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1
/**
2
 * ALAC audio encoder
3
 * Copyright (c) 2008  Jaikrishnan Menon <realityman@gmx.net>
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.
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 *
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 * 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
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21

    
22
#include "avcodec.h"
23
#include "get_bits.h"
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#include "put_bits.h"
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#include "dsputil.h"
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#include "lpc.h"
27
#include "mathops.h"
28

    
29
#define DEFAULT_FRAME_SIZE        4096
30
#define DEFAULT_SAMPLE_SIZE       16
31
#define MAX_CHANNELS              8
32
#define ALAC_EXTRADATA_SIZE       36
33
#define ALAC_FRAME_HEADER_SIZE    55
34
#define ALAC_FRAME_FOOTER_SIZE    3
35

    
36
#define ALAC_ESCAPE_CODE          0x1FF
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#define ALAC_MAX_LPC_ORDER        30
38
#define DEFAULT_MAX_PRED_ORDER    6
39
#define DEFAULT_MIN_PRED_ORDER    4
40
#define ALAC_MAX_LPC_PRECISION    9
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#define ALAC_MAX_LPC_SHIFT        9
42

    
43
#define ALAC_CHMODE_LEFT_RIGHT    0
44
#define ALAC_CHMODE_LEFT_SIDE     1
45
#define ALAC_CHMODE_RIGHT_SIDE    2
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#define ALAC_CHMODE_MID_SIDE      3
47

    
48
typedef struct RiceContext {
49
    int history_mult;
50
    int initial_history;
51
    int k_modifier;
52
    int rice_modifier;
53
} RiceContext;
54

    
55
typedef struct LPCContext {
56
    int lpc_order;
57
    int lpc_coeff[ALAC_MAX_LPC_ORDER+1];
58
    int lpc_quant;
59
} LPCContext;
60

    
61
typedef struct AlacEncodeContext {
62
    int compression_level;
63
    int min_prediction_order;
64
    int max_prediction_order;
65
    int max_coded_frame_size;
66
    int write_sample_size;
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    int32_t sample_buf[MAX_CHANNELS][DEFAULT_FRAME_SIZE];
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    int32_t predictor_buf[DEFAULT_FRAME_SIZE];
69
    int interlacing_shift;
70
    int interlacing_leftweight;
71
    PutBitContext pbctx;
72
    RiceContext rc;
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    LPCContext lpc[MAX_CHANNELS];
74
    DSPContext dspctx;
75
    AVCodecContext *avctx;
76
} AlacEncodeContext;
77

    
78

    
79
static void init_sample_buffers(AlacEncodeContext *s, int16_t *input_samples)
80
{
81
    int ch, i;
82

    
83
    for(ch=0;ch<s->avctx->channels;ch++) {
84
        int16_t *sptr = input_samples + ch;
85
        for(i=0;i<s->avctx->frame_size;i++) {
86
            s->sample_buf[ch][i] = *sptr;
87
            sptr += s->avctx->channels;
88
        }
89
    }
90
}
91

    
92
static void encode_scalar(AlacEncodeContext *s, int x, int k, int write_sample_size)
93
{
94
    int divisor, q, r;
95

    
96
    k = FFMIN(k, s->rc.k_modifier);
97
    divisor = (1<<k) - 1;
98
    q = x / divisor;
99
    r = x % divisor;
100

    
101
    if(q > 8) {
102
        // write escape code and sample value directly
103
        put_bits(&s->pbctx, 9, ALAC_ESCAPE_CODE);
104
        put_bits(&s->pbctx, write_sample_size, x);
105
    } else {
106
        if(q)
107
            put_bits(&s->pbctx, q, (1<<q) - 1);
108
        put_bits(&s->pbctx, 1, 0);
109

    
110
        if(k != 1) {
111
            if(r > 0)
112
                put_bits(&s->pbctx, k, r+1);
113
            else
114
                put_bits(&s->pbctx, k-1, 0);
115
        }
116
    }
117
}
118

