Mercurial > libavcodec.hg
annotate mlpdec.c @ 9202:42e9c5b34503 libavcodec
Support "next parameter flags present" flag.
author | ramiro |
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date | Fri, 20 Mar 2009 13:04:14 +0000 |
parents | 753780c4875e |
children | 5074e89eb15f |
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 /** | |
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23 * @file libavcodec/mlpdec.c |
7194 | 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) | |
9202 | 79 #define PARAM_PRESENCE (1 << 0) |
7194 | 80 //@} |
81 | |
82 //@{ | |
83 /** matrix data */ | |
84 | |
85 //! Number of matrices to be applied. | |
86 uint8_t num_primitive_matrices; | |
87 | |
88 //! matrix output channel | |
89 uint8_t matrix_out_ch[MAX_MATRICES]; | |
90 | |
91 //! Whether the LSBs of the matrix output are encoded in the bitstream. | |
92 uint8_t lsb_bypass[MAX_MATRICES]; | |
93 //! Matrix coefficients, stored as 2.14 fixed point. | |
94 int32_t matrix_coeff[MAX_MATRICES][MAX_CHANNELS+2]; | |
95 //! Left shift to apply to noise values in 0x31eb substreams. | |
96 uint8_t matrix_noise_shift[MAX_MATRICES]; | |
97 //@} | |
98 | |
7198 | 99 //! Left shift to apply to Huffman-decoded residuals. |
7194 | 100 uint8_t quant_step_size[MAX_CHANNELS]; |
101 | |
7198 | 102 //! number of PCM samples in current audio block |
7194 | 103 uint16_t blocksize; |
104 //! Number of PCM samples decoded so far in this frame. | |
105 uint16_t blockpos; | |
106 | |
107 //! Left shift to apply to decoded PCM values to get final 24-bit output. | |
108 int8_t output_shift[MAX_CHANNELS]; | |
109 | |
110 //! Running XOR of all output samples. | |
111 int32_t lossless_check_data; | |
112 | |
113 } SubStream; | |
114 | |
115 typedef struct MLPDecodeContext { | |
116 AVCodecContext *avctx; | |
117 | |
118 //! Set if a valid major sync block has been read. Otherwise no decoding is possible. | |
119 uint8_t params_valid; | |
120 | |
121 //! Number of substreams contained within this stream. | |
122 uint8_t num_substreams; | |
123 | |
124 //! Index of the last substream to decode - further substreams are skipped. | |
125 uint8_t max_decoded_substream; | |
126 | |
7198 | 127 //! number of PCM samples contained in each frame |
7194 | 128 int access_unit_size; |
7198 | 129 //! next power of two above the number of samples in each frame |
7194 | 130 int access_unit_size_pow2; |
131 | |
132 SubStream substream[MAX_SUBSTREAMS]; | |
133 | |
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134 ChannelParams channel_params[MAX_CHANNELS]; |
7194 | 135 |
136 int8_t noise_buffer[MAX_BLOCKSIZE_POW2]; | |
137 int8_t bypassed_lsbs[MAX_BLOCKSIZE][MAX_CHANNELS]; | |
138 int32_t sample_buffer[MAX_BLOCKSIZE][MAX_CHANNELS+2]; | |
139 } MLPDecodeContext; | |
140 | |
141 static VLC huff_vlc[3]; | |
142 | |
143 /** Initialize static data, constant between all invocations of the codec. */ | |
144 | |
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145 static av_cold void init_static(void) |
7194 | 146 { |
147 INIT_VLC_STATIC(&huff_vlc[0], VLC_BITS, 18, | |
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148 &ff_mlp_huffman_tables[0][0][1], 2, 1, |
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149 &ff_mlp_huffman_tables[0][0][0], 2, 1, 512); |
7194 | 150 INIT_VLC_STATIC(&huff_vlc[1], VLC_BITS, 16, |
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151 &ff_mlp_huffman_tables[1][0][1], 2, 1, |
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152 &ff_mlp_huffman_tables[1][0][0], 2, 1, 512); |
7194 | 153 INIT_VLC_STATIC(&huff_vlc[2], VLC_BITS, 15, |
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154 &ff_mlp_huffman_tables[2][0][1], 2, 1, |
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155 &ff_mlp_huffman_tables[2][0][0], 2, 1, 512); |
7194 | 156 |
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157 ff_mlp_init_crc(); |
7194 | 158 } |
159 | |
160 static inline int32_t calculate_sign_huff(MLPDecodeContext *m, | |
161 unsigned int substr, unsigned int ch) | |
162 { | |
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163 ChannelParams *cp = &m->channel_params[ch]; |
7194 | 164 SubStream *s = &m->substream[substr]; |
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165 int lsb_bits = cp->huff_lsbs - s->quant_step_size[ch]; |
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166 int sign_shift = lsb_bits + (cp->codebook ? 2 - cp->codebook : -1); |
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167 int32_t sign_huff_offset = cp->huff_offset; |
7194 | 168 |
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169 if (cp->codebook > 0) |
7194 | 170 sign_huff_offset -= 7 << lsb_bits; |
171 | |
172 if (sign_shift >= 0) | |
173 sign_huff_offset -= 1 << sign_shift; | |
174 | |
175 return sign_huff_offset; | |
176 } | |
177 | |
178 /** Read a sample, consisting of either, both or neither of entropy-coded MSBs | |
179 * and plain LSBs. */ | |
