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
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
|
/*
* Shorten decoder
* Copyright (c) 2005 Jeff Muizelaar
*
* 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
* Shorten decoder
* @author Jeff Muizelaar
*
*/
#include <limits.h>
#include "avcodec.h"
#include "bytestream.h"
#include "get_bits.h"
#include "golomb.h"
#include "internal.h"
#define MAX_CHANNELS 8
#define MAX_BLOCKSIZE 65535
#define OUT_BUFFER_SIZE 16384
#define ULONGSIZE 2
#define WAVE_FORMAT_PCM 0x0001
#define DEFAULT_BLOCK_SIZE 256
#define TYPESIZE 4
#define CHANSIZE 0
#define LPCQSIZE 2
#define ENERGYSIZE 3
#define BITSHIFTSIZE 2
#define TYPE_S8 1
#define TYPE_U8 2
#define TYPE_S16HL 3
#define TYPE_U16HL 4
#define TYPE_S16LH 5
#define TYPE_U16LH 6
#define NWRAP 3
#define NSKIPSIZE 1
#define LPCQUANT 5
#define V2LPCQOFFSET (1 << LPCQUANT)
#define FNSIZE 2
#define FN_DIFF0 0
#define FN_DIFF1 1
#define FN_DIFF2 2
#define FN_DIFF3 3
#define FN_QUIT 4
#define FN_BLOCKSIZE 5
#define FN_BITSHIFT 6
#define FN_QLPC 7
#define FN_ZERO 8
#define FN_VERBATIM 9
/** indicates if the FN_* command is audio or non-audio */
static const uint8_t is_audio_command[10] = { 1, 1, 1, 1, 0, 0, 0, 1, 1, 0 };
#define VERBATIM_CKSIZE_SIZE 5
#define VERBATIM_BYTE_SIZE 8
#define CANONICAL_HEADER_SIZE 44
typedef struct ShortenContext {
AVCodecContext *avctx;
GetBitContext gb;
int min_framesize, max_framesize;
unsigned channels;
int32_t *decoded[MAX_CHANNELS];
int32_t *decoded_base[MAX_CHANNELS];
int32_t *offset[MAX_CHANNELS];
int *coeffs;
uint8_t *bitstream;
int bitstream_size;
int bitstream_index;
unsigned int allocated_bitstream_size;
int header_size;
uint8_t header[OUT_BUFFER_SIZE];
int version;
int cur_chan;
int bitshift;
int nmean;
int internal_ftype;
int nwrap;
int blocksize;
int bitindex;
int32_t lpcqoffset;
int got_header;
int got_quit_command;
} ShortenContext;
static av_cold int shorten_decode_init(AVCodecContext *avctx)
{
ShortenContext *s = avctx->priv_data;
s->avctx = avctx;
return 0;
}
static int allocate_buffers(ShortenContext *s)
{
int i, chan;
int *coeffs;
void *tmp_ptr;
for (chan = 0; chan < s->channels; chan++) {
if (FFMAX(1, s->nmean) >= UINT_MAX / sizeof(int32_t)) {
av_log(s->avctx, AV_LOG_ERROR, "nmean too large\n");
return AVERROR_INVALIDDATA;
}
if (s->blocksize + s->nwrap >= UINT_MAX / sizeof(int32_t) ||
s->blocksize + s->nwrap <= (unsigned)s->nwrap) {
av_log(s->avctx, AV_LOG_ERROR,
"s->blocksize + s->nwrap too large\n");
return AVERROR_INVALIDDATA;
}
tmp_ptr =
av_realloc(s->offset[chan], sizeof(int32_t) * FFMAX(1, s->nmean));
if (!tmp_ptr)
return AVERROR(ENOMEM);
s->offset[chan] = tmp_ptr;
tmp_ptr = av_realloc(s->decoded_base[chan], (s->blocksize + s->nwrap) *
sizeof(s->decoded_base[0][0]));
if (!tmp_ptr)
return AVERROR(ENOMEM);
s->decoded_base[chan] = tmp_ptr;
for (i = 0; i < s->nwrap; i++)
s->decoded_base[chan][i] = 0;
s->decoded[chan] = s->decoded_base[chan] + s->nwrap;
}
coeffs = av_realloc(s->coeffs, s->nwrap * sizeof(*s->coeffs));
if (!coeffs)
return AVERROR(ENOMEM);
s->coeffs = coeffs;
return 0;
}
