Mercurial > libavcodec.hg
comparison mlpdec.c @ 7194:8427f12555a6 libavcodec
MLP/TrueHD decoder.
author | ramiro |
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date | Fri, 04 Jul 2008 15:44:13 +0000 |
parents | |
children | 479fc906650f |
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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 mlpdec.c | |
24 * MLP decoder | |
25 */ | |
26 | |
27 #include "avcodec.h" | |
28 #include "libavutil/intreadwrite.h" | |
29 #include "bitstream.h" | |
30 #include "libavutil/crc.h" | |
31 #include "parser.h" | |
32 #include "mlp_parser.h" | |
33 | |
34 /** Maximum number of channels that can be decoded. */ | |
35 #define MAX_CHANNELS 16 | |
36 | |
37 /** Maximum number of matrices used in decoding. Most streams have one matrix | |
38 * per output channel, but some rematrix a channel (usually 0) more than once. | |
39 */ | |
40 | |
41 #define MAX_MATRICES 15 | |
42 | |
43 /** Maximum number of substreams that can be decoded. This could also be set | |
44 * higher, but again I haven't seen any examples with more than two. */ | |
45 #define MAX_SUBSTREAMS 2 | |
46 | |
47 /** Maximum sample frequency seen in files. */ | |
48 #define MAX_SAMPLERATE 192000 | |
49 | |
50 /** The maximum number of audio samples within one access unit. */ | |
51 #define MAX_BLOCKSIZE (40 * (MAX_SAMPLERATE / 48000)) | |
52 /** The next power of two greater than MAX_BLOCKSIZE. */ | |
53 #define MAX_BLOCKSIZE_POW2 (64 * (MAX_SAMPLERATE / 48000)) | |
54 | |
55 /** Number of allowed filters. */ | |
56 #define NUM_FILTERS 2 | |
57 | |
58 /** The maximum number of taps in either the IIR or FIR filter. | |
59 * I believe MLP actually specifies the maximum order for IIR filters as four, | |
60 * and that the sum of the orders of both filters must be <= 8. */ | |
61 #define MAX_FILTER_ORDER 8 | |
62 | |
63 /** Number of bits used for VLC lookup - longest huffman code is 9. */ | |
64 #define VLC_BITS 9 | |
65 | |
66 | |
67 static const char* sample_message = | |
68 "Please file a bug report following the instructions at " | |
69 "http://ffmpeg.mplayerhq.hu/bugreports.html and include " | |
70 "a sample of this file."; | |
71 | |
72 typedef struct SubStream { | |
73 //! Set if a valid restart header has been read. Otherwise the substream can not be decoded. | |
74 uint8_t restart_seen; | |
75 | |
76 //@{ | |
77 /** restart header data */ | |
78 //! The type of noise to be used in the rematrix stage. | |
79 uint16_t noise_type; | |
80 | |
81 //! The index of the first channel coded in this substream. | |
82 uint8_t min_channel; | |
83 //! The index of the last channel coded in this substream. | |
84 uint8_t max_channel; | |
85 //! The number of channels input into the rematrix stage. | |
86 uint8_t max_matrix_channel; | |
87 | |
88 //! The left shift applied to random noise in 0x31ea substreams. | |
89 uint8_t noise_shift; | |
90 //! The current seed value for the pseudorandom noise generator(s). | |
91 uint32_t noisegen_seed; | |
92 | |
93 //! Set if the substream contains extra info to check the size of VLC blocks. | |
94 uint8_t data_check_present; | |
95 | |
96 //! Bitmask of which parameter sets are conveyed in a decoding parameter block. | |
97 uint8_t param_presence_flags; | |
98 #define PARAM_BLOCKSIZE (1 << 7) | |
99 #define PARAM_MATRIX (1 << 6) | |
100 #define PARAM_OUTSHIFT (1 << 5) | |
101 #define PARAM_QUANTSTEP (1 << 4) | |
102 #define PARAM_FIR (1 << 3) | |
103 #define PARAM_IIR (1 << 2) | |
104 #define PARAM_HUFFOFFSET (1 << 1) | |
105 //@} | |
106 | |
107 //@{ | |
108 /** matrix data */ | |
109 | |
110 //! Number of matrices to be applied. | |
111 uint8_t num_primitive_matrices; | |
112 | |
113 //! matrix output channel | |
114 uint8_t matrix_out_ch[MAX_MATRICES]; | |
115 | |
116 //! Whether the LSBs of the matrix output are encoded in the bitstream. | |
117 uint8_t lsb_bypass[MAX_MATRICES]; | |
118 //! Matrix coefficients, stored as 2.14 fixed point. | |
119 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2]; | |
120 //! Left shift to apply to noise values in 0x31eb substreams. | |
121 uint8_t matrix_noise_shift[MAX_MATRICES]; | |
122 //@} | |
123 | |
124 //! Left shift to apply to huffman-decoded residuals. | |
125 uint8_t quant_step_size[MAX_CHANNELS]; | |
126 | |
127 //! Number of PCM samples in current audio block. | |
128 uint16_t blocksize; | |
129 //! Number of PCM samples decoded so far in this frame. | |
130 uint16_t blockpos; | |
131 | |
132 //! Left shift to apply to decoded PCM values to get final 24-bit output. | |
133 int8_t output_shift[MAX_CHANNELS]; | |
134 | |
135 //! Running XOR of all output samples. | |
136 int32_t lossless_check_data; | |
137 | |
138 } SubStream; | |
139 | |
140 typedef struct MLPDecodeContext { | |
141 AVCodecContext *avctx; | |
142 | |
143 //! Set if a valid major sync block has been read. Otherwise no decoding is possible. | |
144 uint8_t params_valid; | |
145 | |
146 //! Number of substreams contained within this stream. | |
147 uint8_t num_substreams; | |
148 | |
149 //! Index of the last substream to decode - further substreams are skipped. | |