180 | |
181 static inline int read_huff_channels(MLPDecodeContext *m, GetBitContext *gbp, | |
182 unsigned int substr, unsigned int pos) | |
183 { | |
184 SubStream *s = &m->substream[substr]; | |
185 unsigned int mat, channel; | |
186 | |
187 for (mat = 0; mat < s->num_primitive_matrices; mat++) | |
188 if (s->lsb_bypass[mat]) | |
189 m->bypassed_lsbs[pos + s->blockpos][mat] = get_bits1(gbp); | |
190 | |
191 for (channel = s->min_channel; channel <= s->max_channel; channel++) { | |
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192 ChannelParams *cp = &m->channel_params[channel]; |
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193 int codebook = cp->codebook; |
7194 | 194 int quant_step_size = s->quant_step_size[channel]; |
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195 int lsb_bits = cp->huff_lsbs - quant_step_size; |
7194 | 196 int result = 0; |
197 | |
198 if (codebook > 0) | |
199 result = get_vlc2(gbp, huff_vlc[codebook-1].table, | |
200 VLC_BITS, (9 + VLC_BITS - 1) / VLC_BITS); | |
201 | |
202 if (result < 0) | |
203 return -1; | |
204 | |
205 if (lsb_bits > 0) | |
206 result = (result << lsb_bits) + get_bits(gbp, lsb_bits); | |
207 | |
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208 result += cp->sign_huff_offset; |
7194 | 209 result <<= quant_step_size; |
210 | |
211 m->sample_buffer[pos + s->blockpos][channel] = result; | |
212 } | |
213 | |
214 return 0; | |
215 } | |
216 | |
217 static av_cold int mlp_decode_init(AVCodecContext *avctx) | |
218 { | |
219 MLPDecodeContext *m = avctx->priv_data; | |
220 int substr; | |
221 | |
222 init_static(); | |
223 m->avctx = avctx; | |
224 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
225 m->substream[substr].lossless_check_data = 0xffffffff; | |
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226 |
7194 | 227 return 0; |
228 } | |
229 | |
230 /** Read a major sync info header - contains high level information about | |
231 * the stream - sample rate, channel arrangement etc. Most of this | |
232 * information is not actually necessary for decoding, only for playback. | |
233 */ | |
234 | |
235 static int read_major_sync(MLPDecodeContext *m, GetBitContext *gb) | |
236 { | |
237 MLPHeaderInfo mh; | |
238 int substr; | |
239 | |
240 if (ff_mlp_read_major_sync(m->avctx, &mh, gb) != 0) | |
241 return -1; | |
242 | |
243 if (mh.group1_bits == 0) { | |
7198 | 244 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown bits per sample\n"); |
7194 | 245 return -1; |
246 } | |
247 if (mh.group2_bits > mh.group1_bits) { | |
248 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 249 "Channel group 2 cannot have more bits per sample than group 1.\n"); |
7194 | 250 return -1; |
251 } | |
252 | |
253 if (mh.group2_samplerate && mh.group2_samplerate != mh.group1_samplerate) { | |
254 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 255 "Channel groups with differing sample rates are not currently supported.\n"); |
7194 | 256 return -1; |
257 } | |
258 | |
259 if (mh.group1_samplerate == 0) { | |
7198 | 260 av_log(m->avctx, AV_LOG_ERROR, "invalid/unknown sampling rate\n"); |
7194 | 261 return -1; |
262 } | |
263 if (mh.group1_samplerate > MAX_SAMPLERATE) { | |
264 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 265 "Sampling rate %d is greater than the supported maximum (%d).\n", |
7194 | 266 mh.group1_samplerate, MAX_SAMPLERATE); |
267 return -1; | |
268 } | |
269 if (mh.access_unit_size > MAX_BLOCKSIZE) { | |
270 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 271 "Block size %d is greater than the supported maximum (%d).\n", |
7194 | 272 mh.access_unit_size, MAX_BLOCKSIZE); |
273 return -1; | |
274 } | |
275 if (mh.access_unit_size_pow2 > MAX_BLOCKSIZE_POW2) { | |
276 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 277 "Block size pow2 %d is greater than the supported maximum (%d).\n", |
7194 | 278 mh.access_unit_size_pow2, MAX_BLOCKSIZE_POW2); |
279 return -1; | |
280 } | |
281 | |
282 if (mh.num_substreams == 0) | |
283 return -1; | |
9201 | 284 if (m->avctx->codec_id == CODEC_ID_MLP && mh.num_substreams > 2) { |
285 av_log(m->avctx, AV_LOG_ERROR, "MLP only supports up to 2 substreams.\n"); | |
286 return -1; | |
287 } | |
7194 | 288 if (mh.num_substreams > MAX_SUBSTREAMS) { |
289 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 290 "Number of substreams %d is larger than the maximum supported " |
291 "by the decoder. %s\n", mh.num_substreams, sample_message); | |
7194 | 292 return -1; |
293 } | |
294 | |
295 m->access_unit_size = mh.access_unit_size; | |
296 m->access_unit_size_pow2 = mh.access_unit_size_pow2; | |
297 | |
298 m->num_substreams = mh.num_substreams; | |
299 m->max_decoded_substream = m->num_substreams - 1; | |
300 | |
301 m->avctx->sample_rate = mh.group1_samplerate; | |
302 m->avctx->frame_size = mh.access_unit_size; | |
303 | |
8274 | 304 m->avctx->bits_per_raw_sample = mh.group1_bits; |
8277 | 305 if (mh.group1_bits > 16) |