    
119
static void write_frame_header(AlacEncodeContext *s, int is_verbatim)
120
{
121
    put_bits(&s->pbctx, 3,  s->avctx->channels-1);          // No. of channels -1
122
    put_bits(&s->pbctx, 16, 0);                             // Seems to be zero
123
    put_bits(&s->pbctx, 1,  1);                             // Sample count is in the header
124
    put_bits(&s->pbctx, 2,  0);                             // FIXME: Wasted bytes field
125
    put_bits(&s->pbctx, 1,  is_verbatim);                   // Audio block is verbatim
126
    put_bits32(&s->pbctx, s->avctx->frame_size);            // No. of samples in the frame
127
}
128

    
129
static void calc_predictor_params(AlacEncodeContext *s, int ch)
130
{
131
    int32_t coefs[MAX_LPC_ORDER][MAX_LPC_ORDER];
132
    int shift[MAX_LPC_ORDER];
133
    int opt_order;
134

    
135
    if (s->compression_level == 1) {
136
        s->lpc[ch].lpc_order = 6;
137
        s->lpc[ch].lpc_quant = 6;
138
        s->lpc[ch].lpc_coeff[0] =  160;
139
        s->lpc[ch].lpc_coeff[1] = -190;
140
        s->lpc[ch].lpc_coeff[2] =  170;
141
        s->lpc[ch].lpc_coeff[3] = -130;
142
        s->lpc[ch].lpc_coeff[4] =   80;
143
        s->lpc[ch].lpc_coeff[5] =  -25;
144
    } else {
145
        opt_order = ff_lpc_calc_coefs(&s->dspctx, s->sample_buf[ch],
146
                                      s->avctx->frame_size,
147
                                      s->min_prediction_order,
148
                                      s->max_prediction_order,
149
                                      ALAC_MAX_LPC_PRECISION, coefs, shift, 1,
150
                                      ORDER_METHOD_EST, ALAC_MAX_LPC_SHIFT, 1);
151

    
152
        s->lpc[ch].lpc_order = opt_order;
153
        s->lpc[ch].lpc_quant = shift[opt_order-1];
154
        memcpy(s->lpc[ch].lpc_coeff, coefs[opt_order-1], opt_order*sizeof(int));
155
    }
156
}
157

    
158
static int estimate_stereo_mode(int32_t *left_ch, int32_t *right_ch, int n)
159
{
160
    int i, best;
161
    int32_t lt, rt;
162
    uint64_t sum[4];
163
    uint64_t score[4];
164

    
165
    /* calculate sum of 2nd order residual for each channel */
166
    sum[0] = sum[1] = sum[2] = sum[3] = 0;
167
    for(i=2; i<n; i++) {
168
        lt = left_ch[i] - 2*left_ch[i-1] + left_ch[i-2];
169
        rt = right_ch[i] - 2*right_ch[i-1] + right_ch[i-2];
170
        sum[2] += FFABS((lt + rt) >> 1);
171
        sum[3] += FFABS(lt - rt);
172
        sum[0] += FFABS(lt);
173
        sum[1] += FFABS(rt);
174
    }
175

    
176
    /* calculate score for each mode */
177
    score[0] = sum[0] + sum[1];
178
    score[1] = sum[0] + sum[3];
179
    score[2] = sum[1] + sum[3];
180
    score[3] = sum[2] + sum[3];
181

    
182
    /* return mode with lowest score */
183
    best = 0;
184
    for(i=1; i<4; i++) {
185
        if(score[i] < score[best]) {
186
            best = i;
187
        }
188
    }
189
    return best;
190
}
191

    
192
static void alac_stereo_decorrelation(AlacEncodeContext *s)
193
{
194
    int32_t *left = s->sample_buf[0], *right = s->sample_buf[1];
195
    int i, mode, n = s->avctx->frame_size;
196
    int32_t tmp;
197

    
198
    mode = estimate_stereo_mode(left, right, n);
199

    
200
    switch(mode)
201
    {
202
        case ALAC_CHMODE_LEFT_RIGHT:
203
            s->interlacing_leftweight = 0;
204
            s->interlacing_shift = 0;
205
            break;
206

    
207
        case ALAC_CHMODE_LEFT_SIDE:
208
            for(i=0; i<n; i++) {
209
                right[i] = left[i] - right[i];
210
            }
211
            s->interlacing_leftweight = 1;
212
            s->interlacing_shift = 0;
213
            break;
214

