mlpdec: output_shift can be any value from -8 to 7.
[libav.git] / libavcodec / mlpdec.c
1 /*
2 * MLP decoder
3 * Copyright (c) 2007-2008 Ian Caulfield
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file libavcodec/mlpdec.c
24 * MLP decoder
25 */
26
27 #include <stdint.h>
28
29 #include "avcodec.h"
30 #include "libavutil/intreadwrite.h"
31 #include "bitstream.h"
32 #include "libavutil/crc.h"
33 #include "parser.h"
34 #include "mlp_parser.h"
35 #include "mlp.h"
36
37 /** number of bits used for VLC lookup - longest Huffman code is 9 */
38 #define VLC_BITS 9
39
40
41 static const char* sample_message =
42 "Please file a bug report following the instructions at "
43 "http://ffmpeg.org/bugreports.html and include "
44 "a sample of this file.";
45
46 typedef struct SubStream {
47 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded.
48 uint8_t restart_seen;
49
50 //@{
51 /** restart header data */
52 //! The type of noise to be used in the rematrix stage.
53 uint16_t noise_type;
54
55 //! The index of the first channel coded in this substream.
56 uint8_t min_channel;
57 //! The index of the last channel coded in this substream.
58 uint8_t max_channel;
59 //! The number of channels input into the rematrix stage.
60 uint8_t max_matrix_channel;
61 //! For each channel output by the matrix, the output channel to map it to
62 uint8_t ch_assign[MAX_CHANNELS];
63
64 //! The left shift applied to random noise in 0x31ea substreams.
65 uint8_t noise_shift;
66 //! The current seed value for the pseudorandom noise generator(s).
67 uint32_t noisegen_seed;
68
69 //! Set if the substream contains extra info to check the size of VLC blocks.
70 uint8_t data_check_present;
71
72 //! Bitmask of which parameter sets are conveyed in a decoding parameter block.
73 uint8_t param_presence_flags;
74 #define PARAM_BLOCKSIZE (1 << 7)
75 #define PARAM_MATRIX (1 << 6)
76 #define PARAM_OUTSHIFT (1 << 5)
77 #define PARAM_QUANTSTEP (1 << 4)
78 #define PARAM_FIR (1 << 3)
79 #define PARAM_IIR (1 << 2)
80 #define PARAM_HUFFOFFSET (1 << 1)
81 #define PARAM_PRESENCE (1 << 0)
82 //@}
83
84 //@{
85 /** matrix data */
86
87 //! Number of matrices to be applied.
88 uint8_t num_primitive_matrices;
89
90 //! matrix output channel
91 uint8_t matrix_out_ch[MAX_MATRICES];
92
93 //! Whether the LSBs of the matrix output are encoded in the bitstream.
94 uint8_t lsb_bypass[MAX_MATRICES];
95 //! Matrix coefficients, stored as 2.14 fixed point.
96 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2];
97 //! Left shift to apply to noise values in 0x31eb substreams.
98 uint8_t matrix_noise_shift[MAX_MATRICES];
99 //@}
100
101 //! Left shift to apply to Huffman-decoded residuals.
102 uint8_t quant_step_size[MAX_CHANNELS];
103
104 //! number of PCM samples in current audio block
105 uint16_t blocksize;
106 //! Number of PCM samples decoded so far in this frame.
107 uint16_t blockpos;
108
109 //! Left shift to apply to decoded PCM values to get final 24-bit output.
110 int8_t output_shift[MAX_CHANNELS];
111
112 //! Running XOR of all output samples.
113 int32_t lossless_check_data;
114
115 } SubStream;
116
117 typedef struct MLPDecodeContext {
118 AVCodecContext *avctx;
119
120 //! Set if a valid major sync block has been read. Otherwise no decoding is possible.
121 uint8_t params_valid;
122
123 //! Number of substreams contained within this stream.
124 uint8_t num_substreams;
125
126 //! Index of the last substream to decode - further substreams are skipped.