7194 | 306 m->avctx->sample_fmt = SAMPLE_FMT_S32; |
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307 else |
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308 m->avctx->sample_fmt = SAMPLE_FMT_S16; |
7194 | 309 |
310 m->params_valid = 1; | |
311 for (substr = 0; substr < MAX_SUBSTREAMS; substr++) | |
312 m->substream[substr].restart_seen = 0; | |
313 | |
314 return 0; | |
315 } | |
316 | |
317 /** Read a restart header from a block in a substream. This contains parameters | |
318 * required to decode the audio that do not change very often. Generally | |
319 * (always) present only in blocks following a major sync. */ | |
320 | |
321 static int read_restart_header(MLPDecodeContext *m, GetBitContext *gbp, | |
322 const uint8_t *buf, unsigned int substr) | |
323 { | |
324 SubStream *s = &m->substream[substr]; | |
325 unsigned int ch; | |
326 int sync_word, tmp; | |
327 uint8_t checksum; | |
328 uint8_t lossless_check; | |
329 int start_count = get_bits_count(gbp); | |
330 | |
331 sync_word = get_bits(gbp, 13); | |
332 | |
333 if (sync_word != 0x31ea >> 1) { | |
334 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 335 "restart header sync incorrect (got 0x%04x)\n", sync_word); |
7194 | 336 return -1; |
337 } | |
338 s->noise_type = get_bits1(gbp); | |
339 | |
340 skip_bits(gbp, 16); /* Output timestamp */ | |
341 | |
342 s->min_channel = get_bits(gbp, 4); | |
343 s->max_channel = get_bits(gbp, 4); | |
344 s->max_matrix_channel = get_bits(gbp, 4); | |
345 | |
346 if (s->min_channel > s->max_channel) { | |
347 av_log(m->avctx, AV_LOG_ERROR, | |
348 "Substream min channel cannot be greater than max channel.\n"); | |
349 return -1; | |
350 } | |
351 | |
352 if (m->avctx->request_channels > 0 | |
353 && s->max_channel + 1 >= m->avctx->request_channels | |
354 && substr < m->max_decoded_substream) { | |
355 av_log(m->avctx, AV_LOG_INFO, | |
356 "Extracting %d channel downmix from substream %d. " | |
357 "Further substreams will be skipped.\n", | |
358 s->max_channel + 1, substr); | |
359 m->max_decoded_substream = substr; | |
360 } | |
361 | |
362 s->noise_shift = get_bits(gbp, 4); | |
363 s->noisegen_seed = get_bits(gbp, 23); | |
364 | |
365 skip_bits(gbp, 19); | |
366 | |
367 s->data_check_present = get_bits1(gbp); | |
368 lossless_check = get_bits(gbp, 8); | |
369 if (substr == m->max_decoded_substream | |
370 && s->lossless_check_data != 0xffffffff) { | |
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371 tmp = xor_32_to_8(s->lossless_check_data); |
7194 | 372 if (tmp != lossless_check) |
373 av_log(m->avctx, AV_LOG_WARNING, | |
7198 | 374 "Lossless check failed - expected %02x, calculated %02x.\n", |
7194 | 375 lossless_check, tmp); |
376 else | |
7198 | 377 dprintf(m->avctx, "Lossless check passed for substream %d (%x).\n", |
7194 | 378 substr, tmp); |
379 } | |
380 | |
381 skip_bits(gbp, 16); | |
382 | |
383 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
384 int ch_assign = get_bits(gbp, 6); | |
385 dprintf(m->avctx, "ch_assign[%d][%d] = %d\n", substr, ch, | |
386 ch_assign); | |
387 if (ch_assign != ch) { | |
388 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 389 "Non-1:1 channel assignments are used in this stream. %s\n", |
7194 | 390 sample_message); |
391 return -1; | |
392 } | |
393 } | |
394 | |
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395 checksum = ff_mlp_restart_checksum(buf, get_bits_count(gbp) - start_count); |
7194 | 396 |
397 if (checksum != get_bits(gbp, 8)) | |
7198 | 398 av_log(m->avctx, AV_LOG_ERROR, "restart header checksum error\n"); |
7194 | 399 |
7198 | 400 /* Set default decoding parameters. */ |
7194 | 401 s->param_presence_flags = 0xff; |
402 s->num_primitive_matrices = 0; | |
403 s->blocksize = 8; | |
404 s->lossless_check_data = 0; | |
405 | |
406 memset(s->output_shift , 0, sizeof(s->output_shift )); | |
407 memset(s->quant_step_size, 0, sizeof(s->quant_step_size)); | |
408 | |
409 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
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410 ChannelParams *cp = &m->channel_params[ch]; |
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411 cp->filter_params[FIR].order = 0; |
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412 cp->filter_params[IIR].order = 0; |
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413 cp->filter_params[FIR].shift = 0; |
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414 cp->filter_params[IIR].shift = 0; |
7194 | 415 |
7198 | 416 /* Default audio coding is 24-bit raw PCM. */ |
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417 cp->huff_offset = 0; |
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418 cp->sign_huff_offset = (-1) << 23; |
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419 cp->codebook = 0; |
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420 cp->huff_lsbs = 24; |
7194 | 421 } |
422 | |
423 if (substr == m->max_decoded_substream) { | |
