summaryrefslogtreecommitdiffstats
path: root/libavresample/resample.c
blob: 01fc50014db31b711153da89cf55e9402302d10d (plain)
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
/*
 * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
 * 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
 */

#include "libavutil/common.h"
#include "libavutil/libm.h"
#include "libavutil/log.h"
#include "internal.h"
#include "resample.h"
#include "audio_data.h"


/* double template */
#define CONFIG_RESAMPLE_DBL
#include "resample_template.c"
#undef CONFIG_RESAMPLE_DBL

/* float template */
#define CONFIG_RESAMPLE_FLT
#include "resample_template.c"
#undef CONFIG_RESAMPLE_FLT

/* s32 template */
#define CONFIG_RESAMPLE_S32
#include "resample_template.c"
#undef CONFIG_RESAMPLE_S32

/* s16 template */
#include "resample_template.c"


/* 0th order modified bessel function of the first kind. */
static double bessel(double x)
{
    double v     = 1;
    double lastv = 0;
    double t     = 1;
    int i;

    x = x * x / 4;
    for (i = 1; v != lastv; i++) {
        lastv = v;
        t    *= x / (i * i);
        v    += t;
    }
    return v;
}

/* Build a polyphase filterbank. */
static int build_filter(ResampleContext *c, double factor)
{
    int ph, i;
    double x, y, w;
    double *tab;
    int tap_count    = c->filter_length;
    int phase_count  = 1 << c->phase_shift;
    const int center = (tap_count - 1) / 2;

    tab = av_malloc(tap_count * sizeof(*tab));
    if (!tab)
        return AVERROR(ENOMEM);

    for (ph = 0; ph < phase_count; ph++) {
        double norm = 0;
        for (i = 0; i < tap_count; i++) {
            x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
            if (x == 0) y = 1.0;
            else        y = sin(x) / x;
            switch (c->filter_type) {
            case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
                const float d = -0.5; //first order derivative = -0.5
                x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
                if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * (                -x*x + x*x*x);
                else         y =                           d * (-4 + 8 * x - 5 * x*x + x*x*x);
                break;
            }
            case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
                w  = 2.0 * x / (factor * tap_count) + M_PI;
                y *= 0.3635819 - 0.4891775 * cos(    w) +
                                 0.1365995 * cos(2 * w) -
                                 0.0106411 * cos(3 * w);
                break;
            case AV_RESAMPLE_FILTER_TYPE_KAISER:
                w  = 2.0 * x / (factor * tap_count * M_PI);
                y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
                break;
            }

            tab[i] = y;
            norm  += y;
        }
        /* normalize so that an uniform color remains the same */
        for (i = 0; i < tap_count; i++)
            tab[i] = tab[i] / norm;

        c->set_filter(c->filter_bank, tab, ph, tap_count);
    }

    av_free(tab);
    return 0;
}

ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
{
    ResampleContext *c;
    int out_rate    = avr->out_sample_rate;
    int in_rate     = avr->in_sample_rate;
    double factor   = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
    int phase_count = 1 << avr->phase_shift;
    int felem_size;

    if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
        avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
        avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
        avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
        av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
               "resampling: %s\n",
               av_get_sample_fmt_name(avr->internal_sample_fmt));
        return NULL;
    }
    c = av_mallocz(sizeof(*c));
    if (!c)
        return NULL;

    c->avr           = avr;
    c->phase_shift   = avr->phase_shift;
    c->phase_mask    = phase_count - 1;
    c->linear        = avr->linear_interp;
    c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
    c->filter_type   = avr->filter_type;
    c->kaiser_beta   = avr->kaiser_beta;

    switch (avr->internal_sample_fmt) {
    case AV_SAMPLE_FMT_DBLP:
        c->resample_one  = c->linear ? resample_linear_dbl : resample_one_dbl;
        c->resample_nearest = resample_nearest_dbl;
        c->set_filter    = set_filter_dbl;
        break;
    case AV_SAMPLE_FMT_FLTP:
        c->resample_one  = c->linear ? resample_linear_flt : resample_one_flt;
        c->resample_nearest = resample_nearest_flt;
        c->set_filter    = set_filter_flt;
        break;
    case AV_SAMPLE_FMT_S32P:
        c->resample_one  = c->linear ? resample_linear_s32 : resample_one_s32;
        c->resample_nearest = resample_nearest_s32;
        c->set_filter    = set_filter_s32;
        break;
    case AV_SAMPLE_FMT_S16P:
        c->resample_one  = c->linear ? resample_linear_s16 : resample_one_s16;
        c->resample_nearest = resample_nearest_s16;
        c->set_filter    = set_filter_s16;
        break;
    }

    if (ARCH_AARCH64)
        ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
    if (ARCH_ARM)
        ff_audio_resample_init_arm(c, avr->internal_sample_fmt);

    felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
    c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
    if (!c->filter_bank)
        goto error;

    if (build_filter(c, factor) < 0)
        goto error;

    memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
           c->filter_bank, (c->filter_length - 1) * felem_size);
    memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
           &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);

    c->compensation_distance = 0;
    if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
                   in_rate * (int64_t)phase_count, INT32_MAX / 2))
        goto error;
    c->ideal_dst_incr = c->dst_incr;

    c->padding_size   = (c->filter_length - 1) / 2;
    c->initial_padding_filled = 0;
    c->index = 0;
    c->frac  = 0;

