/
mlpdec.c
1146 lines (912 loc) · 37.5 KB
/
mlpdec.c
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
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
/*
* MLP decoder
* Copyright (c) 2007-2008 Ian Caulfield
*
* 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 libavcodec/mlpdec.c
* MLP decoder
*/
#include <stdint.h>
#include "avcodec.h"
#include "dsputil.h"
#include "libavutil/intreadwrite.h"
#include "get_bits.h"
#include "libavutil/crc.h"
#include "parser.h"
#include "mlp_parser.h"
#include "mlp.h"
/** number of bits used for VLC lookup - longest Huffman code is 9 */
#define VLC_BITS 9
static const char* sample_message =
"Please file a bug report following the instructions at "
"http://ffmpeg.org/bugreports.html and include "
"a sample of this file.";
typedef struct SubStream {
//! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
uint8_t restart_seen;
//@{
/** restart header data */
//! The type of noise to be used in the rematrix stage.
uint16_t noise_type;
//! The index of the first channel coded in this substream.
uint8_t min_channel;
//! The index of the last channel coded in this substream.
uint8_t max_channel;
//! The number of channels input into the rematrix stage.
uint8_t max_matrix_channel;
//! For each channel output by the matrix, the output channel to map it to
uint8_t ch_assign[MAX_CHANNELS];
//! The left shift applied to random noise in 0x31ea substreams.
uint8_t noise_shift;
//! The current seed value for the pseudorandom noise generator(s).
uint32_t noisegen_seed;
//! Set if the substream contains extra info to check the size of VLC blocks.
uint8_t data_check_present;
//! Bitmask of which parameter sets are conveyed in a decoding parameter block.
uint8_t param_presence_flags;
#define PARAM_BLOCKSIZE (1 << 7)
#define PARAM_MATRIX (1 << 6)
#define PARAM_OUTSHIFT (1 << 5)
#define PARAM_QUANTSTEP (1 << 4)
#define PARAM_FIR (1 << 3)
#define PARAM_IIR (1 << 2)
#define PARAM_HUFFOFFSET (1 << 1)
#define PARAM_PRESENCE (1 << 0)
//@}
//@{
/** matrix data */
//! Number of matrices to be applied.
uint8_t num_primitive_matrices;
//! matrix output channel
uint8_t matrix_out_ch[MAX_MATRICES];
//! Whether the LSBs of the matrix output are encoded in the bitstream.
uint8_t lsb_bypass[MAX_MATRICES];
//! Matrix coefficients, stored as 2.14 fixed point.
int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
//! Left shift to apply to noise values in 0x31eb substreams.
uint8_t matrix_noise_shift[MAX_MATRICES];
//@}
//! Left shift to apply to Huffman-decoded residuals.
uint8_t quant_step_size[MAX_CHANNELS];
//! number of PCM samples in current audio block
uint16_t blocksize;
//! Number of PCM samples decoded so far in this frame.
uint16_t blockpos;
//! Left shift to apply to decoded PCM values to get final 24-bit output.
int8_t output_shift[MAX_CHANNELS];
//! Running XOR of all output samples.
int32_t lossless_check_data;
} SubStream;
typedef struct MLPDecodeContext {
AVCodecContext *avctx;
//! Current access unit being read has a major sync.
int is_major_sync_unit;
//! Set if a valid major sync block has been read. Otherwise no decoding is possible.
uint8_t params_valid;
//! Number of substreams contained within this stream.
uint8_t num_substreams;
//! Index of the last substream to decode - further substreams are skipped.
