Mercurial > libavcodec.hg
annotate mlpdec.c @ 8105:b122cb798801 libavcodec
Fix illegal identifier, __ are reserved for the system.
author | diego |
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date | Mon, 03 Nov 2008 21:59:35 +0000 |
parents | d73cd240d68c |
children | 82dbfe8d2e0b |
rev | line source |
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7194 | 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 | |
7199 | 27 #include <stdint.h> |
28 | |
7194 | 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" | |
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35 #include "mlp.h" |
7194 | 36 |
7198 | 37 /** number of bits used for VLC lookup - longest Huffman code is 9 */ |
7194 | 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.mplayerhq.hu/bugreports.html and include " | |
44 "a sample of this file."; | |
45 | |
46 typedef struct SubStream { | |
7198 | 47 //! Set if a valid restart header has been read. Otherwise the substream cannot be decoded. |
7194 | 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 | |
62 //! The left shift applied to random noise in 0x31ea substreams. | |
63 uint8_t noise_shift; | |
64 //! The current seed value for the pseudorandom noise generator(s). | |
65 uint32_t noisegen_seed; | |
66 | |
67 //! Set if the substream contains extra info to check the size of VLC blocks. | |
68 uint8_t data_check_present; | |
69 | |
70 //! Bitmask of which parameter sets are conveyed in a decoding parameter block. | |
71 uint8_t param_presence_flags; | |
72 #define PARAM_BLOCKSIZE (1 << 7) | |
73 #define PARAM_MATRIX (1 << 6) | |
74 #define PARAM_OUTSHIFT (1 << 5) | |
75 #define PARAM_QUANTSTEP (1 << 4) | |
76 #define PARAM_FIR (1 << 3) | |
77 #define PARAM_IIR (1 << 2) | |
78 #define PARAM_HUFFOFFSET (1 << 1) | |
79 //@} | |
80 | |
81 //@{ | |
82 /** matrix data */ | |
83 | |
84 //! Number of matrices to be applied. | |
85 uint8_t num_primitive_matrices; | |
86 | |
87 //! matrix output channel | |
88 uint8_t matrix_out_ch[MAX_MATRICES]; | |
89 | |
90 //! Whether the LSBs of the matrix output are encoded in the bitstream. | |
91 uint8_t lsb_bypass[MAX_MATRICES]; | |
92 //! Matrix coefficients, stored as 2.14 fixed point. | |
93 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2]; | |
94 //! Left shift to apply to noise values in 0x31eb substreams. | |
95 uint8_t matrix_noise_shift[MAX_MATRICES]; | |
96 //@} | |
97 | |
7198 | 98 //! Left shift to apply to Huffman-decoded residuals. |
7194 | 99 uint8_t quant_step_size[MAX_CHANNELS]; |
100 | |
7198 | 101 //! number of PCM samples in current audio block |
7194 | 102 uint16_t blocksize; |
103 //! Number of PCM samples decoded so far in this frame. | |
104 uint16_t blockpos; | |
105 | |
106 //! Left shift to apply to decoded PCM values to get final 24-bit output. | |
107 int8_t output_shift[MAX_CHANNELS]; | |
108 | |
109 //! Running XOR of all output samples. | |
110 int32_t lossless_check_data; | |
111 | |
112 } SubStream; | |
113 | |
114 typedef struct MLPDecodeContext { | |
115 AVCodecContext *avctx; | |
116 | |
117 //! Set if a valid major sync block has been read. Otherwise no decoding is possible. | |
118 uint8_t params_valid; | |
119 | |
120 //! Number of substreams contained within this stream. | |
121 uint8_t num_substreams; | |
122 | |
123 //! Index of the last substream to decode - further substreams are skipped. | |
124 uint8_t max_decoded_substream; | |
125 | |
7198 | 126 //! number of PCM samples contained in each frame |
7194 | 127 int access_unit_size; |
7198 | 128 //! next power of two above the number of samples in each frame |
7194 | 129 int access_unit_size_pow2; |
130 | |
131 SubStream substream[MAX_SUBSTREAMS]; | |
132 | |
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133 ChannelParams channel_params[MAX_CHANNELS]; |
7194 | 134 |
135 int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; | |
136 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; | |
137 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2]; | |
138 } MLPDecodeContext; | |
139 | |
140 static VLC huff_vlc[3]; | |
141 | |
142 /** Initialize static data, constant between all invocations of the codec. */ | |
143 | |
144 static av_cold void init_static() | |
145 { | |
146 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, | |
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147 &ff_mlp_huffman_tables[0][0][1], 2, 1, |
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148 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512); |
7194 | 149 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, |
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150 &ff_mlp_huffman_tables[1][0][1], 2, 1, |
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151 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512); |
7194 | 152 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, |
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153 &ff_mlp_huffman_tables[2][0][1], 2, 1, |
