mlp: Simplify adressing of state and coeffs arrays for both filters by making
[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 int32_t *fcoeff = m->channel_params[channel].coeff[filter];
499 int coeff_bits, coeff_shift;
500
501 fp->shift = get_bits(gbp, 4);
502
503 coeff_bits = get_bits(gbp, 5);
504 coeff_shift = get_bits(gbp, 3);
505 if (coeff_bits < 1 || coeff_bits > 16) {
506 av_log(m->avctx, AV_LOG_ERROR,
507 "%cIR filter coeff_bits must be between 1 and 16.\n",
508 fchar);
509 return -1;
510 }
511 if (coeff_bits + coeff_shift > 16) {
512 av_log(m->avctx, AV_LOG_ERROR,
513 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
514 fchar);
515 return -1;
516 }
517
518 for (i = 0; i < order; i++)
519 fcoeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
520
521 if (get_bits1(gbp)) {
522 int state_bits, state_shift;
523
524 if (filter == FIR) {
525 av_log(m->avctx, AV_LOG_ERROR,
526 "FIR filter has state data specified.\n");
527 return -1;
528 }
529
530 state_bits = get_bits(gbp, 4);
531 state_shift = get_bits(gbp, 4);
532
533 /* TODO: Check validity of state data. */
534
535 for (i = 0; i < order; i++)
536 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
537 }
538 }
539
540 return 0;
541 }
542
543 /** Read parameters for primitive matrices. */
544
545 static int read_matrix_params(MLPDecodeContext *m, unsigned int substr, GetBitContext *gbp)
546 {
547 SubStream *s = &m->substream[substr];
548 unsigned int mat, ch;
549 const int max_primitive_matrices = m->avctx->codec_id == CODEC_ID_MLP
550 ? MAX_MATRICES_MLP
551 : MAX_MATRICES_TRUEHD;
552
553 if (m->matrix_changed++ > 1) {
554 av_log(m->avctx, AV_LOG_ERROR, "Matrices may change only once per access unit.\n");
555 return -1;
556 }
557
558 s->num_primitive_matrices = get_bits(gbp, 4);
559
560 if (s->num_primitive_matrices > max_primitive_matrices) {
561 av_log(m->avctx, AV_LOG_ERROR,
562 "Number of primitive matrices cannot be greater than %d.\n",
563 max_primitive_matrices);
564 return -1;
565 }
566
567 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
568 int frac_bits, max_chan;
569 s->matrix_out_ch[mat] = get_bits(gbp, 4);
570 frac_bits = get_bits(gbp, 4);
571 s->lsb_bypass [mat] = get_bits1(gbp);
572
573 if (s->matrix_out_ch[mat] > s->max_matrix_channel) {
574 av_log(m->avctx, AV_LOG_ERROR,
575 "Invalid channel %d specified as output from matrix.\n",
576 s->matrix_out_ch[mat]);
577 return -1;
578 }
579 if (frac_bits > 14) {
580 av_log(m->avctx, AV_LOG_ERROR,
581 "Too many fractional bits specified.\n");
582 return -1;
583 }
584
585 max_chan = s->max_matrix_channel;
586 if (!s->noise_type)
587 max_chan+=2;
588
589 for (ch = 0; ch <= max_chan; ch++) {
590 int coeff_val = 0;
591 if (get_bits1(gbp))
592 coeff_val = get_sbits(gbp, frac_bits + 2);
593
594 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
595 }
596
597 if (s->noise_type)
598 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
599 else
600 s->matrix_noise_shift[mat] = 0;
601 }
602
603 return 0;
604 }
605
606 /** Read channel parameters. */
607
608 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
609 GetBitContext *gbp, unsigned int ch)
610 {
611 ChannelParams *cp = &m->channel_params[ch];
612 FilterParams *fir = &cp->filter_params[FIR];
613 FilterParams *iir = &cp->filter_params[IIR];
614 SubStream *s = &m->substream[substr];
615
616 if (s->param_presence_flags & PARAM_FIR)
617 if (get_bits1(gbp))
618 if (read_filter_params(m, gbp, ch, FIR) < 0)
619 return -1;
620
621 if (s->param_presence_flags & PARAM_IIR)
622 if (get_bits1(gbp))
623 if (read_filter_params(m, gbp, ch, IIR) < 0)
624 return -1;
625
626 if (fir->order + iir->order > 8) {
627 av_log(m->avctx, AV_LOG_ERROR, "Total filter orders too high.\n");
628 return -1;
629 }
630
631 if (fir->order && iir->order &&
632 fir->shift != iir->shift) {
633 av_log(m->avctx, AV_LOG_ERROR,
634 "FIR and IIR filters must use the same precision.\n");
635 return -1;
636 }
637 /* The FIR and IIR filters must have the same precision.
