Mercurial > libavcodec.hg
annotate mlpdec.c @ 8680:83453ea78558 libavcodec
ARM: replace jump tables with conditional branches
author | mru |
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date | Tue, 27 Jan 2009 16:06:47 +0000 |
parents | 5b3c90656fdf |
children | 18737839ed27 |
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 " | |
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43 "http://ffmpeg.org/bugreports.html and include " |
7194 | 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 |
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 | |
8274 | 299 m->avctx->bits_per_raw_sample = mh.group1_bits; |
8277 | 300 if (mh.group1_bits > 16) |
7194 | 301 m->avctx->sample_fmt = SAMPLE_FMT_S32; |
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302 else |
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303 m->avctx->sample_fmt = SAMPLE_FMT_S16; |
7194 | 304 |
305 m->params_valid = 1; | |
306 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
307 m->substream[substr].restart_seen = 0; | |
308 | |
309 return 0; | |
310 } | |
311 | |
312 /** Read a restart header from a block in a substream. This contains parameters | |
313 * required to decode the audio that do not change very often. Generally | |
314 * (always) present only in blocks following a major sync. */ | |
315 | |
316 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, | |
317 const uint8_t *buf, unsigned int substr) | |
318 { | |
319 SubStream *s = &m->substream[substr]; | |
320 unsigned int ch; | |
321 int sync_word, tmp; | |
322 uint8_t checksum; | |
323 uint8_t lossless_check; | |
324 int start_count = get_bits_count(gbp); | |
325 | |
326 sync_word = get_bits(gbp, 13); | |
327 | |
328 if (sync_word != 0x31ea >> 1) { | |
329 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 330 "restart header sync incorrect (got 0x%04x)\n", sync_word); |
7194 | 331 return -1; |
332 } | |
333 s->noise_type = get_bits1(gbp); | |
334 | |
335 skip_bits(gbp, 16); /* Output timestamp */ | |
336 | |
337 s->min_channel = get_bits(gbp, 4); | |
338 s->max_channel = get_bits(gbp, 4); | |
339 s->max_matrix_channel = get_bits(gbp, 4); | |
340 | |
341 if (s->min_channel > s->max_channel) { | |
342 av_log(m->avctx, AV_LOG_ERROR, | |
343 "Substream min channel cannot be greater than max channel.\n"); | |
344 return -1; | |
345 } | |
346 | |
347 if (m->avctx->request_channels > 0 | |
348 && s->max_channel + 1 >= m->avctx->request_channels | |
349 && substr < m->max_decoded_substream) { | |
350 av_log(m->avctx, AV_LOG_INFO, | |
351 "Extracting %d channel downmix from substream %d. " | |
352 "Further substreams will be skipped.\n", | |
353 s->max_channel + 1, substr); | |
354 m->max_decoded_substream = substr; | |
355 } | |
356 | |
357 s->noise_shift = get_bits(gbp, 4); | |
358 s->noisegen_seed = get_bits(gbp, 23); | |
359 | |
360 skip_bits(gbp, 19); | |
361 | |
362 s->data_check_present = get_bits1(gbp); | |
363 lossless_check = get_bits(gbp, 8); | |
364 if (substr == m->max_decoded_substream | |
365 && s->lossless_check_data != 0xffffffff) { | |
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366 tmp = xor_32_to_8(s->lossless_check_data); |
7194 | 367 if (tmp != lossless_check) |
368 av_log(m->avctx, AV_LOG_WARNING, | |
7198 | 369 "Lossless check failed - expected %02x, calculated %02x.\n", |
7194 | 370 lossless_check, tmp); |
371 else | |
7198 | 372 dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n", |
7194 | 373 substr, tmp); |
374 } | |
375 | |
376 skip_bits(gbp, 16); | |
377 | |
378 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
379 int ch_assign = get_bits(gbp, 6); | |
380 dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch, | |
381 ch_assign); | |
382 if (ch_assign != ch) { | |
383 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 384 "Non-1:1 channel assignments are used in this stream. %s\n", |
7194 | 385 sample_message); |
386 return -1; | |
387 } | |
388 } | |
389 | |
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390 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count); |
7194 | 391 |
392 if (checksum != get_bits(gbp, 8)) | |