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154 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512); |
7194 | 155 |
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156 ff_mlp_init_crc(); |
7194 | 157 } |
158 | |
159 static inline int32_t calculate_sign_huff(MLPDecodeContext *m, | |
160 unsigned int substr, unsigned int ch) | |
161 { | |
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162 ChannelParams *cp = &m->channel_params[ch]; |
7194 | 163 SubStream *s = &m->substream[substr]; |
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164 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch]; |
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165 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1); |
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166 int32_t sign_huff_offset = cp->huff_offset; |
7194 | 167 |
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168 if (cp->codebook > 0) |
7194 | 169 sign_huff_offset -= 7 << lsb_bits; |
170 | |
171 if (sign_shift >= 0) | |
172 sign_huff_offset -= 1 << sign_shift; | |
173 | |
174 return sign_huff_offset; | |
175 } | |
176 | |
177 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs | |
178 * and plain LSBs. */ | |
179 | |
180 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, | |
181 unsigned int substr, unsigned int pos) | |
182 { | |
183 SubStream *s = &m->substream[substr]; | |
184 unsigned int mat, channel; | |
185 | |
186 for (mat = 0; mat < s->num_primitive_matrices; mat++) | |
187 if (s->lsb_bypass[mat]) | |
188 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp); | |
189 | |
190 for (channel = s->min_channel; channel <= s->max_channel; channel++) { | |
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191 ChannelParams *cp = &m->channel_params[channel]; |
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192 int codebook = cp->codebook; |
7194 | 193 int quant_step_size = s->quant_step_size[channel]; |
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194 int lsb_bits = cp->huff_lsbs - quant_step_size; |
7194 | 195 int result = 0; |
196 | |
197 if (codebook > 0) | |
198 result = get_vlc2(gbp, huff_vlc[codebook-1].table, | |
199 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); | |
200 | |
201 if (result < 0) | |
202 return -1; | |
203 | |
204 if (lsb_bits > 0) | |
205 result = (result << lsb_bits) + get_bits(gbp, lsb_bits); | |
206 | |
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207 result += cp->sign_huff_offset; |
7194 | 208 result <<= quant_step_size; |
209 | |
210 m->sample_buffer[pos + s->blockpos][channel] = result; | |
211 } | |
212 | |
213 return 0; | |
214 } | |
215 | |
216 static av_cold int mlp_decode_init(AVCodecContext *avctx) | |
217 { | |
218 MLPDecodeContext *m = avctx->priv_data; | |
219 int substr; | |
220 | |
221 init_static(); | |
222 m->avctx = avctx; | |
223 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
224 m->substream[substr].lossless_check_data = 0xffffffff; | |
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225 avctx->sample_fmt = SAMPLE_FMT_S16; |
7194 | 226 return 0; |
227 } | |
228 | |
229 /** Read a major sync info header - contains high level information about | |
230 * the stream - sample rate, channel arrangement etc. Most of this | |
231 * information is not actually necessary for decoding, only for playback. | |
232 */ | |
233 | |
234 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb) | |
235 { | |
236 MLPHeaderInfo mh; | |
237 int substr; | |
238 | |
239 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0) | |
240 return -1; | |
241 | |
242 if (mh.group1_bits == 0) { | |
7198 | 243 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n"); |
7194 | 244 return -1; |
245 } | |
246 if (mh.group2_bits > mh.group1_bits) { | |
247 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 248 "Channel group 2 cannot have more bits per sample than group 1.\n"); |
7194 | 249 return -1; |
250 } | |
251 | |
252 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) { | |
253 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 254 "Channel groups with differing sample rates are not currently supported.\n"); |
7194 | 255 return -1; |
256 } | |
257 | |
258 if (mh.group1_samplerate == 0) { | |
7198 | 259 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n"); |
7194 | 260 return -1; |
261 } | |
262 if (mh.group1_samplerate > MAX_SAMPLERATE) { | |
263 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 264 "Sampling rate %d is greater than the supported maximum (%d).\n", |
7194 | 265 mh.group1_samplerate, MAX_SAMPLERATE); |
266 return -1; | |
267 } | |
268 if (mh.access_unit_size > MAX_BLOCKSIZE) { | |
269 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 270 "Block size %d is greater than the supported maximum (%d).\n", |
7194 | 271 mh.access_unit_size, MAX_BLOCKSIZE); |
272 return -1; | |
273 } | |