424 m->avctx->channels = s->max_channel + 1; | |
425 } | |
426 | |
427 return 0; | |
428 } | |
429 | |
430 /** Read parameters for one of the prediction filters. */ | |
431 | |
432 static int read_filter_params(MLPDecodeContext *m, GetBitContext *gbp, | |
433 unsigned int channel, unsigned int filter) | |
434 { | |
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435 FilterParams *fp = &m->channel_params[channel].filter_params[filter]; |
7194 | 436 const char fchar = filter ? 'I' : 'F'; |
437 int i, order; | |
438 | |
7198 | 439 // Filter is 0 for FIR, 1 for IIR. |
7194 | 440 assert(filter < 2); |
441 | |
442 order = get_bits(gbp, 4); | |
443 if (order > MAX_FILTER_ORDER) { | |
444 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 445 "%cIR filter order %d is greater than maximum %d.\n", |
7194 | 446 fchar, order, MAX_FILTER_ORDER); |
447 return -1; | |
448 } | |
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449 fp->order = order; |
7194 | 450 |
451 if (order > 0) { | |
452 int coeff_bits, coeff_shift; | |
453 | |
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454 fp->shift = get_bits(gbp, 4); |
7194 | 455 |
456 coeff_bits = get_bits(gbp, 5); | |
457 coeff_shift = get_bits(gbp, 3); | |
458 if (coeff_bits < 1 || coeff_bits > 16) { | |
459 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 460 "%cIR filter coeff_bits must be between 1 and 16.\n", |
7194 | 461 fchar); |
462 return -1; | |
463 } | |
464 if (coeff_bits + coeff_shift > 16) { | |
465 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 466 "Sum of coeff_bits and coeff_shift for %cIR filter must be 16 or less.\n", |
7194 | 467 fchar); |
468 return -1; | |
469 } | |
470 | |
471 for (i = 0; i < order; i++) | |
7553 | 472 fp->coeff[i] = get_sbits(gbp, coeff_bits) << coeff_shift; |
7194 | 473 |
474 if (get_bits1(gbp)) { | |
475 int state_bits, state_shift; | |
476 | |
477 if (filter == FIR) { | |
478 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 479 "FIR filter has state data specified.\n"); |
7194 | 480 return -1; |
481 } | |
482 | |
483 state_bits = get_bits(gbp, 4); | |
484 state_shift = get_bits(gbp, 4); | |
485 | |
7198 | 486 /* TODO: Check validity of state data. */ |
7194 | 487 |
488 for (i = 0; i < order; i++) | |
7553 | 489 fp->state[i] = get_sbits(gbp, state_bits) << state_shift; |
7194 | 490 } |
491 } | |
492 | |
493 return 0; | |
494 } | |
495 | |
496 /** Read decoding parameters that change more often than those in the restart | |
497 * header. */ | |
498 | |
499 static int read_decoding_params(MLPDecodeContext *m, GetBitContext *gbp, | |
500 unsigned int substr) | |
501 { | |
502 SubStream *s = &m->substream[substr]; | |
503 unsigned int mat, ch; | |
504 | |
9202 | 505 if (s->param_presence_flags & PARAM_PRESENCE) |
7194 | 506 if (get_bits1(gbp)) |
507 s->param_presence_flags = get_bits(gbp, 8); | |
508 | |
509 if (s->param_presence_flags & PARAM_BLOCKSIZE) | |
510 if (get_bits1(gbp)) { | |
511 s->blocksize = get_bits(gbp, 9); | |
512 if (s->blocksize > MAX_BLOCKSIZE) { | |
7198 | 513 av_log(m->avctx, AV_LOG_ERROR, "block size too large\n"); |
7194 | 514 s->blocksize = 0; |
515 return -1; | |
516 } | |
517 } | |
518 | |
519 if (s->param_presence_flags & PARAM_MATRIX) | |
520 if (get_bits1(gbp)) { | |
521 s->num_primitive_matrices = get_bits(gbp, 4); | |
522 | |
523 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
524 int frac_bits, max_chan; | |
525 s->matrix_out_ch[mat] = get_bits(gbp, 4); | |
526 frac_bits = get_bits(gbp, 4); | |
527 s->lsb_bypass [mat] = get_bits1(gbp); | |
528 | |
529 if (s->matrix_out_ch[mat] > s->max_channel) { | |
530 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 531 "Invalid channel %d specified as output from matrix.\n", |
7194 | 532 s->matrix_out_ch[mat]); |
533 return -1; | |
534 } | |
535 if (frac_bits > 14) { | |
536 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 537 "Too many fractional bits specified.\n"); |
7194 | 538 return -1; |
539 } | |
540 | |
541 max_chan = s->max_matrix_channel; | |
542 if (!s->noise_type) | |
543 max_chan+=2; | |
544 | |
545 for (ch = 0; ch <= max_chan; ch++) { | |
546 int coeff_val = 0; | |
547 if (get_bits1(gbp)) | |
548 coeff_val = get_sbits(gbp, frac_bits + 2); | |
549 | |
550 s->matrix_coeff[mat][ch] = coeff_val << (14 - frac_bits); | |
551 } | |
552 | |
553 if (s->noise_type) | |
554 s->matrix_noise_shift[mat] = get_bits(gbp, 4); | |
555 else | |
556 s->matrix_noise_shift[mat] = 0; | |
557 } | |
558 } | |
559 | |
560 if (s->param_presence_flags & PARAM_OUTSHIFT) | |
561 if (get_bits1(gbp)) | |
562 for (ch = 0; ch <= s->max_matrix_channel; ch++) { | |
563 s->output_shift[ch] = get_bits(gbp, 4); | |
564 dprintf(m->avctx, "output shift[%d] = %d\n", | |
565 ch, s->output_shift[ch]); | |
566 /* TODO: validate */ | |
567 } | |
568 | |
569 if (s->param_presence_flags & PARAM_QUANTSTEP) | |
570 if (get_bits1(gbp)) | |
571 for (ch = 0; ch <= s->max_channel; ch++) { | |