uint8_t max_decoded_substream;
//! number of PCM samples contained in each frame
int access_unit_size;
//! next power of two above the number of samples in each frame
int access_unit_size_pow2;
SubStream substream[MAX_SUBSTREAMS];
ChannelParams channel_params[MAX_CHANNELS];
int matrix_changed;
int filter_changed[MAX_CHANNELS][NUM_FILTERS];
int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
DSPContext dsp;
} MLPDecodeContext;
static VLC huff_vlc[3];
/** Initialize static data, constant between all invocations of the codec. */
static av_cold void init_static(void)
{
INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
&ff_mlp_huffman_tables[0][0][1], 2, 1,
&ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
&ff_mlp_huffman_tables[1][0][1], 2, 1,
&ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
&ff_mlp_huffman_tables[2][0][1], 2, 1,
&ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
ff_mlp_init_crc();
}
static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
unsigned int substr, unsigned int ch)
{
ChannelParams *cp = &m->channel_params[ch];
SubStream *s = &m->substream[substr];
int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
int32_t sign_huff_offset = cp->huff_offset;
if (cp->codebook > 0)
sign_huff_offset -= 7 << lsb_bits;
if (sign_shift >= 0)
sign_huff_offset -= 1 << sign_shift;
return sign_huff_offset;
}
/** Read a sample, consisting of either, both or neither of entropy-coded MSBs
* and plain LSBs. */
static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
unsigned int substr, unsigned int pos)
{
SubStream *s = &m->substream[substr];
unsigned int mat, channel;
for (mat = 0; mat < s->num_primitive_matrices; mat++)
if (s->lsb_bypass[mat])
m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
for (channel = s->min_channel; channel <= s->max_channel; channel++) {
ChannelParams *cp = &m->channel_params[channel];
int codebook = cp->codebook;
int quant_step_size = s->quant_step_size[channel];
int lsb_bits = cp->huff_lsbs - quant_step_size;
int result = 0;
if (codebook > 0)
result = get_vlc2(gbp, huff_vlc[codebook-1].table,
VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
if (result < 0)
return -1;
if (lsb_bits > 0)
result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
result += cp->sign_huff_offset;
result <<= quant_step_size;
m->sample_buffer[pos + s->blockpos][channel] = result;
}
return 0;
}
static av_cold int mlp_decode_init(AVCodecContext *avctx)
{
MLPDecodeContext *m = avctx->priv_data;
int substr;
init_static();
m->avctx = avctx;
for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
m->substream[substr].lossless_check_data = 0xffffffff;
dsputil_init(&m->dsp, avctx);
return 0;
}
/** Read a major sync info header - contains high level information about
* the stream - sample rate, channel arrangement etc. Most of this
* information is not actually necessary for decoding, only for playback.
*/
static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
{
MLPHeaderInfo mh;
int substr;
if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
return -1;
if (mh.group1_bits == 0) {
av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
return -1;
}
if (mh.group2_bits > mh.group1_bits) {
av_log(m->avctx, AV_LOG_ERROR,
"Channel group 2 cannot have more bits per sample than group 1.\n");
return -1;
}
if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
av_log(m->avctx, AV_LOG_ERROR,
"Channel groups with differing sample rates are not currently supported.\n");
return -1;
}
if (mh.group1_samplerate == 0) {
av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
return -1;
}
if (mh.group1_samplerate > MAX_SAMPLERATE) {
av_log(m->avctx, AV_LOG_ERROR,
"Sampling rate %d is greater than the supported maximum (%d).\n",
mh.group1_samplerate, MAX_SAMPLERATE);
return -1;
}
if (mh.access_unit_size > MAX_BLOCKSIZE) {
av_log(m->avctx, AV_LOG_ERROR,
"Block size %d is greater than the supported maximum (%d).\n",
mh.access_unit_size, MAX_BLOCKSIZE);
return -1;
}
if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
av_log(m->avctx, AV_LOG_ERROR,
"Block size pow2 %d is greater than the supported maximum (%d).\n",
mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
return -1;
}
if (mh.num_substreams == 0)
return -1;
if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
return -1;
}
if (mh.num_substreams > MAX_SUBSTREAMS) {
av_log(m->avctx, AV_LOG_ERROR,
"Number of substreams %d is larger than the maximum supported "
