aa3b9cd2cdc6100253bd89b1465a3f71f52fa1e2
[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 if (!huff_vlc[0].bits) {
159 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
160 &ff_mlp_huffman_tables[0][0][1], 2, 1,
161 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
162 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
163 &ff_mlp_huffman_tables[1][0][1], 2, 1,
164 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
165 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
166 &ff_mlp_huffman_tables[2][0][1], 2, 1,
167 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
168 }
169
170 ff_mlp_init_crc();
171 }
172
173 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
174 unsigned int substr, unsigned int ch)
175 {
176 ChannelParams *cp = &m->channel_params[ch];
177 SubStream *s = &m->substream[substr];
178 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
179 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
180 int32_t sign_huff_offset = cp->huff_offset;
181
182 if (cp->codebook > 0)
183 sign_huff_offset -= 7 << lsb_bits;
184
185 if (sign_shift >= 0)
186 sign_huff_offset -= 1 << sign_shift;
187
188 return sign_huff_offset;
189 }
190
191 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
192 * and plain LSBs. */
193
194 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
195 unsigned int substr, unsigned int pos)
196 {
197 SubStream *s = &m->substream[substr];
198 unsigned int mat, channel;
199
200 for (mat = 0; mat < s->num_primitive_matrices; mat++)
201 if (s->lsb_bypass[mat])
202 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
203
204 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
205 ChannelParams *cp = &m->channel_params[channel];
206 int codebook = cp->codebook;
207 int quant_step_size = s->quant_step_size[channel];
208 int lsb_bits = cp->huff_lsbs - quant_step_size;
209 int result = 0;
210
211 if (codebook > 0)
212 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
213 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
214
215 if (result < 0)
216 return -1;
217
218 if (lsb_bits > 0)
219 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
220
221 result += cp->sign_huff_offset;
222 result <<= quant_step_size;
223
224 m->sample_buffer[pos + s->blockpos][channel] = result;
225 }
226
227 return 0;
228 }
229
230 static av_cold int mlp_decode_init(AVCodecContext *avctx)
231 {
232 MLPDecodeContext *m = avctx->priv_data;
233 int substr;
234
235 init_static();
236 m->avctx = avctx;
237 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
238 m->substream[substr].lossless_check_data = 0xffffffff;
239 dsputil_init(&m->dsp, avctx);
240
241 return 0;
242 }
243
244 /** Read a major sync info header - contains high level information about
245 * the stream - sample rate, channel arrangement etc. Most of this
246 * information is not actually necessary for decoding, only for playback.
247 */
248
249 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
250 {
251 MLPHeaderInfo mh;
252 int substr;
253
254 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
255 return -1;
256
257 if (mh.group1_bits == 0) {
258 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
259 return -1;
260 }
261 if (mh.group2_bits > mh.group1_bits) {
262 av_log(m->avctx, AV_LOG_ERROR,
263 "Channel group 2 cannot have more bits per sample than group 1.\n");
264 return -1;
265 }
266
267 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
268 av_log(m->avctx, AV_LOG_ERROR,
269 "Channel groups with differing sample rates are not currently supported.\n");
270 return -1;
271 }
272
273 if (mh.group1_samplerate == 0) {
274 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
275 return -1;
276 }
277 if (mh.group1_samplerate > MAX_SAMPLERATE) {
278 av_log(m->avctx, AV_LOG_ERROR,
279 "Sampling rate %d is greater than the supported maximum (%d).\n",
280 mh.group1_samplerate, MAX_SAMPLERATE);
281 return -1;
282 }
283 if (mh.access_unit_size > MAX_BLOCKSIZE) {
284 av_log(m->avctx, AV_LOG_ERROR,
285 "Block size %d is greater than the supported maximum (%d).\n",
