0a64d79897bf2751ce1a929bdecdcd681ab9b312
[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 "dsputil.h"
31 #include "libavutil/intreadwrite.h"
32 #include "get_bits.h"
33 #include "libavutil/crc.h"
34 #include "parser.h"
35 #include "mlp_parser.h"
36 #include "mlp.h"
37
38 /** number of bits used for VLC lookup - longest Huffman code is 9 */
39 #define VLC_BITS 9
40
41
42 static const char* sample_message =
43 "Please file a bug report following the instructions at "
44 "http://ffmpeg.org/bugreports.html and include "
45 "a sample of this file.";
46
47 typedef struct SubStream {
48 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
49 uint8_t restart_seen;
50
51 //@{
52 /** restart header data */
53 //! The type of noise to be used in the rematrix stage.
54 uint16_t noise_type;
55
56 //! The index of the first channel coded in this substream.
57 uint8_t min_channel;
58 //! The index of the last channel coded in this substream.
59 uint8_t max_channel;
60 //! The number of channels input into the rematrix stage.
61 uint8_t max_matrix_channel;
62 //! For each channel output by the matrix, the output channel to map it to
63 uint8_t ch_assign[MAX_CHANNELS];
64
65 //! The left shift applied to random noise in 0x31ea substreams.
66 uint8_t noise_shift;
67 //! The current seed value for the pseudorandom noise generator(s).
68 uint32_t noisegen_seed;
69
70 //! Set if the substream contains extra info to check the size of VLC blocks.
71 uint8_t data_check_present;
72
73 //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
74 uint8_t param_presence_flags;
75 #define PARAM_BLOCKSIZE (1 << 7)
76 #define PARAM_MATRIX (1 << 6)
77 #define PARAM_OUTSHIFT (1 << 5)
78 #define PARAM_QUANTSTEP (1 << 4)
79 #define PARAM_FIR (1 << 3)
80 #define PARAM_IIR (1 << 2)
81 #define PARAM_HUFFOFFSET (1 << 1)
82 #define PARAM_PRESENCE (1 << 0)
83 //@}
84
85 //@{
86 /** matrix data */
87
88 //! Number of matrices to be applied.
89 uint8_t num_primitive_matrices;
90
91 //! matrix output channel
92 uint8_t matrix_out_ch[MAX_MATRICES];
93
94 //! Whether the LSBs of the matrix output are encoded in the bitstream.
95 uint8_t lsb_bypass[MAX_MATRICES];
96 //! Matrix coefficients, stored as 2.14 fixed point.
97 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS];
98 //! Left shift to apply to noise values in 0x31eb substreams.
99 uint8_t matrix_noise_shift[MAX_MATRICES];
100 //@}
101
102 //! Left shift to apply to Huffman-decoded residuals.
103 uint8_t quant_step_size[MAX_CHANNELS];
104
105 //! number of PCM samples in current audio block
106 uint16_t blocksize;
107 //! Number of PCM samples decoded so far in this frame.
108 uint16_t blockpos;
109
110 //! Left shift to apply to decoded PCM values to get final 24-bit output.
111 int8_t output_shift[MAX_CHANNELS];
112
113 //! Running XOR of all output samples.
114 int32_t lossless_check_data;
115
116 } SubStream;
117
118 typedef struct MLPDecodeContext {
119 AVCodecContext *avctx;
120
121 //! Current access unit being read has a major sync.
122 int is_major_sync_unit;
123
124 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
125 uint8_t params_valid;
126
127 //! Number of substreams contained within this stream.
128 uint8_t num_substreams;
129
130 //! Index of the last substream to decode - further substreams are skipped.