"by the decoder. %s\n", mh.num_substreams, sample_message);
return -1;
}
m->access_unit_size = mh.access_unit_size;
m->access_unit_size_pow2 = mh.access_unit_size_pow2;
m->num_substreams = mh.num_substreams;
m->max_decoded_substream = m->num_substreams - 1;
m->avctx->sample_rate = mh.group1_samplerate;
m->avctx->frame_size = mh.access_unit_size;
m->avctx->bits_per_raw_sample = mh.group1_bits;
if (mh.group1_bits > 16)
m->avctx->sample_fmt = SAMPLE_FMT_S32;
else
m->avctx->sample_fmt = SAMPLE_FMT_S16;
m->params_valid = 1;
for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
m->substream[substr].restart_seen = 0;
return 0;
}
/** Read a restart header from a block in a substream. This contains parameters
* required to decode the audio that do not change very often. Generally
* (always) present only in blocks following a major sync. */
static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
const uint8_t *buf, unsigned int substr)
{
SubStream *s = &m->substream[substr];
unsigned int ch;
int sync_word, tmp;
uint8_t checksum;
uint8_t lossless_check;
int start_count = get_bits_count(gbp);
const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
? MAX_MATRIX_CHANNEL_MLP
: MAX_MATRIX_CHANNEL_TRUEHD;
sync_word = get_bits(gbp, 13);
s->noise_type = get_bits1(gbp);
if ((m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) ||
sync_word != 0x31ea >> 1) {
av_log(m->avctx, AV_LOG_ERROR,
"restart header sync incorrect (got 0x%04x)\n", sync_word);
return -1;
}
skip_bits(gbp, 16); /* Output timestamp */
s->min_channel = get_bits(gbp, 4);
s->max_channel = get_bits(gbp, 4);
s->max_matrix_channel = get_bits(gbp, 4);
if (s->max_matrix_channel > max_matrix_channel) {
av_log(m->avctx, AV_LOG_ERROR,
"Max matrix channel cannot be greater than %d.\n",
max_matrix_channel);
return -1;
}
if (s->max_channel != s->max_matrix_channel) {
av_log(m->avctx, AV_LOG_ERROR,
"Max channel must be equal max matrix channel.\n");
return -1;
}
if (s->min_channel > s->max_channel) {
av_log(m->avctx, AV_LOG_ERROR,
"Substream min channel cannot be greater than max channel.\n");
return -1;
}
if (m->avctx->request_channels > 0
&& s->max_channel + 1 >= m->avctx->request_channels
&& substr < m->max_decoded_substream) {
av_log(m->avctx, AV_LOG_INFO,
"Extracting %d channel downmix from substream %d. "
"Further substreams will be skipped.\n",
s->max_channel + 1, substr);
m->max_decoded_substream = substr;
}
s->noise_shift = get_bits(gbp, 4);
s->noisegen_seed = get_bits(gbp, 23);
skip_bits(gbp, 19);
s->data_check_present = get_bits1(gbp);
lossless_check = get_bits(gbp, 8);
if (substr == m->max_decoded_substream
&& s->lossless_check_data != 0xffffffff) {
tmp = xor_32_to_8(s->lossless_check_data);
if (tmp != lossless_check)
av_log(m->avctx, AV_LOG_WARNING,
"Lossless check failed - expected %02x, calculated %02x.\n",
lossless_check, tmp);
}
skip_bits(gbp, 16);
memset(s->ch_assign, 0, sizeof(s->ch_assign));
for (ch = 0; ch <= s->max_matrix_channel; ch++) {
int ch_assign = get_bits(gbp, 6);
if (ch_assign > s->max_matrix_channel) {
av_log(m->avctx, AV_LOG_ERROR,
"Assignment of matrix channel %d to invalid output channel %d. %s\n",
ch, ch_assign, sample_message);
return -1;
}
s->ch_assign[ch_assign] = ch;
}
checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
if (checksum != get_bits(gbp, 8))
av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
/* Set default decoding parameters. */
s->param_presence_flags = 0xff;
s->num_primitive_matrices = 0;
s->blocksize = 8;
s->lossless_check_data = 0;
memset(s->output_shift , 0, sizeof(s->output_shift ));
memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
for (ch = s->min_channel; ch <= s->max_channel; ch++) {
ChannelParams *cp = &m->channel_params[ch];
cp->filter_params[FIR].order = 0;
cp->filter_params[IIR].order = 0;
cp->filter_params[FIR].shift = 0;
cp->filter_params[IIR].shift = 0;
/* Default audio coding is 24-bit raw PCM. */
cp->huff_offset = 0;
cp->sign_huff_offset = (-1) << 23;
cp->codebook = 0;
cp->huff_lsbs = 24;
}
if (substr == m->max_decoded_substream)
m->avctx->channels = s->max_matrix_channel + 1;
return 0;
}
/** Read parameters for one of the prediction filters. */
static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
unsigned int channel, unsigned int filter)
{
FilterParams *fp = &m->channel_params[channel].filter_params[filter];
const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
const char fchar = filter ? 'I' : 'F';
int i, order;
// Filter is 0 for FIR, 1 for IIR.
