              /**< number of inputs */
    int active_inputs;          /**< number of input currently active */
    int duration_mode;          /**< mode for determining duration */
    float dropout_transition;   /**< transition time when an input drops out */
    char *weights_str;          /**< string for custom weights for every input */
    int normalize;              /**< if inputs are scaled */

    int nb_channels;            /**< number of channels */
    int sample_rate;            /**< sample rate */
    int planar;
    AVAudioFifo **fifos;        /**< audio fifo for each input */
    uint8_t *input_state;       /**< current state of each input */
    float *input_scale;         /**< mixing scale factor for each input */
    float *weights;             /**< custom weights for every input */
    float weight_sum;           /**< sum of custom weights for every input */
    float *scale_norm;          /**< normalization factor for every input */
    int64_t next_pts;           /**< calculated pts for next output frame */
    FrameList *frame_list;      /**< list of frame info for the first input */
} MixContext;

#define OFFSET(x) offsetof(MixContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM
#define T AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption amix_options[] = {
    { "inputs", "Number of inputs.",
            OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, INT16_MAX, A|F },
    { "duration", "How to determine the end-of-stream.",
            OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0,  2, A|F, .unit = "duration" },
        { "longest",  "Duration of longest input.",  0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST  }, 0, 0, A|F, .unit = "duration" },
        { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, 0, 0, A|F, .unit = "duration" },
        { "first",    "Duration of first input.",    0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST    }, 0, 0, A|F, .unit = "duration" },
    { "dropout_transition", "Transition time, in seconds, for volume "
                            "renormalization when an input stream ends.",
            OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F },
    { "weights", "Set weight for each input.",
            OFFSET(weights_str), AV_OPT_TYPE_STRING, {.str="1 1"}, 0, 0, A|F|T },
    { "normalize", "Scale inputs",
            OFFSET(normalize), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A|F|T },
    { NULL }
};

AVFILTER_DEFINE_CLASS(amix);

/**
 * Update the scaling factors to apply to each input during mixing.
 *
 * This balances the full volume range between active inputs and handles
 * volume transitions when EOF is encountered on an input but mixing continues
 * with the remaining inputs.
 */
static void calculate_scales(MixContext *s, int nb_samples)
{
    float weight_sum = 0.f;
    int i;

    for (i = 0; i < s->nb_inputs; i++)
        if (s->input_state[i] & INPUT_ON)
            weight_sum += FFABS(s->weights[i]);

    for (i = 0; i < s->nb_inputs; i++) {
        if (s->input_state[i] & INPUT_ON) {
            if (s->scale_norm[i] > weight_sum / FFABS(s->weights[i])) {
                s->scale_norm[i] -= ((s->weight_sum / FFABS(s->weights[i])) / s->nb_inputs) *
                                    nb_samples / (s->dropout_transition * s->sample_rate);
                s->scale_norm[i] = FFMAX(s->scale_norm[i], weight_sum / FFABS(s->weights[i]));
            }
        }
    }

    for (i = 0; i < s->nb_inputs; i++) {
        if (s->input_state[i] & INPUT_ON) {
            if (!s->normalize)
                s->input_scale[i] = FFABS(s->weights[i]);
            else
                s->input_scale[i] = 1.0f / s->scale_norm[i] * FFSIGN(s->weights[i]);
        } else {
            s->input_scale[i] = 0.0f;
        }
    }
}

static int config_output(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    MixContext *s      = ctx->priv;
    int i;
    char buf[64];

    s->planar          = av_sample_fmt_is_planar(outlink->format);
    s->sample_rate     = outlink->sample_rate;
    outlink->time_base = (AVRational){ 1, outlink->sample_rate };
    s->next_pts        = AV_NOPTS_VALUE;

    s->frame_list = av_mallocz(sizeof(*s->frame_list));
    if (!s->frame_list)
        return AVERROR(ENOMEM);

    s->fifos = av_calloc(s->nb_inputs, sizeof(*s->fifos));
    if (!s->fifos)
        return AVERROR(ENOMEM);

    s->nb_channels = outlink->ch_layout.nb_channels;
    for (i = 0; i < s->nb_inputs; i++) {
        s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024);
        if (!s->fifos[i])
            return AVERROR(ENOMEM);
    }

    s->input_state = av_malloc(s->nb_inputs);
    if (!s->input_state)
        return AVERROR(ENOMEM);
    memset(s->input_state, INPUT_ON, s->nb_inputs);
    s->active_inputs = s->nb_inputs;

    s->input_scale = av_calloc(s->nb_inputs, sizeof(*s->input_scale));
    s->scale_norm  = av_calloc(s->nb_inputs, sizeof(*s->scale_norm));
    if (!s->input_scale || !s->scale_norm)
        return AVERROR(ENOMEM);
    for (i = 0; i < s->nb_inputs; i++)
        s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
    calculate_scales(s, 0);

    av_channel_layout_describe(&outlink->ch_layout, buf, sizeof(buf));