    
215
        case ALAC_CHMODE_RIGHT_SIDE:
216
            for(i=0; i<n; i++) {
217
                tmp = right[i];
218
                right[i] = left[i] - right[i];
219
                left[i] = tmp + (right[i] >> 31);
220
            }
221
            s->interlacing_leftweight = 1;
222
            s->interlacing_shift = 31;
223
            break;
224

    
225
        default:
226
            for(i=0; i<n; i++) {
227
                tmp = left[i];
228
                left[i] = (tmp + right[i]) >> 1;
229
                right[i] = tmp - right[i];
230
            }
231
            s->interlacing_leftweight = 1;
232
            s->interlacing_shift = 1;
233
            break;
234
    }
235
}
236

    
237
static void alac_linear_predictor(AlacEncodeContext *s, int ch)
238
{
239
    int i;
240
    LPCContext lpc = s->lpc[ch];
241

    
242
    if(lpc.lpc_order == 31) {
243
        s->predictor_buf[0] = s->sample_buf[ch][0];
244

    
245
        for(i=1; i<s->avctx->frame_size; i++)
246
            s->predictor_buf[i] = s->sample_buf[ch][i] - s->sample_buf[ch][i-1];
247

    
248
        return;
249
    }
250

    
251
    // generalised linear predictor
252

    
253
    if(lpc.lpc_order > 0) {
254
        int32_t *samples  = s->sample_buf[ch];
255
        int32_t *residual = s->predictor_buf;
256

    
257
        // generate warm-up samples
258
        residual[0] = samples[0];
259
        for(i=1;i<=lpc.lpc_order;i++)
260
            residual[i] = samples[i] - samples[i-1];
261

    
262
        // perform lpc on remaining samples
263
        for(i = lpc.lpc_order + 1; i < s->avctx->frame_size; i++) {
264
            int sum = 1 << (lpc.lpc_quant - 1), res_val, j;
265

    
266
            for (j = 0; j < lpc.lpc_order; j++) {
267
                sum += (samples[lpc.lpc_order-j] - samples[0]) *
268
                        lpc.lpc_coeff[j];
269
            }
270

    
271
            sum >>= lpc.lpc_quant;
272
            sum += samples[0];
273
            residual[i] = sign_extend(samples[lpc.lpc_order+1] - sum,
274
                                      s->write_sample_size);
275
            res_val = residual[i];
276

    
277
            if(res_val) {
278
                int index = lpc.lpc_order - 1;
279
                int neg = (res_val < 0);
280

    
281
                while(index >= 0 && (neg ? (res_val < 0):(res_val > 0))) {
282
                    int val = samples[0] - samples[lpc.lpc_order - index];
283
                    int sign = (val ? FFSIGN(val) : 0);
284

    
285
                    if(neg)
286
                        sign*=-1;
287

    
288
                    lpc.lpc_coeff[index] -= sign;
289
                    val *= sign;
290
                    res_val -= ((val >> lpc.lpc_quant) *
291
                            (lpc.lpc_order - index));
292
                    index--;
293
                }
294
            }
295
            samples++;
296
        }
297
    }
298
}
299

    
300
static void alac_entropy_coder(AlacEncodeContext *s)
301
{
302
    unsigned int history = s->rc.initial_history;
303
    int sign_modifier = 0, i, k;
304
    int32_t *samples = s->predictor_buf;
305

    
306
    for(i=0;i < s->avctx->frame_size;) {
307
        int x;
308

    
309
        k = av_log2((history >> 9) + 3);
310

    
311
        x = -2*(*samples)-1;
312
        x ^= (x>>31);
313

    
314
        samples++;
315
        i++;
316

    
317
        encode_scalar(s, x - sign_modifier, k, s->write_sample_size);
318

    
319
        history += x * s->rc.history_mult
320
                   - ((history * s->rc.history_mult) >> 9);
321

    
322
        sign_modifier = 0;
323
        if(x > 0xFFFF)
324
            history = 0xFFFF;
325

    
326
        if((history < 128) && (i < s->avctx->frame_size)) {
327
            unsigned int block_size = 0;
328

    
329
            k = 7 - av_log2(history) + ((history + 16) >> 6);
330

    
331
            while((*samples == 0) && (i < s->avctx->frame_size)) {
332
                samples++;
333
                i++;
334
                block_size++;
335
            }
336
            encode_scalar(s, block_size, k, 16);
337