638 * To simplify the filtering code, only the precision of the
639 * FIR filter is considered. If only the IIR filter is employed,
640 * the FIR filter precision is set to that of the IIR filter, so
641 * that the filtering code can use it. */
642 if (!fir->order && iir->order)
643 fir->shift = iir->shift;
644
645 if (s->param_presence_flags & PARAM_HUFFOFFSET)
646 if (get_bits1(gbp))
647 cp->huff_offset = get_sbits(gbp, 15);
648
649 cp->codebook = get_bits(gbp, 2);
650 cp->huff_lsbs = get_bits(gbp, 5);
651
652 if (cp->huff_lsbs > 24) {
653 av_log(m->avctx, AV_LOG_ERROR, "Invalid huff_lsbs.\n");
654 return -1;
655 }
656
657 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
658
659 return 0;
660 }
661
662 /** Read decoding parameters that change more often than those in the restart
663 * header. */
664
665 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
666 unsigned int substr)
667 {
668 SubStream *s = &m->substream[substr];
669 unsigned int ch;
670
671 if (s->param_presence_flags & PARAM_PRESENCE)
672 if (get_bits1(gbp))
673 s->param_presence_flags = get_bits(gbp, 8);
674
675 if (s->param_presence_flags & PARAM_BLOCKSIZE)
676 if (get_bits1(gbp)) {
677 s->blocksize = get_bits(gbp, 9);
678 if (s->blocksize < 8 || s->blocksize > m->access_unit_size) {
679 av_log(m->avctx, AV_LOG_ERROR, "Invalid blocksize.");
680 s->blocksize = 0;
681 return -1;
682 }
683 }
684
685 if (s->param_presence_flags & PARAM_MATRIX)
686 if (get_bits1(gbp))
687 if (read_matrix_params(m, substr, gbp) < 0)
688 return -1;
689
690 if (s->param_presence_flags & PARAM_OUTSHIFT)
691 if (get_bits1(gbp))
692 for (ch = 0; ch <= s->max_matrix_channel; ch++)
693 s->output_shift[ch] = get_sbits(gbp, 4);
694
695 if (s->param_presence_flags & PARAM_QUANTSTEP)
696 if (get_bits1(gbp))
697 for (ch = 0; ch <= s->max_channel; ch++) {
698 ChannelParams *cp = &m->channel_params[ch];
699
700 s->quant_step_size[ch] = get_bits(gbp, 4);
701
702 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
703 }
704
705 for (ch = s->min_channel; ch <= s->max_channel; ch++)
706 if (get_bits1(gbp))
707 if (read_channel_params(m, substr, gbp, ch) < 0)
708 return -1;
709
710 return 0;
711 }
712
713 #define MSB_MASK(bits) (-1u << bits)
714
715 /** Generate PCM samples using the prediction filters and residual values
716 * read from the data stream, and update the filter state. */
717
718 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
719 unsigned int channel)
720 {
721 SubStream *s = &m->substream[substr];
722 const int32_t *fircoeff = m->channel_params[channel].coeff[FIR];
723 int32_t state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FIR_ORDER];
724 int32_t *firbuf = state_buffer[FIR] + MAX_BLOCKSIZE;
725 int32_t *iirbuf = state_buffer[IIR] + MAX_BLOCKSIZE;
726 FilterParams *fir = &m->channel_params[channel].filter_params[FIR];
727 FilterParams *iir = &m->channel_params[channel].filter_params[IIR];
728 unsigned int filter_shift = fir->shift;
729 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
730
731 memcpy(firbuf, fir->state, MAX_FIR_ORDER * sizeof(int32_t));
732 memcpy(iirbuf, iir->state, MAX_IIR_ORDER * sizeof(int32_t));
733
734 m->dsp.mlp_filter_channel(firbuf, fircoeff,
735 fir->order, iir->order,
736 filter_shift, mask, s->blocksize,
737 &m->sample_buffer[s->blockpos][channel]);
738
739 memcpy(fir->state, firbuf - s->blocksize, MAX_FIR_ORDER * sizeof(int32_t));
740 memcpy(iir->state, iirbuf - s->blocksize, MAX_IIR_ORDER * sizeof(int32_t));
741 }
742
743 /** Read a block of PCM residual data (or actual if no filtering active). */
744
745 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
746 unsigned int substr)
747 {
748 SubStream *s = &m->substream[substr];
749 unsigned int i, ch, expected_stream_pos = 0;
750
751 if (s->data_check_present) {
752 expected_stream_pos = get_bits_count(gbp);
753 expected_stream_pos += get_bits(gbp, 16);
754 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
755 "we have not tested yet. %s\n", sample_message);