7198 | 393 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n"); |
7194 | 394 |
7198 | 395 /* Set default decoding parameters. */ |
7194 | 396 s->param_presence_flags = 0xff; |
397 s->num_primitive_matrices = 0; | |
398 s->blocksize = 8; | |
399 s->lossless_check_data = 0; | |
400 | |
401 memset(s->output_shift , 0, sizeof(s->output_shift )); | |
402 memset(s->quant_step_size, 0, sizeof(s->quant_step_size)); | |
403 | |
404 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
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405 ChannelParams *cp = &m->channel_params[ch]; |
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406 cp->filter_params[FIR].order = 0; |
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407 cp->filter_params[IIR].order = 0; |
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408 cp->filter_params[FIR].shift = 0; |
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409 cp->filter_params[IIR].shift = 0; |
7194 | 410 |
7198 | 411 /* Default audio coding is 24-bit raw PCM. */ |
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412 cp->huff_offset = 0; |
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413 cp->sign_huff_offset = (-1) << 23; |
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414 cp->codebook = 0; |
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415 cp->huff_lsbs = 24; |
7194 | 416 } |
417 | |
418 if (substr == m->max_decoded_substream) { | |
419 m->avctx->channels = s->max_channel + 1; | |
420 } | |
421 | |
422 return 0; | |
423 } | |
424 | |
425 /** Read parameters for one of the prediction filters. */ | |
426 | |
427 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, | |
428 unsigned int channel, unsigned int filter) | |
429 { | |
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430 FilterParams *fp = &m->channel_params[channel].filter_params[filter]; |
7194 | 431 const char fchar = filter ? 'I' : 'F'; |
432 int i, order; | |
433 | |
7198 | 434 // Filter is 0 for FIR, 1 for IIR. |
7194 | 435 assert(filter < 2); |
436 | |
437 order = get_bits(gbp, 4); | |
438 if (order > MAX_FILTER_ORDER) { | |
439 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 440 "%cIR filter order %d is greater than maximum %d.\n", |
7194 | 441 fchar, order, MAX_FILTER_ORDER); |
442 return -1; | |
443 } | |
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444 fp->order = order; |
7194 | 445 |
446 if (order > 0) { | |
447 int coeff_bits, coeff_shift; | |
448 | |
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449 fp->shift = get_bits(gbp, 4); |
7194 | 450 |
451 coeff_bits = get_bits(gbp, 5); | |
452 coeff_shift = get_bits(gbp, 3); | |
453 if (coeff_bits < 1 || coeff_bits > 16) { | |
454 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 455 "%cIR filter coeff_bits must be between 1 and 16.\n", |
7194 | 456 fchar); |
457 return -1; | |
458 } | |
459 if (coeff_bits + coeff_shift > 16) { | |
460 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 461 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n", |
7194 | 462 fchar); |
463 return -1; | |
464 } | |
465 | |
466 for (i = 0; i < order; i++) | |
7553 | 467 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift; |
7194 | 468 |
469 if (get_bits1(gbp)) { | |
470 int state_bits, state_shift; | |
471 | |
472 if (filter == FIR) { | |
473 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 474 "FIR filter has state data specified.\n"); |
7194 | 475 return -1; |
476 } | |
477 | |
478 state_bits = get_bits(gbp, 4); | |
479 state_shift = get_bits(gbp, 4); | |
480 | |
7198 | 481 /* TODO: Check validity of state data. */ |
7194 | 482 |
483 for (i = 0; i < order; i++) | |
7553 | 484 fp->state[i] = get_sbits(gbp, state_bits) << state_shift; |
7194 | 485 } |
486 } | |
487 | |
488 return 0; | |
489 } | |
490 | |
491 /** Read decoding parameters that change more often than those in the restart | |
492 * header. */ | |
493 | |
494 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, | |
495 unsigned int substr) | |
496 { | |
497 SubStream *s = &m->substream[substr]; | |
498 unsigned int mat, ch; | |
499 | |
