mlpdec: quant_step_size can be any value from 0 to 0xF.
[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
640 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
641 }
642
643 for (ch = s->min_channel; ch <= s->max_channel; ch++)
644 if (get_bits1(gbp)) {
645 if (read_channel_params(m, substr, gbp, ch) < 0)
646 return -1;
647 }
648
649 return 0;
650 }
651
652 #define MSB_MASK(bits) (-1u << bits)
653
654 /** Generate PCM samples using the prediction filters and residual values
655 * read from the data stream, and update the filter state. */
656
657 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
658 unsigned int channel)
659 {
660 SubStream *s = &m->substream[substr];
661 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
662 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
663 &m->channel_params[channel].filter_params[IIR], };
664 unsigned int filter_shift = fp[FIR]->shift;
665 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
666 int index = MAX_BLOCKSIZE;
667 int j, i;
668
669 for (j = 0; j < NUM_FILTERS; j++) {
670 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
671 MAX_FILTER_ORDER * sizeof(int32_t));
672 }
673
674 for (i = 0; i < s->blocksize; i++) {
675 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
676 unsigned int order;
677 int64_t accum = 0;
678 int32_t result;
679
680 /* TODO: Move this code to DSPContext? */
681
682 for (j = 0; j < NUM_FILTERS; j++)
683 for (order = 0; order < fp[j]->order; order++)
684 accum += (int64_t)filter_state_buffer[j][index + order] *
685 fp[j]->coeff[order];
686
687 accum = accum >> filter_shift;
688 result = (accum + residual) & mask;
689
690 --index;
691
692 filter_state_buffer[FIR][index] = result;
693 filter_state_buffer[IIR][index] = result - accum;
694
695 m->sample_buffer[i + s->blockpos][channel] = result;
696 }
697
698 for (j = 0; j < NUM_FILTERS; j++) {
699 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
700 MAX_FILTER_ORDER * sizeof(int32_t));
701 }
702 }
703
704 /** Read a block of PCM residual data (or actual if no filtering active). */
705
706 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
707 unsigned int substr)
708 {
709 SubStream *s = &m->substream[substr];
710 unsigned int i, ch, expected_stream_pos = 0;
711
712 if (s->data_check_present) {
713 expected_stream_pos = get_bits_count(gbp);
714 expected_stream_pos += get_bits(gbp, 16);
715 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
716 "we have not tested yet. %s\n", sample_message);
717 }
718
719 if (s->blockpos + s->blocksize > m->access_unit_size) {
720 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
721 return -1;
722 }
723
724 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
725 s->blocksize * sizeof(m->bypassed_lsbs[0]));
726
727 for (i = 0; i < s->blocksize; i++) {
728 if (read_huff_channels(m, gbp, substr, i) < 0)
729 return -1;
730 }
731
732 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
733 filter_channel(m, substr, ch);
734 }
735
736 s->blockpos += s->blocksize;
737
738 if (s->data_check_present) {
739 if (get_bits_count(gbp) != expected_stream_pos)
740 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
741 skip_bits(gbp, 8);
742 }
743
744 return 0;
745 }
746
747 /** Data table used for TrueHD noise generation function. */
748
749 static const int8_t noise_table[256] = {
750 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
751 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
752 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
753 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
754 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
755 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
756 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
757 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
758 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
759 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
760 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
761 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
762 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
763 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
764 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
765 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
766 };
767
768 /** Noise generation functions.
