Libav 0.7.1
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00001 /* 00002 * ALAC (Apple Lossless Audio Codec) decoder 00003 * Copyright (c) 2005 David Hammerton 00004 * 00005 * This file is part of Libav. 00006 * 00007 * Libav is free software; you can redistribute it and/or 00008 * modify it under the terms of the GNU Lesser General Public 00009 * License as published by the Free Software Foundation; either 00010 * version 2.1 of the License, or (at your option) any later version. 00011 * 00012 * Libav is distributed in the hope that it will be useful, 00013 * but WITHOUT ANY WARRANTY; without even the implied warranty of 00014 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 00015 * Lesser General Public License for more details. 00016 * 00017 * You should have received a copy of the GNU Lesser General Public 00018 * License along with Libav; if not, write to the Free Software 00019 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 00020 */ 00021 00055 #include "avcodec.h" 00056 #include "get_bits.h" 00057 #include "bytestream.h" 00058 #include "unary.h" 00059 #include "mathops.h" 00060 00061 #define ALAC_EXTRADATA_SIZE 36 00062 #define MAX_CHANNELS 2 00063 00064 typedef struct { 00065 00066 AVCodecContext *avctx; 00067 GetBitContext gb; 00068 00069 int numchannels; 00070 int bytespersample; 00071 00072 /* buffers */ 00073 int32_t *predicterror_buffer[MAX_CHANNELS]; 00074 00075 int32_t *outputsamples_buffer[MAX_CHANNELS]; 00076 00077 int32_t *wasted_bits_buffer[MAX_CHANNELS]; 00078 00079 /* stuff from setinfo */ 00080 uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ 00081 uint8_t setinfo_sample_size; /* 0x10 */ 00082 uint8_t setinfo_rice_historymult; /* 0x28 */ 00083 uint8_t setinfo_rice_initialhistory; /* 0x0a */ 00084 uint8_t setinfo_rice_kmodifier; /* 0x0e */ 00085 /* end setinfo stuff */ 00086 00087 int wasted_bits; 00088 } ALACContext; 00089 00090 static void allocate_buffers(ALACContext *alac) 00091 { 00092 int chan; 00093 for (chan = 0; chan < MAX_CHANNELS; chan++) { 00094 alac->predicterror_buffer[chan] = 00095 av_malloc(alac->setinfo_max_samples_per_frame * 4); 00096 00097 alac->outputsamples_buffer[chan] = 00098 av_malloc(alac->setinfo_max_samples_per_frame * 4); 00099 00100 alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4); 00101 } 00102 } 00103 00104 static int alac_set_info(ALACContext *alac) 00105 { 00106 const unsigned char *ptr = alac->avctx->extradata; 00107 00108 ptr += 4; /* size */ 00109 ptr += 4; /* alac */ 00110 ptr += 4; /* 0 ? */ 00111 00112 if(AV_RB32(ptr) >= UINT_MAX/4){ 00113 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n"); 00114 return -1; 00115 } 00116 00117 /* buffer size / 2 ? */ 00118 alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr); 00119 ptr++; /* ??? */ 00120 alac->setinfo_sample_size = *ptr++; 00121 if (alac->setinfo_sample_size > 32) { 00122 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n"); 00123 return -1; 00124 } 00125 alac->setinfo_rice_historymult = *ptr++; 00126 alac->setinfo_rice_initialhistory = *ptr++; 00127 alac->setinfo_rice_kmodifier = *ptr++; 00128 ptr++; /* channels? */ 00129 bytestream_get_be16(&ptr); /* ??? */ 00130 bytestream_get_be32(&ptr); /* max coded frame size */ 00131 bytestream_get_be32(&ptr); /* bitrate ? */ 00132 bytestream_get_be32(&ptr); /* samplerate */ 00133 00134 allocate_buffers(alac); 00135 00136 return 0; 00137 } 00138 00139 static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){ 00140 /* read x - number of 1s before 0 represent the rice */ 00141 int x = get_unary_0_9(gb); 00142 00143 if (x > 8) { /* RICE THRESHOLD */ 00144 /* use alternative encoding */ 00145 x = get_bits(gb, readsamplesize); 00146 } else { 00147 if (k >= limit) 00148 k = limit; 00149 00150 if (k != 1) { 00151 int extrabits = show_bits(gb, k); 00152 00153 /* multiply x by 2^k - 1, as part of their strange algorithm */ 00154 x = (x << k) - x; 00155 00156 if (extrabits > 1) { 00157 x += extrabits - 1; 00158 skip_bits(gb, k); 00159 } else 00160 skip_bits(gb, k - 1); 00161 } 00162 } 00163 return x; 00164 } 00165 00166 static void bastardized_rice_decompress(ALACContext *alac, 00167 int32_t *output_buffer, 00168 int output_size, 00169 int readsamplesize, /* arg_10 */ 00170 int rice_initialhistory, /* arg424->b */ 00171 int rice_kmodifier, /* arg424->d */ 00172 int rice_historymult, /* arg424->c */ 00173 int rice_kmodifier_mask /* arg424->e */ 00174 ) 00175 { 00176 