3be27b40f833f04aa895475eeb0279ea903afda5
[libav.git] / libavcodec / mlpdec.c
1 /*
2 * MLP decoder
3 * Copyright (c) 2007-2008 Ian Caulfield
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.
11 *
12 * 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
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file libavcodec/mlpdec.c
24 * MLP decoder
25 */
26
27 #include <stdint.h>
28
29 #include "avcodec.h"
30 #include "libavutil/intreadwrite.h"
31 #include "bitstream.h"
32 #include "libavutil/crc.h"
33 #include "parser.h"
34 #include "mlp_parser.h"
35 #include "mlp.h"
36
37 /** number of bits used for VLC lookup - longest Huffman code is 9 */
38 #define VLC_BITS 9
39
40
41 static const char* sample_message =
42 "Please file a bug report following the instructions at "
43 "http://ffmpeg.org/bugreports.html and include "
44 "a sample of this file.";
45
46 typedef struct SubStream {
47 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
48 uint8_t restart_seen;
49
50 //@{
51 /** restart header data */
52 //! The type of noise to be used in the rematrix stage.
53 uint16_t noise_type;
54
55 //! The index of the first channel coded in this substream.
56 uint8_t min_channel;
57 //! The index of the last channel coded in this substream.
58 uint8_t max_channel;
59 //! The number of channels input into the rematrix stage.
60 uint8_t max_matrix_channel;
61 //! For each channel output by the matrix, the output channel to map it to
62 uint8_t ch_assign[MAX_CHANNELS];
63
64 //! The left shift applied to random noise in 0x31ea substreams.
65 uint8_t noise_shift;
66 //! The current seed value for the pseudorandom noise generator(s).
67 uint32_t noisegen_seed;
68
69 //! Set if the substream contains extra info to check the size of VLC blocks.
70 uint8_t data_check_present;
71
72 //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
73 uint8_t param_presence_flags;
74 #define PARAM_BLOCKSIZE (1 << 7)
75 #define PARAM_MATRIX (1 << 6)
76 #define PARAM_OUTSHIFT (1 << 5)
77 #define PARAM_QUANTSTEP (1 << 4)
78 #define PARAM_FIR (1 << 3)
79 #define PARAM_IIR (1 << 2)
80 #define PARAM_HUFFOFFSET (1 << 1)
81 #define PARAM_PRESENCE (1 << 0)
82 //@}
83
84 //@{
85 /** matrix data */
86
87 //! Number of matrices to be applied.
88 uint8_t num_primitive_matrices;
89
90 //! matrix output channel
91 uint8_t matrix_out_ch[MAX_MATRICES];
92
93 //! Whether the LSBs of the matrix output are encoded in the bitstream.
94 uint8_t lsb_bypass[MAX_MATRICES];
95 //! Matrix coefficients, stored as 2.14 fixed point.
96 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];
97 //! Left shift to apply to noise values in 0x31eb substreams.
98 uint8_t matrix_noise_shift[MAX_MATRICES];
99 //@}
100
101 //! Left shift to apply to Huffman-decoded residuals.
102 uint8_t quant_step_size[MAX_CHANNELS];
103
104 //! number of PCM samples in current audio block
105 uint16_t blocksize;
106 //! Number of PCM samples decoded so far in this frame.
107 uint16_t blockpos;
108
109 //! Left shift to apply to decoded PCM values to get final 24-bit output.
110 int8_t output_shift[MAX_CHANNELS];
111
112 //! Running XOR of all output samples.
113 int32_t lossless_check_data;
114
115 } SubStream;
116
117 typedef struct MLPDecodeContext {
118 AVCodecContext *avctx;
119
120 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
121 uint8_t params_valid;
122
123 //! Number of substreams contained within this stream.
124 uint8_t num_substreams;
125
126 //! Index of the last substream to decode - further substreams are skipped.