286 mh.access_unit_size, MAX_BLOCKSIZE);
287 return -1;
288 }
289 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
290 av_log(m->avctx, AV_LOG_ERROR,
291 "Block size pow2 %d is greater than the supported maximum (%d).\n",
292 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
293 return -1;
294 }
295
296 if (mh.num_substreams == 0)
297 return -1;
298 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
299 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
300 return -1;
301 }
302 if (mh.num_substreams > MAX_SUBSTREAMS) {
303 av_log(m->avctx, AV_LOG_ERROR,
304 "Number of substreams %d is larger than the maximum supported "
305 "by the decoder. %s\n", mh.num_substreams, sample_message);
306 return -1;
307 }
308
309 m->access_unit_size = mh.access_unit_size;
310 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
311
312 m->num_substreams = mh.num_substreams;
313 m->max_decoded_substream = m->num_substreams - 1;
314
315 m->avctx->sample_rate = mh.group1_samplerate;
316 m->avctx->frame_size = mh.access_unit_size;
317
318 m->avctx->bits_per_raw_sample = mh.group1_bits;
319 if (mh.group1_bits > 16)
320 m->avctx->sample_fmt = SAMPLE_FMT_S32;
321 else
322 m->avctx->sample_fmt = SAMPLE_FMT_S16;
323
324 m->params_valid = 1;
325 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
326 m->substream[substr].restart_seen = 0;
327
328 return 0;
329 }
330
331 /** Read a restart header from a block in a substream. This contains parameters
332 * required to decode the audio that do not change very often. Generally
333 * (always) present only in blocks following a major sync. */
334
335 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
336 const uint8_t *buf, unsigned int substr)
337 {
338 SubStream *s = &m->substream[substr];
339 unsigned int ch;
340 int sync_word, tmp;
341 uint8_t checksum;
342 uint8_t lossless_check;
343 int start_count = get_bits_count(gbp);
344 const int max_matrix_channel = m->avctx->codec_id == CODEC_ID_MLP
345 ? MAX_MATRIX_CHANNEL_MLP
346 : MAX_MATRIX_CHANNEL_TRUEHD;
347
348 sync_word = get_bits(gbp, 13);
349
350 if (sync_word != 0x31ea >> 1) {
351 av_log(m->avctx, AV_LOG_ERROR,
352 "restart header sync incorrect (got 0x%04x)\n", sync_word);
353 return -1;
354 }
355
356 s->noise_type = get_bits1(gbp);
357
358 if (m->avctx->codec_id == CODEC_ID_MLP && s->noise_type) {
359 av_log(m->avctx, AV_LOG_ERROR, "MLP must have 0x31ea sync word.\n");
360 return -1;
361 }
362
363 skip_bits(gbp, 16); /* Output timestamp */
364
365 s->min_channel = get_bits(gbp, 4);
366 s->max_channel = get_bits(gbp, 4);
367 s->max_matrix_channel = get_bits(gbp, 4);
368
369 if (s->max_matrix_channel > max_matrix_channel) {
370 av_log(m->avctx, AV_LOG_ERROR,
371 "Max matrix channel cannot be greater than %d.\n",
372 max_matrix_channel);
373 return -1;
374 }
375
376 if (s->max_channel != s->max_matrix_channel) {
377 av_log(m->avctx, AV_LOG_ERROR,
378 "Max channel must be equal max matrix channel.\n");
379 return -1;
380 }
381
382 /* This should happen for TrueHD streams with >6 channels and MLP's noise
383 * type. It is not yet known if this is allowed. */
384 if (s->max_channel > MAX_MATRIX_CHANNEL_MLP && !s->noise_type) {
385 av_log(m->avctx, AV_LOG_ERROR,
386 "Number of channels %d is larger than the maximum supported "
387 "by the decoder. %s\n", s->max_channel+2, sample_message);
388 return -1;
389 }
390
391 if (s->min_channel > s->max_channel) {
392 av_log(m->avctx, AV_LOG_ERROR,
393 "Substream min channel cannot be greater than max channel.\n");
394 return -1;
395 }
396
397 if (m->avctx->request_channels > 0
398 && s->max_channel + 1 >= m->avctx->request_channels
399 && substr < m->max_decoded_substream) {
400 av_log(m->avctx, AV_LOG_DEBUG,
401 "Extracting %d channel downmix from substream %d. "
402 "Further substreams will be skipped.\n",
403 s->max_channel + 1, substr);
404 m->max_decoded_substream = substr;
405 }
406
407 s->noise_shift = get_bits(gbp, 4);
408 s->noisegen_seed = get_bits(gbp, 23);
409
410 skip_bits(gbp, 19);
411