150 uint8_t max_decoded_substream; | |
151 | |
152 //! Number of PCM samples contained in each frame. | |
153 int access_unit_size; | |
154 //! Next power of two above the number of samples in each frame. | |
155 int access_unit_size_pow2; | |
156 | |
157 SubStream substream[MAX_SUBSTREAMS]; | |
158 | |
159 //@{ | |
160 /** filter data */ | |
161 #define FIR 0 | |
162 #define IIR 1 | |
163 //! Number of taps in filter. | |
164 uint8_t filter_order[MAX_CHANNELS][NUM_FILTERS]; | |
165 //! Right shift to apply to output of filter. | |
166 uint8_t filter_shift[MAX_CHANNELS][NUM_FILTERS]; | |
167 | |
168 int32_t filter_coeff[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER]; | |
169 int32_t filter_state[MAX_CHANNELS][NUM_FILTERS][MAX_FILTER_ORDER]; | |
170 //@} | |
171 | |
172 //@{ | |
173 /** sample data coding infomation */ | |
174 //! Offset to apply to residual values. | |
175 int16_t huff_offset[MAX_CHANNELS]; | |
176 //! Sign/rounding corrected version of huff_offset. | |
177 int32_t sign_huff_offset[MAX_CHANNELS]; | |
178 //! Which VLC codebook to use to read residuals. | |
179 uint8_t codebook[MAX_CHANNELS]; | |
180 //! Size of residual suffix not encoded using VLC. | |
181 uint8_t huff_lsbs[MAX_CHANNELS]; | |
182 //@} | |
183 | |
184 int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; | |
185 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; | |
186 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2]; | |
187 } MLPDecodeContext; | |
188 | |
189 /** Tables defining the huffman codes. | |
190 * There are three entropy coding methods used in MLP (four if you count | |
191 * "none" as a method). These use the same sequences for codes starting with | |
192 * 00 or 01, but have different codes starting with 1. */ | |
193 | |
194 static const uint8_t huffman_tables[3][18][2] = { | |
195 { /* huffman table 0, -7 - +10 */ | |
196 {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, | |
197 {0x04, 3}, {0x05, 3}, {0x06, 3}, {0x07, 3}, | |
198 {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, | |
199 }, { /* huffman table 1, -7 - +8 */ | |
200 {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, | |
201 {0x02, 2}, {0x03, 2}, | |
202 {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, | |
203 }, { /* huffman table 2, -7 - +7 */ | |
204 {0x01, 9}, {0x01, 8}, {0x01, 7}, {0x01, 6}, {0x01, 5}, {0x01, 4}, {0x01, 3}, | |
205 {0x01, 1}, | |
206 {0x03, 3}, {0x05, 4}, {0x09, 5}, {0x11, 6}, {0x21, 7}, {0x41, 8}, {0x81, 9}, | |
207 } | |
208 }; | |
209 | |
210 static VLC huff_vlc[3]; | |
211 | |
212 static int crc_init = 0; | |
213 static AVCRC crc_63[1024]; | |
214 static AVCRC crc_1D[1024]; | |
215 | |
216 | |
217 /** Initialize static data, constant between all invocations of the codec. */ | |
218 | |
219 static av_cold void init_static() | |
220 { | |
221 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, | |
222 &huffman_tables[0][0][1], 2, 1, | |
223 &huffman_tables[0][0][0], 2, 1, 512); | |
224 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, | |
225 &huffman_tables[1][0][1], 2, 1, | |
226 &huffman_tables[1][0][0], 2, 1, 512); | |
227 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, | |
228 &huffman_tables[2][0][1], 2, 1, | |
229 &huffman_tables[2][0][0], 2, 1, 512); | |
230 | |
231 if (!crc_init) { | |
232 av_crc_init(crc_63, 0, 8, 0x63, sizeof(crc_63)); | |
233 av_crc_init(crc_1D, 0, 8, 0x1D, sizeof(crc_1D)); | |
234 crc_init = 1; | |
235 } | |
236 } | |
237 | |
238 | |
239 /** MLP uses checksums that seem to be based on the standard CRC algorithm, | |
240 * but not (in implementation terms, the table lookup and XOR are reversed). | |
241 * We can implement this behavior using a standard av_crc on all but the | |
242 * last element, then XOR that with the last element. */ | |
243 | |
244 static uint8_t mlp_checksum8(const uint8_t *buf, unsigned int buf_size) | |
245 { | |
246 uint8_t checksum = av_crc(crc_63, 0x3c, buf, buf_size - 1); // crc_63[0xa2] == 0x3c | |
247 checksum ^= buf[buf_size-1]; | |
248 return checksum; | |
249 } | |
250 | |
251 /** Calculate an 8-bit checksum over a restart header -- a non-multiple-of-8 | |
252 * number of bits, starting two bits into the first byte of buf. */ | |
253 | |
254 static uint8_t mlp_restart_checksum(const uint8_t *buf, unsigned int bit_size) | |
255 { | |
256 int i; | |
257 int num_bytes = (bit_size + 2) / 8; | |
258 | |
259 int crc = crc_1D[buf[0] & 0x3f]; | |
260 crc = av_crc(crc_1D, crc, buf + 1, num_bytes - 2); | |
261 crc ^= buf[num_bytes - 1]; | |
262 | |
263 for (i = 0; i < ((bit_size + 2) & 7); i++) { | |
264 crc <<= 1; | |
265 if (crc & 0x100) | |
266 crc ^= 0x11D; | |
267 crc ^= (buf[num_bytes] >> (7 - i)) & 1; | |
268 } | |
269 | |
270 return crc; | |
271 } | |
272 | |
273 static inline int32_t calculate_sign_huff(MLPDecodeContext *m, | |
274 unsigned int substr, unsigned int ch) | |
275 { | |
276 SubStream *s = &m->substream[substr]; | |
277 int lsb_bits = m->huff_lsbs[ch] - s->quant_step_size[ch]; | |
278 int sign_shift = lsb_bits + (m->codebook[ch] ? 2 - m->codebook[ch] : -1); | |