127 uint8_t max_decoded_substream;
128
129 //! number of PCM samples contained in each frame
130 int access_unit_size;
131 //! next power of two above the number of samples in each frame
132 int access_unit_size_pow2;
133
134 SubStream substream[MAX_SUBSTREAMS];
135
136 ChannelParams channel_params[MAX_CHANNELS];
137
138 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
139 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
140 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
141 } MLPDecodeContext;
142
143 static VLC huff_vlc[3];
144
145 /** Initialize static data, constant between all invocations of the codec. */
146
147 static av_cold void init_static(void)
148 {
149 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
150 &ff_mlp_huffman_tables[0][0][1], 2, 1,
151 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
152 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
153 &ff_mlp_huffman_tables[1][0][1], 2, 1,
154 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
155 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
156 &ff_mlp_huffman_tables[2][0][1], 2, 1,
157 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
158
159 ff_mlp_init_crc();
160 }
161
162 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
163 unsigned int substr, unsigned int ch)
164 {
165 ChannelParams *cp = &m->channel_params[ch];
166 SubStream *s = &m->substream[substr];
167 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
168 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
169 int32_t sign_huff_offset = cp->huff_offset;
170
171 if (cp->codebook > 0)
172 sign_huff_offset -= 7 << lsb_bits;
173
174 if (sign_shift >= 0)
175 sign_huff_offset -= 1 << sign_shift;
176
177 return sign_huff_offset;
178 }
179
180 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
181 * and plain LSBs. */
182
183 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
184 unsigned int substr, unsigned int pos)
185 {
186 SubStream *s = &m->substream[substr];
187 unsigned int mat, channel;
188
189 for (mat = 0; mat < s->num_primitive_matrices; mat++)
190 if (s->lsb_bypass[mat])
191 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
192
193 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
194 ChannelParams *cp = &m->channel_params[channel];
195 int codebook = cp->codebook;
196 int quant_step_size = s->quant_step_size[channel];
197 int lsb_bits = cp->huff_lsbs - quant_step_size;
198 int result = 0;
199
200 if (codebook > 0)
201 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
202 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
203
204 if (result < 0)
205 return -1;
206
207 if (lsb_bits > 0)
208 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
209
210 result += cp->sign_huff_offset;
211 result <<= quant_step_size;
212
213 m->sample_buffer[pos + s->blockpos][channel] = result;
214 }
215
216 return 0;
217 }
218
219 static av_cold int mlp_decode_init(AVCodecContext *avctx)
220 {
221 MLPDecodeContext *m = avctx->priv_data;
222 int substr;
223
224 init_static();
225 m->avctx = avctx;
226 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
227 m->substream[substr].lossless_check_data = 0xffffffff;
228
229 return 0;
230 }
231
232 /** Read a major sync info header - contains high level information about
233 * the stream - sample rate, channel arrangement etc. Most of this
234 * information is not actually necessary for decoding, only for playback.
235 */
236
237 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
238 {
239 MLPHeaderInfo mh;
240 int substr;
241
242 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
243 return -1;
244
245 if (mh.group1_bits == 0) {
246 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
247 return -1;
248 }
249 if (mh.group2_bits > mh.group1_bits) {
250 av_log(m->avctx, AV_LOG_ERROR,
251 "Channel group 2 cannot have more bits per sample than group 1.\n");
252 return -1;
253 }