131 uint8_t max_decoded_substream;
132
133 //! number of PCM samples contained in each frame
134 int access_unit_size;
135 //! next power of two above the number of samples in each frame
136 int access_unit_size_pow2;
137
138 SubStream substream[MAX_SUBSTREAMS];
139
140 ChannelParams channel_params[MAX_CHANNELS];
141
142 int matrix_changed;
143 int filter_changed[MAX_CHANNELS][NUM_FILTERS];
144
145 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
146 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
147 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS];
148
149 DSPContext dsp;
150 } MLPDecodeContext;
151
152 static VLC huff_vlc[3];
153
154 /** Initialize static data, constant between all invocations of the codec. */
155
156 static av_cold void init_static(void)
157 {
158 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
159 &ff_mlp_huffman_tables[0][0][1], 2, 1,
160 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
161 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
162 &ff_mlp_huffman_tables[1][0][1], 2, 1,
163 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
164 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
165 &ff_mlp_huffman_tables[2][0][1], 2, 1,
166 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
167
168 ff_mlp_init_crc();
169 }
170
171 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
172 unsigned int substr, unsigned int ch)
173 {
174 ChannelParams *cp = &m->channel_params[ch];
175 SubStream *s = &m->substream[substr];
176 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
177 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
178 int32_t sign_huff_offset = cp->huff_offset;
179
180 if (cp->codebook > 0)
181 sign_huff_offset -= 7 << lsb_bits;
182
183 if (sign_shift >= 0)
184 sign_huff_offset -= 1 << sign_shift;
185
186 return sign_huff_offset;
187 }
188
189 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
190 * and plain LSBs. */
191
192 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
193 unsigned int substr, unsigned int pos)
194 {
195 SubStream *s = &m->substream[substr];
196 unsigned int mat, channel;
197
198 for (mat = 0; mat < s->num_primitive_matrices; mat++)
199 if (s->lsb_bypass[mat])
200 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
201
202 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
203 ChannelParams *cp = &m->channel_params[channel];
204 int codebook = cp->codebook;
205 int quant_step_size = s->quant_step_size[channel];
206 int lsb_bits = cp->huff_lsbs - quant_step_size;
207 int result = 0;
208
209 if (codebook > 0)
210 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
211 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
212
213 if (result < 0)
214 return -1;
215
216 if (lsb_bits > 0)
217 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
218
219 result += cp->sign_huff_offset;
220 result <<= quant_step_size;
221
222 m->sample_buffer[pos + s->blockpos][channel] = result;
223 }
224
225 return 0;
226 }
227
228 static av_cold int mlp_decode_init(AVCodecContext *avctx)
229 {
230 MLPDecodeContext *m = avctx->priv_data;
231 int substr;
232
233 init_static();
234 m->avctx = avctx;
235 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
236 m->substream[substr].lossless_check_data = 0xffffffff;
237 dsputil_init(&m->dsp, avctx);
238
239 return 0;
240 }
241
242 /** Read a major sync info header - contains high level information about
243 * the stream - sample rate, channel arrangement etc. Most of this
244 * information is not actually necessary for decoding, only for playback.
245 */
246
247 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
248 {
249 MLPHeaderInfo mh;
250 int substr;
251
252 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
253 return -1;
254
255 if (mh.group1_bits == 0) {
256 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
257 return -1;
258 }
259 if (mh.group2_bits > mh.group1_bits) {
260 av_log(m->avctx, AV_LOG_ERROR,
261 "Channel group 2 cannot have more bits per sample than group 1.\n");
262 return -1;
263 }
264
265 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
266 av_log(m->avctx, AV_LOG_ERROR,
267 "Channel groups with differing sample rates are not currently supported.\n");
268 return -1;
269 }
270
271 if (mh.group1_samplerate == 0) {
272 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
273 return -1;
274 }
275 if (mh.group1_samplerate > MAX_SAMPLERATE) {
276 av_log(m->avctx, AV_LOG_ERROR,