    /* allocate internal buffer */
    c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
                                    avr->internal_sample_fmt,
                                    "resample buffer");
    if (!c->buffer)
        goto error;
    c->buffer->nb_samples      = c->padding_size;
    c->initial_padding_samples = c->padding_size;

    av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
           av_get_sample_fmt_name(avr->internal_sample_fmt),
           avr->in_sample_rate, avr->out_sample_rate);

    return c;

error:
    ff_audio_data_free(&c->buffer);
    av_free(c->filter_bank);
    av_free(c);
    return NULL;
}

void ff_audio_resample_free(ResampleContext **c)
{
    if (!*c)
        return;
    ff_audio_data_free(&(*c)->buffer);
    av_free((*c)->filter_bank);
    av_freep(c);
}

int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
                                int compensation_distance)
{
    ResampleContext *c;
    AudioData *fifo_buf = NULL;
    int ret = 0;

    if (compensation_distance < 0)
        return AVERROR(EINVAL);
    if (!compensation_distance && sample_delta)
        return AVERROR(EINVAL);

    if (!avr->resample_needed) {
#if FF_API_RESAMPLE_CLOSE_OPEN
        /* if resampling was not enabled previously, re-initialize the
           AVAudioResampleContext and force resampling */
        int fifo_samples;
        int restore_matrix = 0;
        double matrix[AVRESAMPLE_MAX_CHANNELS * AVRESAMPLE_MAX_CHANNELS] = { 0 };

        /* buffer any remaining samples in the output FIFO before closing */
        fifo_samples = av_audio_fifo_size(avr->out_fifo);
        if (fifo_samples > 0) {
            fifo_buf = ff_audio_data_alloc(avr->out_channels, fifo_samples,
                                           avr->out_sample_fmt, NULL);
            if (!fifo_buf)
                return AVERROR(EINVAL);
            ret = ff_audio_data_read_from_fifo(avr->out_fifo, fifo_buf,
                                               fifo_samples);
            if (ret < 0)
                goto reinit_fail;
        }
        /* save the channel mixing matrix */
        if (avr->am) {
            ret = avresample_get_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
            if (ret < 0)
                goto reinit_fail;
            restore_matrix = 1;
        }

        /* close the AVAudioResampleContext */
        avresample_close(avr);

        avr->force_resampling = 1;

        /* restore the channel mixing matrix */
        if (restore_matrix) {
            ret = avresample_set_matrix(avr, matrix, AVRESAMPLE_MAX_CHANNELS);
            if (ret < 0)
                goto reinit_fail;
        }

        /* re-open the AVAudioResampleContext */
        ret = avresample_open(avr);
        if (ret < 0)
            goto reinit_fail;

        /* restore buffered samples to the output FIFO */
        if (fifo_samples > 0) {
            ret = ff_audio_data_add_to_fifo(avr->out_fifo, fifo_buf, 0,
                                            fifo_samples);
            if (ret < 0)
                goto reinit_fail;
            ff_audio_data_free(&fifo_buf);
        }
#else
        av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
        return AVERROR(EINVAL);
#endif
    }
    c = avr->resample;
    c->compensation_distance = compensation_distance;
    if (compensation_distance) {
        c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
                      (int64_t)sample_delta / compensation_distance;
    } else {
        c->dst_incr = c->ideal_dst_incr;
    }
    return 0;

reinit_fail:
    ff_audio_data_free(&fifo_buf);
    return ret;
}

static int resample(ResampleContext *c, void *dst, const void *src,
                    int *consumed, int src_size, int dst_size, int update_ctx,
                    int nearest_neighbour)
{
    int dst_index;
    unsigned int index = c->index;
    int frac          = c->frac;
    int dst_incr_frac = c->dst_incr % c->src_incr;
    int dst_incr      = c->dst_incr / c->src_incr;
    int compensation_distance = c->compensation_distance;

    if (!dst != !src)
        return AVERROR(EINVAL);

    if (nearest_neighbour) {
        uint64_t index2 = ((uint64_t)index) << 32;
        int64_t incr   = (1LL << 32) * c->dst_incr / c->src_incr;
        dst_size       = FFMIN(dst_size,
                               (src_size-1-index) * (int64_t)c->src_incr /
                               c->dst_incr);