    av_log(ctx, AV_LOG_VERBOSE,
           "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs,
           av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf);

    return 0;
}

/**
 * Read samples from the input FIFOs, mix, and write to the output link.
 */
static int output_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    MixContext      *s = ctx->priv;
    AVFrame *out_buf, *in_buf;
    int nb_samples, ns, i;

    if (s->input_state[0] & INPUT_ON) {
        /* first input live: use the corresponding frame size */
        nb_samples = frame_list_next_frame_size(s->frame_list);
        for (i = 1; i < s->nb_inputs; i++) {
            if (s->input_state[i] & INPUT_ON) {
                ns = av_audio_fifo_size(s->fifos[i]);
                if (ns < nb_samples) {
                    if (!(s->input_state[i] & INPUT_EOF))
                        /* unclosed input with not enough samples */
                        return 0;
                    /* closed input to drain */
                    nb_samples = ns;
                }
            }
        }

        s->next_pts = frame_list_next_pts(s->frame_list);
    } else {
        /* first input closed: use the available samples */
        nb_samples = INT_MAX;
        for (i = 1; i < s->nb_inputs; i++) {
            if (s->input_state[i] & INPUT_ON) {
                ns = av_audio_fifo_size(s->fifos[i]);
                nb_samples = FFMIN(nb_samples, ns);
            }
        }
        if (nb_samples == INT_MAX) {
            ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
            return 0;
        }
    }

    frame_list_remove_samples(s->frame_list, nb_samples);

    calculate_scales(s, nb_samples);

    if (nb_samples == 0)
        return 0;

    out_buf = ff_get_audio_buffer(outlink, nb_samples);
    if (!out_buf)
        return AVERROR(ENOMEM);

    in_buf = ff_get_audio_buffer(outlink, nb_samples);
    if (!in_buf) {
        av_frame_free(&out_buf);
        return AVERROR(ENOMEM);
    }

    for (i = 0; i < s->nb_inputs; i++) {
        if (s->input_state[i] & INPUT_ON) {
            int planes, plane_size, p;

            av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data,
                               nb_samples);

            planes     = s->planar ? s->nb_channels : 1;
            plane_size = nb_samples * (s->planar ? 1 : s->nb_channels);
            plane_size = FFALIGN(plane_size, 16);

            if (out_buf->format == AV_SAMPLE_FMT_FLT ||
                out_buf->format == AV_SAMPLE_FMT_FLTP) {
                for (p = 0; p < planes; p++) {
                    s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p],
                                                (float *) in_buf->extended_data[p],
                                                s->input_scale[i], plane_size);
                }
            } else {
                for (p = 0; p < planes; p++) {
                    s->fdsp->vector_dmac_scalar((double *)out_buf->extended_data[p],
                                                (double *) in_buf->extended_data[p],
                                                s->input_scale[i], plane_size);
                }
            }
        }
    }
    av_frame_free(&in_buf);

    out_buf->pts = s->next_pts;
    out_buf->duration = av_rescale_q(out_buf->nb_samples, av_make_q(1, outlink->sample_rate),
                                     outlink->time_base);

    if (s->next_pts != AV_NOPTS_VALUE)
        s->next_pts += nb_samples;

    return ff_filter_frame(outlink, out_buf);
}

/**
 * Requests a frame, if needed, from each input link other than the first.
 */
static int request_samples(AVFilterContext *ctx, int min_samples)
{
    MixContext *s = ctx->priv;
    int i;

    av_assert0(s->nb_inputs > 1);
    if (min_samples == 1 && s->duration_mode == DURATION_FIRST)
        min_samples = av_audio_fifo_size(s->fifos[0]);

    for (i = 1; i < s->nb_inputs; i++) {
        if (!(s->input_state[i] & INPUT_ON) ||
             (s->input_state[i] & INPUT_EOF))
            continue;
        if (av_audio_fifo_size(s->fifos[i]) >= min_samples)
            continue;
        ff_inlink_request_frame(ctx->inputs[i]);
        return 0;
    }
    return output_frame(ctx->outputs[0]);
}

/**
 * Calculates the number of active inputs and determines EOF based on the
 * duration option.
 *
 * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop.
 */
static int calc_active_inputs(MixContext *s)
{
    int i;
    int active_inputs = 0;
    for (i = 0; i < s->nb_inputs; i++)
        active_inputs += !!(s->input_state[i] & INPUT_ON);
    s->active_inputs = active_inputs;

    if (!active_inputs ||
        (s->duration_mode == DURATION_FIRST && !(s->input_state[0] & INPUT_ON)) ||
        (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs))
        return AVERROR_EOF;
    return 0;
}

static int activate(AVFilterContext *ctx)
{
    AVFilterLink *outlink = ctx->outputs[0];
    MixContext *s = ctx->priv;
    AVFrame *buf = NULL;
    int i, ret;

    FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);

    for (i = 0; i < s->nb_inputs; i++) {
        AVFilterLink *inlink = ctx->inputs[i];

        if ((ret = ff_inlink_consume_frame(ctx->inputs[i], &buf)) > 0) {
            if (i == 0) {
                int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
                                           outlink->time_base);
                ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts);
                if (ret < 0) {
                    av_frame_free(&buf);
                    return ret;
                }
            }

            ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
                                      buf->nb_samples);
            if (ret < 0) {
                av_frame_free(&buf);
                return ret;
            }

            av_frame_free(&buf);

            ret = output_frame(outlink);
            if (ret < 0)
                return ret;
        }
    }

    for (i = 0; i < s->nb_inputs; i++) {
        int64_t pts;
        int status;

        if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
            if (status == AVERROR_EOF) {
                s->input_state[i] |= INPUT_EOF;
                if (av_audio_fifo_size(s->fifos[i]) == 0) {
                    s->input_state[i] &= ~INPUT_ON;
                    if (s->nb_inputs == 1) {
                        ff_outlink_set_status(outlink, status, pts);
                        return 0;
                    }
                }
            }
        }
    }

    if (calc_active_inputs(s)) {
        ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);
        return 0;
    }

    if (ff_outlink_frame_wanted(outlink)) {
        int wanted_samples;

        if (!(s->input_state[0] & INPUT_ON))
            return request_samples(ctx, 1);

        if (s->frame_list->nb_frames == 0) {
            ff_inlink_request_frame(ctx->inputs[0]);
            return 0;
        }
        av_assert0(s->frame_list->nb_frames > 0);

        wanted_samples = frame_list_next_frame_size(s->frame_list);

        return request_samples(ctx, wanted_samples);
    }

    return 0;
}

static void parse_weights(AVFilterContext *ctx)
{
    MixContext *s = ctx->priv;
    float last_weight = 1.f;
    char *p;
    int i;

    s->weight_sum = 0.f;
    p = s->weights_str;
    for (i = 0; i < s->nb_inputs; i++) {
        last_weight = av_strtod(p, &p);
        s->weights[i] = last_weight;
        s->weight_sum += FFABS(last_weight);
        if (p && *p) {
            p++;
        } else {
            i++;
            break;
        }
    }

    for (; i < s->nb_inputs; i++) {
        s->weights[i] = last_weight;
        s->weight_sum += FFABS(last_weight);
    }
}

static av_cold int init(AVFilterContext *ctx)
{
    MixContext *s = ctx->priv;
    int i, ret;

    for (i = 0; i < s->nb_inputs; i++) {
        AVFilterPad pad = { 0 };

        pad.type           = AVMEDIA_TYPE_AUDIO;
        pad.name           = av_asprintf("input%d", i);
        if (!pad.name)
            return AVERROR(ENOMEM);

        if ((ret = ff_append_inpad_free_name(ctx, &pad)) < 0)
            return ret;
    }

    s->fdsp = avpriv_float_dsp_alloc(0);
    if (!s->fdsp)
        return AVERROR(ENOMEM);

    s->weights = av_calloc(s->nb_inputs, sizeof(*s->weights));
    if (!s->weights)
        return AVERROR(ENOMEM);

    parse_weights(ctx);

    return 0;
}

static av_cold void uninit(AVFilterContext *ctx)
{
    int i;
    MixContext *s = ctx->priv;

    if (s->fifos) {
        for (i = 0; i < s->nb_inputs; i++)
            av_audio_fifo_free(s->fifos[i]);
        av_freep(&s->fifos);
    }
    frame_list_clear(s->frame_list);
    av_freep(&s->frame_list);
    av_freep(&s->input_state);
    av_freep(&s->input_scale);
    av_freep(&s->scale_norm);
    av_freep(&s->weights);
    av_freep(&s->fdsp);
}

static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
                           char *res, int res_len, int flags)
{
    MixContext *s = ctx->priv;
    int ret;

    ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
    if (ret < 0)
        return ret;

    parse_weights(ctx);
    for (int i = 0; i < s->nb_inputs; i++)
        s->scale_norm[i] = s->weight_sum / FFABS(s->weights[i]);
    calculate_scales(s, 0);

    return 0;
}

static const AVFilterPad avfilter_af_amix_outputs[] = {
    {
        .name          = "default",
        .type          = AVMEDIA_TYPE_AUDIO,
        .config_props  = config_output,
    },
};

const AVFilter ff_af_amix = {
    .name           = "amix",
    .description    = NULL_IF_CONFIG_SMALL("Audio mixing."),
    .priv_size      = sizeof(MixContext),
    .priv_class     = &amix_class,
    .init           = init,
    .uninit         = uninit,
    .activate       = activate,
    .inputs         = NULL,
    FILTER_OUTPUTS(avfilter_af_amix_outputs),
    FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
                      AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP),
    .process_command = process_command,
    .flags          = AVFILTER_FLAG_DYNAMIC_INPUTS,
};