277 "Sampling rate %d is greater than the supported maximum (%d).\n",
278 mh.group1_samplerate, MAX_SAMPLERATE);
279 return -1;
280 }
281 if (mh.access_unit_size > MAX_BLOCKSIZE) {
282 av_log(m->avctx, AV_LOG_ERROR,
283 "Block size %d is greater than the supported maximum (%d).\n",
284 mh.access_unit_size, MAX_BLOCKSIZE);
285 return -1;
286 }
287 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
288 av_log(m->avctx, AV_LOG_ERROR,
289 "Block size pow2 %d is greater than the supported maximum (%d).\n",
290 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
291 return -1;
292 }
293
294 if (mh.num_substreams == 0)
295 return -1;
296 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
297 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
298 return -1;
299 }
300 if (mh.num_substreams > MAX_SUBSTREAMS) {
301 av_log(m->avctx, AV_LOG_ERROR,
302 "Number of substreams %d is larger than the maximum supported "
303 "by the decoder. %s\n", mh.num_substreams, sample_message);
304 return -1;
305 }
306
307 m->access_unit_size = mh.access_unit_size;
308 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
309
310 m->num_substreams = mh.num_substreams;
311 m->max_decoded_substream = m->num_substreams - 1;
312
313 m->avctx->sample_rate = mh.group1_samplerate;
314 m->avctx->frame_size = mh.access_unit_size;
315
316 m->avctx->bits_per_raw_sample = mh.group1_bits;
317 if (mh.group1_bits > 16)
318 m->avctx->sample_fmt = SAMPLE_FMT_S32;
319 else
320 m->avctx->sample_fmt = SAMPLE_FMT_S16;
321
322 m->params_valid = 1;
323 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
324 m->substream[substr].restart_seen = 0;
325
326 return 0;
327 }
328
329 /** Read a restart header from a block in a substream. This contains parameters
330 * required to decode the audio that do not change very often. Generally
331 * (always) present only in blocks following a major sync. */
332
333 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
334 const uint8_t *buf, unsigned int substr)
335 {
336 SubStream *s = &m->substream[substr];
337 unsigned int ch;
338 int sync_word, tmp;
339 uint8_t checksum;
340 uint8_t lossless_check;
341 int start_count = get_bits_count(gbp);
342 const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
343 ? MAX_MATRIX_CHANNEL_MLP
344 : MAX_MATRIX_CHANNEL_TRUEHD;
345
346 sync_word = get_bits(gbp, 13);
347
348 if (sync_word != 0x31ea >> 1) {
349 av_log(m->avctx, AV_LOG_ERROR,
350 "restart header sync incorrect (got 0x%04x)\n", sync_word);
351 return -1;
352 }
353
354 s->noise_type = get_bits1(gbp);
355
356 if (m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) {
357 av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
358 return -1;
359 }
360
361 skip_bits(gbp, 16); /* Output timestamp */
362
363 s->min_channel = get_bits(gbp, 4);
364 s->max_channel = get_bits(gbp, 4);
365 s->max_matrix_channel = get_bits(gbp, 4);
366
367 if (s->max_matrix_channel > max_matrix_channel) {
368 av_log(m->avctx, AV_LOG_ERROR,
369 "Max matrix channel cannot be greater than %d.\n",
370 max_matrix_channel);
371 return -1;
372 }
373
374 if (s->max_channel != s->max_matrix_channel) {
375 av_log(m->avctx, AV_LOG_ERROR,
376 "Max channel must be equal max matrix channel.\n");
377 return -1;
378 }
379
380 /* This should happen for TrueHD streams with >6 channels and MLP's noise
381 * type. It is not yet known if this is allowed. */
382 if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
383 av_log(m->avctx, AV_LOG_ERROR,
384 "Number of channels %d is larger than the maximum supported "
385 "by the decoder. %s\n", s->max_channel+2, sample_message);
386 return -1;
387 }
388
389 if (s->min_channel > s->max_channel) {
390 av_log(m->avctx, AV_LOG_ERROR,
391 "Substream min channel cannot be greater than max channel.\n");
392 return -1;
393 }
394
395 if (m->avctx->request_channels > 0
396 && s->max_channel + 1 >= m->avctx->request_channels
397 && substr < m->max_decoded_substream) {
398 av_log(m->avctx, AV_LOG_INFO,
399 "Extracting %d channel downmix from substream %d. "
400 "Further substreams will be skipped.\n",
401 s->max_channel + 1, substr);
402 m->max_decoded_substream = substr;
403 }
404
405 s->noise_shift = get_bits(gbp, 4);
406 s->noisegen_seed = get_bits(gbp, 23);
407
408 skip_bits(gbp, 19);
409
410 s->data_check_present = get_bits1(gbp);
411 lossless_check = get_bits(gbp, 8);
412 if (substr == m->max_decoded_substream
413 && s->lossless_check_data != 0xffffffff) {
