aboutsummaryrefslogtreecommitdiff
path: root/lib/ffmpeg/libavcodec/ra144enc.c
blob: 21d38dcea0aa9c17580743c381fb11bf5507b0ba (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
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
/*
 * Real Audio 1.0 (14.4K) encoder
 * Copyright (c) 2010 Francesco Lavra <francescolavra@interfree.it>
 *
 * 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
 * Real Audio 1.0 (14.4K) encoder
 * @author Francesco Lavra <francescolavra@interfree.it>
 */

#include <float.h>

#include "avcodec.h"
#include "audio_frame_queue.h"
#include "internal.h"
#include "put_bits.h"
#include "celp_filters.h"
#include "ra144.h"


static av_cold int ra144_encode_close(AVCodecContext *avctx)
{
    RA144Context *ractx = avctx->priv_data;
    ff_lpc_end(&ractx->lpc_ctx);
    ff_af_queue_close(&ractx->afq);
#if FF_API_OLD_ENCODE_AUDIO
    av_freep(&avctx->coded_frame);
#endif
    return 0;
}


static av_cold int ra144_encode_init(AVCodecContext * avctx)
{
    RA144Context *ractx;
    int ret;

    if (avctx->channels != 1) {
        av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n",
               avctx->channels);
        return -1;
    }
    avctx->frame_size = NBLOCKS * BLOCKSIZE;
    avctx->delay      = avctx->frame_size;
    avctx->bit_rate = 8000;
    ractx = avctx->priv_data;
    ractx->lpc_coef[0] = ractx->lpc_tables[0];
    ractx->lpc_coef[1] = ractx->lpc_tables[1];
    ractx->avctx = avctx;
    ret = ff_lpc_init(&ractx->lpc_ctx, avctx->frame_size, LPC_ORDER,
                      FF_LPC_TYPE_LEVINSON);
    if (ret < 0)
        goto error;

    ff_af_queue_init(avctx, &ractx->afq);

#if FF_API_OLD_ENCODE_AUDIO
    avctx->coded_frame = avcodec_alloc_frame();
    if (!avctx->coded_frame) {
        ret = AVERROR(ENOMEM);
        goto error;
    }
#endif

    return 0;
error:
    ra144_encode_close(avctx);
    return ret;
}


/**
 * Quantize a value by searching a sorted table for the element with the
 * nearest value
 *
 * @param value value to quantize
 * @param table array containing the quantization table
 * @param size size of the quantization table
 * @return index of the quantization table corresponding to the element with the
 *         nearest value
 */
static int quantize(int value, const int16_t *table, unsigned int size)
{
    unsigned int low = 0, high = size - 1;

    while (1) {
        int index = (low + high) >> 1;
        int error = table[index] - value;

        if (index == low)
            return table[high] + error > value ? low : high;
        if (error > 0) {
            high = index;
        } else {
            low = index;
        }
    }
}


/**
 * Orthogonalize a vector to another vector
 *
 * @param v vector to orthogonalize
 * @param u vector against which orthogonalization is performed
 */
static void orthogonalize(float *v, const float *u)
{
    int i;
    float num = 0, den = 0;

    for (i = 0; i < BLOCKSIZE; i++) {
        num += v[i] * u[i];
        den += u[i] * u[i];
    }
    num /= den;
    for (i = 0; i < BLOCKSIZE; i++)
        v[i] -= num * u[i];
}


/**
 * Calculate match score and gain of an LPC-filtered vector with respect to
 * input data, possibly othogonalizing it to up to 2 other vectors
 *
 * @param work array used to calculate the filtered vector
 * @param coefs coefficients of the LPC filter
 * @param vect original vector
 * @param ortho1 first vector against which orthogonalization is performed
 * @param ortho2 second vector against which orthogonalization is performed
 * @param data input data
 * @param score pointer to variable where match score is returned
 * @param gain pointer to variable where gain is returned
 */
static void get_match_score(float *work, const float *coefs, float *vect,
                            const float *ortho1, const float *ortho2,
                            const float *data, float *score, float *gain)
{
    float c, g;
    int i;

    ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
    if (ortho1)
        orthogonalize(work, ortho1);
    if (ortho2)
        orthogonalize(work, ortho2);
    c = g = 0;
    for (i = 0; i < BLOCKSIZE; i++) {
        g += work[i] * work[i];
        c += data[i] * work[i];
    }
    if (c <= 0) {
        *score = 0;
        return;
    }
    *gain = c / g;
    *score = *gain * c;
}


/**
 * Create a vector from the adaptive codebook at a given lag value
 *
 * @param vect array where vector is stored
 * @param cb adaptive codebook
 * @param lag lag value
 */
static void create_adapt_vect(float *vect, const int16_t *cb, int lag)
{
    int i;

    cb += BUFFERSIZE - lag;
    for (i = 0; i < FFMIN(BLOCKSIZE, lag); i++)
        vect[i] = cb[i];
    if (lag < BLOCKSIZE)
        for (i = 0; i < BLOCKSIZE - lag; i++)
            vect[lag + i] = cb[i];
}


/**
 * Search the adaptive codebook for the best entry and gain and remove its
 * contribution from input data
 *
 * @param adapt_cb array from which the adaptive codebook is extracted
 * @param work array used to calculate LPC-filtered vectors
 * @param coefs coefficients of the LPC filter
 * @param data input data
 * @return index of the best entry of the adaptive codebook
 */
static int adaptive_cb_search(const int16_t *adapt_cb, float *work,
                              const float *coefs, float *data)
{
    int i, av_uninit(best_vect);
    float score, gain, best_score, av_uninit(best_gain);
    float exc[BLOCKSIZE];

    gain = best_score = 0;
    for (i = BLOCKSIZE / 2; i <= BUFFERSIZE; i++) {
        create_adapt_vect(exc, adapt_cb, i);
        get_match_score(work, coefs, exc, NULL, NULL, data, &score, &gain);
        if (score > best_score) {
            best_score = score;
            best_vect = i;
            best_gain = gain;
        }
    }
    if (!best_score)
        return 0;

    /**
     * Re-calculate the filtered vector from the vector with maximum match score
     * and remove its contribution from input data.
     */
    create_adapt_vect(exc, adapt_cb, best_vect);
    ff_celp_lp_synthesis_filterf(work, coefs, exc, BLOCKSIZE, LPC_ORDER);
    for (i = 0; i < BLOCKSIZE; i++)
        data[i] -= best_gain * work[i];
    return best_vect - BLOCKSIZE / 2 + 1;
}


/**
 * Find the best vector of a fixed codebook by applying an LPC filter to
 * codebook entries, possibly othogonalizing them to up to 2 other vectors and
 * matching the results with input data
 *
 * @param work array used to calculate the filtered vectors
 * @param coefs coefficients of the LPC filter
 * @param cb fixed codebook
 * @param ortho1 first vector against which orthogonalization is performed
 * @param ortho2 second vector against which orthogonalization is performed
 * @param data input data
 * @param idx pointer to variable where the index of the best codebook entry is
 *        returned
 * @param gain pointer to variable where the gain of the best codebook entry is
 *        returned
 */
static void find_best_vect(float *work, const float *coefs,
                           const int8_t cb[][BLOCKSIZE], const float *ortho1,
                           const float *ortho2, float *data, int *idx,
                           float *gain)
{
    int i, j;
    float g, score, best_score;
    float vect[BLOCKSIZE];

    *idx = *gain = best_score = 0;
    for (i = 0; i < FIXED_CB_SIZE; i++) {
        for (j = 0; j < BLOCKSIZE; j++)
            vect[j] = cb[i][j];
        get_match_score(work, coefs, vect, ortho1, ortho2, data, &score, &g);
        if (score > best_score) {
            best_score = score;
            *idx = i;
            *gain = g;
        }
    }
}


/**
 * Search the two fixed codebooks for the best entry and gain
 *
 * @param work array used to calculate LPC-filtered vectors
 * @param coefs coefficients of the LPC filter
 * @param data input data
 * @param cba_idx index of the best entry of the adaptive codebook
 * @param cb1_idx pointer to variable where the index of the best entry of the
 *        first fixed codebook is returned
 * @param cb2_idx pointer to variable where the index of the best entry of the
 *        second fixed codebook is returned
 */
static void fixed_cb_search(float *work, const float *coefs, float *data,
                            int cba_idx, int *cb1_idx, int *cb2_idx)
{
    int i, ortho_cb1;
    float gain;
    float cba_vect[BLOCKSIZE], cb1_vect[BLOCKSIZE];
    float vect[BLOCKSIZE];