412 s->data_check_present = get_bits1(gbp);
413 lossless_check = get_bits(gbp, 8);
414 if (substr == m->max_decoded_substream
415 && s->lossless_check_data != 0xffffffff) {
416 tmp = xor_32_to_8(s->lossless_check_data);
417 if (tmp != lossless_check)
418 av_log(m->avctx, AV_LOG_WARNING,
419 "Lossless check failed - expected %02x, calculated %02x.\n",
420 lossless_check, tmp);
421 }
422
423 skip_bits(gbp, 16);
424
425 memset(s->ch_assign, 0, sizeof(s->ch_assign));
426
427 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
428 int ch_assign = get_bits(gbp, 6);
429 if (ch_assign > s->max_matrix_channel) {
430 av_log(m->avctx, AV_LOG_ERROR,
431 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
432 ch, ch_assign, sample_message);
433 return -1;
434 }
435 s->ch_assign[ch_assign] = ch;
436 }
437
438 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
439
440 if (checksum != get_bits(gbp, 8))
441 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
442
443 /* Set default decoding parameters. */
444 s->param_presence_flags = 0xff;
445 s->num_primitive_matrices = 0;
446 s->blocksize = 8;
447 s->lossless_check_data = 0;
448
449 memset(s->output_shift , 0, sizeof(s->output_shift ));
450 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
451
452 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
453 ChannelParams *cp = &m->channel_params[ch];
454 cp->filter_params[FIR].order = 0;
455 cp->filter_params[IIR].order = 0;
456 cp->filter_params[FIR].shift = 0;
457 cp->filter_params[IIR].shift = 0;
458
459 /* Default audio coding is 24-bit raw PCM. */
460 cp->huff_offset = 0;
461 cp->sign_huff_offset = (-1) << 23;
462 cp->codebook = 0;
463 cp->huff_lsbs = 24;
464 }
465
466 if (substr == m->max_decoded_substream)
467 m->avctx->channels = s->max_matrix_channel + 1;
468
469 return 0;
470 }
471
472 /** Read parameters for one of the prediction filters. */
473
474 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
475 unsigned int channel, unsigned int filter)
476 {
477 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
478 const int max_order = filter ? MAX_IIR_ORDER : MAX_FIR_ORDER;
479 const char fchar = filter ? 'I' : 'F';
480 int i, order;
481
482 // Filter is 0 for FIR, 1 for IIR.
483 assert(filter < 2);
484
485 if (m->filter_changed[channel][filter]++ > 1) {
486 av_log(m->avctx, AV_LOG_ERROR, "Filters may change only once per access unit.\n");
487 return -1;
488 }
489
490 order = get_bits(gbp, 4);
491 if (order > max_order) {
492 av_log(m->avctx, AV_LOG_ERROR,
493 "%cIR filter order %d is greater than maximum %d.\n",
494 fchar, order, max_order);
495 return -1;
496 }
497 fp->order = order;
498
499 if (order > 0) {
500 int32_t *fcoeff = m->channel_params[channel].coeff[filter];
501 int coeff_bits, coeff_shift;
502
503 fp->shift = get_bits(gbp, 4);
504
505 coeff_bits = get_bits(gbp, 5);
506 coeff_shift = get_bits(gbp, 3);
507 if (coeff_bits < 1 || coeff_bits > 16) {
508 av_log(m->avctx, AV_LOG_ERROR,
509 "%cIR filter coeff_bits must be between 1 and 16.\n",
510 fchar);
511 return -1;
512 }
513 if (coeff_bits + coeff_shift > 16) {
514 av_log(m->avctx, AV_LOG_ERROR,
515 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
516 fchar);
517 return -1;
518 }
519
520 for (i = 0; i < order; i++)
521 fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
522
523 if (get_bits1(gbp)) {
524 int state_bits, state_shift;
525
526 if (filter == FIR) {
527 av_log(m->avctx, AV_LOG_ERROR,
528 "FIR filter has state data specified.\n");
529 return -1;
530 }
531
532 state_bits = get_bits(gbp, 4);
533 state_shift = get_bits(gbp, 4);
534
535 /* TODO: Check validity of state data. */
536
537 for (i = 0; i < order; i++)
538 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
539 }
540 }
541
542 return 0;
543 }
544
545 /** Read parameters for primitive matrices. */
546
547 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
548 {
549 SubStream *s = &m->substream[substr];
550 unsigned int mat, ch;
551 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
552 ? MAX_MATRICES_MLP