static inline unsigned int get_uint(ShortenContext *s, int k)
{
if (s->version != 0)
k = get_ur_golomb_shorten(&s->gb, ULONGSIZE);
return get_ur_golomb_shorten(&s->gb, k);
}
static void fix_bitshift(ShortenContext *s, int32_t *buffer)
{
int i;
if (s->bitshift != 0)
for (i = 0; i < s->blocksize; i++)
buffer[i] <<= s->bitshift;
}
static int init_offset(ShortenContext *s)
{
int32_t mean = 0;
int chan, i;
int nblock = FFMAX(1, s->nmean);
/* initialise offset */
switch (s->internal_ftype) {
case TYPE_U8:
s->avctx->sample_fmt = AV_SAMPLE_FMT_U8P;
mean = 0x80;
break;
case TYPE_S16HL:
case TYPE_S16LH:
s->avctx->sample_fmt = AV_SAMPLE_FMT_S16P;
break;
default:
av_log(s->avctx, AV_LOG_ERROR, "unknown audio type\n");
return AVERROR_PATCHWELCOME;
}
for (chan = 0; chan < s->channels; chan++)
for (i = 0; i < nblock; i++)
s->offset[chan][i] = mean;
return 0;
}
static int decode_wave_header(AVCodecContext *avctx, const uint8_t *header,
int header_size)
{
int len, bps;
short wave_format;
const uint8_t *end= header + header_size;
if (bytestream_get_le32(&header) != MKTAG('R', 'I', 'F', 'F')) {
av_log(avctx, AV_LOG_ERROR, "missing RIFF tag\n");
return AVERROR_INVALIDDATA;
}
header += 4; /* chunk size */
if (bytestream_get_le32(&header) != MKTAG('W', 'A', 'V', 'E')) {
av_log(avctx, AV_LOG_ERROR, "missing WAVE tag\n");
return AVERROR_INVALIDDATA;
}
while (bytestream_get_le32(&header) != MKTAG('f', 'm', 't', ' ')) {
len = bytestream_get_le32(&header);
if (len<0 || end - header - 8 < len)
return AVERROR_INVALIDDATA;
header += len;
}
len = bytestream_get_le32(&header);
if (len < 16) {
av_log(avctx, AV_LOG_ERROR, "fmt chunk was too short\n");
return AVERROR_INVALIDDATA;
}
wave_format = bytestream_get_le16(&header);
switch (wave_format) {
case WAVE_FORMAT_PCM:
break;
default:
av_log(avctx, AV_LOG_ERROR, "unsupported wave format\n");
return AVERROR(ENOSYS);
}
header += 2; // skip channels (already got from shorten header)
avctx->sample_rate = bytestream_get_le32(&header);
header += 4; // skip bit rate (represents original uncompressed bit rate)
header += 2; // skip block align (not needed)
bps = bytestream_get_le16(&header);
avctx->bits_per_coded_sample = bps;
if (bps != 16 && bps != 8) {
av_log(avctx, AV_LOG_ERROR, "unsupported number of bits per sample: %d\n", bps);
return AVERROR(ENOSYS);
}
len -= 16;
if (len > 0)
av_log(avctx, AV_LOG_INFO, "%d header bytes unparsed\n", len);
return 0;
}
static const int fixed_coeffs[3][3] = {
{ 1, 0, 0 },
{ 2, -1, 0 },
{ 3, -3, 1 }
};
static int decode_subframe_lpc(ShortenContext *s, int command, int channel,
int residual_size, int32_t coffset)
{
int pred_order, sum, qshift, init_sum, i, j;
const int *coeffs;
if (command == FN_QLPC) {
/* read/validate prediction order */
pred_order = get_ur_golomb_shorten(&s->gb, LPCQSIZE);
if (pred_order > s->nwrap) {
av_log(s->avctx, AV_LOG_ERROR, "invalid pred_order %d\n",
pred_order);
return AVERROR(EINVAL);
}
/* read LPC coefficients */
for (i = 0; i < pred_order; i++)
s->coeffs[i] = get_sr_golomb_shorten(&s->gb, LPCQUANT);
coeffs = s->coeffs;
qshift = LPCQUANT;
} else {
/* fixed LPC coeffs */
pred_order = command;