127 uint8_t max_decoded_substream;
128
129 //! number of PCM samples contained in each frame
130 int access_unit_size;
131 //! next power of two above the number of samples in each frame
132 int access_unit_size_pow2;
133
134 SubStream substream[MAX_SUBSTREAMS];
135
136 ChannelParams channel_params[MAX_CHANNELS];
137
138 int8_t noise_buffer[MAX_BLOCKSIZE_POW2];
139 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS];
140 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2];
141 } MLPDecodeContext;
142
143 static VLC huff_vlc[3];
144
145 /** Initialize static data, constant between all invocations of the codec. */
146
147 static av_cold void init_static(void)
148 {
149 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18,
150 &ff_mlp_huffman_tables[0][0][1], 2, 1,
151 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512);
152 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16,
153 &ff_mlp_huffman_tables[1][0][1], 2, 1,
154 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512);
155 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15,
156 &ff_mlp_huffman_tables[2][0][1], 2, 1,
157 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512);
158
159 ff_mlp_init_crc();
160 }
161
162 static inline int32_t calculate_sign_huff(MLPDecodeContext *m,
163 unsigned int substr, unsigned int ch)
164 {
165 ChannelParams *cp = &m->channel_params[ch];
166 SubStream *s = &m->substream[substr];
167 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch];
168 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1);
169 int32_t sign_huff_offset = cp->huff_offset;
170
171 if (cp->codebook > 0)
172 sign_huff_offset -= 7 << lsb_bits;
173
174 if (sign_shift >= 0)
175 sign_huff_offset -= 1 << sign_shift;
176
177 return sign_huff_offset;
178 }
179
180 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs
181 * and plain LSBs. */
182
183 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp,
184 unsigned int substr, unsigned int pos)
185 {
186 SubStream *s = &m->substream[substr];
187 unsigned int mat, channel;
188
189 for (mat = 0; mat < s->num_primitive_matrices; mat++)
190 if (s->lsb_bypass[mat])
191 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp);
192
193 for (channel = s->min_channel; channel <= s->max_channel; channel++) {
194 ChannelParams *cp = &m->channel_params[channel];
195 int codebook = cp->codebook;
196 int quant_step_size = s->quant_step_size[channel];
197 int lsb_bits = cp->huff_lsbs - quant_step_size;
198 int result = 0;
199
200 if (codebook > 0)
201 result = get_vlc2(gbp, huff_vlc[codebook-1].table,
202 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS);
203
204 if (result < 0)
205 return -1;
206
207 if (lsb_bits > 0)
208 result = (result << lsb_bits) + get_bits(gbp, lsb_bits);
209
210 result += cp->sign_huff_offset;
211 result <<= quant_step_size;
212
213 m->sample_buffer[pos + s->blockpos][channel] = result;
214 }
215
216 return 0;
217 }
218
219 static av_cold int mlp_decode_init(AVCodecContext *avctx)
220 {
221 MLPDecodeContext *m = avctx->priv_data;
222 int substr;
223
224 init_static();
225 m->avctx = avctx;
226 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
227 m->substream[substr].lossless_check_data = 0xffffffff;
228
229 return 0;
230 }
231
232 /** Read a major sync info header - contains high level information about
233 * the stream - sample rate, channel arrangement etc. Most of this
234 * information is not actually necessary for decoding, only for playback.
235 */
236
237 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb)
238 {
239 MLPHeaderInfo mh;
240 int substr;
241
242 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0)
243 return -1;
244
245 if (mh.group1_bits == 0) {
246 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n");
247 return -1;
248 }
249 if (mh.group2_bits > mh.group1_bits) {
250 av_log(m->avctx, AV_LOG_ERROR,
251 "Channel group 2 cannot have more bits per sample than group 1.\n");