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572 ChannelParams *cp = &m->channel_params[ch]; |
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573 |
7194 | 574 s->quant_step_size[ch] = get_bits(gbp, 4); |
575 /* TODO: validate */ | |
576 | |
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577 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
7194 | 578 } |
579 | |
580 for (ch = s->min_channel; ch <= s->max_channel; ch++) | |
581 if (get_bits1(gbp)) { | |
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582 ChannelParams *cp = &m->channel_params[ch]; |
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583 FilterParams *fir = &cp->filter_params[FIR]; |
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584 FilterParams *iir = &cp->filter_params[IIR]; |
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585 |
7194 | 586 if (s->param_presence_flags & PARAM_FIR) |
587 if (get_bits1(gbp)) | |
588 if (read_filter_params(m, gbp, ch, FIR) < 0) | |
589 return -1; | |
590 | |
591 if (s->param_presence_flags & PARAM_IIR) | |
592 if (get_bits1(gbp)) | |
593 if (read_filter_params(m, gbp, ch, IIR) < 0) | |
594 return -1; | |
595 | |
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596 if (fir->order && iir->order && |
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597 fir->shift != iir->shift) { |
7194 | 598 av_log(m->avctx, AV_LOG_ERROR, |
7198 | 599 "FIR and IIR filters must use the same precision.\n"); |
7194 | 600 return -1; |
601 } | |
602 /* The FIR and IIR filters must have the same precision. | |
603 * To simplify the filtering code, only the precision of the | |
604 * FIR filter is considered. If only the IIR filter is employed, | |
605 * the FIR filter precision is set to that of the IIR filter, so | |
606 * that the filtering code can use it. */ | |
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607 if (!fir->order && iir->order) |
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608 fir->shift = iir->shift; |
7194 | 609 |
610 if (s->param_presence_flags & PARAM_HUFFOFFSET) | |
611 if (get_bits1(gbp)) | |
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612 cp->huff_offset = get_sbits(gbp, 15); |
7194 | 613 |
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614 cp->codebook = get_bits(gbp, 2); |
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615 cp->huff_lsbs = get_bits(gbp, 5); |
7194 | 616 |
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617 cp->sign_huff_offset = calculate_sign_huff(m, substr, ch); |
7194 | 618 |
619 /* TODO: validate */ | |
620 } | |
621 | |
622 return 0; | |
623 } | |
624 | |
625 #define MSB_MASK(bits) (-1u << bits) | |
626 | |
627 /** Generate PCM samples using the prediction filters and residual values | |
628 * read from the data stream, and update the filter state. */ | |
629 | |
630 static void filter_channel(MLPDecodeContext *m, unsigned int substr, | |
631 unsigned int channel) | |
632 { | |
633 SubStream *s = &m->substream[substr]; | |
634 int32_t filter_state_buffer[NUM_FILTERS][MAX_BLOCKSIZE + MAX_FILTER_ORDER]; | |
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635 FilterParams *fp[NUM_FILTERS] = { &m->channel_params[channel].filter_params[FIR], |
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636 &m->channel_params[channel].filter_params[IIR], }; |
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637 unsigned int filter_shift = fp[FIR]->shift; |
7194 | 638 int32_t mask = MSB_MASK(s->quant_step_size[channel]); |
639 int index = MAX_BLOCKSIZE; | |
640 int j, i; | |
641 | |
642 for (j = 0; j < NUM_FILTERS; j++) { | |
7553 | 643 memcpy(&filter_state_buffer[j][MAX_BLOCKSIZE], &fp[j]->state[0], |
7194 | 644 MAX_FILTER_ORDER * sizeof(int32_t)); |
645 } | |
646 | |
647 for (i = 0; i < s->blocksize; i++) { | |
648 int32_t residual = m->sample_buffer[i + s->blockpos][channel]; | |
649 unsigned int order; | |
650 int64_t accum = 0; | |
651 int32_t result; | |
652 | |
653 /* TODO: Move this code to DSPContext? */ | |
654 | |
655 for (j = 0; j < NUM_FILTERS; j++) | |
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656 for (order = 0; order < fp[j]->order; order++) |
7194 | 657 accum += (int64_t)filter_state_buffer[j][index + order] * |
7553 | 658 fp[j]->coeff[order]; |
7194 | 659 |
660 accum = accum >> filter_shift; | |
661 result = (accum + residual) & mask; | |
662 | |
663 --index; | |
664 | |
665 filter_state_buffer[FIR][index] = result; | |
666 filter_state_buffer[IIR][index] = result - accum; | |
667 | |
668 m->sample_buffer[i + s->blockpos][channel] = result; | |
669 } | |
670 | |
671 for (j = 0; j < NUM_FILTERS; j++) { | |
7553 | 672 memcpy(&fp[j]->state[0], &filter_state_buffer[j][index], |
7194 | 673 MAX_FILTER_ORDER * sizeof(int32_t)); |
674 } | |
675 } | |
676 | |
677 /** Read a block of PCM residual data (or actual if no filtering active). */ | |
678 | |
679 static int read_block_data(MLPDecodeContext *m, GetBitContext *gbp, | |