274 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) { | |
275 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 276 "Block size pow2 %d is greater than the supported maximum (%d).\n", |
7194 | 277 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2); |
278 return -1; | |
279 } | |
280 | |
281 if (mh.num_substreams == 0) | |
282 return -1; | |
283 if (mh.num_substreams > MAX_SUBSTREAMS) { | |
284 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 285 "Number of substreams %d is larger than the maximum supported " |
286 "by the decoder. %s\n", mh.num_substreams, sample_message); | |
7194 | 287 return -1; |
288 } | |
289 | |
290 m->access_unit_size = mh.access_unit_size; | |
291 m->access_unit_size_pow2 = mh.access_unit_size_pow2; | |
292 | |
293 m->num_substreams = mh.num_substreams; | |
294 m->max_decoded_substream = m->num_substreams - 1; | |
295 | |
296 m->avctx->sample_rate = mh.group1_samplerate; | |
297 m->avctx->frame_size = mh.access_unit_size; | |
298 | |
299 #ifdef CONFIG_AUDIO_NONSHORT | |
300 m->avctx->bits_per_sample = mh.group1_bits; | |
301 if (mh.group1_bits > 16) { | |
302 m->avctx->sample_fmt = SAMPLE_FMT_S32; | |
303 } | |
304 #endif | |
305 | |
306 m->params_valid = 1; | |
307 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
308 m->substream[substr].restart_seen = 0; | |
309 | |
310 return 0; | |
311 } | |
312 | |
313 /** Read a restart header from a block in a substream. This contains parameters | |
314 * required to decode the audio that do not change very often. Generally | |
315 * (always) present only in blocks following a major sync. */ | |
316 | |
317 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, | |
318 const uint8_t *buf, unsigned int substr) | |
319 { | |
320 SubStream *s = &m->substream[substr]; | |
321 unsigned int ch; | |
322 int sync_word, tmp; | |
323 uint8_t checksum; | |
324 uint8_t lossless_check; | |
325 int start_count = get_bits_count(gbp); | |
326 | |
327 sync_word = get_bits(gbp, 13); | |
328 | |
329 if (sync_word != 0x31ea >> 1) { | |
330 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 331 "restart header sync incorrect (got 0x%04x)\n", sync_word); |
7194 | 332 return -1; |
333 } | |
334 s->noise_type = get_bits1(gbp); | |
335 | |
336 skip_bits(gbp, 16); /* Output timestamp */ | |
337 | |
338 s->min_channel = get_bits(gbp, 4); | |
339 s->max_channel = get_bits(gbp, 4); | |
340 s->max_matrix_channel = get_bits(gbp, 4); | |
341 | |
342 if (s->min_channel > s->max_channel) { | |
343 av_log(m->avctx, AV_LOG_ERROR, | |
344 "Substream min channel cannot be greater than max channel.\n"); | |
345 return -1; | |
346 } | |
347 | |
348 if (m->avctx->request_channels > 0 | |
349 && s->max_channel + 1 >= m->avctx->request_channels | |
350 && substr < m->max_decoded_substream) { | |
351 av_log(m->avctx, AV_LOG_INFO, | |
352 "Extracting %d channel downmix from substream %d. " | |
353 "Further substreams will be skipped.\n", | |
354 s->max_channel + 1, substr); | |
355 m->max_decoded_substream = substr; | |
356 } | |
357 | |
358 s->noise_shift = get_bits(gbp, 4); | |
359 s->noisegen_seed = get_bits(gbp, 23); | |
360 | |
361 skip_bits(gbp, 19); | |
362 | |
363 s->data_check_present = get_bits1(gbp); | |
364 lossless_check = get_bits(gbp, 8); | |
365 if (substr == m->max_decoded_substream | |
366 && s->lossless_check_data != 0xffffffff) { | |
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367 tmp = xor_32_to_8(s->lossless_check_data); |
7194 | 368 if (tmp != lossless_check) |
369 av_log(m->avctx, AV_LOG_WARNING, | |
7198 | 370 "Lossless check failed - expected %02x, calculated %02x.\n", |
7194 | 371 lossless_check, tmp); |
372 else | |
7198 | 373 dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n", |
7194 | 374 substr, tmp); |
375 } | |
376 | |
377 skip_bits(gbp, 16); | |
378 | |
379 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
380 int ch_assign = get_bits(gbp, 6); | |
381 dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch, | |
382 ch_assign); | |
383 if (ch_assign != ch) { | |
384 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 385 "Non-1:1 channel assignments are used in this stream. %s\n", |
7194 | 386 sample_message); |
387 return -1; | |
388 } | |
389 } | |
390 | |
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391 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count); |
7194 | 392 |
393 if (checksum != get_bits(gbp, 8)) | |
7198 | 394 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n"); |
7194 | 395 |
7198 | 396 /* Set default decoding parameters. */ |
7194 | 397 s->param_presence_flags = 0xff; |
398 s->num_primitive_matrices = 0; | |
399 s->blocksize = 8; | |
400 s->lossless_check_data = 0; | |
401 | |
402 memset(s->output_shift , 0, sizeof(s->output_shift )); | |
403 memset(s->quant_step_size, 0, sizeof(s->quant_step_size)); | |
404 | |
405 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