414 tmp = xor_32_to_8(s->lossless_check_data);
415 if (tmp != lossless_check)
416 av_log(m->avctx, AV_LOG_WARNING,
417 "Lossless check failed - expected %02x, calculated %02x.\n",
418 lossless_check, tmp);
419 }
420
421 skip_bits(gbp, 16);
422
423 memset(s->ch_assign, 0, sizeof(s->ch_assign));
424
425 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
426 int ch_assign = get_bits(gbp, 6);
427 if (ch_assign > s->max_matrix_channel) {
428 av_log(m->avctx, AV_LOG_ERROR,
429 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
430 ch, ch_assign, sample_message);
431 return -1;
432 }
433 s->ch_assign[ch_assign] = ch;
434 }
435
436 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
437
438 if (checksum != get_bits(gbp, 8))
439 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
440
441 /* Set default decoding parameters. */
442 s->param_presence_flags = 0xff;
443 s->num_primitive_matrices = 0;
444 s->blocksize = 8;
445 s->lossless_check_data = 0;
446
447 memset(s->output_shift , 0, sizeof(s->output_shift ));
448 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
449
450 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
451 ChannelParams *cp = &m->channel_params[ch];
452 cp->filter_params[FIR].order = 0;
453 cp->filter_params[IIR].order = 0;
454 cp->filter_params[FIR].shift = 0;
455 cp->filter_params[IIR].shift = 0;
456
457 /* Default audio coding is 24-bit raw PCM. */
458 cp->huff_offset = 0;
459 cp->sign_huff_offset = (-1) << 23;
460 cp->codebook = 0;
461 cp->huff_lsbs = 24;
462 }
463
464 if (substr == m->max_decoded_substream)
465 m->avctx->channels = s->max_matrix_channel + 1;
466
467 return 0;
468 }
469
470 /** Read parameters for one of the prediction filters. */
471
472 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
473 unsigned int channel, unsigned int filter)
474 {
475 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
476 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
477 const char fchar = filter ? 'I' : 'F';
478 int i, order;
479
480 // Filter is 0 for FIR, 1 for IIR.
481 assert(filter < 2);
482
483 if (m->filter_changed[channel][filter]++ > 1) {
484 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
485 return -1;
486 }
487
488 order = get_bits(gbp, 4);
489 if (order > max_order) {
490 av_log(m->avctx, AV_LOG_ERROR,
491 "%cIR filter order %d is greater than maximum %d.\n",
492 fchar, order, max_order);
493 return -1;
494 }
495 fp->order = order;
496
497 if (order > 0) {
498 int coeff_bits, coeff_shift;
499
500 fp->shift = get_bits(gbp, 4);
501
502 coeff_bits = get_bits(gbp, 5);
503 coeff_shift = get_bits(gbp, 3);
504 if (coeff_bits < 1 || coeff_bits > 16) {
505 av_log(m->avctx, AV_LOG_ERROR,
506 "%cIR filter coeff_bits must be between 1 and 16.\n",
507 fchar);
508 return -1;
509 }
510 if (coeff_bits + coeff_shift > 16) {
511 av_log(m->avctx, AV_LOG_ERROR,
512 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
513 fchar);
514 return -1;
515 }
516
517 for (i = 0; i < order; i++)
518 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
519
520 if (get_bits1(gbp)) {
521 int state_bits, state_shift;
522
523 if (filter == FIR) {
524 av_log(m->avctx, AV_LOG_ERROR,
525 "FIR filter has state data specified.\n");
526 return -1;
527 }
528
529 state_bits = get_bits(gbp, 4);
530 state_shift = get_bits(gbp, 4);
531
532 /* TODO: Check validity of state data. */
533
534 for (i = 0; i < order; i++)
535 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
536 }
537 }
538
539 return 0;
540 }
541
542 /** Read parameters for primitive matrices. */
543
544 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
545 {
546 SubStream *s = &m->substream[substr];
547 unsigned int mat, ch;
548 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
549 ? MAX_MATRICES_MLP
550 : MAX_MATRICES_TRUEHD;
551
552 if (m->matrix_changed++ > 1) {
553 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
554 return -1;
555 }
556
557 s->num_primitive_matrices = get_bits(gbp, 4);
558
559 if (s->num_primitive_matrices > max_primitive_matrices) {
560 av_log(m->avctx, AV_LOG_ERROR,
561 "Number of primitive matrices cannot be greater than %d.\n",
562 max_primitive_matrices);
563 return -1;
564 }
565
566 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
567 int frac_bits, max_chan;
568 s->matrix_out_ch[mat] = get_bits(gbp, 4);
569 frac_bits = get_bits(gbp, 4);
570 s->lsb_bypass [mat] = get_bits1(gbp);
571