553 : MAX_MATRICES_TRUEHD;
554
555 if (m->matrix_changed++ > 1) {
556 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
557 return -1;
558 }
559
560 s->num_primitive_matrices = get_bits(gbp, 4);
561
562 if (s->num_primitive_matrices > max_primitive_matrices) {
563 av_log(m->avctx, AV_LOG_ERROR,
564 "Number of primitive matrices cannot be greater than %d.\n",
565 max_primitive_matrices);
566 return -1;
567 }
568
569 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
570 int frac_bits, max_chan;
571 s->matrix_out_ch[mat] = get_bits(gbp, 4);
572 frac_bits = get_bits(gbp, 4);
573 s->lsb_bypass [mat] = get_bits1(gbp);
574
575 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
576 av_log(m->avctx, AV_LOG_ERROR,
577 "Invalid channel %d specified as output from matrix.\n",
578 s->matrix_out_ch[mat]);
579 return -1;
580 }
581 if (frac_bits > 14) {
582 av_log(m->avctx, AV_LOG_ERROR,
583 "Too many fractional bits specified.\n");
584 return -1;
585 }
586
587 max_chan = s->max_matrix_channel;
588 if (!s->noise_type)
589 max_chan+=2;
590
591 for (ch = 0; ch <= max_chan; ch++) {
592 int coeff_val = 0;
593 if (get_bits1(gbp))
594 coeff_val = get_sbits(gbp, frac_bits + 2);
595
596 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
597 }
598
599 if (s->noise_type)
600 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
601 else
602 s->matrix_noise_shift[mat] = 0;
603 }
604
605 return 0;
606 }
607
608 /** Read channel parameters. */
609
610 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
611 GetBitContext *gbp, unsigned int ch)
612 {
613 ChannelParams *cp = &m->channel_params[ch];
614 FilterParams *fir = &cp->filter_params[FIR];
615 FilterParams *iir = &cp->filter_params[IIR];
616 SubStream *s = &m->substream[substr];
617
618 if (s->param_presence_flags & PARAM_FIR)
619 if (get_bits1(gbp))
620 if (read_filter_params(m, gbp, ch, FIR) < 0)
621 return -1;
622
623 if (s->param_presence_flags & PARAM_IIR)
624 if (get_bits1(gbp))
625 if (read_filter_params(m, gbp, ch, IIR) < 0)
626 return -1;
627
628 if (fir->order + iir->order > 8) {
629 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
630 return -1;
631 }
632
633 if (fir->order && iir->order &&
634 fir->shift != iir->shift) {
635 av_log(m->avctx, AV_LOG_ERROR,
636 "FIR and IIR filters must use the same precision.\n");
637 return -1;
638 }
639 /* The FIR and IIR filters must have the same precision.
640 * To simplify the filtering code, only the precision of the
641 * FIR filter is considered. If only the IIR filter is employed,
642 * the FIR filter precision is set to that of the IIR filter, so
643 * that the filtering code can use it. */
644 if (!fir->order && iir->order)
645 fir->shift = iir->shift;
646
647 if (s->param_presence_flags & PARAM_HUFFOFFSET)
648 if (get_bits1(gbp))
649 cp->huff_offset = get_sbits(gbp, 15);
650
651 cp->codebook = get_bits(gbp, 2);
652 cp->huff_lsbs = get_bits(gbp, 5);
653
654 if (cp->huff_lsbs > 24) {
655 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
656 return -1;
657 }
658
659 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
660
661 return 0;
662 }
663
664 /** Read decoding parameters that change more often than those in the restart
665 * header. */
666
667 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
668 unsigned int substr)
669 {
670 SubStream *s = &m->substream[substr];
671 unsigned int ch;
672
673 if (s->param_presence_flags & PARAM_PRESENCE)
674 if (get_bits1(gbp))
675 s->param_presence_flags = get_bits(gbp, 8);
676
677 if (s->param_presence_flags & PARAM_BLOCKSIZE)
678 if (get_bits1(gbp)) {
679 s->blocksize = get_bits(gbp, 9);
680 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
681 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
682 s->blocksize = 0;
683 return -1;
684 }
685 }
686
687 if (s->param_presence_flags & PARAM_MATRIX)
688 if (get_bits1(gbp))
689 if (read_matrix_params(m, substr, gbp) < 0)
690 return -1;
691
692 if (s->param_presence_flags & PARAM_OUTSHIFT)
693 if (get_bits1(gbp))
694 for (ch = 0; ch <= s->max_matrix_channel; ch++)