coeffs = fixed_coeffs[pred_order - 1];
qshift = 0;
}
/* subtract offset from previous samples to use in prediction */
if (command == FN_QLPC && coffset)
for (i = -pred_order; i < 0; i++)
s->decoded[channel][i] -= coffset;
/* decode residual and do LPC prediction */
init_sum = pred_order ? (command == FN_QLPC ? s->lpcqoffset : 0) : coffset;
for (i = 0; i < s->blocksize; i++) {
sum = init_sum;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * s->decoded[channel][i - j - 1];
s->decoded[channel][i] = get_sr_golomb_shorten(&s->gb, residual_size) +
(sum >> qshift);
}
/* add offset to current samples */
if (command == FN_QLPC && coffset)
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] += coffset;
return 0;
}
static int read_header(ShortenContext *s)
{
int i, ret;
int maxnlpc = 0;
/* shorten signature */
if (get_bits_long(&s->gb, 32) != AV_RB32("ajkg")) {
av_log(s->avctx, AV_LOG_ERROR, "missing shorten magic 'ajkg'\n");
return AVERROR_INVALIDDATA;
}
s->lpcqoffset = 0;
s->blocksize = DEFAULT_BLOCK_SIZE;
s->nmean = -1;
s->version = get_bits(&s->gb, 8);
s->internal_ftype = get_uint(s, TYPESIZE);
s->channels = get_uint(s, CHANSIZE);
if (!s->channels) {
av_log(s->avctx, AV_LOG_ERROR, "No channels reported\n");
return AVERROR_INVALIDDATA;
}
if (s->channels > MAX_CHANNELS) {
av_log(s->avctx, AV_LOG_ERROR, "too many channels: %d\n", s->channels);
s->channels = 0;
return AVERROR_INVALIDDATA;
}
s->avctx->channels = s->channels;
/* get blocksize if version > 0 */
if (s->version > 0) {
int skip_bytes;
unsigned blocksize;
blocksize = get_uint(s, av_log2(DEFAULT_BLOCK_SIZE));
if (!blocksize || blocksize > MAX_BLOCKSIZE) {
av_log(s->avctx, AV_LOG_ERROR,
"invalid or unsupported block size: %d\n",
blocksize);
return AVERROR(EINVAL);
}
s->blocksize = blocksize;
maxnlpc = get_uint(s, LPCQSIZE);
s->nmean = get_uint(s, 0);
skip_bytes = get_uint(s, NSKIPSIZE);
for (i = 0; i < skip_bytes; i++)
skip_bits(&s->gb, 8);
}
s->nwrap = FFMAX(NWRAP, maxnlpc);
if ((ret = allocate_buffers(s)) < 0)
return ret;
if ((ret = init_offset(s)) < 0)
return ret;
if (s->version > 1)
s->lpcqoffset = V2LPCQOFFSET;
if (get_ur_golomb_shorten(&s->gb, FNSIZE) != FN_VERBATIM) {
av_log(s->avctx, AV_LOG_ERROR,
"missing verbatim section at beginning of stream\n");
return AVERROR_INVALIDDATA;
}
s->header_size = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
if (s->header_size >= OUT_BUFFER_SIZE ||
s->header_size < CANONICAL_HEADER_SIZE) {
av_log(s->avctx, AV_LOG_ERROR, "header is wrong size: %d\n",
s->header_size);
return AVERROR_INVALIDDATA;
}
for (i = 0; i < s->header_size; i++)
s->header[i] = (char)get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
if ((ret = decode_wave_header(s->avctx, s->header, s->header_size)) < 0)
return ret;
s->cur_chan = 0;
s->bitshift = 0;
s->got_header = 1;
return 0;
}
static int shorten_decode_frame(AVCodecContext *avctx, void *data,
int *got_frame_ptr, AVPacket *avpkt)
{
AVFrame *frame = data;
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
ShortenContext *s = avctx->priv_data;
int i, input_buf_size = 0;
int ret;
/* allocate internal bitstream buffer */