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406 ChannelParams *cp = &m->channel_params[ch]; |
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407 cp->filter_params[FIR].order = 0; |
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408 cp->filter_params[IIR].order = 0; |
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409 cp->filter_params[FIR].shift = 0; |
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410 cp->filter_params[IIR].shift = 0; |
7194 | 411 |
7198 | 412 /* Default audio coding is 24-bit raw PCM. */ |
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413 cp->huff_offset = 0; |
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414 cp->sign_huff_offset = (-1) << 23; |
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415 cp->codebook = 0; |
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416 cp->huff_lsbs = 24; |
7194 | 417 } |
418 | |
419 if (substr == m->max_decoded_substream) { | |
420 m->avctx->channels = s->max_channel + 1; | |
421 } | |
422 | |
423 return 0; | |
424 } | |
425 | |
426 /** Read parameters for one of the prediction filters. */ | |
427 | |
428 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, | |
429 unsigned int channel, unsigned int filter) | |
430 { | |
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431 FilterParams *fp = &m->channel_params[channel].filter_params[filter]; |
7194 | 432 const char fchar = filter ? 'I' : 'F'; |
433 int i, order; | |
434 | |
7198 | 435 // Filter is 0 for FIR, 1 for IIR. |
7194 | 436 assert(filter < 2); |
437 | |
438 order = get_bits(gbp, 4); | |
439 if (order > MAX_FILTER_ORDER) { | |
440 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 441 "%cIR filter order %d is greater than maximum %d.\n", |
7194 | 442 fchar, order, MAX_FILTER_ORDER); |
443 return -1; | |
444 } | |
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445 fp->order = order; |
7194 | 446 |
447 if (order > 0) { | |
448 int coeff_bits, coeff_shift; | |
449 | |
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450 fp->shift = get_bits(gbp, 4); |
7194 | 451 |
452 coeff_bits = get_bits(gbp, 5); | |
453 coeff_shift = get_bits(gbp, 3); | |
454 if (coeff_bits < 1 || coeff_bits > 16) { | |
455 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 456 "%cIR filter coeff_bits must be between 1 and 16.\n", |
7194 | 457 fchar); |
458 return -1; | |
459 } | |
460 if (coeff_bits + coeff_shift > 16) { | |
461 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 462 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n", |
7194 | 463 fchar); |
464 return -1; | |
465 } | |
466 | |
467 for (i = 0; i < order; i++) | |
7553 | 468 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift; |
7194 | 469 |
470 if (get_bits1(gbp)) { | |
471 int state_bits, state_shift; | |
472 | |
473 if (filter == FIR) { | |
474 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 475 "FIR filter has state data specified.\n"); |
7194 | 476 return -1; |
477 } | |
478 | |
479 state_bits = get_bits(gbp, 4); | |
480 state_shift = get_bits(gbp, 4); | |
481 | |
7198 | 482 /* TODO: Check validity of state data. */ |
7194 | 483 |
484 for (i = 0; i < order; i++) | |
7553 | 485 fp->state[i] = get_sbits(gbp, state_bits) << state_shift; |
7194 | 486 } |
487 } | |
488 | |
489 return 0; | |
490 } | |
491 | |
492 /** Read decoding parameters that change more often than those in the restart | |
493 * header. */ | |
494 | |
495 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, | |
496 unsigned int substr) | |
497 { | |
498 SubStream *s = &m->substream[substr]; | |
499 unsigned int mat, ch; | |
500 | |
501 if (get_bits1(gbp)) | |
502 s->param_presence_flags = get_bits(gbp, 8); | |
503 | |
504 if (s->param_presence_flags & PARAM_BLOCKSIZE) | |
505 if (get_bits1(gbp)) { | |
506 s->blocksize = get_bits(gbp, 9); | |
507 if (s->blocksize > MAX_BLOCKSIZE) { | |
7198 | 508 av_log(m->avctx, AV_LOG_ERROR, "block size too large\n"); |
7194 | 509 s->blocksize = 0; |
510 return -1; | |
511 } | |
512 } | |
513 | |
514 if (s->param_presence_flags & PARAM_MATRIX) | |
515 if (get_bits1(gbp)) { | |
516 s->num_primitive_matrices = get_bits(gbp, 4); | |
517 | |
518 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
519 int frac_bits, max_chan; | |
520 s->matrix_out_ch[mat] = get_bits(gbp, 4); | |
521 frac_bits = get_bits(gbp, 4); | |
522 s->lsb_bypass [mat] = get_bits1(gbp); | |
523 | |
524 if (s->matrix_out_ch[mat] > s->max_channel) { | |
525 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 526 "Invalid channel %d specified as output from matrix.\n", |
7194 | 527 s->matrix_out_ch[mat]); |
528 return -1; | |
529 } | |
530 if (frac_bits > 14) { | |
531 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 532 "Too many fractional bits specified.\n"); |
7194 | 533 return -1; |
534 } | |
535 | |
536 max_chan = s->max_matrix_channel; | |
537 if (!s->noise_type) | |
538 max_chan+=2; | |
539 | |
540 for (ch = 0; ch <= max_chan; ch++) { | |