assert(filter < 2);
if (m->filter_changed[channel][filter]++ > 1) {
av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
return -1;
}
order = get_bits(gbp, 4);
if (order > max_order) {
av_log(m->avctx, AV_LOG_ERROR,
"%cIR filter order %d is greater than maximum %d.\n",
fchar, order, max_order);
return -1;
}
fp->order = order;
if (order > 0) {
int coeff_bits, coeff_shift;
fp->shift = get_bits(gbp, 4);
coeff_bits = get_bits(gbp, 5);
coeff_shift = get_bits(gbp, 3);
if (coeff_bits < 1 || coeff_bits > 16) {
av_log(m->avctx, AV_LOG_ERROR,
"%cIR filter coeff_bits must be between 1 and 16.\n",
fchar);
return -1;
}
if (coeff_bits + coeff_shift > 16) {
av_log(m->avctx, AV_LOG_ERROR,
"Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
fchar);
return -1;
}
for (i = 0; i < order; i++)
fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
if (get_bits1(gbp)) {
int state_bits, state_shift;
if (filter == FIR) {
av_log(m->avctx, AV_LOG_ERROR,
"FIR filter has state data specified.\n");
return -1;
}
state_bits = get_bits(gbp, 4);
state_shift = get_bits(gbp, 4);
/* TODO: Check validity of state data. */
for (i = 0; i < order; i++)
fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
}
}
return 0;
}
/** Read parameters for primitive matrices. */
static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
{
SubStream *s = &m->substream[substr];
unsigned int mat, ch;
const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
? MAX_MATRICES_MLP
: MAX_MATRICES_TRUEHD;
if (m->matrix_changed++ > 1) {
av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
return -1;
}
s->num_primitive_matrices = get_bits(gbp, 4);
if (s->num_primitive_matrices > max_primitive_matrices) {
av_log(m->avctx, AV_LOG_ERROR,
"Number of primitive matrices cannot be greater than %d.\n",
max_primitive_matrices);
return -1;
}
for (mat = 0; mat < s->num_primitive_matrices; mat++) {
int frac_bits, max_chan;
s->matrix_out_ch[mat] = get_bits(gbp, 4);
frac_bits = get_bits(gbp, 4);
s->lsb_bypass [mat] = get_bits1(gbp);
if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
av_log(m->avctx, AV_LOG_ERROR,
"Invalid channel %d specified as output from matrix.\n",
s->matrix_out_ch[mat]);
return -1;
}
if (frac_bits > 14) {
av_log(m->avctx, AV_LOG_ERROR,
"Too many fractional bits specified.\n");
return -1;
}
max_chan = s->max_matrix_channel;
if (!s->noise_type)
max_chan+=2;
for (ch = 0; ch <= max_chan; ch++) {
int coeff_val = 0;
if (get_bits1(gbp))
coeff_val = get_sbits(gbp, frac_bits + 2);
s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
}
if (s->noise_type)
s->matrix_noise_shift[mat] = get_bits(gbp, 4);
else
s->matrix_noise_shift[mat] = 0;
}
return 0;
}
/** Read channel parameters. */
static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
GetBitContext *gbp, unsigned int ch)
{
ChannelParams *cp = &m->channel_params[ch];
FilterParams *fir = &cp->filter_params[FIR];
FilterParams *iir = &cp->filter_params[IIR];
SubStream *s = &m->substream[substr];
if (s->param_presence_flags & PARAM_FIR)
if (get_bits1(gbp))
if (read_filter_params(m, gbp, ch, FIR) < 0)
return -1;
if (s->param_presence_flags & PARAM_IIR)
if (get_bits1(gbp))
if (read_filter_params(m, gbp, ch, IIR) < 0)
return -1;
if (fir->order + iir->order > 8) {
av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
return -1;
}
if (fir->order && iir->order &&
fir->shift != iir->shift) {
av_log(m->avctx, AV_LOG_ERROR,
"FIR and IIR filters must use the same precision.\n");
return -1;
}
/* The FIR and IIR filters must have the same precision.