254
255 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
256 av_log(m->avctx, AV_LOG_ERROR,
257 "Channel groups with differing sample rates are not currently supported.\n");
258 return -1;
259 }
260
261 if (mh.group1_samplerate == 0) {
262 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
263 return -1;
264 }
265 if (mh.group1_samplerate > MAX_SAMPLERATE) {
266 av_log(m->avctx, AV_LOG_ERROR,
267 "Sampling rate %d is greater than the supported maximum (%d).\n",
268 mh.group1_samplerate, MAX_SAMPLERATE);
269 return -1;
270 }
271 if (mh.access_unit_size > MAX_BLOCKSIZE) {
272 av_log(m->avctx, AV_LOG_ERROR,
273 "Block size %d is greater than the supported maximum (%d).\n",
274 mh.access_unit_size, MAX_BLOCKSIZE);
275 return -1;
276 }
277 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
278 av_log(m->avctx, AV_LOG_ERROR,
279 "Block size pow2 %d is greater than the supported maximum (%d).\n",
280 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
281 return -1;
282 }
283
284 if (mh.num_substreams == 0)
285 return -1;
286 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
287 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
288 return -1;
289 }
290 if (mh.num_substreams > MAX_SUBSTREAMS) {
291 av_log(m->avctx, AV_LOG_ERROR,
292 "Number of substreams %d is larger than the maximum supported "
293 "by the decoder. %s\n", mh.num_substreams, sample_message);
294 return -1;
295 }
296
297 m->access_unit_size = mh.access_unit_size;
298 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
299
300 m->num_substreams = mh.num_substreams;
301 m->max_decoded_substream = m->num_substreams - 1;
302
303 m->avctx->sample_rate = mh.group1_samplerate;
304 m->avctx->frame_size = mh.access_unit_size;
305
306 m->avctx->bits_per_raw_sample = mh.group1_bits;
307 if (mh.group1_bits > 16)
308 m->avctx->sample_fmt = SAMPLE_FMT_S32;
309 else
310 m->avctx->sample_fmt = SAMPLE_FMT_S16;
311
312 m->params_valid = 1;
313 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
314 m->substream[substr].restart_seen = 0;
315
316 return 0;
317 }
318
319 /** Read a restart header from a block in a substream. This contains parameters
320 * required to decode the audio that do not change very often. Generally
321 * (always) present only in blocks following a major sync. */
322
323 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
324 const uint8_t *buf, unsigned int substr)
325 {
326 SubStream *s = &m->substream[substr];
327 unsigned int ch;
328 int sync_word, tmp;
329 uint8_t checksum;
330 uint8_t lossless_check;
331 int start_count = get_bits_count(gbp);
332
333 sync_word = get_bits(gbp, 13);
334
335 if (sync_word != 0x31ea >> 1) {
336 av_log(m->avctx, AV_LOG_ERROR,
337 "restart header sync incorrect (got 0x%04x)\n", sync_word);
338 return -1;
339 }
340 s->noise_type = get_bits1(gbp);
341
342 skip_bits(gbp, 16); /* Output timestamp */
343
344 s->min_channel = get_bits(gbp, 4);
345 s->max_channel = get_bits(gbp, 4);
346 s->max_matrix_channel = get_bits(gbp, 4);
347
348 if (s->min_channel > s->max_channel) {
349 av_log(m->avctx, AV_LOG_ERROR,
350 "Substream min channel cannot be greater than max channel.\n");
351 return -1;
352 }
353
354 if (m->avctx->request_channels > 0
355 && s->max_channel + 1 >= m->avctx->request_channels
356 && substr < m->max_decoded_substream) {
357 av_log(m->avctx, AV_LOG_INFO,
358 "Extracting %d channel downmix from substream %d. "
359 "Further substreams will be skipped.\n",
360 s->max_channel + 1, substr);
361 m->max_decoded_substream = substr;
362 }
363
364 s->noise_shift = get_bits(gbp, 4);
365 s->noisegen_seed = get_bits(gbp, 23);
366
367 skip_bits(gbp, 19);
368
369 s->data_check_present = get_bits1(gbp);
370 lossless_check = get_bits(gbp, 8);
371 if (substr == m->max_decoded_substream
372 && s->lossless_check_data != 0xffffffff) {
373 tmp = xor_32_to_8(s->lossless_check_data);
374 if (tmp != lossless_check)