541 int coeff_val = 0; | |
542 if (get_bits1(gbp)) | |
543 coeff_val = get_sbits(gbp, frac_bits + 2); | |
544 | |
545 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); | |
546 } | |
547 | |
548 if (s->noise_type) | |
549 s->matrix_noise_shift[mat] = get_bits(gbp, 4); | |
550 else | |
551 s->matrix_noise_shift[mat] = 0; | |
552 } | |
553 } | |
554 | |
555 if (s->param_presence_flags & PARAM_OUTSHIFT) | |
556 if (get_bits1(gbp)) | |
557 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
558 s->output_shift[ch] = get_bits(gbp, 4); | |
559 dprintf(m->avctx, "output shift[%d] = %d\n", | |
560 ch, s->output_shift[ch]); | |
561 /* TODO: validate */ | |
562 } | |
563 | |
564 if (s->param_presence_flags & PARAM_QUANTSTEP) | |
565 if (get_bits1(gbp)) | |
566 for (ch = 0; ch <= s->max_channel; ch++) { | |
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567 ChannelParams *cp = &m->channel_params[ch]; |
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568 |
7194 | 569 s->quant_step_size[ch] = get_bits(gbp, 4); |
570 /* TODO: validate */ | |
571 | |
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572 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
7194 | 573 } |
574 | |
575 for (ch = s->min_channel; ch <= s->max_channel; ch++) | |
576 if (get_bits1(gbp)) { | |
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577 ChannelParams *cp = &m->channel_params[ch]; |
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578 FilterParams *fir = &cp->filter_params[FIR]; |
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579 FilterParams *iir = &cp->filter_params[IIR]; |
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580 |
7194 | 581 if (s->param_presence_flags & PARAM_FIR) |
582 if (get_bits1(gbp)) | |
583 if (read_filter_params(m, gbp, ch, FIR) < 0) | |
584 return -1; | |
585 | |
586 if (s->param_presence_flags & PARAM_IIR) | |
587 if (get_bits1(gbp)) | |
588 if (read_filter_params(m, gbp, ch, IIR) < 0) | |
589 return -1; | |
590 | |
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591 if (fir->order && iir->order && |
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592 fir->shift != iir->shift) { |
7194 | 593 av_log(m->avctx, AV_LOG_ERROR, |
7198 | 594 "FIR and IIR filters must use the same precision.\n"); |
7194 | 595 return -1; |
596 } | |
597 /* The FIR and IIR filters must have the same precision. | |
598 * To simplify the filtering code, only the precision of the | |
599 * FIR filter is considered. If only the IIR filter is employed, | |
600 * the FIR filter precision is set to that of the IIR filter, so | |
601 * that the filtering code can use it. */ | |
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602 if (!fir->order && iir->order) |
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603 fir->shift = iir->shift; |
7194 | 604 |
605 if (s->param_presence_flags & PARAM_HUFFOFFSET) | |
606 if (get_bits1(gbp)) | |
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607 cp->huff_offset = get_sbits(gbp, 15); |
7194 | 608 |
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609 cp->codebook = get_bits(gbp, 2); |
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610 cp->huff_lsbs = get_bits(gbp, 5); |
7194 | 611 |
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612 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
7194 | 613 |
614 /* TODO: validate */ | |
615 } | |
616 | |
617 return 0; | |
618 } | |
619 | |
620 #define MSB_MASK(bits) (-1u << bits) | |
621 | |
622 /** Generate PCM samples using the prediction filters and residual values | |
623 * read from the data stream, and update the filter state. */ | |
624 | |
625 static void filter_channel(MLPDecodeContext *m, unsigned int substr, | |
626 unsigned int channel) | |
627 { | |
628 SubStream *s = &m->substream[substr]; | |
629 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER]; | |
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630 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR], |
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631 &m->channel_params[channel].filter_params[IIR], }; |
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632 unsigned int filter_shift = fp[FIR]->shift; |
7194 | 633 int32_t mask = MSB_MASK(s->quant_step_size[channel]); |
634 int index = MAX_BLOCKSIZE; | |
635 int j, i; | |
636 | |
637 for (j = 0; j < NUM_FILTERS; j++) { | |
7553 | 638 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0], |
7194 | 639 MAX_FILTER_ORDER * sizeof(int32_t)); |
640 } | |
641 | |
642 for (i = 0; i < s->blocksize; i++) { | |
643 int32_t residual = m->sample_buffer[i + s->blockpos][channel]; | |
644 unsigned int order; | |
645 int64_t accum = 0; | |
646 int32_t result; | |
647 | |
648 /* TODO: Move this code to DSPContext? */ | |
649 | |
650 for (j = 0; j < NUM_FILTERS; j++) | |
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651 for (order = 0; order < fp[j]->order; order++) |