695 s->output_shift[ch] = get_sbits(gbp, 4);
696
697 if (s->param_presence_flags & PARAM_QUANTSTEP)
698 if (get_bits1(gbp))
699 for (ch = 0; ch <= s->max_channel; ch++) {
700 ChannelParams *cp = &m->channel_params[ch];
701
702 s->quant_step_size[ch] = get_bits(gbp, 4);
703
704 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
705 }
706
707 for (ch = s->min_channel; ch <= s->max_channel; ch++)
708 if (get_bits1(gbp))
709 if (read_channel_params(m, substr, gbp, ch) < 0)
710 return -1;
711
712 return 0;
713 }
714
715 #define MSB_MASK(bits) (-1u << bits)
716
717 /** Generate PCM samples using the prediction filters and residual values
718 * read from the data stream, and update the filter state. */
719
720 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
721 unsigned int channel)
722 {
723 SubStream *s = &m->substream[substr];
724 const int32_t *fircoeff = m->channel_params[channel].coeff[FIR];
725 int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
726 int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
727 int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
728 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
729 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
730 unsigned int filter_shift = fir->shift;
731 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
732
733 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
734 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
735
736 m->dsp.mlp_filter_channel(firbuf, fircoeff,
737 fir->order, iir->order,
738 filter_shift, mask, s->blocksize,
739 &m->sample_buffer[s->blockpos][channel]);
740
741 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
742 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
743 }
744
745 /** Read a block of PCM residual data (or actual if no filtering active). */
746
747 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
748 unsigned int substr)
749 {
750 SubStream *s = &m->substream[substr];
751 unsigned int i, ch, expected_stream_pos = 0;
752
753 if (s->data_check_present) {
754 expected_stream_pos = get_bits_count(gbp);
755 expected_stream_pos += get_bits(gbp, 16);
756 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
757 "we have not tested yet. %s\n", sample_message);
758 }
759
760 if (s->blockpos + s->blocksize > m->access_unit_size) {
761 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
762 return -1;
763 }
764
765 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
766 s->blocksize * sizeof(m->bypassed_lsbs[0]));
767
768 for (i = 0; i < s->blocksize; i++)
769 if (read_huff_channels(m, gbp, substr, i) < 0)
770 return -1;
771
772 for (ch = s->min_channel; ch <= s->max_channel; ch++)
773 filter_channel(m, substr, ch);
774
775 s->blockpos += s->blocksize;
776
777 if (s->data_check_present) {
778 if (get_bits_count(gbp) != expected_stream_pos)
779 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
780 skip_bits(gbp, 8);
781 }
782
783 return 0;
784 }
785
786 /** Data table used for TrueHD noise generation function. */
787
788 static const int8_t noise_table[256] = {
789 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
790 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
791 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
792 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
793 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
794 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
795 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
796 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
797 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
798 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
799 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
800 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
801 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
802 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
803 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
804 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
805 };
806
807 /** Noise generation functions.