769 * I'm not sure what these are for - they seem to be some kind of pseudorandom
770 * sequence generators, used to generate noise data which is used when the
771 * channels are rematrixed. I'm not sure if they provide a practical benefit
772 * to compression, or just obfuscate the decoder. Are they for some kind of
773 * dithering? */
774
775 /** Generate two channels of noise, used in the matrix when
776 * restart sync word == 0x31ea. */
777
778 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
779 {
780 SubStream *s = &m->substream[substr];
781 unsigned int i;
782 uint32_t seed = s->noisegen_seed;
783 unsigned int maxchan = s->max_matrix_channel;
784
785 for (i = 0; i < s->blockpos; i++) {
786 uint16_t seed_shr7 = seed >> 7;
787 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
788 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
789
790 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
791 }
792
793 s->noisegen_seed = seed;
794 }
795
796 /** Generate a block of noise, used when restart sync word == 0x31eb. */
797
798 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
799 {
800 SubStream *s = &m->substream[substr];
801 unsigned int i;
802 uint32_t seed = s->noisegen_seed;
803
804 for (i = 0; i < m->access_unit_size_pow2; i++) {
805 uint8_t seed_shr15 = seed >> 15;
806 m->noise_buffer[i] = noise_table[seed_shr15];
807 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
808 }
809
810 s->noisegen_seed = seed;
811 }
812
813
814 /** Apply the channel matrices in turn to reconstruct the original audio
815 * samples. */
816
817 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
818 {
819 SubStream *s = &m->substream[substr];
820 unsigned int mat, src_ch, i;
821 unsigned int maxchan;
822
823 maxchan = s->max_matrix_channel;
824 if (!s->noise_type) {
825 generate_2_noise_channels(m, substr);
826 maxchan += 2;
827 } else {
828 fill_noise_buffer(m, substr);
829 }
830
831 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
832 int matrix_noise_shift = s->matrix_noise_shift[mat];
833 unsigned int dest_ch = s->matrix_out_ch[mat];
834 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
835
836 /* TODO: DSPContext? */
837
838 for (i = 0; i < s->blockpos; i++) {
839 int64_t accum = 0;
840 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
841 accum += (int64_t)m->sample_buffer[i][src_ch]
842 * s->matrix_coeff[mat][src_ch];
843 }
844 if (matrix_noise_shift) {
845 uint32_t index = s->num_primitive_matrices - mat;
846 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
847 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
848 }
849 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
850 + m->bypassed_lsbs[i][mat];
851 }
852 }
853 }
854
855 /** Write the audio data into the output buffer. */
856
857 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
858 uint8_t *data, unsigned int *data_size, int is32)
859 {
860 SubStream *s = &m->substream[substr];
861 unsigned int i, out_ch = 0;
862 int32_t *data_32 = (int32_t*) data;
863 int16_t *data_16 = (int16_t*) data;
864
865 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
866 return -1;
867
868 for (i = 0; i < s->blockpos; i++) {
869 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
870 int mat_ch = s->ch_assign[out_ch];
871 int32_t sample = m->sample_buffer[i][mat_ch]
872 << s->output_shift[mat_ch];
873 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
874 if (is32) *data_32++ = sample << 8;
875 else *data_16++ = sample >> 8;
876 }
877 }
878
879 *data_size = i * out_ch * (is32 ? 4 : 2);
880
881 return 0;
882 }
883
884 static int output_data(MLPDecodeContext *m, unsigned int substr,
885 uint8_t *data, unsigned int *data_size)
886 {
887 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
888 return output_data_internal(m, substr, data, data_size, 1);
889 else
890 return output_data_internal(m, substr, data, data_size, 0);
891 }
892
893
894 /** Read an access unit from the stream.
895 * Returns < 0 on error, 0 if not enough data is present in the input stream
896 * otherwise returns the number of bytes consumed. */
897
898 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
899 const uint8_t *buf, int buf_size)
900 {
901 MLPDecodeContext *m = avctx->priv_data;
902 GetBitContext gb;
903 unsigned int length, substr;
904 unsigned int substream_start;
905 unsigned int header_size = 4;
906 unsigned int substr_header_size = 0;
907 uint8_t substream_parity_present[MAX_SUBSTREAMS];
908 uint16_t substream_data_len[MAX_SUBSTREAMS];
909 uint8_t parity_bits;
910
911 if (buf_size < 4)
912 return 0;
913
914 length = (AV_RB16(buf) & 0xfff) * 2;
915
916 if (length > buf_size)
917 return -1;
918
919 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
920
921 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
922 dprintf(m->avctx, "Found major sync.\n");
