mlpdec: output_shift is signed
[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 /* TODO: validate */
632 }
633
634 if (s->param_presence_flags & PARAM_QUANTSTEP)
635 if (get_bits1(gbp))
636 for (ch = 0; ch <= s->max_channel; ch++) {
637 ChannelParams *cp = &m->channel_params[ch];
638
639 s->quant_step_size[ch] = get_bits(gbp, 4);
640 /* TODO: validate */
641
642 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch);
643 }
644
645 for (ch = s->min_channel; ch <= s->max_channel; ch++)
646 if (get_bits1(gbp)) {
647 if (read_channel_params(m, substr, gbp, ch) < 0)
648 return -1;
649 }
650
651 return 0;
652 }
653
654 #define MSB_MASK(bits) (-1u << bits)
655
656 /** Generate PCM samples using the prediction filters and residual values
657 * read from the data stream, and update the filter state. */
658
659 static void filter_channel(MLPDecodeContext *m, unsigned int substr,
660 unsigned int channel)
661 {
662 SubStream *s = &m->substream[substr];
663 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER];
664 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR],
665 &m->channel_params[channel].filter_params[IIR], };
666 unsigned int filter_shift = fp[FIR]->shift;
667 int32_t mask = MSB_MASK(s->quant_step_size[channel]);
668 int index = MAX_BLOCKSIZE;
669 int j, i;
670
671 for (j = 0; j < NUM_FILTERS; j++) {
672 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0],
673 MAX_FILTER_ORDER * sizeof(int32_t));
674 }
675
676 for (i = 0; i < s->blocksize; i++) {
677 int32_t residual = m->sample_buffer[i + s->blockpos][channel];
678 unsigned int order;
679 int64_t accum = 0;
680 int32_t result;
681
682 /* TODO: Move this code to DSPContext? */
683
684 for (j = 0; j < NUM_FILTERS; j++)
685 for (order = 0; order < fp[j]->order; order++)
686 accum += (int64_t)filter_state_buffer[j][index + order] *
687 fp[j]->coeff[order];
688
689 accum = accum >> filter_shift;
690 result = (accum + residual) & mask;
691
692 --index;
693
694 filter_state_buffer[FIR][index] = result;
695 filter_state_buffer[IIR][index] = result - accum;
696
697 m->sample_buffer[i + s->blockpos][channel] = result;
698 }
699
700 for (j = 0; j < NUM_FILTERS; j++) {
701 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index],
702 MAX_FILTER_ORDER * sizeof(int32_t));
703 }
704 }
705
706 /** Read a block of PCM residual data (or actual if no filtering active). */
707
708 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp,
709 unsigned int substr)
710 {
711 SubStream *s = &m->substream[substr];
712 unsigned int i, ch, expected_stream_pos = 0;
713
714 if (s->data_check_present) {
715 expected_stream_pos = get_bits_count(gbp);
716 expected_stream_pos += get_bits(gbp, 16);
717 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features "
718 "we have not tested yet. %s\n", sample_message);
719 }
720
721 if (s->blockpos + s->blocksize > m->access_unit_size) {
722 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n");
723 return -1;
724 }
725
726 memset(&m->bypassed_lsbs[s->blockpos][0], 0,
727 s->blocksize * sizeof(m->bypassed_lsbs[0]));
728
729 for (i = 0; i < s->blocksize; i++) {
730 if (read_huff_channels(m, gbp, substr, i) < 0)
731 return -1;
732 }
733
734 for (ch = s->min_channel; ch <= s->max_channel; ch++) {
735 filter_channel(m, substr, ch);
736 }
737
738 s->blockpos += s->blocksize;
739
740 if (s->data_check_present) {
741 if (get_bits_count(gbp) != expected_stream_pos)
742 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n");
743 skip_bits(gbp, 8);
744 }
745
746 return 0;
747 }
748
749 /** Data table used for TrueHD noise generation function. */
750
751 static const int8_t noise_table[256] = {
752 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2,
753 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62,
754 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5,
755 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40,
756 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34,
757 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30,
758 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36,
759 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69,
760 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24,
761 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20,
762 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23,
763 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8,
764 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40,
765 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37,
766 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52,
767 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70,
768 };
769
770 /** Noise generation functions.