279 int32_t sign_huff_offset = m->huff_offset[ch]; | |
280 | |
281 if (m->codebook[ch] > 0) | |
282 sign_huff_offset -= 7 << lsb_bits; | |
283 | |
284 if (sign_shift >= 0) | |
285 sign_huff_offset -= 1 << sign_shift; | |
286 | |
287 return sign_huff_offset; | |
288 } | |
289 | |
290 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs | |
291 * and plain LSBs. */ | |
292 | |
293 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, | |
294 unsigned int substr, unsigned int pos) | |
295 { | |
296 SubStream *s = &m->substream[substr]; | |
297 unsigned int mat, channel; | |
298 | |
299 for (mat = 0; mat < s->num_primitive_matrices; mat++) | |
300 if (s->lsb_bypass[mat]) | |
301 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp); | |
302 | |
303 for (channel = s->min_channel; channel <= s->max_channel; channel++) { | |
304 int codebook = m->codebook[channel]; | |
305 int quant_step_size = s->quant_step_size[channel]; | |
306 int lsb_bits = m->huff_lsbs[channel] - quant_step_size; | |
307 int result = 0; | |
308 | |
309 if (codebook > 0) | |
310 result = get_vlc2(gbp, huff_vlc[codebook-1].table, | |
311 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); | |
312 | |
313 if (result < 0) | |
314 return -1; | |
315 | |
316 if (lsb_bits > 0) | |
317 result = (result << lsb_bits) + get_bits(gbp, lsb_bits); | |
318 | |
319 result += m->sign_huff_offset[channel]; | |
320 result <<= quant_step_size; | |
321 | |
322 m->sample_buffer[pos + s->blockpos][channel] = result; | |
323 } | |
324 | |
325 return 0; | |
326 } | |
327 | |
328 static av_cold int mlp_decode_init(AVCodecContext *avctx) | |
329 { | |
330 MLPDecodeContext *m = avctx->priv_data; | |
331 int substr; | |
332 | |
333 init_static(); | |
334 m->avctx = avctx; | |
335 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
336 m->substream[substr].lossless_check_data = 0xffffffff; | |
337 return 0; | |
338 } | |
339 | |
340 /** Read a major sync info header - contains high level information about | |
341 * the stream - sample rate, channel arrangement etc. Most of this | |
342 * information is not actually necessary for decoding, only for playback. | |
343 */ | |
344 | |
345 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb) | |
346 { | |
347 MLPHeaderInfo mh; | |
348 int substr; | |
349 | |
350 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0) | |
351 return -1; | |
352 | |
353 if (mh.group1_bits == 0) { | |
354 av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown bits per sample\n"); | |
355 return -1; | |
356 } | |
357 if (mh.group2_bits > mh.group1_bits) { | |
358 av_log(m->avctx, AV_LOG_ERROR, | |
359 "Channel group 2 cannot have more bits per sample than group 1\n"); | |
360 return -1; | |
361 } | |
362 | |
363 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) { | |
364 av_log(m->avctx, AV_LOG_ERROR, | |
365 "Channel groups with differing sample rates not currently supported\n"); | |
366 return -1; | |
367 } | |
368 | |
369 if (mh.group1_samplerate == 0) { | |
370 av_log(m->avctx, AV_LOG_ERROR, "Invalid/unknown sampling rate\n"); | |
371 return -1; | |
372 } | |
373 if (mh.group1_samplerate > MAX_SAMPLERATE) { | |
374 av_log(m->avctx, AV_LOG_ERROR, | |
375 "Sampling rate %d is greater than maximum supported (%d)\n", | |
376 mh.group1_samplerate, MAX_SAMPLERATE); | |
377 return -1; | |
378 } | |
379 if (mh.access_unit_size > MAX_BLOCKSIZE) { | |
380 av_log(m->avctx, AV_LOG_ERROR, | |
381 "Block size %d is greater than maximum supported (%d)\n", | |
382 mh.access_unit_size, MAX_BLOCKSIZE); | |
383 return -1; | |
384 } | |
385 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) { | |
386 av_log(m->avctx, AV_LOG_ERROR, | |
387 "Block size pow2 %d is greater than maximum supported (%d)\n", | |
388 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2); | |
389 return -1; | |
390 } | |
391 | |
392 if (mh.num_substreams == 0) | |
393 return -1; | |
394 if (mh.num_substreams > MAX_SUBSTREAMS) { | |
395 av_log(m->avctx, AV_LOG_ERROR, | |
396 "Number of substreams %d is more than maximum supported by " | |
397 "decoder. %s\n", mh.num_substreams, sample_message); | |
398 return -1; | |
399 } | |
400 | |
401 m->access_unit_size = mh.access_unit_size; | |
402 m->access_unit_size_pow2 = mh.access_unit_size_pow2; | |
403 | |
404 m->num_substreams = mh.num_substreams; | |
405 m->max_decoded_substream = m->num_substreams - 1; | |
406 | |
407 m->avctx->sample_rate = mh.group1_samplerate; | |
408 m->avctx->frame_size = mh.access_unit_size; | |
409 | |
410 #ifdef CONFIG_AUDIO_NONSHORT | |
411 m->avctx->bits_per_sample = mh.group1_bits; | |
412 if (mh.group1_bits > 16) { | |
413 m->avctx->sample_fmt = SAMPLE_FMT_S32; | |
414 } | |
415 #endif | |
416 | |
417 m->params_valid = 1; | |
418 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
419 m->substream[substr].restart_seen = 0; | |
420 | |
421 return 0; | |
422 } | |
423 | |
424 /** Read a restart header from a block in a substream. This contains parameters | |
425 * required to decode the audio that do not change very often. Generally | |