375 av_log(m->avctx, AV_LOG_WARNING,
376 "Lossless check failed - expected %02x, calculated %02x.\n",
377 lossless_check, tmp);
378 }
379
380 skip_bits(gbp, 16);
381
382 memset(s->ch_assign, 0, sizeof(s->ch_assign));
383
384 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
385 int ch_assign = get_bits(gbp, 6);
386 if (ch_assign > s->max_matrix_channel) {
387 av_log(m->avctx, AV_LOG_ERROR,
388 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
389 ch, ch_assign, sample_message);
390 return -1;
391 }
392 s->ch_assign[ch_assign] = ch;
393 }
394
395 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
396
397 if (checksum != get_bits(gbp, 8))
398 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
399
400 /* Set default decoding parameters. */
401 s->param_presence_flags = 0xff;
402 s->num_primitive_matrices = 0;
403 s->blocksize = 8;
404 s->lossless_check_data = 0;
405
406 memset(s->output_shift , 0, sizeof(s->output_shift ));
407 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
408
409 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
410 ChannelParams *cp = &m->channel_params[ch];
411 cp->filter_params[FIR].order = 0;
412 cp->filter_params[IIR].order = 0;
413 cp->filter_params[FIR].shift = 0;
414 cp->filter_params[IIR].shift = 0;
415
416 /* Default audio coding is 24-bit raw PCM. */
417 cp->huff_offset = 0;
418 cp->sign_huff_offset = (-1) << 23;
419 cp->codebook = 0;
420 cp->huff_lsbs = 24;
421 }
422
423 if (substr == m->max_decoded_substream) {
424 m->avctx->channels = s->max_matrix_channel + 1;
425 }
426
427 return 0;
428 }
429
430 /** Read parameters for one of the prediction filters. */
431
432 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
433 unsigned int channel, unsigned int filter)
434 {
435 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
436 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
437 const char fchar = filter ? 'I' : 'F';
438 int i, order;
439
440 // Filter is 0 for FIR, 1 for IIR.
441 assert(filter < 2);
442
443 order = get_bits(gbp, 4);
444 if (order > max_order) {
445 av_log(m->avctx, AV_LOG_ERROR,
446 "%cIR filter order %d is greater than maximum %d.\n",
447 fchar, order, max_order);
448 return -1;
449 }
450 fp->order = order;
451
452 if (order > 0) {
453 int coeff_bits, coeff_shift;
454
455 fp->shift = get_bits(gbp, 4);
456
457 coeff_bits = get_bits(gbp, 5);
458 coeff_shift = get_bits(gbp, 3);
459 if (coeff_bits < 1 || coeff_bits > 16) {
460 av_log(m->avctx, AV_LOG_ERROR,
461 "%cIR filter coeff_bits must be between 1 and 16.\n",
462 fchar);
463 return -1;
464 }
465 if (coeff_bits + coeff_shift > 16) {
466 av_log(m->avctx, AV_LOG_ERROR,
467 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
468 fchar);
469 return -1;
470 }
471
472 for (i = 0; i < order; i++)
473 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
474
475 if (get_bits1(gbp)) {
476 int state_bits, state_shift;
477
478 if (filter == FIR) {
479 av_log(m->avctx, AV_LOG_ERROR,
480 "FIR filter has state data specified.\n");
481 return -1;
482 }
483
484 state_bits = get_bits(gbp, 4);
485 state_shift = get_bits(gbp, 4);
486
487 /* TODO: Check validity of state data. */
488
489 for (i = 0; i < order; i++)
490 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
491 }
492 }
493
494 return 0;
495 }
496
497 /** Read parameters for primitive matrices. */
498
499 static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
500 {
501 unsigned int mat, ch;
502
503 s->num_primitive_matrices = get_bits(gbp, 4);
504
505 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
506 int frac_bits, max_chan;
507 s->matrix_out_ch[mat] = get_bits(gbp, 4);
508 frac_bits = get_bits(gbp, 4);
509 s->lsb_bypass [mat] = get_bits1(gbp);
510
511 if (s->matrix_out_ch[mat] > s->max_channel) {
512 av_log(m->avctx, AV_LOG_ERROR,
513 "Invalid channel %d specified as output from matrix.\n",
514 s->matrix_out_ch[mat]);