252 return -1;
253 }
254
255 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) {
256 av_log(m->avctx, AV_LOG_ERROR,
257 "Channel groups with differing sample rates are not currently supported.\n");
258 return -1;
259 }
260
261 if (mh.group1_samplerate == 0) {
262 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n");
263 return -1;
264 }
265 if (mh.group1_samplerate > MAX_SAMPLERATE) {
266 av_log(m->avctx, AV_LOG_ERROR,
267 "Sampling rate %d is greater than the supported maximum (%d).\n",
268 mh.group1_samplerate, MAX_SAMPLERATE);
269 return -1;
270 }
271 if (mh.access_unit_size > MAX_BLOCKSIZE) {
272 av_log(m->avctx, AV_LOG_ERROR,
273 "Block size %d is greater than the supported maximum (%d).\n",
274 mh.access_unit_size, MAX_BLOCKSIZE);
275 return -1;
276 }
277 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) {
278 av_log(m->avctx, AV_LOG_ERROR,
279 "Block size pow2 %d is greater than the supported maximum (%d).\n",
280 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2);
281 return -1;
282 }
283
284 if (mh.num_substreams == 0)
285 return -1;
286 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) {
287 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n");
288 return -1;
289 }
290 if (mh.num_substreams > MAX_SUBSTREAMS) {
291 av_log(m->avctx, AV_LOG_ERROR,
292 "Number of substreams %d is larger than the maximum supported "
293 "by the decoder. %s\n", mh.num_substreams, sample_message);
294 return -1;
295 }
296
297 m->access_unit_size = mh.access_unit_size;
298 m->access_unit_size_pow2 = mh.access_unit_size_pow2;
299
300 m->num_substreams = mh.num_substreams;
301 m->max_decoded_substream = m->num_substreams - 1;
302
303 m->avctx->sample_rate = mh.group1_samplerate;
304 m->avctx->frame_size = mh.access_unit_size;
305
306 m->avctx->bits_per_raw_sample = mh.group1_bits;
307 if (mh.group1_bits > 16)
308 m->avctx->sample_fmt = SAMPLE_FMT_S32;
309 else
310 m->avctx->sample_fmt = SAMPLE_FMT_S16;
311
312 m->params_valid = 1;
313 for (substr = 0; substr < MAX_SUBSTREAMS; substr++)
314 m->substream[substr].restart_seen = 0;
315
316 return 0;
317 }
318
319 /** Read a restart header from a block in a substream. This contains parameters
320 * required to decode the audio that do not change very often. Generally
321 * (always) present only in blocks following a major sync. */
322
323 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp,
324 const uint8_t *buf, unsigned int substr)
325 {
326 SubStream *s = &m->substream[substr];
327 unsigned int ch;
328 int sync_word, tmp;
329 uint8_t checksum;
330 uint8_t lossless_check;
331 int start_count = get_bits_count(gbp);
332
333 sync_word = get_bits(gbp, 13);
334
335 if (sync_word != 0x31ea >> 1) {
336 av_log(m->avctx, AV_LOG_ERROR,
337 "restart header sync incorrect (got 0x%04x)\n", sync_word);
338 return -1;
339 }
340 s->noise_type = get_bits1(gbp);
341
342 skip_bits(gbp, 16); /* Output timestamp */
343
344 s->min_channel = get_bits(gbp, 4);
345 s->max_channel = get_bits(gbp, 4);
346 s->max_matrix_channel = get_bits(gbp, 4);
347
348 if (s->min_channel > s->max_channel) {
349 av_log(m->avctx, AV_LOG_ERROR,
350 "Substream min channel cannot be greater than max channel.\n");
351 return -1;
352 }
353
354 if (m->avctx->request_channels > 0
355 && s->max_channel + 1 >= m->avctx->request_channels
356 && substr < m->max_decoded_substream) {
357 av_log(m->avctx, AV_LOG_INFO,
358 "Extracting %d channel downmix from substream %d. "
359 "Further substreams will be skipped.\n",
360 s->max_channel + 1, substr);
361 m->max_decoded_substream = substr;
362 }
363
364 s->noise_shift = get_bits(gbp, 4);
365 s->noisegen_seed = get_bits(gbp, 23);
366
367 skip_bits(gbp, 19);
368
369 s->data_check_present = get_bits1(gbp);
370 lossless_check = get_bits(gbp, 8);
371 if (substr == m->max_decoded_substream
372 && s->lossless_check_data != 0xffffffff) {
373 tmp = xor_32_to_8(s->lossless_check_data);