500 if (get_bits1(gbp)) | |
501 s->param_presence_flags = get_bits(gbp, 8); | |
502 | |
503 if (s->param_presence_flags & PARAM_BLOCKSIZE) | |
504 if (get_bits1(gbp)) { | |
505 s->blocksize = get_bits(gbp, 9); | |
506 if (s->blocksize > MAX_BLOCKSIZE) { | |
7198 | 507 av_log(m->avctx, AV_LOG_ERROR, "block size too large\n"); |
7194 | 508 s->blocksize = 0; |
509 return -1; | |
510 } | |
511 } | |
512 | |
513 if (s->param_presence_flags & PARAM_MATRIX) | |
514 if (get_bits1(gbp)) { | |
515 s->num_primitive_matrices = get_bits(gbp, 4); | |
516 | |
517 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
518 int frac_bits, max_chan; | |
519 s->matrix_out_ch[mat] = get_bits(gbp, 4); | |
520 frac_bits = get_bits(gbp, 4); | |
521 s->lsb_bypass [mat] = get_bits1(gbp); | |
522 | |
523 if (s->matrix_out_ch[mat] > s->max_channel) { | |
524 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 525 "Invalid channel %d specified as output from matrix.\n", |
7194 | 526 s->matrix_out_ch[mat]); |
527 return -1; | |
528 } | |
529 if (frac_bits > 14) { | |
530 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 531 "Too many fractional bits specified.\n"); |
7194 | 532 return -1; |
533 } | |
534 | |
535 max_chan = s->max_matrix_channel; | |
536 if (!s->noise_type) | |
537 max_chan+=2; | |
538 | |
539 for (ch = 0; ch <= max_chan; ch++) { | |
540 int coeff_val = 0; | |
541 if (get_bits1(gbp)) | |
542 coeff_val = get_sbits(gbp, frac_bits + 2); | |
543 | |
544 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); | |
545 } | |
546 | |
547 if (s->noise_type) | |
548 s->matrix_noise_shift[mat] = get_bits(gbp, 4); | |
549 else | |
550 s->matrix_noise_shift[mat] = 0; | |
551 } | |
552 } | |
553 | |
554 if (s->param_presence_flags & PARAM_OUTSHIFT) | |
555 if (get_bits1(gbp)) | |
556 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
557 s->output_shift[ch] = get_bits(gbp, 4); | |
558 dprintf(m->avctx, "output shift[%d] = %d\n", | |
559 ch, s->output_shift[ch]); | |
560 /* TODO: validate */ | |
561 } | |
562 | |
563 if (s->param_presence_flags & PARAM_QUANTSTEP) | |
564 if (get_bits1(gbp)) | |
565 for (ch = 0; ch <= s->max_channel; ch++) { | |
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566 ChannelParams *cp = &m->channel_params[ch]; |
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567 |
7194 | 568 s->quant_step_size[ch] = get_bits(gbp, 4); |
569 /* TODO: validate */ | |
570 | |
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571 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
7194 | 572 } |
573 | |
574 for (ch = s->min_channel; ch <= s->max_channel; ch++) | |
575 if (get_bits1(gbp)) { | |
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576 ChannelParams *cp = &m->channel_params[ch]; |
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577 FilterParams *fir = &cp->filter_params[FIR]; |
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578 FilterParams *iir = &cp->filter_params[IIR]; |
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579 |
7194 | 580 if (s->param_presence_flags & PARAM_FIR) |
581 if (get_bits1(gbp)) | |
582 if (read_filter_params(m, gbp, ch, FIR) < 0) | |
583 return -1; | |
584 | |
585 if (s->param_presence_flags & PARAM_IIR) | |
586 if (get_bits1(gbp)) | |
587 if (read_filter_params(m, gbp, ch, IIR) < 0) | |
588 return -1; | |
589 | |
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590 if (fir->order && iir->order && |
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591 fir->shift != iir->shift) { |
7194 | 592 av_log(m->avctx, AV_LOG_ERROR, |
7198 | 593 "FIR and IIR filters must use the same precision.\n"); |
7194 | 594 return -1; |
595 } | |
596 /* The FIR and IIR filters must have the same precision. | |
597 * To simplify the filtering code, only the precision of the | |
598 * FIR filter is considered. If only the IIR filter is employed, | |
599 * the FIR filter precision is set to that of the IIR filter, so | |
600 * that the filtering code can use it. */ | |
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601 if (!fir->order && iir->order) |