    /**
     * The filtered vector from the adaptive codebook can be retrieved from
     * work, because this function is called just after adaptive_cb_search().
     */
    if (cba_idx)
        memcpy(cba_vect, work, sizeof(cba_vect));

    find_best_vect(work, coefs, ff_cb1_vects, cba_idx ? cba_vect : NULL, NULL,
                   data, cb1_idx, &gain);

    /**
     * Re-calculate the filtered vector from the vector with maximum match score
     * and remove its contribution from input data.
     */
    if (gain) {
        for (i = 0; i < BLOCKSIZE; i++)
            vect[i] = ff_cb1_vects[*cb1_idx][i];
        ff_celp_lp_synthesis_filterf(work, coefs, vect, BLOCKSIZE, LPC_ORDER);
        if (cba_idx)
            orthogonalize(work, cba_vect);
        for (i = 0; i < BLOCKSIZE; i++)
            data[i] -= gain * work[i];
        memcpy(cb1_vect, work, sizeof(cb1_vect));
        ortho_cb1 = 1;
    } else
        ortho_cb1 = 0;

    find_best_vect(work, coefs, ff_cb2_vects, cba_idx ? cba_vect : NULL,
                   ortho_cb1 ? cb1_vect : NULL, data, cb2_idx, &gain);
}


/**
 * Encode a subblock of the current frame
 *
 * @param ractx encoder context
 * @param sblock_data input data of the subblock
 * @param lpc_coefs coefficients of the LPC filter
 * @param rms RMS of the reflection coefficients
 * @param pb pointer to PutBitContext of the current frame
 */
static void ra144_encode_subblock(RA144Context *ractx,
                                  const int16_t *sblock_data,
                                  const int16_t *lpc_coefs, unsigned int rms,
                                  PutBitContext *pb)
{
    float data[BLOCKSIZE] = { 0 }, work[LPC_ORDER + BLOCKSIZE];
    float coefs[LPC_ORDER];
    float zero[BLOCKSIZE], cba[BLOCKSIZE], cb1[BLOCKSIZE], cb2[BLOCKSIZE];
    int16_t cba_vect[BLOCKSIZE];
    int cba_idx, cb1_idx, cb2_idx, gain;
    int i, n;
    unsigned m[3];
    float g[3];
    float error, best_error;

    for (i = 0; i < LPC_ORDER; i++) {
        work[i] = ractx->curr_sblock[BLOCKSIZE + i];
        coefs[i] = lpc_coefs[i] * (1/4096.0);
    }

    /**
     * Calculate the zero-input response of the LPC filter and subtract it from
     * input data.
     */
    ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, data, BLOCKSIZE,
                                 LPC_ORDER);
    for (i = 0; i < BLOCKSIZE; i++) {
        zero[i] = work[LPC_ORDER + i];
        data[i] = sblock_data[i] - zero[i];
    }

    /**
     * Codebook search is performed without taking into account the contribution
     * of the previous subblock, since it has been just subtracted from input
     * data.
     */
    memset(work, 0, LPC_ORDER * sizeof(*work));

    cba_idx = adaptive_cb_search(ractx->adapt_cb, work + LPC_ORDER, coefs,
                                 data);
    if (cba_idx) {
        /**
         * The filtered vector from the adaptive codebook can be retrieved from
         * work, see implementation of adaptive_cb_search().
         */
        memcpy(cba, work + LPC_ORDER, sizeof(cba));