515 return -1;
516 }
517 if (frac_bits > 14) {
518 av_log(m->avctx, AV_LOG_ERROR,
519 "Too many fractional bits specified.\n");
520 return -1;
521 }
522
523 max_chan = s->max_matrix_channel;
524 if (!s->noise_type)
525 max_chan+=2;
526
527 for (ch = 0; ch <= max_chan; ch++) {
528 int coeff_val = 0;
529 if (get_bits1(gbp))
530 coeff_val = get_sbits(gbp, frac_bits + 2);
531
532 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
533 }
534
535 if (s->noise_type)
536 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
537 else
538 s->matrix_noise_shift[mat] = 0;
539 }
540
541 return 0;
542 }
543
544 /** Read channel parameters. */
545
546 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
547 GetBitContext *gbp, unsigned int ch)
548 {
549 ChannelParams *cp = &m->channel_params[ch];
550 FilterParams *fir = &cp->filter_params[FIR];
551 FilterParams *iir = &cp->filter_params[IIR];
552 SubStream *s = &m->substream[substr];
553
554 if (s->param_presence_flags & PARAM_FIR)
555 if (get_bits1(gbp))
556 if (read_filter_params(m, gbp, ch, FIR) < 0)
557 return -1;
558
559 if (s->param_presence_flags & PARAM_IIR)
560 if (get_bits1(gbp))
561 if (read_filter_params(m, gbp, ch, IIR) < 0)
562 return -1;
563
564 if (fir->order + iir->order > 8) {
565 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
566 return -1;
567 }
568
569 if (fir->order && iir->order &&
570 fir->shift != iir->shift) {
571 av_log(m->avctx, AV_LOG_ERROR,
572 "FIR and IIR filters must use the same precision.\n");
573 return -1;
574 }
575 /* The FIR and IIR filters must have the same precision.
576 * To simplify the filtering code, only the precision of the
577 * FIR filter is considered. If only the IIR filter is employed,
578 * the FIR filter precision is set to that of the IIR filter, so
579 * that the filtering code can use it. */
580 if (!fir->order && iir->order)
581 fir->shift = iir->shift;
582
583 if (s->param_presence_flags & PARAM_HUFFOFFSET)
584 if (get_bits1(gbp))
585 cp->huff_offset = get_sbits(gbp, 15);
586
587 cp->codebook = get_bits(gbp, 2);
588 cp->huff_lsbs = get_bits(gbp, 5);
589
590 if (cp->huff_lsbs > 24) {
591 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
592 return -1;
593 }
594
595 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
596
597 return 0;
598 }
599
600 /** Read decoding parameters that change more often than those in the restart
601 * header. */
602
603 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
604 unsigned int substr)
605 {
606 SubStream *s = &m->substream[substr];
607 unsigned int ch;
608
609 if (s->param_presence_flags & PARAM_PRESENCE)
610 if (get_bits1(gbp))
611 s->param_presence_flags = get_bits(gbp, 8);
612
613 if (s->param_presence_flags & PARAM_BLOCKSIZE)
614 if (get_bits1(gbp)) {
615 s->blocksize = get_bits(gbp, 9);
616 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
617 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
618 s->blocksize = 0;
619 return -1;
620 }
621 }
622
623 if (s->param_presence_flags & PARAM_MATRIX)
624 if (get_bits1(gbp)) {
625 if (read_matrix_params(m, s, gbp) < 0)
626 return -1;
627 }
628
629 if (s->param_presence_flags & PARAM_OUTSHIFT)
630 if (get_bits1(gbp))
631 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
632 s->output_shift[ch] = get_sbits(gbp, 4);
633 }
634
635 if (s->param_presence_flags & PARAM_QUANTSTEP)
636 if (get_bits1(gbp))
637 for (ch = 0; ch <= s->max_channel; ch++) {
638 ChannelParams *cp = &m->channel_params[ch];
639
640 s->quant_step_size[ch] = get_bits(gbp, 4);
641
642 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
643 }
644
645 for (ch = s->min_channel; ch <= s->max_channel; ch++)
646 if (get_bits1(gbp)) {
647 if (read_channel_params(m, substr, gbp, ch) < 0)
648 return -1;
649 }
650
651 return 0;
652 }
653
654 #define MSB_MASK(bits) (-1u << bits)
655
656 /** Generate PCM samples using the prediction filters and residual values