374 if (tmp != lossless_check)
375 av_log(m->avctx, AV_LOG_WARNING,
376 "Lossless check failed - expected %02x, calculated %02x.\n",
377 lossless_check, tmp);
378 else
379 dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n",
380 substr, tmp);
381 }
382
383 skip_bits(gbp, 16);
384
385 memset(s->ch_assign, 0, sizeof(s->ch_assign));
386
387 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
388 int ch_assign = get_bits(gbp, 6);
389 dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch,
390 ch_assign);
391 if (ch_assign > s->max_matrix_channel) {
392 av_log(m->avctx, AV_LOG_ERROR,
393 "Assignment of matrix channel %d to invalid output channel %d. %s\n",
394 ch, ch_assign, sample_message);
395 return -1;
396 }
397 s->ch_assign[ch_assign] = ch;
398 }
399
400 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count);
401
402 if (checksum != get_bits(gbp, 8))
403 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n");
404
405 /* Set default decoding parameters. */
406 s->param_presence_flags = 0xff;
407 s->num_primitive_matrices = 0;
408 s->blocksize = 8;
409 s->lossless_check_data = 0;
410
411 memset(s->output_shift , 0, sizeof(s->output_shift ));
412 memset(s->quant_step_size, 0, sizeof(s->quant_step_size));
413
414 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
415 ChannelParams *cp = &m->channel_params[ch];
416 cp->filter_params[FIR].order = 0;
417 cp->filter_params[IIR].order = 0;
418 cp->filter_params[FIR].shift = 0;
419 cp->filter_params[IIR].shift = 0;
420
421 /* Default audio coding is 24-bit raw PCM. */
422 cp->huff_offset = 0;
423 cp->sign_huff_offset = (-1) << 23;
424 cp->codebook = 0;
425 cp->huff_lsbs = 24;
426 }
427
428 if (substr == m->max_decoded_substream) {
429 m->avctx->channels = s->max_matrix_channel + 1;
430 }
431
432 return 0;
433 }
434
435 /** Read parameters for one of the prediction filters. */
436
437 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp,
438 unsigned int channel, unsigned int filter)
439 {
440 FilterParams *fp = &m->channel_params[channel].filter_params[filter];
441 const char fchar = filter ? 'I' : 'F';
442 int i, order;
443
444 // Filter is 0 for FIR, 1 for IIR.
445 assert(filter < 2);
446
447 order = get_bits(gbp, 4);
448 if (order > MAX_FILTER_ORDER) {
449 av_log(m->avctx, AV_LOG_ERROR,
450 "%cIR filter order %d is greater than maximum %d.\n",
451 fchar, order, MAX_FILTER_ORDER);
452 return -1;
453 }
454 fp->order = order;
455
456 if (order > 0) {
457 int coeff_bits, coeff_shift;
458
459 fp->shift = get_bits(gbp, 4);
460
461 coeff_bits = get_bits(gbp, 5);
462 coeff_shift = get_bits(gbp, 3);
463 if (coeff_bits < 1 || coeff_bits > 16) {
464 av_log(m->avctx, AV_LOG_ERROR,
465 "%cIR filter coeff_bits must be between 1 and 16.\n",
466 fchar);
467 return -1;
468 }
469 if (coeff_bits + coeff_shift > 16) {
470 av_log(m->avctx, AV_LOG_ERROR,
471 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n",
472 fchar);
473 return -1;
474 }
475
476 for (i = 0; i < order; i++)
477 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift;
478
479 if (get_bits1(gbp)) {
480 int state_bits, state_shift;
481
482 if (filter == FIR) {
483 av_log(m->avctx, AV_LOG_ERROR,
484 "FIR filter has state data specified.\n");
485 return -1;
486 }
487
488 state_bits = get_bits(gbp, 4);
489 state_shift = get_bits(gbp, 4);
490
491 /* TODO: Check validity of state data. */
492
493 for (i = 0; i < order; i++)
494 fp->state[i] = get_sbits(gbp, state_bits) << state_shift;
495 }
496 }
497
498 return 0;
499 }
500
501 /** Read parameters for primitive matrices. */
502
503 static int read_matrix_params(MLPDecodeContext *m, SubStream *s, GetBitContext *gbp)
504 {
505 unsigned int mat, ch;
506
507 s->num_primitive_matrices = get_bits(gbp, 4);
508
509 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
510 int frac_bits, max_chan;
511 s->matrix_out_ch[mat] = get_bits(gbp, 4);
512 frac_bits = get_bits(gbp, 4);