7194 | 652 accum += (int64_t)filter_state_buffer[j][index + order] * |
7553 | 653 fp[j]->coeff[order]; |
7194 | 654 |
655 accum = accum >> filter_shift; | |
656 result = (accum + residual) & mask; | |
657 | |
658 --index; | |
659 | |
660 filter_state_buffer[FIR][index] = result; | |
661 filter_state_buffer[IIR][index] = result - accum; | |
662 | |
663 m->sample_buffer[i + s->blockpos][channel] = result; | |
664 } | |
665 | |
666 for (j = 0; j < NUM_FILTERS; j++) { | |
7553 | 667 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index], |
7194 | 668 MAX_FILTER_ORDER * sizeof(int32_t)); |
669 } | |
670 } | |
671 | |
672 /** Read a block of PCM residual data (or actual if no filtering active). */ | |
673 | |
674 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, | |
675 unsigned int substr) | |
676 { | |
677 SubStream *s = &m->substream[substr]; | |
678 unsigned int i, ch, expected_stream_pos = 0; | |
679 | |
680 if (s->data_check_present) { | |
681 expected_stream_pos = get_bits_count(gbp); | |
682 expected_stream_pos += get_bits(gbp, 16); | |
683 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features " | |
684 "we have not tested yet. %s\n", sample_message); | |
685 } | |
686 | |
687 if (s->blockpos + s->blocksize > m->access_unit_size) { | |
7198 | 688 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n"); |
7194 | 689 return -1; |
690 } | |
691 | |
692 memset(&m->bypassed_lsbs[s->blockpos][0], 0, | |
693 s->blocksize * sizeof(m->bypassed_lsbs[0])); | |
694 | |
695 for (i = 0; i < s->blocksize; i++) { | |
696 if (read_huff_channels(m, gbp, substr, i) < 0) | |
697 return -1; | |
698 } | |
699 | |
700 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
701 filter_channel(m, substr, ch); | |
702 } | |
703 | |
704 s->blockpos += s->blocksize; | |
705 | |
706 if (s->data_check_present) { | |
707 if (get_bits_count(gbp) != expected_stream_pos) | |
7198 | 708 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n"); |
7194 | 709 skip_bits(gbp, 8); |
710 } | |
711 | |
712 return 0; | |
713 } | |
714 | |
7198 | 715 /** Data table used for TrueHD noise generation function. */ |
7194 | 716 |
717 static const int8_t noise_table[256] = { | |
718 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2, | |
719 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62, | |
720 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5, | |
721 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40, | |
722 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34, | |
723 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30, | |
724 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36, | |
725 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69, | |
726 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24, | |
727 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20, | |
728 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23, | |
729 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8, | |
730 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40, | |
731 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37, | |
732 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52, | |
733 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70, | |
734 }; | |
735 | |
736 /** Noise generation functions. | |
737 * I'm not sure what these are for - they seem to be some kind of pseudorandom | |
738 * sequence generators, used to generate noise data which is used when the | |
739 * channels are rematrixed. I'm not sure if they provide a practical benefit | |
740 * to compression, or just obfuscate the decoder. Are they for some kind of | |
741 * dithering? */ | |
742 | |
743 /** Generate two channels of noise, used in the matrix when | |
744 * restart sync word == 0x31ea. */ | |
745 | |
746 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr) | |
747 { | |
748 SubStream *s = &m->substream[substr]; | |
749 unsigned int i; | |
750 uint32_t seed = s->noisegen_seed; | |
751 unsigned int maxchan = s->max_matrix_channel; | |
752 | |
753 for (i = 0; i < s->blockpos; i++) { | |
754 uint16_t seed_shr7 = seed >> 7; | |
755 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift; | |
756 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift; | |
757 | |
758 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5); | |
759 } | |
760 | |
761 s->noisegen_seed = seed; | |
762 } | |
763 | |
764 /** Generate a block of noise, used when restart sync word == 0x31eb. */ | |
765 | |
766 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr) | |
767 { | |
768 SubStream *s = &m->substream[substr]; | |
769 unsigned int i; | |
770 uint32_t seed = s->noisegen_seed; | |
771 | |
772 for (i = 0; i < m->access_unit_size_pow2; i++) { | |
773 uint8_t seed_shr15 = seed >> 15; | |
774 m->noise_buffer[i] = noise_table[seed_shr15]; | |
775 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5); | |
776 } | |
777 | |
778 s->noisegen_seed = seed; | |