923 if (read_major_sync(m, &gb) < 0)
924 goto error;
925 header_size += 28;
926 }
927
928 if (!m->params_valid) {
929 av_log(m->avctx, AV_LOG_WARNING,
930 "Stream parameters not seen; skipping frame.\n");
931 *data_size = 0;
932 return length;
933 }
934
935 substream_start = 0;
936
937 for (substr = 0; substr < m->num_substreams; substr++) {
938 int extraword_present, checkdata_present, end;
939
940 extraword_present = get_bits1(&gb);
941 skip_bits1(&gb);
942 checkdata_present = get_bits1(&gb);
943 skip_bits1(&gb);
944
945 end = get_bits(&gb, 12) * 2;
946
947 substr_header_size += 2;
948
949 if (extraword_present) {
950 skip_bits(&gb, 16);
951 substr_header_size += 2;
952 }
953
954 if (end + header_size + substr_header_size > length) {
955 av_log(m->avctx, AV_LOG_ERROR,
956 "Indicated length of substream %d data goes off end of "
957 "packet.\n", substr);
958 end = length - header_size - substr_header_size;
959 }
960
961 if (end < substream_start) {
962 av_log(avctx, AV_LOG_ERROR,
963 "Indicated end offset of substream %d data "
964 "is smaller than calculated start offset.\n",
965 substr);
966 goto error;
967 }
968
969 if (substr > m->max_decoded_substream)
970 continue;
971
972 substream_parity_present[substr] = checkdata_present;
973 substream_data_len[substr] = end - substream_start;
974 substream_start = end;
975 }
976
977 parity_bits = ff_mlp_calculate_parity(buf, 4);
978 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
979
980 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
981 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
982 goto error;
983 }
984
985 buf += header_size + substr_header_size;
986
987 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
988 SubStream *s = &m->substream[substr];
989 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
990
991 s->blockpos = 0;
992 do {
993 if (get_bits1(&gb)) {
994 if (get_bits1(&gb)) {
995 /* A restart header should be present. */
996 if (read_restart_header(m, &gb, buf, substr) < 0)
997 goto next_substr;
998 s->restart_seen = 1;
999 }
1000
1001 if (!s->restart_seen) {
1002 av_log(m->avctx, AV_LOG_ERROR,
1003 "No restart header present in substream %d.\n",
1004 substr);
1005 goto next_substr;
1006 }
1007
1008 if (read_decoding_params(m, &gb, substr) < 0)
1009 goto next_substr;
1010 }
1011
1012 if (!s->restart_seen) {
1013 av_log(m->avctx, AV_LOG_ERROR,
1014 "No restart header present in substream %d.\n",
1015 substr);
1016 goto next_substr;
1017 }
1018
1019 if (read_block_data(m, &gb, substr) < 0)
1020 return -1;
1021
1022 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1023 && get_bits1(&gb) == 0);
1024
1025 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1026 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
1027 (show_bits_long(&gb, 32) == END_OF_STREAM ||
1028 show_bits_long(&gb, 20) == 0xd234e)) {
1029 skip_bits(&gb, 18);
1030 if (substr == m->max_decoded_substream)
1031 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1032
1033 if (get_bits1(&gb)) {
1034 int shorten_by = get_bits(&gb, 13);
1035 shorten_by = FFMIN(shorten_by, s->blockpos);
1036 s->blockpos -= shorten_by;
1037 } else
1038 skip_bits(&gb, 13);
1039 }
1040 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1041 substream_parity_present[substr]) {
1042 uint8_t parity, checksum;
1043
1044 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1045 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1046 av_log(m->avctx, AV_LOG_ERROR,
1047 "Substream %d parity check failed.\n", substr);
1048
1049 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
1050 if (checksum != get_bits(&gb, 8))
1051 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
1052 substr);
1053 }
1054 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1055 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
1056 substr);
1057 return -1;
1058 }
1059
1060 next_substr:
1061 buf += substream_data_len[substr];
1062 }
1063
1064 rematrix_channels(m, m->max_decoded_substream);
1065
1066 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1067 return -1;
1068
1069 return length;
1070
1071 error:
1072 m->params_valid = 0;
1073 return -1;
1074 }
1075
1076 #if CONFIG_MLP_DECODER
1077 AVCodec mlp_decoder = {
1078 "mlp",
1079 CODEC_TYPE_AUDIO,
1080 CODEC_ID_MLP,
1081 sizeof(MLPDecodeContext),
1082 mlp_decode_init,
1083 NULL,
1084 NULL,
1085 read_access_unit,
1086 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1087 };
1088 #endif /* CONFIG_MLP_DECODER */
1089
1090 #if CONFIG_TRUEHD_DECODER
1091 AVCodec truehd_decoder = {
1092 "truehd",
1093 CODEC_TYPE_AUDIO,
1094 CODEC_ID_TRUEHD,
1095 sizeof(MLPDecodeContext),
1096 mlp_decode_init,
1097 NULL,
1098 NULL,
1099 read_access_unit,
1100 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1101 };
1102 #endif /* CONFIG_TRUEHD_DECODER */