513 s->lsb_bypass [mat] = get_bits1(gbp);
514
515 if (s->matrix_out_ch[mat] > s->max_channel) {
516 av_log(m->avctx, AV_LOG_ERROR,
517 "Invalid channel %d specified as output from matrix.\n",
518 s->matrix_out_ch[mat]);
519 return -1;
520 }
521 if (frac_bits > 14) {
522 av_log(m->avctx, AV_LOG_ERROR,
523 "Too many fractional bits specified.\n");
524 return -1;
525 }
526
527 max_chan = s->max_matrix_channel;
528 if (!s->noise_type)
529 max_chan+=2;
530
531 for (ch = 0; ch <= max_chan; ch++) {
532 int coeff_val = 0;
533 if (get_bits1(gbp))
534 coeff_val = get_sbits(gbp, frac_bits + 2);
535
536 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits);
537 }
538
539 if (s->noise_type)
540 s->matrix_noise_shift[mat] = get_bits(gbp, 4);
541 else
542 s->matrix_noise_shift[mat] = 0;
543 }
544
545 return 0;
546 }
547
548 /** Read channel parameters. */
549
550 static int read_channel_params(MLPDecodeContext *m, unsigned int substr,
551 GetBitContext *gbp, unsigned int ch)
552 {
553 ChannelParams *cp = &m->channel_params[ch];
554 FilterParams *fir = &cp->filter_params[FIR];
555 FilterParams *iir = &cp->filter_params[IIR];
556 SubStream *s = &m->substream[substr];
557
558 if (s->param_presence_flags & PARAM_FIR)
559 if (get_bits1(gbp))
560 if (read_filter_params(m, gbp, ch, FIR) < 0)
561 return -1;
562
563 if (s->param_presence_flags & PARAM_IIR)
564 if (get_bits1(gbp))
565 if (read_filter_params(m, gbp, ch, IIR) < 0)
566 return -1;
567
568 if (fir->order && iir->order &&
569 fir->shift != iir->shift) {
570 av_log(m->avctx, AV_LOG_ERROR,
571 "FIR and IIR filters must use the same precision.\n");
572 return -1;
573 }
574 /* The FIR and IIR filters must have the same precision.
575 * To simplify the filtering code, only the precision of the
576 * FIR filter is considered. If only the IIR filter is employed,
577 * the FIR filter precision is set to that of the IIR filter, so
578 * that the filtering code can use it. */
579 if (!fir->order && iir->order)
580 fir->shift = iir->shift;
581
582 if (s->param_presence_flags & PARAM_HUFFOFFSET)
583 if (get_bits1(gbp))
584 cp->huff_offset = get_sbits(gbp, 15);
585
586 cp->codebook = get_bits(gbp, 2);
587 cp->huff_lsbs = get_bits(gbp, 5);
588
589 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
590
591 /* TODO: validate */
592
593 return 0;
594 }
595
596 /** Read decoding parameters that change more often than those in the restart
597 * header. */
598
599 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp,
600 unsigned int substr)
601 {
602 SubStream *s = &m->substream[substr];
603 unsigned int ch;
604
605 if (s->param_presence_flags & PARAM_PRESENCE)
606 if (get_bits1(gbp))
607 s->param_presence_flags = get_bits(gbp, 8);
608
609 if (s->param_presence_flags & PARAM_BLOCKSIZE)
610 if (get_bits1(gbp)) {
611 s->blocksize = get_bits(gbp, 9);
612 if (s->blocksize > MAX_BLOCKSIZE) {
613 av_log(m->avctx, AV_LOG_ERROR, "block size too large\n");
614 s->blocksize = 0;
615 return -1;
616 }
617 }
618
619 if (s->param_presence_flags & PARAM_MATRIX)
620 if (get_bits1(gbp)) {
621 if (read_matrix_params(m, s, gbp) < 0)
622 return -1;
623 }
624
625 if (s->param_presence_flags & PARAM_OUTSHIFT)
626 if (get_bits1(gbp))
627 for (ch = 0; ch <= s->max_matrix_channel; ch++) {
628 s->output_shift[ch] = get_sbits(gbp, 4);
629 dprintf(m->avctx, "output shift[%d] = %d\n",
630 ch, s->output_shift[ch]);
631 }
632
633 if (s->param_presence_flags & PARAM_QUANTSTEP)
634 if (get_bits1(gbp))
635 for (ch = 0; ch <= s->max_channel; ch++) {
636 ChannelParams *cp = &m->channel_params[ch];
637
638 s->quant_step_size[ch] = get_bits(gbp, 4);
639 /* TODO: validate */
640
641 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
642 }
643
644 for (ch = s->min_channel; ch <= s->max_channel; ch++)
645 if (get_bits1(gbp)) {
646 if (read_channel_params(m, substr, gbp, ch) < 0)
647 return -1;
648 }
649
650 return 0;
651 }
652
653 #define MSB_MASK(bits) (-1u << bits)