572 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
573 av_log(m->avctx, AV_LOG_ERROR,
574 "Invalid channel %d specified as output from matrix.\n",
575 s->matrix_out_ch[mat]);
576 return -1;
577 }
578 if (frac_bits > 14) {
579 av_log(m->avctx, AV_LOG_ERROR,
580 "Too many fractional bits specified.\n");
581 return -1;
582 }
583
584 max_chan = s->max_matrix_channel;
585 if (!s->noise_type)
586 max_chan+=2;
587
588 for (ch = 0; ch <= max_chan; ch++) {
589 int coeff_val = 0;
590 if (get_bits1(gbp))
591 coeff_val = get_sbits(gbp, frac_bits + 2);
592
593 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
594 }
595
596 if (s->noise_type)
597 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
598 else
599 s->matrix_noise_shift[mat] = 0;
600 }
601
602 return 0;
603 }
604
605 /** Read channel parameters. */
606
607 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
608 GetBitContext *gbp, unsigned int ch)
609 {
610 ChannelParams *cp = &m->channel_params[ch];
611 FilterParams *fir = &cp->filter_params[FIR];
612 FilterParams *iir = &cp->filter_params[IIR];
613 SubStream *s = &m->substream[substr];
614
615 if (s->param_presence_flags & PARAM_FIR)
616 if (get_bits1(gbp))
617 if (read_filter_params(m, gbp, ch, FIR) < 0)
618 return -1;
619
620 if (s->param_presence_flags & PARAM_IIR)
621 if (get_bits1(gbp))
622 if (read_filter_params(m, gbp, ch, IIR) < 0)
623 return -1;
624
625 if (fir->order + iir->order > 8) {
626 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
627 return -1;
628 }
629
630 if (fir->order && iir->order &&
631 fir->shift != iir->shift) {
632 av_log(m->avctx, AV_LOG_ERROR,
633 "FIR and IIR filters must use the same precision.\n");
634 return -1;
635 }
636 /* The FIR and IIR filters must have the same precision.
637 * To simplify the filtering code, only the precision of the
638 * FIR filter is considered. If only the IIR filter is employed,
639 * the FIR filter precision is set to that of the IIR filter, so
640 * that the filtering code can use it. */
641 if (!fir->order && iir->order)
642 fir->shift = iir->shift;
643
644 if (s->param_presence_flags & PARAM_HUFFOFFSET)
645 if (get_bits1(gbp))
646 cp->huff_offset = get_sbits(gbp, 15);
647
648 cp->codebook = get_bits(gbp, 2);
649 cp->huff_lsbs = get_bits(gbp, 5);
650
651 if (cp->huff_lsbs > 24) {
652 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
653 return -1;
654 }
655
656 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
657
658 return 0;
659 }
660
661 /** Read decoding parameters that change more often than those in the restart
662 * header. */
663
664 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
665 unsigned int substr)
666 {
667 SubStream *s = &m->substream[substr];
668 unsigned int ch;
669
670 if (s->param_presence_flags & PARAM_PRESENCE)
671 if (get_bits1(gbp))
672 s->param_presence_flags = get_bits(gbp, 8);
673
674 if (s->param_presence_flags & PARAM_BLOCKSIZE)
675 if (get_bits1(gbp)) {
676 s->blocksize = get_bits(gbp, 9);
677 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
678 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
679 s->blocksize = 0;
680 return -1;
681 }
682 }
683
684 if (s->param_presence_flags & PARAM_MATRIX)
685 if (get_bits1(gbp))
686 if (read_matrix_params(m, substr, gbp) < 0)
687 return -1;
688
689 if (s->param_presence_flags & PARAM_OUTSHIFT)
690 if (get_bits1(gbp))
691 for (ch = 0; ch <= s->max_matrix_channel; ch++)
692 s->output_shift[ch] = get_sbits(gbp, 4);
693
694 if (s->param_presence_flags & PARAM_QUANTSTEP)
695 if (get_bits1(gbp))
696 for (ch = 0; ch <= s->max_channel; ch++) {
697 ChannelParams *cp = &m->channel_params[ch];
698
699 s->quant_step_size[ch] = get_bits(gbp, 4);
700
701 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
702 }
703
704 for (ch = s->min_channel; ch <= s->max_channel; ch++)
705 if (get_bits1(gbp))
706 if (read_channel_params(m, substr, gbp, ch) < 0)
707 return -1;
708
709 return 0;
710 }
711
712 #define MSB_MASK(bits) (-1u << bits)
713
714 /** Generate PCM samples using the prediction filters and residual values
715 * read from the data stream, and update the filter state. */
716
717 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
718 unsigned int channel)
719 {
720 SubStream *s = &m->substream[substr];
721 int32_t fir_state_buffer[MAX_BLOCKSIZE + MAX_FIR_ORDER];
722 int32_t iir_state_buffer[MAX_BLOCKSIZE + MAX_IIR_ORDER];
723 int32_t *firbuf = fir_state_buffer + MAX_BLOCKSIZE;