* To simplify the filtering code, only the precision of the
* FIR filter is considered. If only the IIR filter is employed,
* the FIR filter precision is set to that of the IIR filter, so
* that the filtering code can use it. */
if (!fir->order && iir->order)
fir->shift = iir->shift;
if (s->param_presence_flags & PARAM_HUFFOFFSET)
if (get_bits1(gbp))
cp->huff_offset = get_sbits(gbp, 15);
cp->codebook = get_bits(gbp, 2);
cp->huff_lsbs = get_bits(gbp, 5);
if (cp->huff_lsbs > 24) {
av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
return -1;
}
cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
return 0;
}
/** Read decoding parameters that change more often than those in the restart
* header. */
static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
unsigned int substr)
{
SubStream *s = &m->substream[substr];
unsigned int ch;
if (s->param_presence_flags & PARAM_PRESENCE)
if (get_bits1(gbp))
s->param_presence_flags = get_bits(gbp, 8);
if (s->param_presence_flags & PARAM_BLOCKSIZE)
if (get_bits1(gbp)) {
s->blocksize = get_bits(gbp, 9);
if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
s->blocksize = 0;
return -1;
}
}
if (s->param_presence_flags & PARAM_MATRIX)
if (get_bits1(gbp))
if (read_matrix_params(m, substr, gbp) < 0)
return -1;
if (s->param_presence_flags & PARAM_OUTSHIFT)
if (get_bits1(gbp))
for (ch = 0; ch <= s->max_matrix_channel; ch++)
s->output_shift[ch] = get_sbits(gbp, 4);
if (s->param_presence_flags & PARAM_QUANTSTEP)
if (get_bits1(gbp))
for (ch = 0; ch <= s->max_channel; ch++) {
ChannelParams *cp = &m->channel_params[ch];
s->quant_step_size[ch] = get_bits(gbp, 4);
cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
}
for (ch = s->min_channel; ch <= s->max_channel; ch++)
if (get_bits1(gbp))
if (read_channel_params(m, substr, gbp, ch) < 0)
return -1;
return 0;
}
#define MSB_MASK(bits) (-1u << bits)
/** Generate PCM samples using the prediction filters and residual values
* read from the data stream, and update the filter state. */
static void filter_channel(MLPDecodeContext *m, unsigned int substr,
unsigned int channel)
{
SubStream *s = &m->substream[substr];
int32_t fir_state_buffer[MAX_BLOCKSIZE + MAX_FIR_ORDER];
int32_t iir_state_buffer[MAX_BLOCKSIZE + MAX_IIR_ORDER];
int32_t *firbuf = fir_state_buffer + MAX_BLOCKSIZE;
int32_t *iirbuf = iir_state_buffer + MAX_BLOCKSIZE;
FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
unsigned int filter_shift = fir->shift;
int32_t mask = MSB_MASK(s->quant_step_size[channel]);
memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
m->dsp.mlp_filter_channel(firbuf, fir->coeff, fir->order,
iirbuf, iir->coeff, iir->order,
filter_shift, mask, s->blocksize,
&m->sample_buffer[s->blockpos][channel]);
memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
}
/** Read a block of PCM residual data (or actual if no filtering active). */
static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
unsigned int substr)
{
SubStream *s = &m->substream[substr];
unsigned int i, ch, expected_stream_pos = 0;
if (s->data_check_present) {
expected_stream_pos = get_bits_count(gbp);
expected_stream_pos += get_bits(gbp, 16);
av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
"we have not tested yet. %s\n", sample_message);
}
if (s->blockpos + s->blocksize > m->access_unit_size) {
av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
return -1;
}
memset(&m->bypassed_lsbs[s->blockpos][0], 0,
s->blocksize * sizeof(m->bypassed_lsbs[0]));
for (i = 0; i < s->blocksize; i++)
if (read_huff_channels(m, gbp, substr, i) < 0)
return -1;
for (ch = s->min_channel; ch <= s->max_channel; ch++)
filter_channel(m, substr, ch);
s->blockpos += s->blocksize;
if (s->data_check_present) {
if (get_bits_count(gbp) != expected_stream_pos)
av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
skip_bits(gbp, 8);
}
return 0;
}
/** Data table used for TrueHD noise generation function. */
static const int8_t noise_table[256] = {
30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
-25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
};
/** Noise generation functions.