654
655 /** Generate PCM samples using the prediction filters and residual values
656 * read from the data stream, and update the filter state. */
657
658 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
659 unsigned int channel)
660 {
661 SubStream *s = &m->substream[substr];
662 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
663 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
664 &m->channel_params[channel].filter_params[IIR], };
665 unsigned int filter_shift = fp[FIR]->shift;
666 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
667 int index = MAX_BLOCKSIZE;
668 int j, i;
669
670 for (j = 0; j < NUM_FILTERS; j++) {
671 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
672 MAX_FILTER_ORDER * sizeof(int32_t));
673 }
674
675 for (i = 0; i < s->blocksize; i++) {
676 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
677 unsigned int order;
678 int64_t accum = 0;
679 int32_t result;
680
681 /* TODO: Move this code to DSPContext? */
682
683 for (j = 0; j < NUM_FILTERS; j++)
684 for (order = 0; order < fp[j]->order; order++)
685 accum += (int64_t)filter_state_buffer[j][index + order] *
686 fp[j]->coeff[order];
687
688 accum = accum >> filter_shift;
689 result = (accum + residual) & mask;
690
691 --index;
692
693 filter_state_buffer[FIR][index] = result;
694 filter_state_buffer[IIR][index] = result - accum;
695
696 m->sample_buffer[i + s->blockpos][channel] = result;
697 }
698
699 for (j = 0; j < NUM_FILTERS; j++) {
700 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
701 MAX_FILTER_ORDER * sizeof(int32_t));
702 }
703 }
704
705 /** Read a block of PCM residual data (or actual if no filtering active). */
706
707 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
708 unsigned int substr)
709 {
710 SubStream *s = &m->substream[substr];
711 unsigned int i, ch, expected_stream_pos = 0;
712
713 if (s->data_check_present) {
714 expected_stream_pos = get_bits_count(gbp);
715 expected_stream_pos += get_bits(gbp, 16);
716 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
717 "we have not tested yet. %s\n", sample_message);
718 }
719
720 if (s->blockpos + s->blocksize > m->access_unit_size) {
721 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
722 return -1;
723 }
724
725 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
726 s->blocksize * sizeof(m->bypassed_lsbs[0]));
727
728 for (i = 0; i < s->blocksize; i++) {
729 if (read_huff_channels(m, gbp, substr, i) < 0)
730 return -1;
731 }
732
733 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
734 filter_channel(m, substr, ch);
735 }
736
737 s->blockpos += s->blocksize;
738
739 if (s->data_check_present) {
740 if (get_bits_count(gbp) != expected_stream_pos)
741 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
742 skip_bits(gbp, 8);
743 }
744
745 return 0;
746 }
747
748 /** Data table used for TrueHD noise generation function. */
749
750 static const int8_t noise_table[256] = {
751 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
752 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
753 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
754 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
755 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
756 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
757 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
758 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
759 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
760 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
761 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
762 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
763 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
764 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
765 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
766 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
767 };
768
769 /** Noise generation functions.