808 * I'm not sure what these are for - they seem to be some kind of pseudorandom
809 * sequence generators, used to generate noise data which is used when the
810 * channels are rematrixed. I'm not sure if they provide a practical benefit
811 * to compression, or just obfuscate the decoder. Are they for some kind of
812 * dithering? */
813
814 /** Generate two channels of noise, used in the matrix when
815 * restart sync word == 0x31ea. */
816
817 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
818 {
819 SubStream *s = &m->substream[substr];
820 unsigned int i;
821 uint32_t seed = s->noisegen_seed;
822 unsigned int maxchan = s->max_matrix_channel;
823
824 for (i = 0; i < s->blockpos; i++) {
825 uint16_t seed_shr7 = seed >> 7;
826 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
827 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
828
829 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
830 }
831
832 s->noisegen_seed = seed;
833 }
834
835 /** Generate a block of noise, used when restart sync word == 0x31eb. */
836
837 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
838 {
839 SubStream *s = &m->substream[substr];
840 unsigned int i;
841 uint32_t seed = s->noisegen_seed;
842
843 for (i = 0; i < m->access_unit_size_pow2; i++) {
844 uint8_t seed_shr15 = seed >> 15;
845 m->noise_buffer[i] = noise_table[seed_shr15];
846 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
847 }
848
849 s->noisegen_seed = seed;
850 }
851
852
853 /** Apply the channel matrices in turn to reconstruct the original audio
854 * samples. */
855
856 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
857 {
858 SubStream *s = &m->substream[substr];
859 unsigned int mat, src_ch, i;
860 unsigned int maxchan;
861
862 maxchan = s->max_matrix_channel;
863 if (!s->noise_type) {
864 generate_2_noise_channels(m, substr);
865 maxchan += 2;
866 } else {
867 fill_noise_buffer(m, substr);
868 }
869
870 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
871 int matrix_noise_shift = s->matrix_noise_shift[mat];
872 unsigned int dest_ch = s->matrix_out_ch[mat];
873 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
874 int32_t *coeffs = s->matrix_coeff[mat];
875 int index = s->num_primitive_matrices - mat;
876 int index2 = 2 * index + 1;
877
878 /* TODO: DSPContext? */
879
880 for (i = 0; i < s->blockpos; i++) {
881 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
882 int32_t *samples = m->sample_buffer[i];
883 int64_t accum = 0;
884
885 for (src_ch = 0; src_ch <= maxchan; src_ch++)
886 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
887
888 if (matrix_noise_shift) {
889 index &= m->access_unit_size_pow2 - 1;
890 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
891 index += index2;
892 }
893
894 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
895 }
896 }
897 }
898
899 /** Write the audio data into the output buffer. */
900
901 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
902 uint8_t *data, unsigned int *data_size, int is32)
903 {
904 SubStream *s = &m->substream[substr];
905 unsigned int i, out_ch = 0;
906 int32_t *data_32 = (int32_t*) data;
907 int16_t *data_16 = (int16_t*) data;
908
909 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
910 return -1;
911
912 for (i = 0; i < s->blockpos; i++) {
913 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
914 int mat_ch = s->ch_assign[out_ch];
915 int32_t sample = m->sample_buffer[i][mat_ch]
916 << s->output_shift[mat_ch];
917 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
918 if (is32) *data_32++ = sample << 8;
919 else *data_16++ = sample >> 8;
920 }
921 }
922
923 *data_size = i * out_ch * (is32 ? 4 : 2);
924
925 return 0;
926 }
927
928 static int output_data(MLPDecodeContext *m, unsigned int substr,
929 uint8_t *data, unsigned int *data_size)
930 {
931 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
932 return output_data_internal(m, substr, data, data_size, 1);
933 else
934 return output_data_internal(m, substr, data, data_size, 0);
935 }
936
937
938 /** Read an access unit from the stream.
939 * Returns < 0 on error, 0 if not enough data is present in the input stream
940 * otherwise returns the number of bytes consumed. */
941
942 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
943 AVPacket *avpkt)
944 {
945 const uint8_t *buf = avpkt->data;
946 int buf_size = avpkt->size;
947 MLPDecodeContext *m = avctx->priv_data;
948 GetBitContext gb;
949 unsigned int length, substr;
950 unsigned int substream_start;
951 unsigned int header_size = 4;
952 unsigned int substr_header_size = 0;
953 uint8_t substream_parity_present[MAX_SUBSTREAMS];
954 uint16_t substream_data_len[MAX_SUBSTREAMS];
955 uint8_t parity_bits;
956
957 if (buf_size < 4)
958 return 0;
959
960 length = (AV_RB16(buf) & 0xfff) * 2;
961
962 if (length < 4 || length > buf_size)
963 return -1;
964
965 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
966
967 m->is_major_sync_unit = 0;
968 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
969 if (read_major_sync(m, &gb) < 0)
970 goto error;
971 m->is_major_sync_unit = 1;
972 header_size += 28;
973 }
974