771 * I'm not sure what these are for - they seem to be some kind of pseudorandom
772 * sequence generators, used to generate noise data which is used when the
773 * channels are rematrixed. I'm not sure if they provide a practical benefit
774 * to compression, or just obfuscate the decoder. Are they for some kind of
775 * dithering? */
776
777 /** Generate two channels of noise, used in the matrix when
778 * restart sync word == 0x31ea. */
779
780 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr)
781 {
782 SubStream *s = &m->substream[substr];
783 unsigned int i;
784 uint32_t seed = s->noisegen_seed;
785 unsigned int maxchan = s->max_matrix_channel;
786
787 for (i = 0; i < s->blockpos; i++) {
788 uint16_t seed_shr7 = seed >> 7;
789 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift;
790 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift;
791
792 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5);
793 }
794
795 s->noisegen_seed = seed;
796 }
797
798 /** Generate a block of noise, used when restart sync word == 0x31eb. */
799
800 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr)
801 {
802 SubStream *s = &m->substream[substr];
803 unsigned int i;
804 uint32_t seed = s->noisegen_seed;
805
806 for (i = 0; i < m->access_unit_size_pow2; i++) {
807 uint8_t seed_shr15 = seed >> 15;
808 m->noise_buffer[i] = noise_table[seed_shr15];
809 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5);
810 }
811
812 s->noisegen_seed = seed;
813 }
814
815
816 /** Apply the channel matrices in turn to reconstruct the original audio
817 * samples. */
818
819 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr)
820 {
821 SubStream *s = &m->substream[substr];
822 unsigned int mat, src_ch, i;
823 unsigned int maxchan;
824
825 maxchan = s->max_matrix_channel;
826 if (!s->noise_type) {
827 generate_2_noise_channels(m, substr);
828 maxchan += 2;
829 } else {
830 fill_noise_buffer(m, substr);
831 }
832
833 for (mat = 0; mat < s->num_primitive_matrices; mat++) {
834 int matrix_noise_shift = s->matrix_noise_shift[mat];
835 unsigned int dest_ch = s->matrix_out_ch[mat];
836 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]);
837
838 /* TODO: DSPContext? */
839
840 for (i = 0; i < s->blockpos; i++) {
841 int64_t accum = 0;
842 for (src_ch = 0; src_ch <= maxchan; src_ch++) {
843 accum += (int64_t)m->sample_buffer[i][src_ch]
844 * s->matrix_coeff[mat][src_ch];
845 }
846 if (matrix_noise_shift) {
847 uint32_t index = s->num_primitive_matrices - mat;
848 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1);
849 accum += m->noise_buffer[index] << (matrix_noise_shift + 7);
850 }
851 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask)
852 + m->bypassed_lsbs[i][mat];
853 }
854 }
855 }
856
857 /** Write the audio data into the output buffer. */
858
859 static int output_data_internal(MLPDecodeContext *m, unsigned int substr,
860 uint8_t *data, unsigned int *data_size, int is32)
861 {
862 SubStream *s = &m->substream[substr];
863 unsigned int i, out_ch = 0;
864 int32_t *data_32 = (int32_t*) data;
865 int16_t *data_16 = (int16_t*) data;
866
867 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2))
868 return -1;
869
870 for (i = 0; i < s->blockpos; i++) {
871 for (out_ch = 0; out_ch <= s->max_matrix_channel; out_ch++) {
872 int mat_ch = s->ch_assign[out_ch];
873 int32_t sample = m->sample_buffer[i][mat_ch]
874 << s->output_shift[mat_ch];
875 s->lossless_check_data ^= (sample & 0xffffff) << mat_ch;
876 if (is32) *data_32++ = sample << 8;
877 else *data_16++ = sample >> 8;
878 }
879 }
880
881 *data_size = i * out_ch * (is32 ? 4 : 2);
882
883 return 0;
884 }
885
886 static int output_data(MLPDecodeContext *m, unsigned int substr,
887 uint8_t *data, unsigned int *data_size)
888 {
889 if (m->avctx->sample_fmt == SAMPLE_FMT_S32)
890 return output_data_internal(m, substr, data, data_size, 1);
891 else
892 return output_data_internal(m, substr, data, data_size, 0);
893 }
894
895
896 /** Read an access unit from the stream.
897 * Returns < 0 on error, 0 if not enough data is present in the input stream
898 * otherwise returns the number of bytes consumed. */
899
900 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size,
901 const uint8_t *buf, int buf_size)
902 {
903 MLPDecodeContext *m = avctx->priv_data;
904 GetBitContext gb;
905 unsigned int length, substr;
906 unsigned int substream_start;
907 unsigned int header_size = 4;
908 unsigned int substr_header_size = 0;
909 uint8_t substream_parity_present[MAX_SUBSTREAMS];
910 uint16_t substream_data_len[MAX_SUBSTREAMS];
911 uint8_t parity_bits;
912
913 if (buf_size < 4)
914 return 0;
915
916 length = (AV_RB16(buf) & 0xfff) * 2;
917
918 if (length > buf_size)
919 return -1;
920
921 init_get_bits(&gb, (buf + 4), (length - 4) * 8);
922
923 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) {
924 dprintf(m->avctx, "Found major sync.\n");