426 * (always) present only in blocks following a major sync. */ | |
427 | |
428 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, | |
429 const uint8_t *buf, unsigned int substr) | |
430 { | |
431 SubStream *s = &m->substream[substr]; | |
432 unsigned int ch; | |
433 int sync_word, tmp; | |
434 uint8_t checksum; | |
435 uint8_t lossless_check; | |
436 int start_count = get_bits_count(gbp); | |
437 | |
438 sync_word = get_bits(gbp, 13); | |
439 | |
440 if (sync_word != 0x31ea >> 1) { | |
441 av_log(m->avctx, AV_LOG_ERROR, | |
442 "Restart header sync incorrect (got 0x%04x)\n", sync_word); | |
443 return -1; | |
444 } | |
445 s->noise_type = get_bits1(gbp); | |
446 | |
447 skip_bits(gbp, 16); /* Output timestamp */ | |
448 | |
449 s->min_channel = get_bits(gbp, 4); | |
450 s->max_channel = get_bits(gbp, 4); | |
451 s->max_matrix_channel = get_bits(gbp, 4); | |
452 | |
453 if (s->min_channel > s->max_channel) { | |
454 av_log(m->avctx, AV_LOG_ERROR, | |
455 "Substream min channel cannot be greater than max channel.\n"); | |
456 return -1; | |
457 } | |
458 | |
459 if (m->avctx->request_channels > 0 | |
460 && s->max_channel + 1 >= m->avctx->request_channels | |
461 && substr < m->max_decoded_substream) { | |
462 av_log(m->avctx, AV_LOG_INFO, | |
463 "Extracting %d channel downmix from substream %d. " | |
464 "Further substreams will be skipped.\n", | |
465 s->max_channel + 1, substr); | |
466 m->max_decoded_substream = substr; | |
467 } | |
468 | |
469 s->noise_shift = get_bits(gbp, 4); | |
470 s->noisegen_seed = get_bits(gbp, 23); | |
471 | |
472 skip_bits(gbp, 19); | |
473 | |
474 s->data_check_present = get_bits1(gbp); | |
475 lossless_check = get_bits(gbp, 8); | |
476 if (substr == m->max_decoded_substream | |
477 && s->lossless_check_data != 0xffffffff) { | |
478 tmp = s->lossless_check_data; | |
479 tmp ^= tmp >> 16; | |
480 tmp ^= tmp >> 8; | |
481 tmp &= 0xff; | |
482 if (tmp != lossless_check) | |
483 av_log(m->avctx, AV_LOG_WARNING, | |
484 "Lossless check failed - expected %02x, calculated %02x\n", | |
485 lossless_check, tmp); | |
486 else | |
487 dprintf(m->avctx, "Lossless check passed for substream %d (%x)\n", | |
488 substr, tmp); | |
489 } | |
490 | |
491 skip_bits(gbp, 16); | |
492 | |
493 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
494 int ch_assign = get_bits(gbp, 6); | |
495 dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch, | |
496 ch_assign); | |
497 if (ch_assign != ch) { | |
498 av_log(m->avctx, AV_LOG_ERROR, | |
499 "Non 1:1 channel assignments are used in this stream. %s\n", | |
500 sample_message); | |
501 return -1; | |
502 } | |
503 } | |
504 | |
505 checksum = mlp_restart_checksum(buf, get_bits_count(gbp) - start_count); | |
506 | |
507 if (checksum != get_bits(gbp, 8)) | |
508 av_log(m->avctx, AV_LOG_ERROR, "Restart header checksum error\n"); | |
509 | |
510 /* Set default decoding parameters */ | |
511 s->param_presence_flags = 0xff; | |
512 s->num_primitive_matrices = 0; | |
513 s->blocksize = 8; | |
514 s->lossless_check_data = 0; | |
515 | |
516 memset(s->output_shift , 0, sizeof(s->output_shift )); | |
517 memset(s->quant_step_size, 0, sizeof(s->quant_step_size)); | |
518 | |
519 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
520 m->filter_order[ch][FIR] = 0; | |
521 m->filter_order[ch][IIR] = 0; | |
522 m->filter_shift[ch][FIR] = 0; | |
523 m->filter_shift[ch][IIR] = 0; | |
524 | |
525 /* Default audio coding is 24-bit raw PCM */ | |
526 m->huff_offset [ch] = 0; | |
527 m->sign_huff_offset[ch] = (-1) << 23; | |
528 m->codebook [ch] = 0; | |
529 m->huff_lsbs [ch] = 24; | |
530 } | |
531 | |
532 if (substr == m->max_decoded_substream) { | |
533 m->avctx->channels = s->max_channel + 1; | |
534 } | |
535 | |
536 return 0; | |
537 } | |
538 | |
539 /** Read parameters for one of the prediction filters. */ | |
540 | |
541 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, | |
542 unsigned int channel, unsigned int filter) | |
543 { | |
544 const char fchar = filter ? 'I' : 'F'; | |
545 int i, order; | |
546 | |
547 // filter is 0 for FIR, 1 for IIR | |
548 assert(filter < 2); | |
549 | |
550 order = get_bits(gbp, 4); | |
551 if (order > MAX_FILTER_ORDER) { | |
552 av_log(m->avctx, AV_LOG_ERROR, | |
553 "%cIR filter order %d is greater than maximum %d\n", | |
554 fchar, order, MAX_FILTER_ORDER); | |
555 return -1; | |
556 } | |
557 m->filter_order[channel][filter] = order; | |
558 | |
559 if (order > 0) { | |
560 int coeff_bits, coeff_shift; | |
561 | |
562 m->filter_shift[channel][filter] = get_bits(gbp, 4); | |
563 | |
564 coeff_bits = get_bits(gbp, 5); | |
565 coeff_shift = get_bits(gbp, 3); | |
566 if (coeff_bits < 1 || coeff_bits > 16) { | |
567 av_log(m->avctx, AV_LOG_ERROR, | |
568 "%cIR filter coeff_bits must be between 1 and 16\n", | |
569 fchar); | |
570 return -1; | |
571 } | |
572 if (coeff_bits + coeff_shift > 16) { | |
573 av_log(m->avctx, AV_LOG_ERROR, | |
574 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less\n", | |
575 fchar); | |
576 return -1; | |
577 } | |
578 | |
579 for (i = 0; i < order; i++) | |
580 m->filter_coeff[channel][filter][i] = | |