if (s->max_framesize == 0) {
void *tmp_ptr;
s->max_framesize = 8192; // should hopefully be enough for the first header
tmp_ptr = av_fast_realloc(s->bitstream, &s->allocated_bitstream_size,
s->max_framesize);
if (!tmp_ptr) {
av_log(avctx, AV_LOG_ERROR, "error allocating bitstream buffer\n");
return AVERROR(ENOMEM);
}
s->bitstream = tmp_ptr;
}
/* append current packet data to bitstream buffer */
if (1 && s->max_framesize) { //FIXME truncated
buf_size = FFMIN(buf_size, s->max_framesize - s->bitstream_size);
input_buf_size = buf_size;
if (s->bitstream_index + s->bitstream_size + buf_size >
s->allocated_bitstream_size) {
memmove(s->bitstream, &s->bitstream[s->bitstream_index],
s->bitstream_size);
s->bitstream_index = 0;
}
if (buf)
memcpy(&s->bitstream[s->bitstream_index + s->bitstream_size], buf,
buf_size);
buf = &s->bitstream[s->bitstream_index];
buf_size += s->bitstream_size;
s->bitstream_size = buf_size;
/* do not decode until buffer has at least max_framesize bytes or
* the end of the file has been reached */
if (buf_size < s->max_framesize && avpkt->data) {
*got_frame_ptr = 0;
return input_buf_size;
}
}
/* init and position bitstream reader */
init_get_bits(&s->gb, buf, buf_size * 8);
skip_bits(&s->gb, s->bitindex);
/* process header or next subblock */
if (!s->got_header) {
if ((ret = read_header(s)) < 0)
return ret;
*got_frame_ptr = 0;
goto finish_frame;
}
/* if quit command was read previously, don't decode anything */
if (s->got_quit_command) {
*got_frame_ptr = 0;
return avpkt->size;
}
s->cur_chan = 0;
while (s->cur_chan < s->channels) {
unsigned cmd;
int len;
if (get_bits_left(&s->gb) < 3 + FNSIZE) {
*got_frame_ptr = 0;
break;
}
cmd = get_ur_golomb_shorten(&s->gb, FNSIZE);
if (cmd > FN_VERBATIM) {
av_log(avctx, AV_LOG_ERROR, "unknown shorten function %d\n", cmd);
*got_frame_ptr = 0;
break;
}
if (!is_audio_command[cmd]) {
/* process non-audio command */
switch (cmd) {
case FN_VERBATIM:
len = get_ur_golomb_shorten(&s->gb, VERBATIM_CKSIZE_SIZE);
while (len--)
get_ur_golomb_shorten(&s->gb, VERBATIM_BYTE_SIZE);
break;
case FN_BITSHIFT:
s->bitshift = get_ur_golomb_shorten(&s->gb, BITSHIFTSIZE);
break;
case FN_BLOCKSIZE: {
unsigned blocksize = get_uint(s, av_log2(s->blocksize));
if (blocksize > s->blocksize) {
av_log(avctx, AV_LOG_ERROR,
"Increasing block size is not supported\n");
return AVERROR_PATCHWELCOME;
}
if (!blocksize || blocksize > MAX_BLOCKSIZE) {
av_log(avctx, AV_LOG_ERROR, "invalid or unsupported "
"block size: %d\n", blocksize);
return AVERROR(EINVAL);
}
s->blocksize = blocksize;
break;
}
case FN_QUIT:
s->got_quit_command = 1;
break;
}
if (cmd == FN_BLOCKSIZE || cmd == FN_QUIT) {
*got_frame_ptr = 0;
break;
}
} else {
/* process audio command */
int residual_size = 0;
int channel = s->cur_chan;
int32_t coffset;
/* get Rice code for residual decoding */
if (cmd != FN_ZERO) {
residual_size = get_ur_golomb_shorten(&s->gb, ENERGYSIZE);
/* This is a hack as version 0 differed in the definition
* of get_sr_golomb_shorten(). */
if (s->version == 0)
residual_size--;
}
/* calculate sample offset using means from previous blocks */