    
338
            sign_modifier = (block_size <= 0xFFFF);
339

    
340
            history = 0;
341
        }
342

    
343
    }
344
}
345

    
346
static void write_compressed_frame(AlacEncodeContext *s)
347
{
348
    int i, j;
349

    
350
    if(s->avctx->channels == 2)
351
        alac_stereo_decorrelation(s);
352
    put_bits(&s->pbctx, 8, s->interlacing_shift);
353
    put_bits(&s->pbctx, 8, s->interlacing_leftweight);
354

    
355
    for(i=0;i<s->avctx->channels;i++) {
356

    
357
        calc_predictor_params(s, i);
358

    
359
        put_bits(&s->pbctx, 4, 0);  // prediction type : currently only type 0 has been RE'd
360
        put_bits(&s->pbctx, 4, s->lpc[i].lpc_quant);
361

    
362
        put_bits(&s->pbctx, 3, s->rc.rice_modifier);
363
        put_bits(&s->pbctx, 5, s->lpc[i].lpc_order);
364
        // predictor coeff. table
365
        for(j=0;j<s->lpc[i].lpc_order;j++) {
366
            put_sbits(&s->pbctx, 16, s->lpc[i].lpc_coeff[j]);
367
        }
368
    }
369

    
370
    // apply lpc and entropy coding to audio samples
371

    
372
    for(i=0;i<s->avctx->channels;i++) {
373
        alac_linear_predictor(s, i);
374
        alac_entropy_coder(s);
375
    }
376
}
377

    
378
static av_cold int alac_encode_init(AVCodecContext *avctx)
379
{
380
    AlacEncodeContext *s    = avctx->priv_data;
381
    uint8_t *alac_extradata = av_mallocz(ALAC_EXTRADATA_SIZE+1);
382

    
383
    avctx->frame_size      = DEFAULT_FRAME_SIZE;
384
    avctx->bits_per_coded_sample = DEFAULT_SAMPLE_SIZE;
385

    
386
    if(avctx->sample_fmt != SAMPLE_FMT_S16) {
387
        av_log(avctx, AV_LOG_ERROR, "only pcm_s16 input samples are supported\n");
388
        return -1;
389
    }
390

    
391
    // Set default compression level
392
    if(avctx->compression_level == FF_COMPRESSION_DEFAULT)
393
        s->compression_level = 2;
394
    else
395
        s->compression_level = av_clip(avctx->compression_level, 0, 2);
396

    
397
    // Initialize default Rice parameters
398
    s->rc.history_mult    = 40;
399
    s->rc.initial_history = 10;
400
    s->rc.k_modifier      = 14;
401
    s->rc.rice_modifier   = 4;
402

    
403
    s->max_coded_frame_size = 8 + (avctx->frame_size*avctx->channels*avctx->bits_per_coded_sample>>3);
404

    
405
    s->write_sample_size  = avctx->bits_per_coded_sample + avctx->channels - 1; // FIXME: consider wasted_bytes
406

    
407
    AV_WB32(alac_extradata,    ALAC_EXTRADATA_SIZE);
408
    AV_WB32(alac_extradata+4,  MKBETAG('a','l','a','c'));
409
    AV_WB32(alac_extradata+12, avctx->frame_size);
410
    AV_WB8 (alac_extradata+17, avctx->bits_per_coded_sample);
411
    AV_WB8 (alac_extradata+21, avctx->channels);
412
    AV_WB32(alac_extradata+24, s->max_coded_frame_size);
413
    AV_WB32(alac_extradata+28, avctx->sample_rate*avctx->channels*avctx->bits_per_coded_sample); // average bitrate
414
    AV_WB32(alac_extradata+32, avctx->sample_rate);
415

    
416
    // Set relevant extradata fields
417
    if(s->compression_level > 0) {
418
        AV_WB8(alac_extradata+18, s->rc.history_mult);
419
        AV_WB8(alac_extradata+19, s->rc.initial_history);
420
        AV_WB8(alac_extradata+20, s->rc.k_modifier);
421
    }
422