581 get_sbits(gbp, coeff_bits) << coeff_shift; | |
582 | |
583 if (get_bits1(gbp)) { | |
584 int state_bits, state_shift; | |
585 | |
586 if (filter == FIR) { | |
587 av_log(m->avctx, AV_LOG_ERROR, | |
588 "FIR filter has state data specified\n"); | |
589 return -1; | |
590 } | |
591 | |
592 state_bits = get_bits(gbp, 4); | |
593 state_shift = get_bits(gbp, 4); | |
594 | |
595 /* TODO: check validity of state data */ | |
596 | |
597 for (i = 0; i < order; i++) | |
598 m->filter_state[channel][filter][i] = | |
599 get_sbits(gbp, state_bits) << state_shift; | |
600 } | |
601 } | |
602 | |
603 return 0; | |
604 } | |
605 | |
606 /** Read decoding parameters that change more often than those in the restart | |
607 * header. */ | |
608 | |
609 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, | |
610 unsigned int substr) | |
611 { | |
612 SubStream *s = &m->substream[substr]; | |
613 unsigned int mat, ch; | |
614 | |
615 if (get_bits1(gbp)) | |
616 s->param_presence_flags = get_bits(gbp, 8); | |
617 | |
618 if (s->param_presence_flags & PARAM_BLOCKSIZE) | |
619 if (get_bits1(gbp)) { | |
620 s->blocksize = get_bits(gbp, 9); | |
621 if (s->blocksize > MAX_BLOCKSIZE) { | |
622 av_log(m->avctx, AV_LOG_ERROR, "Block size too large\n"); | |
623 s->blocksize = 0; | |
624 return -1; | |
625 } | |
626 } | |
627 | |
628 if (s->param_presence_flags & PARAM_MATRIX) | |
629 if (get_bits1(gbp)) { | |
630 s->num_primitive_matrices = get_bits(gbp, 4); | |
631 | |
632 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
633 int frac_bits, max_chan; | |
634 s->matrix_out_ch[mat] = get_bits(gbp, 4); | |
635 frac_bits = get_bits(gbp, 4); | |
636 s->lsb_bypass [mat] = get_bits1(gbp); | |
637 | |
638 if (s->matrix_out_ch[mat] > s->max_channel) { | |
639 av_log(m->avctx, AV_LOG_ERROR, | |
640 "Invalid channel %d specified as output from matrix\n", | |
641 s->matrix_out_ch[mat]); | |
642 return -1; | |
643 } | |
644 if (frac_bits > 14) { | |
645 av_log(m->avctx, AV_LOG_ERROR, | |
646 "Too many fractional bits specified\n"); | |
647 return -1; | |
648 } | |
649 | |
650 max_chan = s->max_matrix_channel; | |
651 if (!s->noise_type) | |
652 max_chan+=2; | |
653 | |
654 for (ch = 0; ch <= max_chan; ch++) { | |
655 int coeff_val = 0; | |
656 if (get_bits1(gbp)) | |
657 coeff_val = get_sbits(gbp, frac_bits + 2); | |
658 | |
659 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); | |
660 } | |
661 | |
662 if (s->noise_type) | |
663 s->matrix_noise_shift[mat] = get_bits(gbp, 4); | |
664 else | |
665 s->matrix_noise_shift[mat] = 0; | |
666 } | |
667 } | |
668 | |
669 if (s->param_presence_flags & PARAM_OUTSHIFT) | |
670 if (get_bits1(gbp)) | |
671 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
672 s->output_shift[ch] = get_bits(gbp, 4); | |
673 dprintf(m->avctx, "output shift[%d] = %d\n", | |
674 ch, s->output_shift[ch]); | |
675 /* TODO: validate */ | |
676 } | |
677 | |
678 if (s->param_presence_flags & PARAM_QUANTSTEP) | |
679 if (get_bits1(gbp)) | |
680 for (ch = 0; ch <= s->max_channel; ch++) { | |
681 s->quant_step_size[ch] = get_bits(gbp, 4); | |
682 /* TODO: validate */ | |
683 | |
684 m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch); | |
685 } | |
686 | |
687 for (ch = s->min_channel; ch <= s->max_channel; ch++) | |
688 if (get_bits1(gbp)) { | |
689 if (s->param_presence_flags & PARAM_FIR) | |
690 if (get_bits1(gbp)) | |
691 if (read_filter_params(m, gbp, ch, FIR) < 0) | |
692 return -1; | |
693 | |
694 if (s->param_presence_flags & PARAM_IIR) | |
695 if (get_bits1(gbp)) | |
696 if (read_filter_params(m, gbp, ch, IIR) < 0) | |
697 return -1; | |
698 | |
699 if (m->filter_order[ch][FIR] && m->filter_order[ch][IIR] && | |
700 m->filter_shift[ch][FIR] != m->filter_shift[ch][IIR]) { | |
701 av_log(m->avctx, AV_LOG_ERROR, | |
702 "FIR and IIR filters must use same precision\n"); | |
703 return -1; | |
704 } | |
705 /* The FIR and IIR filters must have the same precision. | |
706 * To simplify the filtering code, only the precision of the | |
707 * FIR filter is considered. If only the IIR filter is employed, | |
708 * the FIR filter precision is set to that of the IIR filter, so | |
709 * that the filtering code can use it. */ | |
710 if (!m->filter_order[ch][FIR] && m->filter_order[ch][IIR]) | |
711 m->filter_shift[ch][FIR] = m->filter_shift[ch][IIR]; | |
712 | |
713 if (s->param_presence_flags & PARAM_HUFFOFFSET) | |
714 if (get_bits1(gbp)) | |
715 m->huff_offset[ch] = get_sbits(gbp, 15); | |
716 | |
717 m->codebook [ch] = get_bits(gbp, 2); | |
718 m->huff_lsbs[ch] = get_bits(gbp, 5); | |
719 | |
720 m->sign_huff_offset[ch] = calculate_sign_huff(m, substr, ch); | |
721 | |
722 /* TODO: validate */ | |
723 } | |
724 | |
725 return 0; | |
726 } | |
727 | |
728 #define MSB_MASK(bits) (-1u << bits) | |
729 | |
730 /** Generate PCM samples using the prediction filters and residual values | |
731 * read from the data stream, and update the filter state. */ | |
732 | |
733 static void filter_channel(MLPDecodeContext *m, unsigned int substr, | |
734 unsigned int channel) | |
735 { | |