        if (dst) {
            for(dst_index = 0; dst_index < dst_size; dst_index++) {
                c->resample_nearest(dst, dst_index, src, index2 >> 32);
                index2 += incr;
            }
        } else {
            dst_index = dst_size;
        }
        index += dst_index * dst_incr;
        index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
        frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
    } else {
        for (dst_index = 0; dst_index < dst_size; dst_index++) {
            int sample_index = index >> c->phase_shift;

            if (sample_index + c->filter_length > src_size)
                break;

            if (dst)
                c->resample_one(c, dst, dst_index, src, index, frac);

            frac  += dst_incr_frac;
            index += dst_incr;
            if (frac >= c->src_incr) {
                frac -= c->src_incr;
                index++;
            }
            if (dst_index + 1 == compensation_distance) {
                compensation_distance = 0;
                dst_incr_frac = c->ideal_dst_incr % c->src_incr;
                dst_incr      = c->ideal_dst_incr / c->src_incr;
            }
        }
    }
    if (consumed)
        *consumed = index >> c->phase_shift;

    if (update_ctx) {
        index &= c->phase_mask;

        if (compensation_distance) {
            compensation_distance -= dst_index;
            if (compensation_distance <= 0)
                return AVERROR_BUG;
        }
        c->frac     = frac;
        c->index    = index;
        c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
        c->compensation_distance = compensation_distance;
    }

    return dst_index;
}

int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
{
    int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
    int ret = AVERROR(EINVAL);
    int nearest_neighbour = (c->compensation_distance == 0 &&
                             c->filter_length == 1 &&
                             c->phase_shift == 0);

    in_samples  = src ? src->nb_samples : 0;
    in_leftover = c->buffer->nb_samples;

    /* add input samples to the internal buffer */
    if (src) {
        ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
        if (ret < 0)
            return ret;
    } else if (in_leftover <= c->final_padding_samples) {
        /* no remaining samples to flush */
        return 0;
    }

    if (!c->initial_padding_filled) {
        int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
        int i;

        if (src && c->buffer->nb_samples < 2 * c->padding_size)
            return 0;

        for (i = 0; i < c->padding_size; i++)
            for (ch = 0; ch < c->buffer->channels; ch++) {
                if (c->buffer->nb_samples > 2 * c->padding_size - i) {
                    memcpy(c->buffer->data[ch] + bps * i,
                           c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
                } else {
                    memset(c->buffer->data[ch] + bps * i, 0, bps);
                }
            }
        c->initial_padding_filled = 1;
    }

    if (!src && !c->final_padding_filled) {
        int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
        int i;

        ret = ff_audio_data_realloc(c->buffer,
                                    FFMAX(in_samples, in_leftover) +
                                    c->padding_size);
        if (ret < 0) {
            av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
            return AVERROR(ENOMEM);
        }

        for (i = 0; i < c->padding_size; i++)
            for (ch = 0; ch < c->buffer->channels; ch++) {
                if (in_leftover > i) {
                    memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
                           c->buffer->data[ch] + bps * (in_leftover - i - 1),
                           bps);
                } else {
                    memset(c->buffer->data[ch] + bps * (in_leftover + i),
                           0, bps);
                }
            }
        c->buffer->nb_samples   += c->padding_size;
        c->final_padding_samples = c->padding_size;
        c->final_padding_filled  = 1;
    }


    /* calculate output size and reallocate output buffer if needed */
    /* TODO: try to calculate this without the dummy resample() run */
    if (!dst->read_only && dst->allow_realloc) {
        out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
                               INT_MAX, 0, nearest_neighbour);
        ret = ff_audio_data_realloc(dst, out_samples);
        if (ret < 0) {
            av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
            return ret;
        }
    }

    /* resample each channel plane */
    for (ch = 0; ch < c->buffer->channels; ch++) {
        out_samples = resample(c, (void *)dst->data[ch],
                               (const void *)c->buffer->data[ch], &consumed,
                               c->buffer->nb_samples, dst->allocated_samples,
                               ch + 1 == c->buffer->channels, nearest_neighbour);
    }
    if (out_samples < 0) {
        av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
        return out_samples;
    }

    /* drain consumed samples from the internal buffer */
    ff_audio_data_drain(c->buffer, consumed);
    c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);

    av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",
            in_samples, in_leftover, out_samples, c->buffer->nb_samples);

    dst->nb_samples = out_samples;
    return 0;
}

int avresample_get_delay(AVAudioResampleContext *avr)
{
    ResampleContext *c = avr->resample;

    if (!avr->resample_needed || !avr->resample)
        return 0;

    return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
}
OpenPOWER on IntegriCloud