if (s->nmean == 0)
coffset = s->offset[channel][0];
else {
int32_t sum = (s->version < 2) ? 0 : s->nmean / 2;
for (i = 0; i < s->nmean; i++)
sum += s->offset[channel][i];
coffset = sum / s->nmean;
if (s->version >= 2)
coffset = s->bitshift == 0 ? coffset : coffset >> s->bitshift - 1 >> 1;
}
/* decode samples for this channel */
if (cmd == FN_ZERO) {
for (i = 0; i < s->blocksize; i++)
s->decoded[channel][i] = 0;
} else {
if ((ret = decode_subframe_lpc(s, cmd, channel,
residual_size, coffset)) < 0)
return ret;
}
/* update means with info from the current block */
if (s->nmean > 0) {
int32_t sum = (s->version < 2) ? 0 : s->blocksize / 2;
for (i = 0; i < s->blocksize; i++)
sum += s->decoded[channel][i];
for (i = 1; i < s->nmean; i++)
s->offset[channel][i - 1] = s->offset[channel][i];
if (s->version < 2)
s->offset[channel][s->nmean - 1] = sum / s->blocksize;
else
s->offset[channel][s->nmean - 1] = (sum / s->blocksize) << s->bitshift;
}
/* copy wrap samples for use with next block */
for (i = -s->nwrap; i < 0; i++)
s->decoded[channel][i] = s->decoded[channel][i + s->blocksize];
/* shift samples to add in unused zero bits which were removed
* during encoding */
fix_bitshift(s, s->decoded[channel]);
/* if this is the last channel in the block, output the samples */
s->cur_chan++;
if (s->cur_chan == s->channels) {
uint8_t *samples_u8;
int16_t *samples_s16;
int chan;
/* get output buffer */
frame->nb_samples = s->blocksize;
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
return ret;
for (chan = 0; chan < s->channels; chan++) {
samples_u8 = ((uint8_t **)frame->extended_data)[chan];
samples_s16 = ((int16_t **)frame->extended_data)[chan];
for (i = 0; i < s->blocksize; i++) {
switch (s->internal_ftype) {
case TYPE_U8:
*samples_u8++ = av_clip_uint8(s->decoded[chan][i]);
break;
case TYPE_S16HL:
case TYPE_S16LH:
*samples_s16++ = av_clip_int16(s->decoded[chan][i]);
break;
}
}
}
*got_frame_ptr = 1;
}
}
}
if (s->cur_chan < s->channels)
*got_frame_ptr = 0;
finish_frame:
s->bitindex = get_bits_count(&s->gb) - 8 * (get_bits_count(&s->gb) / 8);
i = get_bits_count(&s->gb) / 8;
if (i > buf_size) {
av_log(s->avctx, AV_LOG_ERROR, "overread: %d\n", i - buf_size);
s->bitstream_size = 0;
s->bitstream_index = 0;
return AVERROR_INVALIDDATA;
}
if (s->bitstream_size) {
s->bitstream_index += i;
s->bitstream_size -= i;
return input_buf_size;
} else
return i;
}
static av_cold int shorten_decode_close(AVCodecContext *avctx)
{
ShortenContext *s = avctx->priv_data;
int i;
for (i = 0; i < s->channels; i++) {
s->decoded[i] = NULL;
av_freep(&s->decoded_base[i]);
av_freep(&s->offset[i]);
}
av_freep(&s->bitstream);
av_freep(&s->coeffs);
return 0;
}
AVCodec ff_shorten_decoder = {
.name = "shorten",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_SHORTEN,
.priv_data_size = sizeof(ShortenContext),
.init = shorten_decode_init,
.close = shorten_decode_close,
.decode = shorten_decode_frame,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_DR1,
.long_name = NULL_IF_CONFIG_SMALL("Shorten"),
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_U8P,
AV_SAMPLE_FMT_NONE },
};
|