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602 fir->shift = iir->shift; |
7194 | 603 |
604 if (s->param_presence_flags & PARAM_HUFFOFFSET) | |
605 if (get_bits1(gbp)) | |
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606 cp->huff_offset = get_sbits(gbp, 15); |
7194 | 607 |
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608 cp->codebook = get_bits(gbp, 2); |
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609 cp->huff_lsbs = get_bits(gbp, 5); |
7194 | 610 |
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611 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
7194 | 612 |
613 /* TODO: validate */ | |
614 } | |
615 | |
616 return 0; | |
617 } | |
618 | |
619 #define MSB_MASK(bits) (-1u << bits) | |
620 | |
621 /** Generate PCM samples using the prediction filters and residual values | |
622 * read from the data stream, and update the filter state. */ | |
623 | |
624 static void filter_channel(MLPDecodeContext *m, unsigned int substr, | |
625 unsigned int channel) | |
626 { | |
627 SubStream *s = &m->substream[substr]; | |
628 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER]; | |
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629 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR], |
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630 &m->channel_params[channel].filter_params[IIR], }; |
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631 unsigned int filter_shift = fp[FIR]->shift; |
7194 | 632 int32_t mask = MSB_MASK(s->quant_step_size[channel]); |
633 int index = MAX_BLOCKSIZE; | |
634 int j, i; | |
635 | |
636 for (j = 0; j < NUM_FILTERS; j++) { | |
7553 | 637 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0], |
7194 | 638 MAX_FILTER_ORDER * sizeof(int32_t)); |
639 } | |
640 | |
641 for (i = 0; i < s->blocksize; i++) { | |
642 int32_t residual = m->sample_buffer[i + s->blockpos][channel]; | |
643 unsigned int order; | |
644 int64_t accum = 0; | |
645 int32_t result; | |
646 | |
647 /* TODO: Move this code to DSPContext? */ | |
648 | |
649 for (j = 0; j < NUM_FILTERS; j++) | |
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650 for (order = 0; order < fp[j]->order; order++) |
7194 | 651 accum += (int64_t)filter_state_buffer[j][index + order] * |
7553 | 652 fp[j]->coeff[order]; |
7194 | 653 |
654 accum = accum >> filter_shift; | |
655 result = (accum + residual) & mask; | |
656 | |
657 --index; | |
658 | |
659 filter_state_buffer[FIR][index] = result; | |
660 filter_state_buffer[IIR][index] = result - accum; | |
661 | |
662 m->sample_buffer[i + s->blockpos][channel] = result; | |
663 } | |
664 | |
665 for (j = 0; j < NUM_FILTERS; j++) { | |
7553 | 666 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index], |
7194 | 667 MAX_FILTER_ORDER * sizeof(int32_t)); |
668 } | |
669 } | |
670 | |
671 /** Read a block of PCM residual data (or actual if no filtering active). */ | |
672 | |
673 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, | |
674 unsigned int substr) | |
675 { | |
676 SubStream *s = &m->substream[substr]; | |
677 unsigned int i, ch, expected_stream_pos = 0; | |
678 | |
679 if (s->data_check_present) { | |
680 expected_stream_pos = get_bits_count(gbp); | |
681 expected_stream_pos += get_bits(gbp, 16); | |
682 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features " | |
683 "we have not tested yet. %s\n", sample_message); | |
684 } | |
685 | |
686 if (s->blockpos + s->blocksize > m->access_unit_size) { | |
7198 | 687 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n"); |
7194 | 688 return -1; |
689 } | |
690 | |
691 memset(&m->bypassed_lsbs[s->blockpos][0], 0, | |
692 s->blocksize * sizeof(m->bypassed_lsbs[0])); | |
693 | |
694 for (i = 0; i < s->blocksize; i++) { | |
695 if (read_huff_channels(m, gbp, substr, i) < 0) | |
696 return -1; | |
697 } | |
698 | |
699 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
700 filter_channel(m, substr, ch); | |
701 } | |
702 | |
703 s->blockpos += s->blocksize; | |
704 | |
705 if (s->data_check_present) { | |
706 if (get_bits_count(gbp) != expected_stream_pos) | |
7198 | 707 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n"); |