925 if (read_major_sync(m, &gb) < 0)
926 goto error;
927 header_size += 28;
928 }
929
930 if (!m->params_valid) {
931 av_log(m->avctx, AV_LOG_WARNING,
932 "Stream parameters not seen; skipping frame.\n");
933 *data_size = 0;
934 return length;
935 }
936
937 substream_start = 0;
938
939 for (substr = 0; substr < m->num_substreams; substr++) {
940 int extraword_present, checkdata_present, end;
941
942 extraword_present = get_bits1(&gb);
943 skip_bits1(&gb);
944 checkdata_present = get_bits1(&gb);
945 skip_bits1(&gb);
946
947 end = get_bits(&gb, 12) * 2;
948
949 substr_header_size += 2;
950
951 if (extraword_present) {
952 skip_bits(&gb, 16);
953 substr_header_size += 2;
954 }
955
956 if (end + header_size + substr_header_size > length) {
957 av_log(m->avctx, AV_LOG_ERROR,
958 "Indicated length of substream %d data goes off end of "
959 "packet.\n", substr);
960 end = length - header_size - substr_header_size;
961 }
962
963 if (end < substream_start) {
964 av_log(avctx, AV_LOG_ERROR,
965 "Indicated end offset of substream %d data "
966 "is smaller than calculated start offset.\n",
967 substr);
968 goto error;
969 }
970
971 if (substr > m->max_decoded_substream)
972 continue;
973
974 substream_parity_present[substr] = checkdata_present;
975 substream_data_len[substr] = end - substream_start;
976 substream_start = end;
977 }
978
979 parity_bits = ff_mlp_calculate_parity(buf, 4);
980 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size);
981
982 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) {
983 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n");
984 goto error;
985 }
986
987 buf += header_size + substr_header_size;
988
989 for (substr = 0; substr <= m->max_decoded_substream; substr++) {
990 SubStream *s = &m->substream[substr];
991 init_get_bits(&gb, buf, substream_data_len[substr] * 8);
992
993 s->blockpos = 0;
994 do {
995 if (get_bits1(&gb)) {
996 if (get_bits1(&gb)) {
997 /* A restart header should be present. */
998 if (read_restart_header(m, &gb, buf, substr) < 0)
999 goto next_substr;
1000 s->restart_seen = 1;
1001 }
1002
1003 if (!s->restart_seen) {
1004 av_log(m->avctx, AV_LOG_ERROR,
1005 "No restart header present in substream %d.\n",
1006 substr);
1007 goto next_substr;
1008 }
1009
1010 if (read_decoding_params(m, &gb, substr) < 0)
1011 goto next_substr;
1012 }
1013
1014 if (!s->restart_seen) {
1015 av_log(m->avctx, AV_LOG_ERROR,
1016 "No restart header present in substream %d.\n",
1017 substr);
1018 goto next_substr;
1019 }
1020
1021 if (read_block_data(m, &gb, substr) < 0)
1022 return -1;
1023
1024 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8)
1025 && get_bits1(&gb) == 0);
1026
1027 skip_bits(&gb, (-get_bits_count(&gb)) & 15);
1028 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 &&
1029 (show_bits_long(&gb, 32) == END_OF_STREAM ||
1030 show_bits_long(&gb, 20) == 0xd234e)) {
1031 skip_bits(&gb, 18);
1032 if (substr == m->max_decoded_substream)
1033 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n");
1034
1035 if (get_bits1(&gb)) {
1036 int shorten_by = get_bits(&gb, 13);
1037 shorten_by = FFMIN(shorten_by, s->blockpos);
1038 s->blockpos -= shorten_by;
1039 } else
1040 skip_bits(&gb, 13);
1041 }
1042 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 &&
1043 substream_parity_present[substr]) {
1044 uint8_t parity, checksum;
1045
1046 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2);
1047 if ((parity ^ get_bits(&gb, 8)) != 0xa9)
1048 av_log(m->avctx, AV_LOG_ERROR,
1049 "Substream %d parity check failed.\n", substr);
1050
1051 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2);
1052 if (checksum != get_bits(&gb, 8))
1053 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n",
1054 substr);
1055 }
1056 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) {
1057 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n",
1058 substr);
1059 return -1;
1060 }
1061
1062 next_substr:
1063 buf += substream_data_len[substr];
1064 }
1065
1066 rematrix_channels(m, m->max_decoded_substream);
1067
1068 if (output_data(m, m->max_decoded_substream, data, data_size) < 0)
1069 return -1;
1070
1071 return length;
1072
1073 error:
1074 m->params_valid = 0;
1075 return -1;
1076 }
1077
1078 #if CONFIG_MLP_DECODER
1079 AVCodec mlp_decoder = {
1080 "mlp",
1081 CODEC_TYPE_AUDIO,
1082 CODEC_ID_MLP,
1083 sizeof(MLPDecodeContext),
1084 mlp_decode_init,
1085 NULL,
1086 NULL,
1087 read_access_unit,
1088 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"),
1089 };
1090 #endif /* CONFIG_MLP_DECODER */
1091
1092 #if CONFIG_TRUEHD_DECODER
1093 AVCodec truehd_decoder = {
1094 "truehd",
1095 CODEC_TYPE_AUDIO,
1096 CODEC_ID_TRUEHD,
1097 sizeof(MLPDecodeContext),
1098 mlp_decode_init,
1099 NULL,
1100 NULL,
1101 read_access_unit,
1102 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"),
1103 };
1104 #endif /* CONFIG_TRUEHD_DECODER */