975 if (!m->params_valid) {
976 av_log(m->avctx, AV_LOG_WARNING,
977 "Stream parameters not seen; skipping frame.\n");
978 *data_size = 0;
979 return length;
980 }
981
982 substream_start = 0;
983
984 for (substr = 0; substr < m->num_substreams; substr++) {
985 int extraword_present, checkdata_present, end, nonrestart_substr;
986
987 extraword_present = get_bits1(&gb);
988 nonrestart_substr = get_bits1(&gb);
989 checkdata_present = get_bits1(&gb);
990 skip_bits1(&gb);
991
992 end = get_bits(&gb, 12) * 2;
993
994 substr_header_size += 2;
995
996 if (extraword_present) {
997 if (m->avctx->codec_id == CODEC_ID_MLP) {
998 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
999 goto error;
1000 }
1001 skip_bits(&gb, 16);
1002 substr_header_size += 2;
1003 }
1004
1005 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1006 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1007 goto error;
1008 }
1009
1010 if (end + header_size + substr_header_size > length) {
1011 av_log(m->avctx, AV_LOG_ERROR,
1012 "Indicated length of substream %d data goes off end of "
1013 "packet.\n", substr);
1014 end = length - header_size - substr_header_size;
1015 }
1016
1017 if (end < substream_start) {
1018 av_log(avctx, AV_LOG_ERROR,
1019 "Indicated end offset of substream %d data "
1020 "is smaller than calculated start offset.\n",
1021 substr);
1022 goto error;
1023 }
1024
1025 if (substr > m->max_decoded_substream)
1026 continue;
1027
1028 substream_parity_present[substr] = checkdata_present;
1029 substream_data_len[substr] = end - substream_start;
1030 substream_start = end;
1031 }
1032
1033 parity_bits = ff_mlp_calculate_parity(buf, 4);
1034 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1035
1036 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1037 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1038 goto error;
1039 }
1040
1041 buf += header_size + substr_header_size;
1042
1043 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1044 SubStream *s = &m->substream[substr];
1045 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1046
1047 m->matrix_changed = 0;
1048 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1049
1050 s->blockpos = 0;
1051 do {
1052 if (get_bits1(&gb)) {
1053 if (get_bits1(&gb)) {
1054 /* A restart header should be present. */
1055 if (read_restart_header(m, &gb, buf, substr) < 0)
1056 goto next_substr;
1057 s->restart_seen = 1;
1058 }
1059
1060 if (!s->restart_seen)
1061 goto next_substr;
1062 if (read_decoding_params(m, &gb, substr) < 0)
1063 goto next_substr;
1064 }
1065
1066 if (!s->restart_seen)
1067 goto next_substr;
1068
1069 if (read_block_data(m, &gb, substr) < 0)
1070 return -1;
1071
1072 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1073 goto substream_length_mismatch;
1074
1075 } while (!get_bits1(&gb));
1076
1077 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1078
1079 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1080 int shorten_by;
1081
1082 if (get_bits(&gb, 16) != 0xD234)
1083 return -1;
1084
1085 shorten_by = get_bits(&gb, 16);
1086 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1087 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1088 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1089 return -1;
1090
1091 if (substr == m->max_decoded_substream)
1092 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1093 }
1094
1095 if (substream_parity_present[substr]) {
1096 uint8_t parity, checksum;
1097
1098 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1099 goto substream_length_mismatch;
1100
1101 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1102 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1103
1104 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1105 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1106 if ( get_bits(&gb, 8) != checksum)
1107 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1108 }
1109
1110 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1111 goto substream_length_mismatch;
1112
1113 next_substr:
1114 if (!s->restart_seen)
1115 av_log(m->avctx, AV_LOG_ERROR,
1116 "No restart header present in substream %d.\n", substr);
1117
1118 buf += substream_data_len[substr];
1119 }
1120
1121 rematrix_channels(m, m->max_decoded_substream);
1122
1123 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1124 return -1;
1125
1126 return length;
1127
1128 substream_length_mismatch:
1129 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1130 return -1;
1131
1132 error:
1133 m->params_valid = 0;
1134 return -1;
1135 }
1136
1137 AVCodec mlp_decoder = {
1138 "mlp",
1139 CODEC_TYPE_AUDIO,
1140 CODEC_ID_MLP,
1141 sizeof(MLPDecodeContext),
1142 mlp_decode_init,
1143 NULL,
1144 NULL,
1145 read_access_unit,
1146 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1147 };
1148
1149 #if CONFIG_TRUEHD_DECODER
1150 AVCodec truehd_decoder = {
1151 "truehd",
1152 CODEC_TYPE_AUDIO,
1153 CODEC_ID_TRUEHD,
1154 sizeof(MLPDecodeContext),
1155 mlp_decode_init,
1156 NULL,
1157 NULL,
1158 read_access_unit,
1159 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1160 };
1161 #endif /* CONFIG_TRUEHD_DECODER */