770 * I'm not sure what these are for - they seem to be some kind of pseudorandom
771 * sequence generators, used to generate noise data which is used when the
772 * channels are rematrixed. I'm not sure if they provide a practical benefit
773 * to compression, or just obfuscate the decoder. Are they for some kind of
774 * dithering? */
775
776 /** Generate two channels of noise, used in the matrix when
777 * restart sync word == 0x31ea. */
778
779 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
780 {
781 SubStream *s = &m->substream[substr];
782 unsigned int i;
783 uint32_t seed = s->noisegen_seed;
784 unsigned int maxchan = s->max_matrix_channel;
785
786 for (i = 0; i < s->blockpos; i++) {
787 uint16_t seed_shr7 = seed >> 7;
788 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
789 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
790
791 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
792 }
793
794 s->noisegen_seed = seed;
795 }
796
797 /** Generate a block of noise, used when restart sync word == 0x31eb. */
798
799 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
800 {
801 SubStream *s = &m->substream[substr];
802 unsigned int i;
803 uint32_t seed = s->noisegen_seed;
804
805 for (i = 0; i < m->access_unit_size_pow2; i++) {
806 uint8_t seed_shr15 = seed >> 15;
807 m->noise_buffer[i] = noise_table[seed_shr15];
808 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
809 }
810
811 s->noisegen_seed = seed;
812 }
813
814
815 /** Apply the channel matrices in turn to reconstruct the original audio
816 * samples. */
817
818 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
819 {
820 SubStream *s = &m->substream[substr];
821 unsigned int mat, src_ch, i;
822 unsigned int maxchan;
823
824 maxchan = s->max_matrix_channel;
825 if (!s->noise_type) {
826 generate_2_noise_channels(m, substr);
827 maxchan += 2;
828 } else {
829 fill_noise_buffer(m, substr);
830 }
831
832 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
833 int matrix_noise_shift = s->matrix_noise_shift[mat];
834 unsigned int dest_ch = s->matrix_out_ch[mat];
835 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
836
837 /* TODO: DSPContext? */
838
839 for (i = 0; i < s->blockpos; i++) {
840 int64_t accum = 0;
841 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
842 accum += (int64_t)m->sample_buffer[i][src_ch]
843 * s->matrix_coeff[mat][src_ch];
844 }
845 if (matrix_noise_shift) {
846 uint32_t index = s->num_primitive_matrices - mat;
847 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
848 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
849 }
850 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
851 + m->bypassed_lsbs[i][mat];
852 }
853 }
854 }
855
856 /** Write the audio data into the output buffer. */
857
858 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
859 uint8_t *data, unsigned int *data_size, int is32)
860 {
861 SubStream *s = &m->substream[substr];
862 unsigned int i, out_ch = 0;
863 int32_t *data_32 = (int32_t*) data;
864 int16_t *data_16 = (int16_t*) data;
865
866 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
867 return -1;
868
869 for (i = 0; i < s->blockpos; i++) {
870 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
871 int mat_ch = s->ch_assign[out_ch];
872 int32_t sample = m->sample_buffer[i][mat_ch]
873 << s->output_shift[mat_ch];
874 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
875 if (is32) *data_32++ = sample << 8;
876 else *data_16++ = sample >> 8;
877 }
878 }
879
880 *data_size = i * out_ch * (is32 ? 4 : 2);
881
882 return 0;
883 }
884
885 static int output_data(MLPDecodeContext *m, unsigned int substr,
886 uint8_t *data, unsigned int *data_size)
887 {
888 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
889 return output_data_internal(m, substr, data, data_size, 1);
890 else
891 return output_data_internal(m, substr, data, data_size, 0);
892 }
893
894
895 /** Read an access unit from the stream.
896 * Returns < 0 on error, 0 if not enough data is present in the input stream
897 * otherwise returns the number of bytes consumed. */
898
899 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
900 const uint8_t *buf, int buf_size)
901 {
902 MLPDecodeContext *m = avctx->priv_data;
903 GetBitContext gb;
904 unsigned int length, substr;
905 unsigned int substream_start;
906 unsigned int header_size = 4;
907 unsigned int substr_header_size = 0;
908 uint8_t substream_parity_present[MAX_SUBSTREAMS];
909 uint16_t substream_data_len[MAX_SUBSTREAMS];
910 uint8_t parity_bits;
911
912 if (buf_size < 4)
913 return 0;
914
915 length = (AV_RB16(buf) & 0xfff) * 2;
916
917 if (length > buf_size)
918 return -1;
919
920 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
921
922 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
923 dprintf(m->avctx, "Found major sync.\n");