724 int32_t *iirbuf = iir_state_buffer + MAX_BLOCKSIZE;
725 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
726 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
727 unsigned int filter_shift = fir->shift;
728 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
729
730 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
731 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
732
733 m->dsp.mlp_filter_channel(firbuf, fir->coeff, fir->order,
734 iirbuf, iir->coeff, iir->order,
735 filter_shift, mask, s->blocksize,
736 &m->sample_buffer[s->blockpos][channel]);
737
738 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
739 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
740 }
741
742 /** Read a block of PCM residual data (or actual if no filtering active). */
743
744 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
745 unsigned int substr)
746 {
747 SubStream *s = &m->substream[substr];
748 unsigned int i, ch, expected_stream_pos = 0;
749
750 if (s->data_check_present) {
751 expected_stream_pos = get_bits_count(gbp);
752 expected_stream_pos += get_bits(gbp, 16);
753 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
754 "we have not tested yet. %s\n", sample_message);
755 }
756
757 if (s->blockpos + s->blocksize > m->access_unit_size) {
758 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
759 return -1;
760 }
761
762 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
763 s->blocksize * sizeof(m->bypassed_lsbs[0]));
764
765 for (i = 0; i < s->blocksize; i++)
766 if (read_huff_channels(m, gbp, substr, i) < 0)
767 return -1;
768
769 for (ch = s->min_channel; ch <= s->max_channel; ch++)
770 filter_channel(m, substr, ch);
771
772 s->blockpos += s->blocksize;
773
774 if (s->data_check_present) {
775 if (get_bits_count(gbp) != expected_stream_pos)
776 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
777 skip_bits(gbp, 8);
778 }
779
780 return 0;
781 }
782
783 /** Data table used for TrueHD noise generation function. */
784
785 static const int8_t noise_table[256] = {
786 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
787 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
788 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
789 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
790 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
791 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
792 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
793 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
794 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
795 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
796 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
797 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
798 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
799 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
800 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
801 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
802 };
803
804 /** Noise generation functions.
805 * I'm not sure what these are for - they seem to be some kind of pseudorandom
806 * sequence generators, used to generate noise data which is used when the
807 * channels are rematrixed. I'm not sure if they provide a practical benefit
808 * to compression, or just obfuscate the decoder. Are they for some kind of
809 * dithering? */
810
811 /** Generate two channels of noise, used in the matrix when
812 * restart sync word == 0x31ea. */
813
814 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
815 {
816 SubStream *s = &m->substream[substr];
817 unsigned int i;
818 uint32_t seed = s->noisegen_seed;
819 unsigned int maxchan = s->max_matrix_channel;
820
821 for (i = 0; i < s->blockpos; i++) {
822 uint16_t seed_shr7 = seed >> 7;
823 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
824 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
825
826 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
827 }
828
829 s->noisegen_seed = seed;
830 }
831
832 /** Generate a block of noise, used when restart sync word == 0x31eb. */
833
834 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
835 {
836 SubStream *s = &m->substream[substr];
837 unsigned int i;
838 uint32_t seed = s->noisegen_seed;
839
840 for (i = 0; i < m->access_unit_size_pow2; i++) {
841 uint8_t seed_shr15 = seed >> 15;
842 m->noise_buffer[i] = noise_table[seed_shr15];
843 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