657 * read from the data stream, and update the filter state. */
658
659 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
660 unsigned int channel)
661 {
662 SubStream *s = &m->substream[substr];
663 int32_t firbuf[MAX_BLOCKSIZE + MAX_FIR_ORDER];
664 int32_t iirbuf[MAX_BLOCKSIZE + MAX_IIR_ORDER];
665 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
666 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
667 unsigned int filter_shift = fir->shift;
668 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
669 int index = MAX_BLOCKSIZE;
670 int i;
671
672 memcpy(&firbuf[index], fir->state, MAX_FIR_ORDER * sizeof(int32_t));
673 memcpy(&iirbuf[index], iir->state, MAX_IIR_ORDER * sizeof(int32_t));
674
675 for (i = 0; i < s->blocksize; i++) {
676 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
677 unsigned int order;
678 int64_t accum = 0;
679 int32_t result;
680
681 /* TODO: Move this code to DSPContext? */
682
683 for (order = 0; order < fir->order; order++)
684 accum += (int64_t) firbuf[index + order] * fir->coeff[order];
685 for (order = 0; order < iir->order; order++)
686 accum += (int64_t) iirbuf[index + order] * iir->coeff[order];
687
688 accum = accum >> filter_shift;
689 result = (accum + residual) & mask;
690
691 --index;
692
693 firbuf[index] = result;
694 iirbuf[index] = result - accum;
695
696 m->sample_buffer[i + s->blockpos][channel] = result;
697 }
698
699 memcpy(fir->state, &firbuf[index], MAX_FIR_ORDER * sizeof(int32_t));
700 memcpy(iir->state, &iirbuf[index], MAX_IIR_ORDER * sizeof(int32_t));
701 }
702
703 /** Read a block of PCM residual data (or actual if no filtering active). */
704
705 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
706 unsigned int substr)
707 {
708 SubStream *s = &m->substream[substr];
709 unsigned int i, ch, expected_stream_pos = 0;
710
711 if (s->data_check_present) {
712 expected_stream_pos = get_bits_count(gbp);
713 expected_stream_pos += get_bits(gbp, 16);
714 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
715 "we have not tested yet. %s\n", sample_message);
716 }
717
718 if (s->blockpos + s->blocksize > m->access_unit_size) {
719 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
720 return -1;
721 }
722
723 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
724 s->blocksize * sizeof(m->bypassed_lsbs[0]));
725
726 for (i = 0; i < s->blocksize; i++) {
727 if (read_huff_channels(m, gbp, substr, i) < 0)
728 return -1;
729 }
730
731 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
732 filter_channel(m, substr, ch);
733 }
734
735 s->blockpos += s->blocksize;
736
737 if (s->data_check_present) {
738 if (get_bits_count(gbp) != expected_stream_pos)
739 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
740 skip_bits(gbp, 8);
741 }
742
743 return 0;
744 }
745
746 /** Data table used for TrueHD noise generation function. */
747
748 static const int8_t noise_table[256] = {
749 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
750 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
751 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
752 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
753 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
754 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
755 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
756 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
757 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
758 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
759 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
760 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
761 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
762 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
763 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
764 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
765 };
766
767 /** Noise generation functions.