779 } | |
780 | |
781 | |
782 /** Apply the channel matrices in turn to reconstruct the original audio | |
783 * samples. */ | |
784 | |
785 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr) | |
786 { | |
787 SubStream *s = &m->substream[substr]; | |
788 unsigned int mat, src_ch, i; | |
789 unsigned int maxchan; | |
790 | |
791 maxchan = s->max_matrix_channel; | |
792 if (!s->noise_type) { | |
793 generate_2_noise_channels(m, substr); | |
794 maxchan += 2; | |
795 } else { | |
796 fill_noise_buffer(m, substr); | |
797 } | |
798 | |
799 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
800 int matrix_noise_shift = s->matrix_noise_shift[mat]; | |
801 unsigned int dest_ch = s->matrix_out_ch[mat]; | |
802 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]); | |
803 | |
804 /* TODO: DSPContext? */ | |
805 | |
806 for (i = 0; i < s->blockpos; i++) { | |
807 int64_t accum = 0; | |
808 for (src_ch = 0; src_ch <= maxchan; src_ch++) { | |
809 accum += (int64_t)m->sample_buffer[i][src_ch] | |
810 * s->matrix_coeff[mat][src_ch]; | |
811 } | |
812 if (matrix_noise_shift) { | |
813 uint32_t index = s->num_primitive_matrices - mat; | |
814 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1); | |
815 accum += m->noise_buffer[index] << (matrix_noise_shift + 7); | |
816 } | |
817 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask) | |
818 + m->bypassed_lsbs[i][mat]; | |
819 } | |
820 } | |
821 } | |
822 | |
823 /** Write the audio data into the output buffer. */ | |
824 | |
825 static int output_data_internal(MLPDecodeContext *m, unsigned int substr, | |
826 uint8_t *data, unsigned int *data_size, int is32) | |
827 { | |
828 SubStream *s = &m->substream[substr]; | |
829 unsigned int i, ch = 0; | |
830 int32_t *data_32 = (int32_t*) data; | |
831 int16_t *data_16 = (int16_t*) data; | |
832 | |
833 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2)) | |
834 return -1; | |
835 | |
836 for (i = 0; i < s->blockpos; i++) { | |
837 for (ch = 0; ch <= s->max_channel; ch++) { | |
838 int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch]; | |
839 s->lossless_check_data ^= (sample & 0xffffff) << ch; | |
840 if (is32) *data_32++ = sample << 8; | |
841 else *data_16++ = sample >> 8; | |
842 } | |
843 } | |
844 | |
845 *data_size = i * ch * (is32 ? 4 : 2); | |
846 | |
847 return 0; | |
848 } | |
849 | |
850 static int output_data(MLPDecodeContext *m, unsigned int substr, | |
851 uint8_t *data, unsigned int *data_size) | |
852 { | |
853 if (m->avctx->sample_fmt == SAMPLE_FMT_S32) | |
854 return output_data_internal(m, substr, data, data_size, 1); | |
855 else | |
856 return output_data_internal(m, substr, data, data_size, 0); | |
857 } | |
858 | |
859 | |
860 /** Read an access unit from the stream. | |
861 * Returns < 0 on error, 0 if not enough data is present in the input stream | |
862 * otherwise returns the number of bytes consumed. */ | |
863 | |
864 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, | |
865 const uint8_t *buf, int buf_size) | |
866 { | |
867 MLPDecodeContext *m = avctx->priv_data; | |
868 GetBitContext gb; | |
869 unsigned int length, substr; | |
870 unsigned int substream_start; | |
871 unsigned int header_size = 4; | |
872 unsigned int substr_header_size = 0; | |
873 uint8_t substream_parity_present[MAX_SUBSTREAMS]; | |
874 uint16_t substream_data_len[MAX_SUBSTREAMS]; | |
875 uint8_t parity_bits; | |
876 | |
877 if (buf_size < 4) | |
878 return 0; | |
879 | |
880 length = (AV_RB16(buf) & 0xfff) * 2; | |
881 | |
882 if (length > buf_size) | |
883 return -1; | |
884 | |
885 init_get_bits(&gb, (buf + 4), (length - 4) * 8); | |
886 | |
887 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) { | |
7198 | 888 dprintf(m->avctx, "Found major sync.\n"); |
7194 | 889 if (read_major_sync(m, &gb) < 0) |
890 goto error; | |
891 header_size += 28; | |
892 } | |
893 | |
894 if (!m->params_valid) { | |
895 av_log(m->avctx, AV_LOG_WARNING, | |
7198 | 896 "Stream parameters not seen; skipping frame.\n"); |
7194 | 897 *data_size = 0; |
898 return length; | |
899 } | |
900 | |
901 substream_start = 0; | |
902 | |
903 for (substr = 0; substr < m->num_substreams; substr++) { | |
904 int extraword_present, checkdata_present, end; | |
905 | |
906 extraword_present = get_bits1(&gb); | |
907 skip_bits1(&gb); | |
908 checkdata_present = get_bits1(&gb); | |
909 skip_bits1(&gb); | |
910 | |
911 end = get_bits(&gb, 12) * 2; | |
912 | |
913 substr_header_size += 2; | |
914 | |
915 if (extraword_present) { | |
916 skip_bits(&gb, 16); | |
917 substr_header_size += 2; | |
918 } | |
919 | |
920 if (end + header_size + substr_header_size > length) { | |
921 av_log(m->avctx, AV_LOG_ERROR, | |
922 "Indicated length of substream %d data goes off end of " | |
923 "packet.\n", substr); | |
924 end = length - header_size - substr_header_size; | |
925 } | |
926 | |
927 if (end < substream_start) { | |