736 SubStream *s = &m->substream[substr]; | |
737 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER]; | |
738 unsigned int filter_shift = m->filter_shift[channel][FIR]; | |
739 int32_t mask = MSB_MASK(s->quant_step_size[channel]); | |
740 int index = MAX_BLOCKSIZE; | |
741 int j, i; | |
742 | |
743 for (j = 0; j < NUM_FILTERS; j++) { | |
744 memcpy(& filter_state_buffer [j][MAX_BLOCKSIZE], | |
745 &m->filter_state[channel][j][0], | |
746 MAX_FILTER_ORDER * sizeof(int32_t)); | |
747 } | |
748 | |
749 for (i = 0; i < s->blocksize; i++) { | |
750 int32_t residual = m->sample_buffer[i + s->blockpos][channel]; | |
751 unsigned int order; | |
752 int64_t accum = 0; | |
753 int32_t result; | |
754 | |
755 /* TODO: Move this code to DSPContext? */ | |
756 | |
757 for (j = 0; j < NUM_FILTERS; j++) | |
758 for (order = 0; order < m->filter_order[channel][j]; order++) | |
759 accum += (int64_t)filter_state_buffer[j][index + order] * | |
760 m->filter_coeff[channel][j][order]; | |
761 | |
762 accum = accum >> filter_shift; | |
763 result = (accum + residual) & mask; | |
764 | |
765 --index; | |
766 | |
767 filter_state_buffer[FIR][index] = result; | |
768 filter_state_buffer[IIR][index] = result - accum; | |
769 | |
770 m->sample_buffer[i + s->blockpos][channel] = result; | |
771 } | |
772 | |
773 for (j = 0; j < NUM_FILTERS; j++) { | |
774 memcpy(&m->filter_state[channel][j][0], | |
775 & filter_state_buffer [j][index], | |
776 MAX_FILTER_ORDER * sizeof(int32_t)); | |
777 } | |
778 } | |
779 | |
780 /** Read a block of PCM residual data (or actual if no filtering active). */ | |
781 | |
782 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, | |
783 unsigned int substr) | |
784 { | |
785 SubStream *s = &m->substream[substr]; | |
786 unsigned int i, ch, expected_stream_pos = 0; | |
787 | |
788 if (s->data_check_present) { | |
789 expected_stream_pos = get_bits_count(gbp); | |
790 expected_stream_pos += get_bits(gbp, 16); | |
791 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features " | |
792 "we have not tested yet. %s\n", sample_message); | |
793 } | |
794 | |
795 if (s->blockpos + s->blocksize > m->access_unit_size) { | |
796 av_log(m->avctx, AV_LOG_ERROR, "Too many audio samples in frame\n"); | |
797 return -1; | |
798 } | |
799 | |
800 memset(&m->bypassed_lsbs[s->blockpos][0], 0, | |
801 s->blocksize * sizeof(m->bypassed_lsbs[0])); | |
802 | |
803 for (i = 0; i < s->blocksize; i++) { | |
804 if (read_huff_channels(m, gbp, substr, i) < 0) | |
805 return -1; | |
806 } | |
807 | |
808 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
809 filter_channel(m, substr, ch); | |
810 } | |
811 | |
812 s->blockpos += s->blocksize; | |
813 | |
814 if (s->data_check_present) { | |
815 if (get_bits_count(gbp) != expected_stream_pos) | |
816 av_log(m->avctx, AV_LOG_ERROR, "Block data length mismatch\n"); | |
817 skip_bits(gbp, 8); | |
818 } | |
819 | |
820 return 0; | |
821 } | |
822 | |
823 /** Data table used for TrueHD noise generation function */ | |
824 | |
825 static const int8_t noise_table[256] = { | |
826 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2, | |
827 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62, | |
828 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5, | |
829 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40, | |
830 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34, | |
831 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30, | |
832 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36, | |
833 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69, | |
834 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24, | |
835 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20, | |
836 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23, | |
837 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8, | |
838 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40, | |
839 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37, | |
840 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52, | |
841 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70, | |
842 }; | |
843 | |
844 /** Noise generation functions. | |
845 * I'm not sure what these are for - they seem to be some kind of pseudorandom | |
846 * sequence generators, used to generate noise data which is used when the | |
847 * channels are rematrixed. I'm not sure if they provide a practical benefit | |
848 * to compression, or just obfuscate the decoder. Are they for some kind of | |
849 * dithering? */ | |
850 | |
851 /** Generate two channels of noise, used in the matrix when | |
852 * restart sync word == 0x31ea. */ | |
853 | |
854 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr) | |
855 { | |
856 SubStream *s = &m->substream[substr]; | |
857 unsigned int i; | |
858 uint32_t seed = s->noisegen_seed; | |
859 unsigned int maxchan = s->max_matrix_channel; | |
860 | |
861 for (i = 0; i < s->blockpos; i++) { | |
862 uint16_t seed_shr7 = seed >> 7; | |
863 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift; | |
864 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift; | |