768 * I'm not sure what these are for - they seem to be some kind of pseudorandom
769 * sequence generators, used to generate noise data which is used when the
770 * channels are rematrixed. I'm not sure if they provide a practical benefit
771 * to compression, or just obfuscate the decoder. Are they for some kind of
772 * dithering? */
773
774 /** Generate two channels of noise, used in the matrix when
775 * restart sync word == 0x31ea. */
776
777 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
778 {
779 SubStream *s = &m->substream[substr];
780 unsigned int i;
781 uint32_t seed = s->noisegen_seed;
782 unsigned int maxchan = s->max_matrix_channel;
783
784 for (i = 0; i < s->blockpos; i++) {
785 uint16_t seed_shr7 = seed >> 7;
786 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
787 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
788
789 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
790 }
791
792 s->noisegen_seed = seed;
793 }
794
795 /** Generate a block of noise, used when restart sync word == 0x31eb. */
796
797 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
798 {
799 SubStream *s = &m->substream[substr];
800 unsigned int i;
801 uint32_t seed = s->noisegen_seed;
802
803 for (i = 0; i < m->access_unit_size_pow2; i++) {
804 uint8_t seed_shr15 = seed >> 15;
805 m->noise_buffer[i] = noise_table[seed_shr15];
806 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
807 }
808
809 s->noisegen_seed = seed;
810 }
811
812
813 /** Apply the channel matrices in turn to reconstruct the original audio
814 * samples. */
815
816 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
817 {
818 SubStream *s = &m->substream[substr];
819 unsigned int mat, src_ch, i;
820 unsigned int maxchan;
821
822 maxchan = s->max_matrix_channel;
823 if (!s->noise_type) {
824 generate_2_noise_channels(m, substr);
825 maxchan += 2;
826 } else {
827 fill_noise_buffer(m, substr);
828 }
829
830 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
831 int matrix_noise_shift = s->matrix_noise_shift[mat];
832 unsigned int dest_ch = s->matrix_out_ch[mat];
833 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
834
835 /* TODO: DSPContext? */
836
837 for (i = 0; i < s->blockpos; i++) {
838 int64_t accum = 0;
839 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
840 accum += (int64_t)m->sample_buffer[i][src_ch]
841 * s->matrix_coeff[mat][src_ch];
842 }
843 if (matrix_noise_shift) {
844 uint32_t index = s->num_primitive_matrices - mat;
845 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
846 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
847 }
848 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
849 + m->bypassed_lsbs[i][mat];
850 }
851 }
852 }
853
854 /** Write the audio data into the output buffer. */
855
856 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
857 uint8_t *data, unsigned int *data_size, int is32)
858 {
859 SubStream *s = &m->substream[substr];
860 unsigned int i, out_ch = 0;
861 int32_t *data_32 = (int32_t*) data;
862 int16_t *data_16 = (int16_t*) data;
863
864 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
865 return -1;
866
867 for (i = 0; i < s->blockpos; i++) {
868 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
869 int mat_ch = s->ch_assign[out_ch];
870 int32_t sample = m->sample_buffer[i][mat_ch]
871 << s->output_shift[mat_ch];
872 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
873 if (is32) *data_32++ = sample << 8;
874 else *data_16++ = sample >> 8;
875 }
876 }
877
878 *data_size = i * out_ch * (is32 ? 4 : 2);
879
880 return 0;
881 }
882
883 static int output_data(MLPDecodeContext *m, unsigned int substr,
884 uint8_t *data, unsigned int *data_size)
885 {
886 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
887 return output_data_internal(m, substr, data, data_size, 1);
888 else
889 return output_data_internal(m, substr, data, data_size, 0);
890 }
891
892
893 /** Read an access unit from the stream.
894 * Returns < 0 on error, 0 if not enough data is present in the input stream
895 * otherwise returns the number of bytes consumed. */
896
897 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
898 const uint8_t *buf, int buf_size)
899 {
900 MLPDecodeContext *m = avctx->priv_data;
901 GetBitContext gb;
902 unsigned int length, substr;
903 unsigned int substream_start;
904 unsigned int header_size = 4;
905 unsigned int substr_header_size = 0;
906 uint8_t substream_parity_present[MAX_SUBSTREAMS];
907 uint16_t substream_data_len[MAX_SUBSTREAMS];
908 uint8_t parity_bits;
909
910 if (buf_size < 4)
911 return 0;
912
913 length = (AV_RB16(buf) & 0xfff) * 2;
914
915 if (length > buf_size)
916 return -1;
917
918 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
919
920 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