680 unsigned int substr) | |
681 { | |
682 SubStream *s = &m->substream[substr]; | |
683 unsigned int i, ch, expected_stream_pos = 0; | |
684 | |
685 if (s->data_check_present) { | |
686 expected_stream_pos = get_bits_count(gbp); | |
687 expected_stream_pos += get_bits(gbp, 16); | |
688 av_log(m->avctx, AV_LOG_WARNING, "This file contains some features " | |
689 "we have not tested yet. %s\n", sample_message); | |
690 } | |
691 | |
692 if (s->blockpos + s->blocksize > m->access_unit_size) { | |
7198 | 693 av_log(m->avctx, AV_LOG_ERROR, "too many audio samples in frame\n"); |
7194 | 694 return -1; |
695 } | |
696 | |
697 memset(&m->bypassed_lsbs[s->blockpos][0], 0, | |
698 s->blocksize * sizeof(m->bypassed_lsbs[0])); | |
699 | |
700 for (i = 0; i < s->blocksize; i++) { | |
701 if (read_huff_channels(m, gbp, substr, i) < 0) | |
702 return -1; | |
703 } | |
704 | |
705 for (ch = s->min_channel; ch <= s->max_channel; ch++) { | |
706 filter_channel(m, substr, ch); | |
707 } | |
708 | |
709 s->blockpos += s->blocksize; | |
710 | |
711 if (s->data_check_present) { | |
712 if (get_bits_count(gbp) != expected_stream_pos) | |
7198 | 713 av_log(m->avctx, AV_LOG_ERROR, "block data length mismatch\n"); |
7194 | 714 skip_bits(gbp, 8); |
715 } | |
716 | |
717 return 0; | |
718 } | |
719 | |
7198 | 720 /** Data table used for TrueHD noise generation function. */ |
7194 | 721 |
722 static const int8_t noise_table[256] = { | |
723 30, 51, 22, 54, 3, 7, -4, 38, 14, 55, 46, 81, 22, 58, -3, 2, | |
724 52, 31, -7, 51, 15, 44, 74, 30, 85, -17, 10, 33, 18, 80, 28, 62, | |
725 10, 32, 23, 69, 72, 26, 35, 17, 73, 60, 8, 56, 2, 6, -2, -5, | |
726 51, 4, 11, 50, 66, 76, 21, 44, 33, 47, 1, 26, 64, 48, 57, 40, | |
727 38, 16, -10, -28, 92, 22, -18, 29, -10, 5, -13, 49, 19, 24, 70, 34, | |
728 61, 48, 30, 14, -6, 25, 58, 33, 42, 60, 67, 17, 54, 17, 22, 30, | |
729 67, 44, -9, 50, -11, 43, 40, 32, 59, 82, 13, 49, -14, 55, 60, 36, | |
730 48, 49, 31, 47, 15, 12, 4, 65, 1, 23, 29, 39, 45, -2, 84, 69, | |
731 0, 72, 37, 57, 27, 41, -15, -16, 35, 31, 14, 61, 24, 0, 27, 24, | |
732 16, 41, 55, 34, 53, 9, 56, 12, 25, 29, 53, 5, 20, -20, -8, 20, | |
733 13, 28, -3, 78, 38, 16, 11, 62, 46, 29, 21, 24, 46, 65, 43, -23, | |
734 89, 18, 74, 21, 38, -12, 19, 12, -19, 8, 15, 33, 4, 57, 9, -8, | |
735 36, 35, 26, 28, 7, 83, 63, 79, 75, 11, 3, 87, 37, 47, 34, 40, | |
736 39, 19, 20, 42, 27, 34, 39, 77, 13, 42, 59, 64, 45, -1, 32, 37, | |
737 45, -5, 53, -6, 7, 36, 50, 23, 6, 32, 9, -21, 18, 71, 27, 52, | |
738 -25, 31, 35, 42, -1, 68, 63, 52, 26, 43, 66, 37, 41, 25, 40, 70, | |
739 }; | |
740 | |
741 /** Noise generation functions. | |
742 * I'm not sure what these are for - they seem to be some kind of pseudorandom | |
743 * sequence generators, used to generate noise data which is used when the | |
744 * channels are rematrixed. I'm not sure if they provide a practical benefit | |
745 * to compression, or just obfuscate the decoder. Are they for some kind of | |
746 * dithering? */ | |
747 | |
748 /** Generate two channels of noise, used in the matrix when | |
749 * restart sync word == 0x31ea. */ | |
750 | |
751 static void generate_2_noise_channels(MLPDecodeContext *m, unsigned int substr) | |
752 { | |
753 SubStream *s = &m->substream[substr]; | |
754 unsigned int i; | |
755 uint32_t seed = s->noisegen_seed; | |
756 unsigned int maxchan = s->max_matrix_channel; | |
757 | |
758 for (i = 0; i < s->blockpos; i++) { | |
759 uint16_t seed_shr7 = seed >> 7; | |
760 m->sample_buffer[i][maxchan+1] = ((int8_t)(seed >> 15)) << s->noise_shift; | |
761 m->sample_buffer[i][maxchan+2] = ((int8_t) seed_shr7) << s->noise_shift; | |
762 | |
763 seed = (seed << 16) ^ seed_shr7 ^ (seed_shr7 << 5); | |
764 } | |
765 | |
766 s->noisegen_seed = seed; | |
767 } | |
768 | |
769 /** Generate a block of noise, used when restart sync word == 0x31eb. */ | |
770 | |
771 static void fill_noise_buffer(MLPDecodeContext *m, unsigned int substr) | |
772 { | |
773 SubStream *s = &m->substream[substr]; | |
774 unsigned int i; | |
775 uint32_t seed = s->noisegen_seed; | |
776 | |
777 for (i = 0; i < m->access_unit_size_pow2; i++) { | |
778 uint8_t seed_shr15 = seed >> 15; | |
779 m->noise_buffer[i] = noise_table[seed_shr15]; | |
780 seed = (seed << 8) ^ seed_shr15 ^ (seed_shr15 << 5); | |
781 } | |
782 | |
783 s->noisegen_seed = seed; | |
784 } | |
785 | |
786 | |
787 /** Apply the channel matrices in turn to reconstruct the original audio | |
788 * samples. */ | |
789 | |
790 static void rematrix_channels(MLPDecodeContext *m, unsigned int substr) | |
791 { | |
792 SubStream *s = &m->substream[substr]; | |
793 unsigned int mat, src_ch, i; | |
794 unsigned int maxchan; | |
795 | |
796 maxchan = s->max_matrix_channel; | |
797 if (!s->noise_type) { | |
798 generate_2_noise_channels(m, substr); | |
799 maxchan += 2; | |
800 } else { | |
801 fill_noise_buffer(m, substr); | |
802 } | |
803 | |