756 }
757
758 if (s->blockpos + s->blocksize > m->access_unit_size) {
759 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
760 return -1;
761 }
762
763 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
764 s->blocksize * sizeof(m->bypassed_lsbs[0]));
765
766 for (i = 0; i < s->blocksize; i++)
767 if (read_huff_channels(m, gbp, substr, i) < 0)
768 return -1;
769
770 for (ch = s->min_channel; ch <= s->max_channel; ch++)
771 filter_channel(m, substr, ch);
772
773 s->blockpos += s->blocksize;
774
775 if (s->data_check_present) {
776 if (get_bits_count(gbp) != expected_stream_pos)
777 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
778 skip_bits(gbp, 8);
779 }
780
781 return 0;
782 }
783
784 /** Data table used for TrueHD noise generation function. */
785
786 static const int8_t noise_table[256] = {
787 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
788 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
789 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
790 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
791 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
792 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
793 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
794 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
795 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
796 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
797 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
798 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
799 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
800 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
801 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
802 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
803 };
804
805 /** Noise generation functions.
806 * I'm not sure what these are for - they seem to be some kind of pseudorandom
807 * sequence generators, used to generate noise data which is used when the
808 * channels are rematrixed. I'm not sure if they provide a practical benefit
809 * to compression, or just obfuscate the decoder. Are they for some kind of
810 * dithering? */
811
812 /** Generate two channels of noise, used in the matrix when
813 * restart sync word == 0x31ea. */
814
815 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
816 {
817 SubStream *s = &m->substream[substr];
818 unsigned int i;
819 uint32_t seed = s->noisegen_seed;
820 unsigned int maxchan = s->max_matrix_channel;
821
822 for (i = 0; i < s->blockpos; i++) {
823 uint16_t seed_shr7 = seed >> 7;
824 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
825 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
826
827 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
828 }
829
830 s->noisegen_seed = seed;
831 }
832
833 /** Generate a block of noise, used when restart sync word == 0x31eb. */
834
835 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
836 {
837 SubStream *s = &m->substream[substr];
838 unsigned int i;
839 uint32_t seed = s->noisegen_seed;
840
841 for (i = 0; i < m->access_unit_size_pow2; i++) {
842 uint8_t seed_shr15 = seed >> 15;
843 m->noise_buffer[i] = noise_table[seed_shr15];
844 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
845 }
846
847 s->noisegen_seed = seed;
848 }
849
850
851 /** Apply the channel matrices in turn to reconstruct the original audio
852 * samples. */
853
854 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
855 {
856 SubStream *s = &m->substream[substr];
857 unsigned int mat, src_ch, i;
858 unsigned int maxchan;
859
860 maxchan = s->max_matrix_channel;
861 if (!s->noise_type) {
862 generate_2_noise_channels(m, substr);
863 maxchan += 2;
864 } else {
865 fill_noise_buffer(m, substr);
866 }
867
868 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
869 int matrix_noise_shift = s->matrix_noise_shift[mat];
870 unsigned int dest_ch = s->matrix_out_ch[mat];
871 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
872 int32_t *coeffs = s->matrix_coeff[mat];