        ff_copy_and_dup(cba_vect, ractx->adapt_cb, cba_idx + BLOCKSIZE / 2 - 1);
        m[0] = (ff_irms(cba_vect) * rms) >> 12;
    }
    fixed_cb_search(work + LPC_ORDER, coefs, data, cba_idx, &cb1_idx, &cb2_idx);
    for (i = 0; i < BLOCKSIZE; i++) {
        cb1[i] = ff_cb1_vects[cb1_idx][i];
        cb2[i] = ff_cb2_vects[cb2_idx][i];
    }
    ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb1, BLOCKSIZE,
                                 LPC_ORDER);
    memcpy(cb1, work + LPC_ORDER, sizeof(cb1));
    m[1] = (ff_cb1_base[cb1_idx] * rms) >> 8;
    ff_celp_lp_synthesis_filterf(work + LPC_ORDER, coefs, cb2, BLOCKSIZE,
                                 LPC_ORDER);
    memcpy(cb2, work + LPC_ORDER, sizeof(cb2));
    m[2] = (ff_cb2_base[cb2_idx] * rms) >> 8;
    best_error = FLT_MAX;
    gain = 0;
    for (n = 0; n < 256; n++) {
        g[1] = ((ff_gain_val_tab[n][1] * m[1]) >> ff_gain_exp_tab[n]) *
               (1/4096.0);
        g[2] = ((ff_gain_val_tab[n][2] * m[2]) >> ff_gain_exp_tab[n]) *
               (1/4096.0);
        error = 0;
        if (cba_idx) {
            g[0] = ((ff_gain_val_tab[n][0] * m[0]) >> ff_gain_exp_tab[n]) *
                   (1/4096.0);
            for (i = 0; i < BLOCKSIZE; i++) {
                data[i] = zero[i] + g[0] * cba[i] + g[1] * cb1[i] +
                          g[2] * cb2[i];
                error += (data[i] - sblock_data[i]) *
                         (data[i] - sblock_data[i]);
            }
        } else {
            for (i = 0; i < BLOCKSIZE; i++) {
                data[i] = zero[i] + g[1] * cb1[i] + g[2] * cb2[i];
                error += (data[i] - sblock_data[i]) *
                         (data[i] - sblock_data[i]);
            }
        }
        if (error < best_error) {
            best_error = error;
            gain = n;
        }
    }
    put_bits(pb, 7, cba_idx);
    put_bits(pb, 8, gain);
    put_bits(pb, 7, cb1_idx);
    put_bits(pb, 7, cb2_idx);
    ff_subblock_synthesis(ractx, lpc_coefs, cba_idx, cb1_idx, cb2_idx, rms,
                          gain);
}


static int ra144_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                              const AVFrame *frame, int *got_packet_ptr)
{
    static const uint8_t sizes[LPC_ORDER] = {64, 32, 32, 16, 16, 8, 8, 8, 8, 4};
    static const uint8_t bit_sizes[LPC_ORDER] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2};
    RA144Context *ractx = avctx->priv_data;
    PutBitContext pb;
    int32_t lpc_data[NBLOCKS * BLOCKSIZE];
    int32_t lpc_coefs[LPC_ORDER][MAX_LPC_ORDER];
    int shift[LPC_ORDER];
    int16_t block_coefs[NBLOCKS][LPC_ORDER];
    int lpc_refl[LPC_ORDER];    /**< reflection coefficients of the frame */
    unsigned int refl_rms[NBLOCKS]; /**< RMS of the reflection coefficients */
    const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
    int energy = 0;
    int i, idx, ret;

    if (ractx->last_frame)
        return 0;

    if ((ret = ff_alloc_packet2(avctx, avpkt, FRAMESIZE)) < 0)
        return ret;

    /**
     * Since the LPC coefficients are calculated on a frame centered over the
     * fourth subframe, to encode a given frame, data from the next frame is
     * needed. In each call to this function, the previous frame (whose data are
     * saved in the encoder context) is encoded, and data from the current frame
     * are saved in the encoder context to be used in the next function call.
     */
    for (i = 0; i < (2 * BLOCKSIZE + BLOCKSIZE / 2); i++) {
        lpc_data[i] = ractx->curr_block[BLOCKSIZE + BLOCKSIZE / 2 + i];
        energy += (lpc_data[i] * lpc_data[i]) >> 4;
    }
    if (frame) {
        int j;
        for (j = 0; j < frame->nb_samples && i < NBLOCKS * BLOCKSIZE; i++, j++) {
            lpc_data[i] = samples[j] >> 2;
            energy += (lpc_data[i] * lpc_data[i]) >> 4;
        }
    }
    if (i < NBLOCKS * BLOCKSIZE)
        memset(&lpc_data[i], 0, (NBLOCKS * BLOCKSIZE - i) * sizeof(*lpc_data));
    energy = ff_energy_tab[quantize(ff_t_sqrt(energy >> 5) >> 10, ff_energy_tab,
                                    32)];