804 for (mat = 0; mat < s->num_primitive_matrices; mat++) { | |
805 int matrix_noise_shift = s->matrix_noise_shift[mat]; | |
806 unsigned int dest_ch = s->matrix_out_ch[mat]; | |
807 int32_t mask = MSB_MASK(s->quant_step_size[dest_ch]); | |
808 | |
809 /* TODO: DSPContext? */ | |
810 | |
811 for (i = 0; i < s->blockpos; i++) { | |
812 int64_t accum = 0; | |
813 for (src_ch = 0; src_ch <= maxchan; src_ch++) { | |
814 accum += (int64_t)m->sample_buffer[i][src_ch] | |
815 * s->matrix_coeff[mat][src_ch]; | |
816 } | |
817 if (matrix_noise_shift) { | |
818 uint32_t index = s->num_primitive_matrices - mat; | |
819 index = (i * (index * 2 + 1) + index) & (m->access_unit_size_pow2 - 1); | |
820 accum += m->noise_buffer[index] << (matrix_noise_shift + 7); | |
821 } | |
822 m->sample_buffer[i][dest_ch] = ((accum >> 14) & mask) | |
823 + m->bypassed_lsbs[i][mat]; | |
824 } | |
825 } | |
826 } | |
827 | |
828 /** Write the audio data into the output buffer. */ | |
829 | |
830 static int output_data_internal(MLPDecodeContext *m, unsigned int substr, | |
831 uint8_t *data, unsigned int *data_size, int is32) | |
832 { | |
833 SubStream *s = &m->substream[substr]; | |
834 unsigned int i, ch = 0; | |
835 int32_t *data_32 = (int32_t*) data; | |
836 int16_t *data_16 = (int16_t*) data; | |
837 | |
838 if (*data_size < (s->max_channel + 1) * s->blockpos * (is32 ? 4 : 2)) | |
839 return -1; | |
840 | |
841 for (i = 0; i < s->blockpos; i++) { | |
842 for (ch = 0; ch <= s->max_channel; ch++) { | |
843 int32_t sample = m->sample_buffer[i][ch] << s->output_shift[ch]; | |
844 s->lossless_check_data ^= (sample & 0xffffff) << ch; | |
845 if (is32) *data_32++ = sample << 8; | |
846 else *data_16++ = sample >> 8; | |
847 } | |
848 } | |
849 | |
850 *data_size = i * ch * (is32 ? 4 : 2); | |
851 | |
852 return 0; | |
853 } | |
854 | |
855 static int output_data(MLPDecodeContext *m, unsigned int substr, | |
856 uint8_t *data, unsigned int *data_size) | |
857 { | |
858 if (m->avctx->sample_fmt == SAMPLE_FMT_S32) | |
859 return output_data_internal(m, substr, data, data_size, 1); | |
860 else | |
861 return output_data_internal(m, substr, data, data_size, 0); | |
862 } | |
863 | |
864 | |
865 /** Read an access unit from the stream. | |
866 * Returns < 0 on error, 0 if not enough data is present in the input stream | |
867 * otherwise returns the number of bytes consumed. */ | |
868 | |
869 static int read_access_unit(AVCodecContext *avctx, void* data, int *data_size, | |
870 const uint8_t *buf, int buf_size) | |
871 { | |
872 MLPDecodeContext *m = avctx->priv_data; | |
873 GetBitContext gb; | |
874 unsigned int length, substr; | |
875 unsigned int substream_start; | |
876 unsigned int header_size = 4; | |
877 unsigned int substr_header_size = 0; | |
878 uint8_t substream_parity_present[MAX_SUBSTREAMS]; | |
879 uint16_t substream_data_len[MAX_SUBSTREAMS]; | |
880 uint8_t parity_bits; | |
881 | |
882 if (buf_size < 4) | |
883 return 0; | |
884 | |
885 length = (AV_RB16(buf) & 0xfff) * 2; | |
886 | |
887 if (length > buf_size) | |
888 return -1; | |
889 | |
890 init_get_bits(&gb, (buf + 4), (length - 4) * 8); | |
891 | |
892 if (show_bits_long(&gb, 31) == (0xf8726fba >> 1)) { | |
7198 | 893 dprintf(m->avctx, "Found major sync.\n"); |
7194 | 894 if (read_major_sync(m, &gb) < 0) |
895 goto error; | |
896 header_size += 28; | |
897 } | |
898 | |
899 if (!m->params_valid) { | |
900 av_log(m->avctx, AV_LOG_WARNING, | |
7198 | 901 "Stream parameters not seen; skipping frame.\n"); |
7194 | 902 *data_size = 0; |
903 return length; | |
904 } | |
905 | |
906 substream_start = 0; | |
907 | |
908 for (substr = 0; substr < m->num_substreams; substr++) { | |
909 int extraword_present, checkdata_present, end; | |
910 | |
911 extraword_present = get_bits1(&gb); | |
912 skip_bits1(&gb); | |
913 checkdata_present = get_bits1(&gb); | |
914 skip_bits1(&gb); | |
915 | |
916 end = get_bits(&gb, 12) * 2; | |
917 | |
918 substr_header_size += 2; | |
919 | |
920 if (extraword_present) { | |
921 skip_bits(&gb, 16); | |
922 substr_header_size += 2; | |
923 } | |
924 | |
925 if (end + header_size + substr_header_size > length) { | |
926 av_log(m->avctx, AV_LOG_ERROR, | |
927 "Indicated length of substream %d data goes off end of " | |
928 "packet.\n", substr); | |
929 end = length - header_size - substr_header_size; | |
930 } | |
931 | |
932 if (end < substream_start) { | |
933 av_log(avctx, AV_LOG_ERROR, | |
934 "Indicated end offset of substream %d data " | |
935 "is smaller than calculated start offset.\n", | |
936 substr); | |
937 goto error; | |
938 } | |
939 | |
940 if (substr > m->max_decoded_substream) | |
941 continue; | |
942 | |
943 substream_parity_present[substr] = checkdata_present; | |
944 substream_data_len[substr] = end - substream_start; | |
945 substream_start = end; | |
946 } | |
947 | |
7559
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mlp: Split common code from parser and decoder to be used by encoder.
