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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 |