7194 | 708 skip_bits(gbp, 8); |
709 } | |
710 | |
711 return 0; | |
712 } | |
713 | |
7198 | 714 /** Data table used for TrueHD noise generation function. */ |
7194 | 715 |
716 static const int8_t noise_table[256] = { | |
717 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2, | |
718 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62, | |
719 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5, | |
720 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40, | |
721 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34, | |
722 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30, | |
723 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36, | |
724 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69, | |
725 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24, | |
726 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20, | |
727 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23, | |
728 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8, | |
729 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40, | |
730 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37, | |
731 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52, | |
732 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70, | |
733 }; | |
734 | |
735 /** Noise generation functions. | |
736 * I'm not sure what these are for - they seem to be some kind of pseudorandom | |
737 * sequence generators, used to generate noise data which is used when the | |
738 * channels are rematrixed. I'm not sure if they provide a practical benefit | |
739 * to compression, or just obfuscate the decoder. Are they for some kind of | |
740 * dithering? */ | |
741 | |
742 /** Generate two channels of noise, used in the matrix when | |
743 * restart sync word == 0x31ea. */ | |
744 | |
745 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr) | |
746 { | |
747 SubStream *s = &m->substream[substr]; | |
748 unsigned int i; | |
749 uint32_t seed = s->noisegen_seed; | |
750 unsigned int maxchan = s->max_matrix_channel; | |
751 | |
752 for (i = 0; i < s->blockpos; i++) { | |
753 uint16_t seed_shr7 = seed >> 7; | |
754 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift; | |
755 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift; | |
756 | |
757 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5); | |
758 } | |
759 | |
760 s->noisegen_seed = seed; | |
761 } | |
762 | |
763 /** Generate a block of noise, used when restart sync word == 0x31eb. */ | |
764 | |
765 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr) | |
766 { | |
767 SubStream *s = &m->substream[substr]; | |
768 unsigned int i; | |
769 uint32_t seed = s->noisegen_seed; | |
770 | |
771 for (i = 0; i < m->access_unit_size_pow2; i++) { | |
772 uint8_t seed_shr15 = seed >> 15; | |
773 m->noise_buffer[i] = noise_table[seed_shr15]; | |
774 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5); | |
775 } | |
776 | |
777 s->noisegen_seed = seed; | |
778 } | |
779 | |
780 | |
781 /** Apply the channel matrices in turn to reconstruct the original audio | |
782 * samples. */ | |
783 | |
784 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr) | |
785 { | |
786 SubStream *s = &m->substream[substr]; | |
787 unsigned int mat, src_ch, i; | |
788 unsigned int maxchan; | |
789 | |
790 maxchan = s->max_matrix_channel; | |
791 if (!s->noise_type) { | |
792 generate_2_noise_channels(m, substr); | |
793 maxchan += 2; | |
794 } else { | |
795 fill_noise_buffer(m, substr); | |
796 } | |
797 | |
798 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
799 int matrix_noise_shift = s->matrix_noise_shift[mat]; | |
800 unsigned int dest_ch = s->matrix_out_ch[mat]; | |
801 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]); | |
802 | |
803 /* TODO: DSPContext? */ | |
804 | |
805 for (i = 0; i < s->blockpos; i++) { | |
806 int64_t accum = 0; | |
807 for (src_ch = 0; src_ch <= maxchan; src_ch++) { | |
808 accum += (int64_t)m->sample_buffer[i][src_ch] | |
809 * s->matrix_coeff[mat][src_ch]; | |
810 } | |
811 if (matrix_noise_shift) { | |
812 uint32_t index = s->num_primitive_matrices - mat; | |