844 }
845
846 s->noisegen_seed = seed;
847 }
848
849
850 /** Apply the channel matrices in turn to reconstruct the original audio
851 * samples. */
852
853 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
854 {
855 SubStream *s = &m->substream[substr];
856 unsigned int mat, src_ch, i;
857 unsigned int maxchan;
858
859 maxchan = s->max_matrix_channel;
860 if (!s->noise_type) {
861 generate_2_noise_channels(m, substr);
862 maxchan += 2;
863 } else {
864 fill_noise_buffer(m, substr);
865 }
866
867 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
868 int matrix_noise_shift = s->matrix_noise_shift[mat];
869 unsigned int dest_ch = s->matrix_out_ch[mat];
870 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
871 int32_t *coeffs = s->matrix_coeff[mat];
872 int index = s->num_primitive_matrices - mat;
873 int index2 = 2 * index + 1;
874
875 /* TODO: DSPContext? */
876
877 for (i = 0; i < s->blockpos; i++) {
878 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
879 int32_t *samples = m->sample_buffer[i];
880 int64_t accum = 0;
881
882 for (src_ch = 0; src_ch <= maxchan; src_ch++)
883 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
884
885 if (matrix_noise_shift) {
886 index &= m->access_unit_size_pow2 - 1;
887 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
888 index += index2;
889 }
890
891 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
892 }
893 }
894 }
895
896 /** Write the audio data into the output buffer. */
897
898 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
899 uint8_t *data, unsigned int *data_size, int is32)
900 {
901 SubStream *s = &m->substream[substr];
902 unsigned int i, out_ch = 0;
903 int32_t *data_32 = (int32_t*) data;
904 int16_t *data_16 = (int16_t*) data;
905
906 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
907 return -1;
908
909 for (i = 0; i < s->blockpos; i++) {
910 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
911 int mat_ch = s->ch_assign[out_ch];
912 int32_t sample = m->sample_buffer[i][mat_ch]
913 << s->output_shift[mat_ch];
914 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
915 if (is32) *data_32++ = sample << 8;
916 else *data_16++ = sample >> 8;
917 }
918 }
919
920 *data_size = i * out_ch * (is32 ? 4 : 2);
921
922 return 0;
923 }
924
925 static int output_data(MLPDecodeContext *m, unsigned int substr,
926 uint8_t *data, unsigned int *data_size)
927 {
928 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
929 return output_data_internal(m, substr, data, data_size, 1);
930 else
931 return output_data_internal(m, substr, data, data_size, 0);
932 }
933
934
935 /** Read an access unit from the stream.
936 * Returns < 0 on error, 0 if not enough data is present in the input stream
937 * otherwise returns the number of bytes consumed. */
938
939 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
940 AVPacket *avpkt)
941 {
942 const uint8_t *buf = avpkt->data;
943 int buf_size = avpkt->size;
944 MLPDecodeContext *m = avctx->priv_data;
945 GetBitContext gb;
946 unsigned int length, substr;
947 unsigned int substream_start;
948 unsigned int header_size = 4;
949 unsigned int substr_header_size = 0;
950 uint8_t substream_parity_present[MAX_SUBSTREAMS];
951 uint16_t substream_data_len[MAX_SUBSTREAMS];
952 uint8_t parity_bits;
953
954 if (buf_size < 4)
955 return 0;
956
957 length = (AV_RB16(buf) & 0xfff) * 2;
958
959 if (length > buf_size)
960 return -1;
961
962 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
963
964 m->is_major_sync_unit = 0;
965 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
966 if (read_major_sync(m, &gb) < 0)
967 goto error;
968 m->is_major_sync_unit = 1;
969 header_size += 28;
970 }
971
972 if (!m->params_valid) {
973 av_log(m->avctx, AV_LOG_WARNING,
974 "Stream parameters not seen; skipping frame.\n");
975 *data_size = 0;
976 return length;
977 }
978
979 substream_start = 0;
980
981 for (substr = 0; substr < m->num_substreams; substr++) {
982 int extraword_present, checkdata_present, end, nonrestart_substr;
983
984 extraword_present = get_bits1(&gb);
985 nonrestart_substr = get_bits1(&gb);
986 checkdata_present = get_bits1(&gb);
987 skip_bits1(&gb);
988
989 end = get_bits(&gb, 12) * 2;
990
991 substr_header_size += 2;
992
993 if (extraword_present) {
994 if (m->avctx->codec_id == CODEC_ID_MLP) {
995 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
996 goto error;
997 }
998 skip_bits(&gb, 16);
999 substr_header_size += 2;