* I'm not sure what these are for - they seem to be some kind of pseudorandom
* sequence generators, used to generate noise data which is used when the
* channels are rematrixed. I'm not sure if they provide a practical benefit
* to compression, or just obfuscate the decoder. Are they for some kind of
* dithering? */
/** Generate two channels of noise, used in the matrix when
* restart sync word == 0x31ea. */
static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
{
SubStream *s = &m->substream[substr];
unsigned int i;
uint32_t seed = s->noisegen_seed;
unsigned int maxchan = s->max_matrix_channel;
for (i = 0; i < s->blockpos; i++) {
uint16_t seed_shr7 = seed >> 7;
m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
}
s->noisegen_seed = seed;
}
/** Generate a block of noise, used when restart sync word == 0x31eb. */
static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
{
SubStream *s = &m->substream[substr];
unsigned int i;
uint32_t seed = s->noisegen_seed;
for (i = 0; i < m->access_unit_size_pow2; i++) {
uint8_t seed_shr15 = seed >> 15;
m->noise_buffer[i] = noise_table[seed_shr15];
seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
}
s->noisegen_seed = seed;
}
/** Apply the channel matrices in turn to reconstruct the original audio
* samples. */
static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
{
SubStream *s = &m->substream[substr];
unsigned int mat, src_ch, i;
unsigned int maxchan;
maxchan = s->max_matrix_channel;
if (!s->noise_type) {
generate_2_noise_channels(m, substr);
maxchan += 2;
} else {
fill_noise_buffer(m, substr);
}
for (mat = 0; mat < s->num_primitive_matrices; mat++) {
int matrix_noise_shift = s->matrix_noise_shift[mat];
unsigned int dest_ch = s->matrix_out_ch[mat];
int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
int32_t *coeffs = s->matrix_coeff[mat];
int index = s->num_primitive_matrices - mat;
int index2 = 2 * index + 1;
/* TODO: DSPContext? */
for (i = 0; i < s->blockpos; i++) {
int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
int32_t *samples = m->sample_buffer[i];
int64_t accum = 0;
for (src_ch = 0; src_ch <= maxchan; src_ch++)
accum += (int64_t) samples[src_ch] * coeffs[src_ch];
if (matrix_noise_shift) {
index &= m->access_unit_size_pow2 - 1;
accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
index += index2;
}
samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
}
}
}
/** Write the audio data into the output buffer. */
static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
uint8_t *data, unsigned int *data_size, int is32)
{
SubStream *s = &m->substream[substr];
unsigned int i, out_ch = 0;
int32_t *data_32 = (int32_t*) data;
int16_t *data_16 = (int16_t*) data;
if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
return -1;
for (i = 0; i < s->blockpos; i++) {
for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
int mat_ch = s->ch_assign[out_ch];
int32_t sample = m->sample_buffer[i][mat_ch]
<< s->output_shift[mat_ch];
s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
if (is32) *data_32++ = sample << 8;
else *data_16++ = sample >> 8;
}
}
*data_size = i * out_ch * (is32 ? 4 : 2);
return 0;
}
static int output_data(MLPDecodeContext *m, unsigned int substr,
uint8_t *data, unsigned int *data_size)
{
if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
return output_data_internal(m, substr, data, data_size, 1);
else
return output_data_internal(m, substr, data, data_size, 0);
}
/** Read an access unit from the stream.
* Returns < 0 on error, 0 if not enough data is present in the input stream
* otherwise returns the number of bytes consumed. */
static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
AVPacket *avpkt)
{
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
MLPDecodeContext *m = avctx->priv_data;
GetBitContext gb;
unsigned int length, substr;
unsigned int substream_start;
unsigned int header_size = 4;
unsigned int substr_header_size = 0;
uint8_t substream_parity_present[MAX_SUBSTREAMS];
uint16_t substream_data_len[MAX_SUBSTREAMS];
uint8_t parity_bits;
if (buf_size < 4)
return 0;
length = (AV_RB16(buf) & 0xfff) * 2;
if (length > buf_size)
return -1;
init_get_bits(&gb, (buf + 4), (length - 4) * 8);
m->is_major_sync_unit = 0;
if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
if (read_major_sync(m, &gb) < 0)
goto error;
m->is_major_sync_unit = 1;
header_size += 28;
}
if (!m->params_valid) {
av_log(m->avctx, AV_LOG_WARNING,
"Stream parameters not seen; skipping frame.\n");
*data_size = 0;
return length;
}
substream_start = 0;
for (substr = 0; substr < m->num_substreams; substr++) {
int extraword_present, checkdata_present, end, nonrestart_substr;
extraword_present = get_bits1(&gb);
nonrestart_substr = get_bits1(&gb);
checkdata_present = get_bits1(&gb);
skip_bits1(&gb);
end = get_bits(&gb, 12) * 2;
substr_header_size += 2;
if (extraword_present) {
if (m->avctx->codec_id == CODEC_ID_MLP) {
av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
goto error;
}
skip_bits(&gb, 16);
substr_header_size += 2;
}
if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
goto error;
}
if (end + header_size + substr_header_size > length) {
av_log(m->avctx, AV_LOG_ERROR,
"Indicated length of substream %d data goes off end of "
"packet.\n", substr);
end = length - header_size - substr_header_size;
}
if (end < substream_start) {