921 if (read_major_sync(m, &gb) < 0)
922 goto error;
923 header_size += 28;
924 }
925
926 if (!m->params_valid) {
927 av_log(m->avctx, AV_LOG_WARNING,
928 "Stream parameters not seen; skipping frame.\n");
929 *data_size = 0;
930 return length;
931 }
932
933 substream_start = 0;
934
935 for (substr = 0; substr < m->num_substreams; substr++) {
936 int extraword_present, checkdata_present, end;
937
938 extraword_present = get_bits1(&gb);
939 skip_bits1(&gb);
940 checkdata_present = get_bits1(&gb);
941 skip_bits1(&gb);
942
943 end = get_bits(&gb, 12) * 2;
944
945 substr_header_size += 2;
946
947 if (extraword_present) {
948 skip_bits(&gb, 16);
949 substr_header_size += 2;
950 }
951
952 if (end + header_size + substr_header_size > length) {
953 av_log(m->avctx, AV_LOG_ERROR,
954 "Indicated length of substream %d data goes off end of "
955 "packet.\n", substr);
956 end = length - header_size - substr_header_size;
957 }
958
959 if (end < substream_start) {
960 av_log(avctx, AV_LOG_ERROR,
961 "Indicated end offset of substream %d data "
962 "is smaller than calculated start offset.\n",
963 substr);
964 goto error;
965 }
966
967 if (substr > m->max_decoded_substream)
968 continue;
969
970 substream_parity_present[substr] = checkdata_present;
971 substream_data_len[substr] = end - substream_start;
972 substream_start = end;
973 }
974
975 parity_bits = ff_mlp_calculate_parity(buf, 4);
976 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
977
978 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
979 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
980 goto error;
981 }
982
983 buf += header_size + substr_header_size;
984
985 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
986 SubStream *s = &m->substream[substr];
987 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
988
989 s->blockpos = 0;
990 do {
991 if (get_bits1(&gb)) {
992 if (get_bits1(&gb)) {
993 /* A restart header should be present. */
994 if (read_restart_header(m, &gb, buf, substr) < 0)
995 goto next_substr;
996 s->restart_seen = 1;
997 }
998
999 if (!s->restart_seen) {
1000 av_log(m->avctx, AV_LOG_ERROR,
1001 "No restart header present in substream %d.\n",
1002 substr);
1003 goto next_substr;
1004 }
1005
1006 if (read_decoding_params(m, &gb, substr) < 0)
1007 goto next_substr;
1008 }
1009
1010 if (!s->restart_seen) {
1011 av_log(m->avctx, AV_LOG_ERROR,
1012 "No restart header present in substream %d.\n",
1013 substr);
1014 goto next_substr;
1015 }
1016
1017 if (read_block_data(m, &gb, substr) < 0)
1018 return -1;
1019
1020 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1021 && get_bits1(&gb) == 0);
1022
1023 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1024 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1025 int shorten_by;
1026
1027 if (get_bits(&gb, 16) != 0xD234)
1028 return -1;
1029
1030 shorten_by = get_bits(&gb, 16);
1031 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1032 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1033 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1034 return -1;
1035
1036 if (substr == m->max_decoded_substream)
1037 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1038 }
1039 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1040 substream_parity_present[substr]) {
1041 uint8_t parity, checksum;
1042
1043 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1044 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1045 av_log(m->avctx, AV_LOG_ERROR,
1046 "Substream %d parity check failed.\n", substr);
1047
1048 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
1049 if (checksum != get_bits(&gb, 8))
1050 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
1051 substr);
1052 }
1053 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1054 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
1055 substr);
1056 return -1;
1057 }
1058
1059 next_substr:
1060 buf += substream_data_len[substr];
1061 }
1062
1063 rematrix_channels(m, m->max_decoded_substream);
1064
1065 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1066 return -1;
1067
1068 return length;
1069
1070 error:
1071 m->params_valid = 0;
1072 return -1;
1073 }
1074
1075 #if CONFIG_MLP_DECODER
1076 AVCodec mlp_decoder = {
1077 "mlp",
1078 CODEC_TYPE_AUDIO,
1079 CODEC_ID_MLP,
1080 sizeof(MLPDecodeContext),
1081 mlp_decode_init,
1082 NULL,
1083 NULL,
1084 read_access_unit,
1085 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1086 };
1087 #endif /* CONFIG_MLP_DECODER */
1088
1089 #if CONFIG_TRUEHD_DECODER
1090 AVCodec truehd_decoder = {
1091 "truehd",
1092 CODEC_TYPE_AUDIO,
1093 CODEC_ID_TRUEHD,
1094 sizeof(MLPDecodeContext),
1095 mlp_decode_init,
1096 NULL,
1097 NULL,
1098 read_access_unit,
1099 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1100 };
1101 #endif /* CONFIG_TRUEHD_DECODER */