873 int index = s->num_primitive_matrices - mat;
874 int index2 = 2 * index + 1;
875
876 /* TODO: DSPContext? */
877
878 for (i = 0; i < s->blockpos; i++) {
879 int32_t bypassed_lsb = m->bypassed_lsbs[i][mat];
880 int32_t *samples = m->sample_buffer[i];
881 int64_t accum = 0;
882
883 for (src_ch = 0; src_ch <= maxchan; src_ch++)
884 accum += (int64_t) samples[src_ch] * coeffs[src_ch];
885
886 if (matrix_noise_shift) {
887 index &= m->access_unit_size_pow2 - 1;
888 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
889 index += index2;
890 }
891
892 samples[dest_ch] = ((accum >> 14) & mask) + bypassed_lsb;
893 }
894 }
895 }
896
897 /** Write the audio data into the output buffer. */
898
899 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
900 uint8_t *data, unsigned int *data_size, int is32)
901 {
902 SubStream *s = &m->substream[substr];
903 unsigned int i, out_ch = 0;
904 int32_t *data_32 = (int32_t*) data;
905 int16_t *data_16 = (int16_t*) data;
906
907 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
908 return -1;
909
910 for (i = 0; i < s->blockpos; i++) {
911 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
912 int mat_ch = s->ch_assign[out_ch];
913 int32_t sample = m->sample_buffer[i][mat_ch]
914 << s->output_shift[mat_ch];
915 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
916 if (is32) *data_32++ = sample << 8;
917 else *data_16++ = sample >> 8;
918 }
919 }
920
921 *data_size = i * out_ch * (is32 ? 4 : 2);
922
923 return 0;
924 }
925
926 static int output_data(MLPDecodeContext *m, unsigned int substr,
927 uint8_t *data, unsigned int *data_size)
928 {
929 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
930 return output_data_internal(m, substr, data, data_size, 1);
931 else
932 return output_data_internal(m, substr, data, data_size, 0);
933 }
934
935
936 /** Read an access unit from the stream.
937 * Returns < 0 on error, 0 if not enough data is present in the input stream
938 * otherwise returns the number of bytes consumed. */
939
940 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
941 AVPacket *avpkt)
942 {
943 const uint8_t *buf = avpkt->data;
944 int buf_size = avpkt->size;
945 MLPDecodeContext *m = avctx->priv_data;
946 GetBitContext gb;
947 unsigned int length, substr;
948 unsigned int substream_start;
949 unsigned int header_size = 4;
950 unsigned int substr_header_size = 0;
951 uint8_t substream_parity_present[MAX_SUBSTREAMS];
952 uint16_t substream_data_len[MAX_SUBSTREAMS];
953 uint8_t parity_bits;
954
955 if (buf_size < 4)
956 return 0;
957
958 length = (AV_RB16(buf) & 0xfff) * 2;
959
960 if (length > buf_size)
961 return -1;
962
963 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
964
965 m->is_major_sync_unit = 0;
966 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
967 if (read_major_sync(m, &gb) < 0)
968 goto error;
969 m->is_major_sync_unit = 1;
970 header_size += 28;
971 }
972
973 if (!m->params_valid) {
974 av_log(m->avctx, AV_LOG_WARNING,
975 "Stream parameters not seen; skipping frame.\n");
976 *data_size = 0;
977 return length;
978 }
979
980 substream_start = 0;
981
982 for (substr = 0; substr < m->num_substreams; substr++) {
983 int extraword_present, checkdata_present, end, nonrestart_substr;
984
985 extraword_present = get_bits1(&gb);
986 nonrestart_substr = get_bits1(&gb);
987 checkdata_present = get_bits1(&gb);
988 skip_bits1(&gb);
989
990 end = get_bits(&gb, 12) * 2;
991
992 substr_header_size += 2;
993
994 if (extraword_present) {
995 if (m->avctx->codec_id == CODEC_ID_MLP) {
996 av_log(m->avctx, AV_LOG_ERROR, "There must be no extraword for MLP.\n");
997 goto error;
998 }
999 skip_bits(&gb, 16);
1000 substr_header_size += 2;
1001 }
1002
1003 if (!(nonrestart_substr ^ m->is_major_sync_unit)) {
1004 av_log(m->avctx, AV_LOG_ERROR, "Invalid nonrestart_substr.\n");
1005 goto error;
1006 }
1007
1008 if (end + header_size + substr_header_size > length) {
1009 av_log(m->avctx, AV_LOG_ERROR,