ramiro
parents:
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diff
changeset
|
948 parity_bits = ff_mlp_calculate_parity(buf, 4); |
fd24c8628221
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ramiro
parents:
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diff
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|
949 parity_bits ^= ff_mlp_calculate_parity(buf + header_size, substr_header_size); |
7194 | 950 |
951 if ((((parity_bits >> 4) ^ parity_bits) & 0xF) != 0xF) { | |
952 av_log(avctx, AV_LOG_ERROR, "Parity check failed.\n"); | |
953 goto error; | |
954 } | |
955 | |
956 buf += header_size + substr_header_size; | |
957 | |
958 for (substr = 0; substr <= m->max_decoded_substream; substr++) { | |
959 SubStream *s = &m->substream[substr]; | |
960 init_get_bits(&gb, buf, substream_data_len[substr] * 8); | |
961 | |
962 s->blockpos = 0; | |
963 do { | |
964 if (get_bits1(&gb)) { | |
965 if (get_bits1(&gb)) { | |
7198 | 966 /* A restart header should be present. */ |
7194 | 967 if (read_restart_header(m, &gb, buf, substr) < 0) |
968 goto next_substr; | |
969 s->restart_seen = 1; | |
970 } | |
971 | |
972 if (!s->restart_seen) { | |
973 av_log(m->avctx, AV_LOG_ERROR, | |
974 "No restart header present in substream %d.\n", | |
975 substr); | |
976 goto next_substr; | |
977 } | |
978 | |
979 if (read_decoding_params(m, &gb, substr) < 0) | |
980 goto next_substr; | |
981 } | |
982 | |
983 if (!s->restart_seen) { | |
984 av_log(m->avctx, AV_LOG_ERROR, | |
985 "No restart header present in substream %d.\n", | |
986 substr); | |
987 goto next_substr; | |
988 } | |
989 | |
990 if (read_block_data(m, &gb, substr) < 0) | |
991 return -1; | |
992 | |
993 } while ((get_bits_count(&gb) < substream_data_len[substr] * 8) | |
994 && get_bits1(&gb) == 0); | |
995 | |
996 skip_bits(&gb, (-get_bits_count(&gb)) & 15); | |
7262
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mlpdec: Check for bits left before each read of End-of-Stream indicator and
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7199
diff
changeset
|
997 if (substream_data_len[substr] * 8 - get_bits_count(&gb) >= 32 && |
7575
d73cd240d68c
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7566
diff
changeset
|
998 (show_bits_long(&gb, 32) == END_OF_STREAM || |
7194 | 999 show_bits_long(&gb, 20) == 0xd234e)) { |
1000 skip_bits(&gb, 18); | |
1001 if (substr == m->max_decoded_substream) | |
7198 | 1002 av_log(m->avctx, AV_LOG_INFO, "End of stream indicated.\n"); |
7194 | 1003 |
1004 if (get_bits1(&gb)) { | |
1005 int shorten_by = get_bits(&gb, 13); | |
1006 shorten_by = FFMIN(shorten_by, s->blockpos); | |
1007 s->blockpos -= shorten_by; | |
1008 } else | |
1009 skip_bits(&gb, 13); | |
1010 } | |
7262
e3822c61f2e4
mlpdec: Check for bits left before each read of End-of-Stream indicator and
ramiro
parents:
7199
diff
changeset
|
1011 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
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changeset
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1012 substream_parity_present[substr]) { |
7194 | 1013 uint8_t parity, checksum; |
1014 | |
7559
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ramiro
parents:
7556
diff
changeset
|
1015 parity = ff_mlp_calculate_parity(buf, substream_data_len[substr] - 2); |
7194 | 1016 if ((parity ^ get_bits(&gb, 8)) != 0xa9) |
1017 av_log(m->avctx, AV_LOG_ERROR, | |
7198 | 1018 "Substream %d parity check failed.\n", substr); |
7194 | 1019 |
7559
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ramiro
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7556
diff
changeset
|
1020 checksum = ff_mlp_checksum8(buf, substream_data_len[substr] - 2); |
7194 | 1021 if (checksum != get_bits(&gb, 8)) |
7198 | 1022 av_log(m->avctx, AV_LOG_ERROR, "Substream %d checksum failed.\n", |
7194 | 1023 substr); |
1024 } | |
1025 if (substream_data_len[substr] * 8 != get_bits_count(&gb)) { | |
7198 | 1026 av_log(m->avctx, AV_LOG_ERROR, "substream %d length mismatch\n", |
7194 | 1027 substr); |
1028 return -1; | |
1029 } | |
1030 | |
1031 next_substr: | |
1032 buf += substream_data_len[substr]; | |
1033 } | |
1034 | |
1035 rematrix_channels(m, m->max_decoded_substream); | |
1036 | |
1037 if (output_data(m, m->max_decoded_substream, data, data_size) < 0) | |
1038 return -1; | |
1039 | |
1040 return length; | |
1041 | |
1042 error: | |
1043 m->params_valid = 0; | |
1044 return -1; | |
1045 } | |
1046 | |
9190 | 1047 #if CONFIG_MLP_DECODER |
7194 | 1048 AVCodec mlp_decoder = { |
1049 "mlp", | |
1050 CODEC_TYPE_AUDIO, | |
1051 CODEC_ID_MLP, | |
1052 sizeof(MLPDecodeContext), | |
1053 mlp_decode_init, | |
1054 NULL, | |
1055 NULL, | |
1056 read_access_unit, | |
9190 | 1057 .long_name = NULL_IF_CONFIG_SMALL("MLP (Meridian Lossless Packing)"), |
7194 | 1058 }; |
9190 | 1059 #endif /* CONFIG_MLP_DECODER */ |
7194 | 1060 |
9190 | 1061 #if CONFIG_TRUEHD_DECODER |
1062 AVCodec truehd_decoder = { | |
1063 "truehd", | |
1064 CODEC_TYPE_AUDIO, | |
1065 CODEC_ID_TRUEHD, | |
1066 sizeof(MLPDecodeContext), | |
1067 mlp_decode_init, | |
1068 NULL, | |
1069 NULL, | |
1070 read_access_unit, | |
1071 .long_name = NULL_IF_CONFIG_SMALL("TrueHD"), | |
1072 }; | |
1073 #endif /* CONFIG_TRUEHD_DECODER */ |