    ff_lpc_calc_coefs(&ractx->lpc_ctx, lpc_data, NBLOCKS * BLOCKSIZE, LPC_ORDER,
                      LPC_ORDER, 16, lpc_coefs, shift, FF_LPC_TYPE_LEVINSON,
                      0, ORDER_METHOD_EST, 12, 0);
    for (i = 0; i < LPC_ORDER; i++)
        block_coefs[NBLOCKS - 1][i] = -(lpc_coefs[LPC_ORDER - 1][i] <<
                                        (12 - shift[LPC_ORDER - 1]));

    /**
     * TODO: apply perceptual weighting of the input speech through bandwidth
     * expansion of the LPC filter.
     */

    if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
        /**
         * The filter is unstable: use the coefficients of the previous frame.
         */
        ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[1]);
        if (ff_eval_refl(lpc_refl, block_coefs[NBLOCKS - 1], avctx)) {
            /* the filter is still unstable. set reflection coeffs to zero. */
            memset(lpc_refl, 0, sizeof(lpc_refl));
        }
    }
    init_put_bits(&pb, avpkt->data, avpkt->size);
    for (i = 0; i < LPC_ORDER; i++) {
        idx = quantize(lpc_refl[i], ff_lpc_refl_cb[i], sizes[i]);
        put_bits(&pb, bit_sizes[i], idx);
        lpc_refl[i] = ff_lpc_refl_cb[i][idx];
    }
    ractx->lpc_refl_rms[0] = ff_rms(lpc_refl);
    ff_eval_coefs(ractx->lpc_coef[0], lpc_refl);
    refl_rms[0] = ff_interp(ractx, block_coefs[0], 1, 1, ractx->old_energy);
    refl_rms[1] = ff_interp(ractx, block_coefs[1], 2,
                            energy <= ractx->old_energy,
                            ff_t_sqrt(energy * ractx->old_energy) >> 12);
    refl_rms[2] = ff_interp(ractx, block_coefs[2], 3, 0, energy);
    refl_rms[3] = ff_rescale_rms(ractx->lpc_refl_rms[0], energy);
    ff_int_to_int16(block_coefs[NBLOCKS - 1], ractx->lpc_coef[0]);
    put_bits(&pb, 5, quantize(energy, ff_energy_tab, 32));
    for (i = 0; i < NBLOCKS; i++)
        ra144_encode_subblock(ractx, ractx->curr_block + i * BLOCKSIZE,
                              block_coefs[i], refl_rms[i], &pb);
    flush_put_bits(&pb);
    ractx->old_energy = energy;
    ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0];
    FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]);

    /* copy input samples to current block for processing in next call */
    i = 0;
    if (frame) {
        for (; i < frame->nb_samples; i++)
            ractx->curr_block[i] = samples[i] >> 2;

        if ((ret = ff_af_queue_add(&ractx->afq, frame)) < 0)
            return ret;
    } else
        ractx->last_frame = 1;
    memset(&ractx->curr_block[i], 0,
           (NBLOCKS * BLOCKSIZE - i) * sizeof(*ractx->curr_block));

    /* Get the next frame pts/duration */
    ff_af_queue_remove(&ractx->afq, avctx->frame_size, &avpkt->pts,
                       &avpkt->duration);

    avpkt->size = FRAMESIZE;
    *got_packet_ptr = 1;
    return 0;
}


AVCodec ff_ra_144_encoder = {
    .name           = "real_144",
    .type           = AVMEDIA_TYPE_AUDIO,
    .id             = AV_CODEC_ID_RA_144,
    .priv_data_size = sizeof(RA144Context),
    .init           = ra144_encode_init,
    .encode2        = ra144_encode_frame,
    .close          = ra144_encode_close,
    .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_SMALL_LAST_FRAME,
    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                     AV_SAMPLE_FMT_NONE },
    .supported_samplerates = (const int[]){ 8000, 0 },
    .long_name      = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"),
};