813 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1); | |
814 accum += m->noise_buffer[index] << (matrix_noise_shift + 7); | |
815 } | |
816 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask) | |
817 + m->bypassed_lsbs[i][mat]; | |
818 } | |
819 } | |
820 } | |
821 | |
822 /** Write the audio data into the output buffer. */ | |
823 | |
824 static int output_data_internal(MLPDecodeContext *m, unsigned int substr, | |
825 uint8_t *data, unsigned int *data_size, int is32) | |
826 { | |
827 SubStream *s = &m->substream[substr]; | |
828 unsigned int i, ch = 0; | |
829 int32_t *data_32 = (int32_t*) data; | |
830 int16_t *data_16 = (int16_t*) data; | |
831 | |
832 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2)) | |
833 return -1; | |
834 | |
835 for (i = 0; i < s->blockpos; i++) { | |
836 for (ch = 0; ch <= s->max_channel; ch++) { | |
837 int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch]; | |
838 s->lossless_check_data ^= (sample & 0xffffff) << ch; | |
839 if (is32) *data_32++ = sample << 8; | |
840 else *data_16++ = sample >> 8; | |
841 } | |
842 } | |
843 | |
844 *data_size = i * ch * (is32 ? 4 : 2); | |
845 | |
846 return 0; | |
847 } | |
848 | |
849 static int output_data(MLPDecodeContext *m, unsigned int substr, | |
850 uint8_t *data, unsigned int *data_size) | |
851 { | |
852 if (m->avctx->sample_fmt == SAMPLE_FMT_S32) | |
853 return output_data_internal(m, substr, data, data_size, 1); | |
854 else | |
855 return output_data_internal(m, substr, data, data_size, 0); | |
856 } | |
857 | |
858 | |
859 /** Read an access unit from the stream. | |
860 * Returns < 0 on error, 0 if not enough data is present in the input stream | |
861 * otherwise returns the number of bytes consumed. */ | |
862 | |
863 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, | |
864 const uint8_t *buf, int buf_size) | |
865 { | |
866 MLPDecodeContext *m = avctx->priv_data; | |
867 GetBitContext gb; | |
868 unsigned int length, substr; | |
869 unsigned int substream_start; | |
870 unsigned int header_size = 4; | |
871 unsigned int substr_header_size = 0; | |
872 uint8_t substream_parity_present[MAX_SUBSTREAMS]; | |
873 uint16_t substream_data_len[MAX_SUBSTREAMS]; | |
874 uint8_t parity_bits; | |
875 | |
876 if (buf_size < 4) | |
877 return 0; | |
878 | |
879 length = (AV_RB16(buf) & 0xfff) * 2; | |
880 | |
881 if (length > buf_size) | |
882 return -1; | |
883 | |
884 init_get_bits(&gb, (buf + 4), (length - 4) * 8); | |
885 | |
886 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) { | |
7198 | 887 dprintf(m->avctx, "Found major sync.\n"); |
7194 | 888 if (read_major_sync(m, &gb) < 0) |
889 goto error; | |
890 header_size += 28; | |
891 } | |
892 | |
893 if (!m->params_valid) { | |
894 av_log(m->avctx, AV_LOG_WARNING, | |
7198 | 895 "Stream parameters not seen; skipping frame.\n"); |
7194 | 896 *data_size = 0; |
897 return length; | |
898 } | |
899 | |
900 substream_start = 0; | |
901 | |
902 for (substr = 0; substr < m->num_substreams; substr++) { | |
903 int extraword_present, checkdata_present, end; | |
904 | |
905 extraword_present = get_bits1(&gb); | |
906 skip_bits1(&gb); | |
907 checkdata_present = get_bits1(&gb); | |
908 skip_bits1(&gb); | |
909 | |
910 end = get_bits(&gb, 12) * 2; | |
911 | |
912 substr_header_size += 2; | |
913 | |
914 if (extraword_present) { | |
915 skip_bits(&gb, 16); | |
916 substr_header_size += 2; | |
917 } | |
918 | |
919 if (end + header_size + substr_header_size > length) { | |
920 av_log(m->avctx, AV_LOG_ERROR, | |
921 "Indicated length of substream %d data goes off end of " | |
922 "packet.\n", substr); | |
923 end = length - header_size - substr_header_size; | |
924 } | |
925 | |
926 if (end < substream_start) { | |
927 av_log(avctx, AV_LOG_ERROR, | |
928 "Indicated end offset of substream %d data " | |
929 "is smaller than calculated start offset.\n", | |
930 substr); | |
931 goto error; | |
932 } | |
933 | |
934 if (substr > m->max_decoded_substream) | |
935 continue; | |
936 | |
937 substream_parity_present[substr] = checkdata_present; | |