    
423
    s->min_prediction_order = DEFAULT_MIN_PRED_ORDER;
424
    if(avctx->min_prediction_order >= 0) {
425
        if(avctx->min_prediction_order < MIN_LPC_ORDER ||
426
           avctx->min_prediction_order > ALAC_MAX_LPC_ORDER) {
427
            av_log(avctx, AV_LOG_ERROR, "invalid min prediction order: %d\n", avctx->min_prediction_order);
428
                return -1;
429
        }
430

    
431
        s->min_prediction_order = avctx->min_prediction_order;
432
    }
433

    
434
    s->max_prediction_order = DEFAULT_MAX_PRED_ORDER;
435
    if(avctx->max_prediction_order >= 0) {
436
        if(avctx->max_prediction_order < MIN_LPC_ORDER ||
437
           avctx->max_prediction_order > ALAC_MAX_LPC_ORDER) {
438
            av_log(avctx, AV_LOG_ERROR, "invalid max prediction order: %d\n", avctx->max_prediction_order);
439
                return -1;
440
        }
441

    
442
        s->max_prediction_order = avctx->max_prediction_order;
443
    }
444

    
445
    if(s->max_prediction_order < s->min_prediction_order) {
446
        av_log(avctx, AV_LOG_ERROR, "invalid prediction orders: min=%d max=%d\n",
447
               s->min_prediction_order, s->max_prediction_order);
448
        return -1;
449
    }
450

    
451
    avctx->extradata = alac_extradata;
452
    avctx->extradata_size = ALAC_EXTRADATA_SIZE;
453

    
454
    avctx->coded_frame = avcodec_alloc_frame();
455
    avctx->coded_frame->key_frame = 1;
456

    
457
    s->avctx = avctx;
458
    dsputil_init(&s->dspctx, avctx);
459

    
460
    return 0;
461
}
462

    
463
static int alac_encode_frame(AVCodecContext *avctx, uint8_t *frame,
464
                             int buf_size, void *data)
465
{
466
    AlacEncodeContext *s = avctx->priv_data;
467
    PutBitContext *pb = &s->pbctx;
468
    int i, out_bytes, verbatim_flag = 0;
469

    
470
    if(avctx->frame_size > DEFAULT_FRAME_SIZE) {
471
        av_log(avctx, AV_LOG_ERROR, "input frame size exceeded\n");
472
        return -1;
473
    }
474

    
475
    if(buf_size < 2*s->max_coded_frame_size) {
476
        av_log(avctx, AV_LOG_ERROR, "buffer size is too small\n");
477
        return -1;
478
    }
479

    
480
verbatim:
481
    init_put_bits(pb, frame, buf_size);
482

    
483
    if((s->compression_level == 0) || verbatim_flag) {
484
        // Verbatim mode
485
        int16_t *samples = data;
486
        write_frame_header(s, 1);
487
        for(i=0; i<avctx->frame_size*avctx->channels; i++) {
488
            put_sbits(pb, 16, *samples++);
489
        }
490
    } else {
491
        init_sample_buffers(s, data);
492
        write_frame_header(s, 0);
493
        write_compressed_frame(s);
494
    }
495

    
496
    put_bits(pb, 3, 7);
497
    flush_put_bits(pb);
498
    out_bytes = put_bits_count(pb) >> 3;
499

    
500
    if(out_bytes > s->max_coded_frame_size) {
501
        /* frame too large. use verbatim mode */
502
        if(verbatim_flag || (s->compression_level == 0)) {
503
            /* still too large. must be an error. */
504
            av_log(avctx, AV_LOG_ERROR, "error encoding frame\n");
505
            return -1;
506
        }
507
        verbatim_flag = 1;
508
        goto verbatim;
509
    }
510

    
511
    return out_bytes;
512
}
513

    
514
static av_cold int alac_encode_close(AVCodecContext *avctx)
515
{
516
    av_freep(&avctx->extradata);
517
    avctx->extradata_size = 0;
518
    av_freep(&avctx->coded_frame);
519
    return 0;
520
}
521

    
522
AVCodec alac_encoder = {
523
    "alac",
524
    AVMEDIA_TYPE_AUDIO,
525
    CODEC_ID_ALAC,
526
    sizeof(AlacEncodeContext),
527
    alac_encode_init,
528
    alac_encode_frame,
529
    alac_encode_close,
530
    .capabilities = CODEC_CAP_SMALL_LAST_FRAME,
531
    .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"),
532
};