865 | |
866 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5); | |
867 } | |
868 | |
869 s->noisegen_seed = seed; | |
870 } | |
871 | |
872 /** Generate a block of noise, used when restart sync word == 0x31eb. */ | |
873 | |
874 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr) | |
875 { | |
876 SubStream *s = &m->substream[substr]; | |
877 unsigned int i; | |
878 uint32_t seed = s->noisegen_seed; | |
879 | |
880 for (i = 0; i < m->access_unit_size_pow2; i++) { | |
881 uint8_t seed_shr15 = seed >> 15; | |
882 m->noise_buffer[i] = noise_table[seed_shr15]; | |
883 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5); | |
884 } | |
885 | |
886 s->noisegen_seed = seed; | |
887 } | |
888 | |
889 | |
890 /** Apply the channel matrices in turn to reconstruct the original audio | |
891 * samples. */ | |
892 | |
893 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr) | |
894 { | |
895 SubStream *s = &m->substream[substr]; | |
896 unsigned int mat, src_ch, i; | |
897 unsigned int maxchan; | |
898 | |
899 maxchan = s->max_matrix_channel; | |
900 if (!s->noise_type) { | |
901 generate_2_noise_channels(m, substr); | |
902 maxchan += 2; | |
903 } else { | |
904 fill_noise_buffer(m, substr); | |
905 } | |
906 | |
907 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
908 int matrix_noise_shift = s->matrix_noise_shift[mat]; | |
909 unsigned int dest_ch = s->matrix_out_ch[mat]; | |
910 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]); | |
911 | |
912 /* TODO: DSPContext? */ | |
913 | |
914 for (i = 0; i < s->blockpos; i++) { | |
915 int64_t accum = 0; | |
916 for (src_ch = 0; src_ch <= maxchan; src_ch++) { | |
917 accum += (int64_t)m->sample_buffer[i][src_ch] | |
918 * s->matrix_coeff[mat][src_ch]; | |
919 } | |
920 if (matrix_noise_shift) { | |
921 uint32_t index = s->num_primitive_matrices - mat; | |
922 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1); | |
923 accum += m->noise_buffer[index] << (matrix_noise_shift + 7); | |
924 } | |
925 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask) | |
926 + m->bypassed_lsbs[i][mat]; | |
927 } | |
928 } | |
929 } | |
930 | |
931 /** Write the audio data into the output buffer. */ | |
932 | |
933 static int output_data_internal(MLPDecodeContext *m, unsigned int substr, | |
934 uint8_t *data, unsigned int *data_size, int is32) | |
935 { | |
936 SubStream *s = &m->substream[substr]; | |
937 unsigned int i, ch = 0; | |
938 int32_t *data_32 = (int32_t*) data; | |
939 int16_t *data_16 = (int16_t*) data; | |
940 | |
941 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2)) | |
942 return -1; | |
943 | |
944 for (i = 0; i < s->blockpos; i++) { | |
945 for (ch = 0; ch <= s->max_channel; ch++) { | |
946 int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch]; | |
947 s->lossless_check_data ^= (sample & 0xffffff) << ch; | |
948 if (is32) *data_32++ = sample << 8; | |
949 else *data_16++ = sample >> 8; | |
950 } | |
951 } | |
952 | |
953 *data_size = i * ch * (is32 ? 4 : 2); | |
954 | |
955 return 0; | |
956 } | |
957 | |
958 static int output_data(MLPDecodeContext *m, unsigned int substr, | |
959 uint8_t *data, unsigned int *data_size) | |
960 { | |
961 if (m->avctx->sample_fmt == SAMPLE_FMT_S32) | |
962 return output_data_internal(m, substr, data, data_size, 1); | |
963 else | |
964 return output_data_internal(m, substr, data, data_size, 0); | |
965 } | |
966 | |
967 | |
968 /** XOR together all the bytes of a buffer. | |
969 * Does this belong in dspcontext? */ | |
970 | |
971 static uint8_t calculate_parity(const uint8_t *buf, unsigned int buf_size) | |
972 { | |
973 uint32_t scratch = 0; | |
974 const uint8_t *buf_end = buf + buf_size; | |
975 | |
976 for (; buf < buf_end - 3; buf += 4) | |
977 scratch ^= *((const uint32_t*)buf); | |
978 | |
979 scratch ^= scratch >> 16; | |
980 scratch ^= scratch >> 8; | |
981 | |
982 for (; buf < buf_end; buf++) | |
983 scratch ^= *buf; | |
984 | |
985 return scratch; | |
986 } | |
987 | |
988 /** Read an access unit from the stream. | |
989 * Returns < 0 on error, 0 if not enough data is present in the input stream | |
990 * otherwise returns the number of bytes consumed. */ | |
991 | |
992 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, | |
993 const uint8_t *buf, int buf_size) | |
994 { | |
995 MLPDecodeContext *m = avctx->priv_data; | |
996 GetBitContext gb; | |
997 unsigned int length, substr; | |
998 unsigned int substream_start; | |
999 unsigned int header_size = 4; | |
1000 unsigned int substr_header_size = 0; | |
1001 uint8_t substream_parity_present[MAX_SUBSTREAMS]; | |
1002 uint16_t substream_data_len[MAX_SUBSTREAMS]; | |
1003 uint8_t parity_bits; | |
1004 | |
1005 if (buf_size < 4) | |
1006 return 0; | |
1007 | |
1008 length = (AV_RB16(buf) & 0xfff) * 2; | |
1009 | |
1010 if (length > buf_size) | |
1011 return -1; | |
1012 | |
1013 init_get_bits(&gb, (buf + 4), (length - 4) * 8); | |
1014 | |
1015 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) { | |
1016 dprintf(m->avctx, "Found major sync\n"); | |
1017 if (read_major_sync(m, &gb) < 0) | |
1018 goto error; | |