938 substream_data_len[substr] = end - substream_start; | |
939 substream_start = end; | |
940 } | |
941 | |
7559
fd24c8628221
mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
7556
diff
changeset
|
942 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
|
943 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size); |
7194 | 944 |
945 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) { | |
946 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n"); | |
947 goto error; | |
948 } | |
949 | |
950 buf += header_size + substr_header_size; | |
951 | |
952 for (substr = 0; substr <= m->max_decoded_substream; substr++) { | |
953 SubStream *s = &m->substream[substr]; | |
954 init_get_bits(&gb, buf, substream_data_len[substr] * 8); | |
955 | |
956 s->blockpos = 0; | |
957 do { | |
958 if (get_bits1(&gb)) { | |
959 if (get_bits1(&gb)) { | |
7198 | 960 /* A restart header should be present. */ |
7194 | 961 if (read_restart_header(m, &gb, buf, substr) < 0) |
962 goto next_substr; | |
963 s->restart_seen = 1; | |
964 } | |
965 | |
966 if (!s->restart_seen) { | |
967 av_log(m->avctx, AV_LOG_ERROR, | |
968 "No restart header present in substream %d.\n", | |
969 substr); | |
970 goto next_substr; | |
971 } | |
972 | |
973 if (read_decoding_params(m, &gb, substr) < 0) | |
974 goto next_substr; | |
975 } | |
976 | |
977 if (!s->restart_seen) { | |
978 av_log(m->avctx, AV_LOG_ERROR, | |
979 "No restart header present in substream %d.\n", | |
980 substr); | |
981 goto next_substr; | |
982 } | |
983 | |
984 if (read_block_data(m, &gb, substr) < 0) | |
985 return -1; | |
986 | |
987 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8) | |
988 && get_bits1(&gb) == 0); | |
989 | |
990 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
|
991 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
|
992 (show_bits_long(&gb, 32) == END_OF_STREAM || |
7194 | 993 show_bits_long(&gb, 20) == 0xd234e)) { |
994 skip_bits(&gb, 18); | |
995 if (substr == m->max_decoded_substream) | |
7198 | 996 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n"); |
7194 | 997 |
998 if (get_bits1(&gb)) { | |
999 int shorten_by = get_bits(&gb, 13); | |
1000 shorten_by = FFMIN(shorten_by, s->blockpos); | |
1001 s->blockpos -= shorten_by; | |
1002 } else | |
1003 skip_bits(&gb, 13); | |
1004 } | |
7262
e3822c61f2e4
mlpdec: Check for bits left before each read of End-of-Stream indicator and
ramiro
parents:
7199
diff
changeset
|
1005 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
|
1006 substream_parity_present[substr]) { |
7194 | 1007 uint8_t parity, checksum; |
1008 | |
7559
fd24c8628221
mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
7556
diff
changeset
|
1009 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2); |
7194 | 1010 if ((parity ^ get_bits(&gb, 8)) != 0xa9) |
1011 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 1012 "Substream %d parity check failed.\n", substr); |
7194 | 1013 |
7559
fd24c8628221
mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
7556
diff
changeset
|
1014 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2); |
7194 | 1015 if (checksum != get_bits(&gb, 8)) |
7198 | 1016 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n", |
7194 | 1017 substr); |
1018 } | |
1019 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) { | |
7198 | 1020 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", |
7194 | 1021 substr); |
1022 return -1; | |
1023 } | |
1024 | |
1025 next_substr: | |
1026 buf += substream_data_len[substr]; | |
1027 } | |
1028 | |
1029 rematrix_channels(m, m->max_decoded_substream); | |
1030 | |
1031 if (output_data(m, m->max_decoded_substream, data, data_size) < 0) | |
1032 return -1; | |
1033 | |
1034 return length; | |
1035 | |
1036 error: | |
1037 m->params_valid = 0; | |
1038 return -1; | |
1039 } | |
1040 | |
1041 AVCodec mlp_decoder = { | |
1042 "mlp", | |
1043 CODEC_TYPE_AUDIO, | |
1044 CODEC_ID_MLP, | |
1045 sizeof(MLPDecodeContext), | |
1046 mlp_decode_init, | |
1047 NULL, | |
1048 NULL, | |
1049 read_access_unit, | |
1050 .long_name = NULL_IF_CONFIG_SMALL("Meridian Lossless Packing"), | |
1051 }; | |
1052 |