924 if (read_major_sync(m, &gb) < 0)
925 goto error;
926 header_size += 28;
927 }
928
929 if (!m->params_valid) {
930 av_log(m->avctx, AV_LOG_WARNING,
931 "Stream parameters not seen; skipping frame.\n");
932 *data_size = 0;
933 return length;
934 }
935
936 substream_start = 0;
937
938 for (substr = 0; substr < m->num_substreams; substr++) {
939 int extraword_present, checkdata_present, end;
940
941 extraword_present = get_bits1(&gb);
942 skip_bits1(&gb);
943 checkdata_present = get_bits1(&gb);
944 skip_bits1(&gb);
945
946 end = get_bits(&gb, 12) * 2;
947
948 substr_header_size += 2;
949
950 if (extraword_present) {
951 skip_bits(&gb, 16);
952 substr_header_size += 2;
953 }
954
955 if (end + header_size + substr_header_size > length) {
956 av_log(m->avctx, AV_LOG_ERROR,
957 "Indicated length of substream %d data goes off end of "
958 "packet.\n", substr);
959 end = length - header_size - substr_header_size;
960 }
961
962 if (end < substream_start) {
963 av_log(avctx, AV_LOG_ERROR,
964 "Indicated end offset of substream %d data "
965 "is smaller than calculated start offset.\n",
966 substr);
967 goto error;
968 }
969
970 if (substr > m->max_decoded_substream)
971 continue;
972
973 substream_parity_present[substr] = checkdata_present;
974 substream_data_len[substr] = end - substream_start;
975 substream_start = end;
976 }
977
978 parity_bits = ff_mlp_calculate_parity(buf, 4);
979 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
980
981 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
982 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
983 goto error;
984 }
985
986 buf += header_size + substr_header_size;
987
988 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
989 SubStream *s = &m->substream[substr];
990 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
991
992 s->blockpos = 0;
993 do {
994 if (get_bits1(&gb)) {
995 if (get_bits1(&gb)) {
996 /* A restart header should be present. */
997 if (read_restart_header(m, &gb, buf, substr) < 0)
998 goto next_substr;
999 s->restart_seen = 1;
1000 }
1001
1002 if (!s->restart_seen) {
1003 av_log(m->avctx, AV_LOG_ERROR,
1004 "No restart header present in substream %d.\n",
1005 substr);
1006 goto next_substr;
1007 }
1008
1009 if (read_decoding_params(m, &gb, substr) < 0)
1010 goto next_substr;
1011 }
1012
1013 if (!s->restart_seen) {
1014 av_log(m->avctx, AV_LOG_ERROR,
1015 "No restart header present in substream %d.\n",
1016 substr);
1017 goto next_substr;
1018 }
1019
1020 if (read_block_data(m, &gb, substr) < 0)
1021 return -1;
1022
1023 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1024 && get_bits1(&gb) == 0);
1025
1026 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1027 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
1028 (show_bits_long(&gb, 32) == END_OF_STREAM ||
1029 show_bits_long(&gb, 20) == 0xd234e)) {
1030 skip_bits(&gb, 18);
1031 if (substr == m->max_decoded_substream)
1032 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1033
1034 if (get_bits1(&gb)) {
1035 int shorten_by = get_bits(&gb, 13);
1036 shorten_by = FFMIN(shorten_by, s->blockpos);
1037 s->blockpos -= shorten_by;
1038 } else
1039 skip_bits(&gb, 13);
1040 }
1041 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1042 substream_parity_present[substr]) {
1043 uint8_t parity, checksum;
1044
1045 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1046 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1047 av_log(m->avctx, AV_LOG_ERROR,
1048 "Substream %d parity check failed.\n", substr);
1049
1050 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
1051 if (checksum != get_bits(&gb, 8))
1052 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
1053 substr);
1054 }
1055 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1056 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
1057 substr);
1058 return -1;
1059 }
1060
1061 next_substr:
1062 buf += substream_data_len[substr];
1063 }
1064
1065 rematrix_channels(m, m->max_decoded_substream);
1066
1067 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1068 return -1;
1069
1070 return length;
1071
1072 error:
1073 m->params_valid = 0;
1074 return -1;
1075 }
1076
1077 #if CONFIG_MLP_DECODER
1078 AVCodec mlp_decoder = {
1079 "mlp",
1080 CODEC_TYPE_AUDIO,
1081 CODEC_ID_MLP,
1082 sizeof(MLPDecodeContext),
1083 mlp_decode_init,
1084 NULL,
1085 NULL,
1086 read_access_unit,
1087 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1088 };
1089 #endif /* CONFIG_MLP_DECODER */
1090
1091 #if CONFIG_TRUEHD_DECODER
1092 AVCodec truehd_decoder = {
1093 "truehd",
1094 CODEC_TYPE_AUDIO,
1095 CODEC_ID_TRUEHD,
1096 sizeof(MLPDecodeContext),
1097 mlp_decode_init,
1098 NULL,
1099 NULL,
1100 read_access_unit,
1101 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1102 };
1103 #endif /* CONFIG_TRUEHD_DECODER */