1019 header_size += 28; | |
1020 } | |
1021 | |
1022 if (!m->params_valid) { | |
1023 av_log(m->avctx, AV_LOG_WARNING, | |
1024 "Stream parameters not seen; skipping frame\n"); | |
1025 *data_size = 0; | |
1026 return length; | |
1027 } | |
1028 | |
1029 substream_start = 0; | |
1030 | |
1031 for (substr = 0; substr < m->num_substreams; substr++) { | |
1032 int extraword_present, checkdata_present, end; | |
1033 | |
1034 extraword_present = get_bits1(&gb); | |
1035 skip_bits1(&gb); | |
1036 checkdata_present = get_bits1(&gb); | |
1037 skip_bits1(&gb); | |
1038 | |
1039 end = get_bits(&gb, 12) * 2; | |
1040 | |
1041 substr_header_size += 2; | |
1042 | |
1043 if (extraword_present) { | |
1044 skip_bits(&gb, 16); | |
1045 substr_header_size += 2; | |
1046 } | |
1047 | |
1048 if (end + header_size + substr_header_size > length) { | |
1049 av_log(m->avctx, AV_LOG_ERROR, | |
1050 "Indicated length of substream %d data goes off end of " | |
1051 "packet.\n", substr); | |
1052 end = length - header_size - substr_header_size; | |
1053 } | |
1054 | |
1055 if (end < substream_start) { | |
1056 av_log(avctx, AV_LOG_ERROR, | |
1057 "Indicated end offset of substream %d data " | |
1058 "is smaller than calculated start offset.\n", | |
1059 substr); | |
1060 goto error; | |
1061 } | |
1062 | |
1063 if (substr > m->max_decoded_substream) | |
1064 continue; | |
1065 | |
1066 substream_parity_present[substr] = checkdata_present; | |
1067 substream_data_len[substr] = end - substream_start; | |
1068 substream_start = end; | |
1069 } | |
1070 | |
1071 parity_bits = calculate_parity(buf, 4); | |
1072 parity_bits ^= calculate_parity(buf + header_size, substr_header_size); | |
1073 | |
1074 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) { | |
1075 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n"); | |
1076 goto error; | |
1077 } | |
1078 | |
1079 buf += header_size + substr_header_size; | |
1080 | |
1081 for (substr = 0; substr <= m->max_decoded_substream; substr++) { | |
1082 SubStream *s = &m->substream[substr]; | |
1083 init_get_bits(&gb, buf, substream_data_len[substr] * 8); | |
1084 | |
1085 s->blockpos = 0; | |
1086 do { | |
1087 if (get_bits1(&gb)) { | |
1088 if (get_bits1(&gb)) { | |
1089 /* A restart header should be present */ | |
1090 if (read_restart_header(m, &gb, buf, substr) < 0) | |
1091 goto next_substr; | |
1092 s->restart_seen = 1; | |
1093 } | |
1094 | |
1095 if (!s->restart_seen) { | |
1096 av_log(m->avctx, AV_LOG_ERROR, | |
1097 "No restart header present in substream %d.\n", | |
1098 substr); | |
1099 goto next_substr; | |
1100 } | |
1101 | |
1102 if (read_decoding_params(m, &gb, substr) < 0) | |
1103 goto next_substr; | |
1104 } | |
1105 | |
1106 if (!s->restart_seen) { | |
1107 av_log(m->avctx, AV_LOG_ERROR, | |
1108 "No restart header present in substream %d.\n", | |
1109 substr); | |
1110 goto next_substr; | |
1111 } | |
1112 | |
1113 if (read_block_data(m, &gb, substr) < 0) | |
1114 return -1; | |
1115 | |
1116 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8) | |
1117 && get_bits1(&gb) == 0); | |
1118 | |
1119 skip_bits(&gb, (-get_bits_count(&gb)) & 15); | |
1120 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 48 && | |
1121 (show_bits_long(&gb, 32) == 0xd234d234 || | |
1122 show_bits_long(&gb, 20) == 0xd234e)) { | |
1123 skip_bits(&gb, 18); | |
1124 if (substr == m->max_decoded_substream) | |
1125 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated\n"); | |
1126 | |
1127 if (get_bits1(&gb)) { | |
1128 int shorten_by = get_bits(&gb, 13); | |
1129 shorten_by = FFMIN(shorten_by, s->blockpos); | |
1130 s->blockpos -= shorten_by; | |
1131 } else | |
1132 skip_bits(&gb, 13); | |
1133 } | |
1134 if (substream_parity_present[substr]) { | |
1135 uint8_t parity, checksum; | |
1136 | |
1137 parity = calculate_parity(buf, substream_data_len[substr] - 2); | |
1138 if ((parity ^ get_bits(&gb, 8)) != 0xa9) | |
1139 av_log(m->avctx, AV_LOG_ERROR, | |
1140 "Substream %d parity check failed\n", substr); | |
1141 | |
1142 checksum = mlp_checksum8(buf, substream_data_len[substr] - 2); | |
1143 if (checksum != get_bits(&gb, 8)) | |
1144 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed\n", | |
1145 substr); | |
1146 } | |
1147 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) { | |
1148 av_log(m->avctx, AV_LOG_ERROR, "Substream %d length mismatch.\n", | |
1149 substr); | |
1150 return -1; | |
1151 } | |
1152 | |
1153 next_substr: | |
1154 buf += substream_data_len[substr]; | |
1155 } | |
1156 | |
1157 rematrix_channels(m, m->max_decoded_substream); | |
1158 | |
1159 if (output_data(m, m->max_decoded_substream, data, data_size) < 0) | |
1160 return -1; | |
1161 | |
1162 return length; | |
1163 | |
1164 error: | |
1165 m->params_valid = 0; | |
1166 return -1; | |
1167 } | |
1168 | |
1169 AVCodec mlp_decoder = { | |
1170 "mlp", | |
1171 CODEC_TYPE_AUDIO, | |
1172 CODEC_ID_MLP, | |
1173 sizeof(MLPDecodeContext), | |
1174 mlp_decode_init, | |
1175 NULL, | |
1176 NULL, | |
1177 read_access_unit, | |
1178 .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"), | |
1179 }; | |
1180 |