1010 "Indicated length of substream %d data goes off end of "
1011 "packet.\n", substr);
1012 end = length - header_size - substr_header_size;
1013 }
1014
1015 if (end < substream_start) {
1016 av_log(avctx, AV_LOG_ERROR,
1017 "Indicated end offset of substream %d data "
1018 "is smaller than calculated start offset.\n",
1019 substr);
1020 goto error;
1021 }
1022
1023 if (substr > m->max_decoded_substream)
1024 continue;
1025
1026 substream_parity_present[substr] = checkdata_present;
1027 substream_data_len[substr] = end - substream_start;
1028 substream_start = end;
1029 }
1030
1031 parity_bits = ff_mlp_calculate_parity(buf, 4);
1032 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
1033
1034 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
1035 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
1036 goto error;
1037 }
1038
1039 buf += header_size + substr_header_size;
1040
1041 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
1042 SubStream *s = &m->substream[substr];
1043 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
1044
1045 m->matrix_changed = 0;
1046 memset(m->filter_changed, 0, sizeof(m->filter_changed));
1047
1048 s->blockpos = 0;
1049 do {
1050 if (get_bits1(&gb)) {
1051 if (get_bits1(&gb)) {
1052 /* A restart header should be present. */
1053 if (read_restart_header(m, &gb, buf, substr) < 0)
1054 goto next_substr;
1055 s->restart_seen = 1;
1056 }
1057
1058 if (!s->restart_seen)
1059 goto next_substr;
1060 if (read_decoding_params(m, &gb, substr) < 0)
1061 goto next_substr;
1062 }
1063
1064 if (!s->restart_seen)
1065 goto next_substr;
1066
1067 if (read_block_data(m, &gb, substr) < 0)
1068 return -1;
1069
1070 if (get_bits_count(&gb) >= substream_data_len[substr] * 8)
1071 goto substream_length_mismatch;
1072
1073 } while (!get_bits1(&gb));
1074
1075 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1076
1077 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32) {
1078 int shorten_by;
1079
1080 if (get_bits(&gb, 16) != 0xD234)
1081 return -1;
1082
1083 shorten_by = get_bits(&gb, 16);
1084 if (m->avctx->codec_id == CODEC_ID_TRUEHD && shorten_by & 0x2000)
1085 s->blockpos -= FFMIN(shorten_by & 0x1FFF, s->blockpos);
1086 else if (m->avctx->codec_id == CODEC_ID_MLP && shorten_by != 0xD234)
1087 return -1;
1088
1089 if (substr == m->max_decoded_substream)
1090 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1091 }
1092
1093 if (substream_parity_present[substr]) {
1094 uint8_t parity, checksum;
1095
1096 if (substream_data_len[substr] * 8 - get_bits_count(&gb) != 16)
1097 goto substream_length_mismatch;
1098
1099 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1100 checksum = ff_mlp_checksum8 (buf, substream_data_len[substr] - 2);
1101
1102 if ((get_bits(&gb, 8) ^ parity) != 0xa9 )
1103 av_log(m->avctx, AV_LOG_ERROR, "Substream %d parity check failed.\n", substr);
1104 if ( get_bits(&gb, 8) != checksum)
1105 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n" , substr);
1106 }
1107
1108 if (substream_data_len[substr] * 8 != get_bits_count(&gb))
1109 goto substream_length_mismatch;
1110
1111 next_substr:
1112 if (!s->restart_seen)
1113 av_log(m->avctx, AV_LOG_ERROR,
1114 "No restart header present in substream %d.\n", substr);
1115
1116 buf += substream_data_len[substr];
1117 }
1118
1119 rematrix_channels(m, m->max_decoded_substream);
1120
1121 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1122 return -1;
1123
1124 return length;
1125
1126 substream_length_mismatch:
1127 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", substr);
1128 return -1;
1129
1130 error:
1131 m->params_valid = 0;
1132 return -1;
1133 }
1134
1135 #if CONFIG_MLP_DECODER
1136 AVCodec mlp_decoder = {
1137 "mlp",
1138 CODEC_TYPE_AUDIO,
1139 CODEC_ID_MLP,
1140 sizeof(MLPDecodeContext),
1141 mlp_decode_init,
1142 NULL,
1143 NULL,
1144 read_access_unit,
1145 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1146 };
1147 #endif /* CONFIG_MLP_DECODER */
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 */