1000 }
1001
1002 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1003 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1004 goto error;
1005 }
1006
1007 if (end + header_size + substr_header_size > length) {
1008 av_log(m->avctx, AV_LOG_ERROR,
1009 "Indicated length of substream %d data goes off end of "
1010 "packet.\n", substr);
1011 end = length - header_size - substr_header_size;
1012 }
1013
1014 if (end < substream_start) {
1015 av_log(avctx, AV_LOG_ERROR,
1016 "Indicated end offset of substream %d data "
1017 "is smaller than calculated start offset.\n",
1018 substr);
1019 goto error;
1020 }
1021
1022 if (substr > m->max_decoded_substream)
1023 continue;
1024
1025 substream_parity_present[substr] = checkdata_present;
1026 substream_data_len[substr] = end - substream_start;
1027 substream_start = end;
1028 }
1029
1030 parity_bits = ff_mlp_calculate_parity(buf, 4);
1031 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1032
1033 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1034 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1035 goto error;
1036 }
1037
1038 buf += header_size + substr_header_size;
1039
1040 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1041 SubStream *s = &m->substream[substr];
1042 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1043
1044 m->matrix_changed = 0;
1045 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1046
1047 s->blockpos = 0;
1048 do {
1049 if (get_bits1(&gb)) {
1050 if (get_bits1(&gb)) {
1051 /* A restart header should be present. */
1052 if (read_restart_header(m, &gb, buf, substr) < 0)
1053 goto next_substr;
1054 s->restart_seen = 1;
1055 }
1056
1057 if (!s->restart_seen)
1058 goto next_substr;
1059 if (read_decoding_params(m, &gb, substr) < 0)
1060 goto next_substr;
1061 }
1062
1063 if (!s->restart_seen)
1064 goto next_substr;
1065
1066 if (read_block_data(m, &gb, substr) < 0)
1067 return -1;
1068
1069 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1070 goto substream_length_mismatch;
1071
1072 } while (!get_bits1(&gb));
1073
1074 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1075
1076 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1077 int shorten_by;
1078
1079 if (get_bits(&gb, 16) != 0xD234)
1080 return -1;
1081
1082 shorten_by = get_bits(&gb, 16);
1083 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1084 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1085 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1086 return -1;
1087
1088 if (substr == m->max_decoded_substream)
1089 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1090 }
1091
1092 if (substream_parity_present[substr]) {
1093 uint8_t parity, checksum;
1094
1095 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1096 goto substream_length_mismatch;
1097
1098 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1099 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1100
1101 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1102 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1103 if ( get_bits(&gb, 8) != checksum)
1104 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1105 }
1106
1107 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1108 goto substream_length_mismatch;
1109
1110 next_substr:
1111 if (!s->restart_seen)
1112 av_log(m->avctx, AV_LOG_ERROR,
1113 "No restart header present in substream %d.\n", substr);
1114
1115 buf += substream_data_len[substr];
1116 }
1117
1118 rematrix_channels(m, m->max_decoded_substream);
1119
1120 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1121 return -1;
1122
1123 return length;
1124
1125 substream_length_mismatch:
1126 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1127 return -1;
1128
1129 error:
1130 m->params_valid = 0;
1131 return -1;
1132 }
1133
1134 #if CONFIG_MLP_DECODER
1135 AVCodec mlp_decoder = {
1136 "mlp",
1137 CODEC_TYPE_AUDIO,
1138 CODEC_ID_MLP,
1139 sizeof(MLPDecodeContext),
1140 mlp_decode_init,
1141 NULL,
1142 NULL,
1143 read_access_unit,
1144 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1145 };
1146 #endif /* CONFIG_MLP_DECODER */
1147
1148 #if CONFIG_TRUEHD_DECODER
1149 AVCodec truehd_decoder = {
1150 "truehd",
1151 CODEC_TYPE_AUDIO,
1152 CODEC_ID_TRUEHD,
1153 sizeof(MLPDecodeContext),
1154 mlp_decode_init,
1155 NULL,
1156 NULL,
1157 read_access_unit,
1158 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1159 };
1160 #endif /* CONFIG_TRUEHD_DECODER */