928 av_log(avctx, AV_LOG_ERROR, | |
929 "Indicated end offset of substream %d data " | |
930 "is smaller than calculated start offset.\n", | |
931 substr); | |
932 goto error; | |
933 } | |
934 | |
935 if (substr > m->max_decoded_substream) | |
936 continue; | |
937 | |
938 substream_parity_present[substr] = checkdata_present; | |
939 substream_data_len[substr] = end - substream_start; | |
940 substream_start = end; | |
941 } | |
942 | |
7559
fd24c8628221
mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
7556
diff
changeset
|
943 parity_bits = ff_mlp_calculate_parity(buf, 4); |
fd24c8628221
mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
7556
diff
changeset
|
944 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size); |
7194 | 945 |
946 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) { | |
947 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n"); | |
948 goto error; | |
949 } | |
950 | |
951 buf += header_size + substr_header_size; | |
952 | |
953 for (substr = 0; substr <= m->max_decoded_substream; substr++) { | |
954 SubStream *s = &m->substream[substr]; | |
955 init_get_bits(&gb, buf, substream_data_len[substr] * 8); | |
956 | |
957 s->blockpos = 0; | |
958 do { | |
959 if (get_bits1(&gb)) { | |
960 if (get_bits1(&gb)) { | |
7198 | 961 /* A restart header should be present. */ |
7194 | 962 if (read_restart_header(m, &gb, buf, substr) < 0) |
963 goto next_substr; | |
964 s->restart_seen = 1; | |
965 } | |
966 | |
967 if (!s->restart_seen) { | |
968 av_log(m->avctx, AV_LOG_ERROR, | |
969 "No restart header present in substream %d.\n", | |
970 substr); | |
971 goto next_substr; | |
972 } | |
973 | |
974 if (read_decoding_params(m, &gb, substr) < 0) | |
975 goto next_substr; | |
976 } | |
977 | |
978 if (!s->restart_seen) { | |
979 av_log(m->avctx, AV_LOG_ERROR, | |
980 "No restart header present in substream %d.\n", | |
981 substr); | |
982 goto next_substr; | |
983 } | |
984 | |
985 if (read_block_data(m, &gb, substr) < 0) | |
986 return -1; | |
987 | |
988 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8) | |
989 && get_bits1(&gb) == 0); | |
990 | |
991 skip_bits(&gb, (-get_bits_count(&gb)) & 15); | |
7262
e3822c61f2e4
mlpdec: Check for bits left before each read of End-of-Stream indicator and
ramiro
parents:
7199
diff
changeset
|
992 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 && |
7575
d73cd240d68c
mlp: Define End-of-Stream code in common header file and use it in decoder.
ramiro
parents:
7566
diff
changeset
|
993 (show_bits_long(&gb, 32) == END_OF_STREAM || |
7194 | 994 show_bits_long(&gb, 20) == 0xd234e)) { |
995 skip_bits(&gb, 18); | |
996 if (substr == m->max_decoded_substream) | |
7198 | 997 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n"); |
7194 | 998 |
999 if (get_bits1(&gb)) { | |
1000 int shorten_by = get_bits(&gb, 13); | |
1001 shorten_by = FFMIN(shorten_by, s->blockpos); | |
1002 s->blockpos -= shorten_by; | |
1003 } else | |
1004 skip_bits(&gb, 13); | |
1005 } | |
7262
e3822c61f2e4
mlpdec: Check for bits left before each read of End-of-Stream indicator and
ramiro
parents:
7199
diff
changeset
|
1006 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 16 && |
e3822c61f2e4
mlpdec: Check for bits left before each read of End-of-Stream indicator and
ramiro
parents:
7199
diff
changeset
|
1007 substream_parity_present[substr]) { |
7194 | 1008 uint8_t parity, checksum; |
1009 | |
7559
fd24c8628221
mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
7556
diff
changeset
|
1010 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2); |
7194 | 1011 if ((parity ^ get_bits(&gb, 8)) != 0xa9) |
1012 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 1013 "Substream %d parity check failed.\n", substr); |
7194 | 1014 |
7559
fd24c8628221
mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
7556
diff
changeset
|
1015 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2); |
7194 | 1016 if (checksum != get_bits(&gb, 8)) |
7198 | 1017 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n", |
7194 | 1018 substr); |
1019 } | |
1020 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) { | |
7198 | 1021 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", |
7194 | 1022 substr); |
1023 return -1; | |
1024 } | |
1025 | |
1026 next_substr: | |
1027 buf += substream_data_len[substr]; | |
1028 } | |
1029 | |
1030 rematrix_channels(m, m->max_decoded_substream); | |
1031 | |
1032 if (output_data(m, m->max_decoded_substream, data, data_size) < 0) | |
1033 return -1; | |
1034 | |
1035 return length; | |
1036 | |
1037 error: | |
1038 m->params_valid = 0; | |
1039 return -1; | |
1040 } | |
1041 | |
1042 AVCodec mlp_decoder = { | |
1043 "mlp", | |
1044 CODEC_TYPE_AUDIO, | |
1045 CODEC_ID_MLP, | |
1046 sizeof(MLPDecodeContext), | |
1047 mlp_decode_init, | |
1048 NULL, | |
1049 NULL, | |
1050 read_access_unit, | |
1051 .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"), | |
1052 }; | |
1053 |