int output_count; 00177 unsigned int history = rice_initialhistory; 00178 int sign_modifier = 0; 00179 00180 for (output_count = 0; output_count < output_size; output_count++) { 00181 int32_t x; 00182 int32_t x_modified; 00183 int32_t final_val; 00184 00185 /* standard rice encoding */ 00186 int k; /* size of extra bits */ 00187 00188 /* read k, that is bits as is */ 00189 k = av_log2((history >> 9) + 3); 00190 x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize); 00191 00192 x_modified = sign_modifier + x; 00193 final_val = (x_modified + 1) / 2; 00194 if (x_modified & 1) final_val *= -1; 00195 00196 output_buffer[output_count] = final_val; 00197 00198 sign_modifier = 0; 00199 00200 /* now update the history */ 00201 history += x_modified * rice_historymult 00202 - ((history * rice_historymult) >> 9); 00203 00204 if (x_modified > 0xffff) 00205 history = 0xffff; 00206 00207 /* special case: there may be compressed blocks of 0 */ 00208 if ((history < 128) && (output_count+1 < output_size)) { 00209 int k; 00210 unsigned int block_size; 00211 00212 sign_modifier = 1; 00213 00214 k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */); 00215 00216 block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16); 00217 00218 if (block_size > 0) { 00219 if(block_size >= output_size - output_count){ 00220 av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count); 00221 block_size= output_size - output_count - 1; 00222 } 00223 memset(&output_buffer[output_count+1], 0, block_size * 4); 00224 output_count += block_size; 00225 } 00226 00227 if (block_size > 0xffff) 00228 sign_modifier = 0; 00229 00230 history = 0; 00231 } 00232 } 00233 } 00234 00235 static inline int sign_only(int v) 00236 { 00237 return v ? FFSIGN(v) : 0; 00238 } 00239 00240 static void predictor_decompress_fir_adapt(int32_t *error_buffer, 00241 int32_t *buffer_out, 00242 int output_size, 00243 int readsamplesize, 00244 int16_t *predictor_coef_table, 00245 int predictor_coef_num, 00246 int predictor_quantitization) 00247 { 00248 int i; 00249 00250 /* first sample always copies */ 00251 *buffer_out = *error_buffer; 00252 00253 if (!predictor_coef_num) { 00254 if (output_size <= 1) 00255 return; 00256 00257 memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4); 00258 return; 00259 } 00260 00261 if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */ 00262 /* second-best case scenario for fir decompression, 00263 * error describes a small difference from the previous sample only 00264 */ 00265 if (output_size <= 1) 00266 return; 00267 for (i = 0; i < output_size - 1; i++) { 00268 int32_t prev_value; 00269 int32_t error_value; 00270 00271 prev_value = buffer_out[i]; 00272 error_value = error_buffer[i+1]; 00273 buffer_out[i+1] = 00274 sign_extend((prev_value + error_value), readsamplesize); 00275 } 00276 return; 00277 } 00278 00279 /* read warm-up samples */ 00280 if (predictor_coef_num > 0) 00281 for (i = 0; i < predictor_coef_num; i++) { 00282 int32_t val; 00283 00284 val = buffer_out[i] + error_buffer[i+1]; 00285 val = sign_extend(val, readsamplesize); 00286 buffer_out[i+1] = val; 00287 } 00288 00289 #if 0 00290 /* 4 and 8 are very common cases (the only ones i've seen). these 00291 * should be unrolled and optimized 00292 */ 00293 if (predictor_coef_num == 4) { 00294 /* FIXME: optimized general case */ 00295 return; 00296 } 00297 00298 if (predictor_coef_table == 8) { 00299 /* FIXME: optimized general case */ 00300 return; 00301 } 00302 #endif 00303 00304 /* general case */ 00305 if (predictor_coef_num > 0) { 00306 for (i = predictor_coef_num + 1; i < output_size; i++) { 00307 int j; 00308 int sum = 0; 00309 int outval; 00310 int error_val = error_buffer[i]; 00311 00312 for (j = 0; j < predictor_coef_num; j++) { 00313 sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) * 00314 predictor_coef_table[j]; 00315 } 00316 00317 outval = (1 << (predictor_quantitization-1)) + sum; 00318 outval = outval >> predictor_quantitization; 00319 outval = outval + buffer_out[0] + error_val; 00320 outval = sign_extend(outval, readsamplesize); 00321 00322 buffer_out[predictor_coef_num+1] = outval; 00323 00324 if (error_val > 0) { 00325 int predictor_num = predictor_coef_num - 1; 00326 00327 while (predictor_num >= 0 && error_val > 0) { 00328 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; 00329 int sign = sign_only(val); 00330 00331 predictor_coef_table[predictor_num] -= sign; 00332 00333 val *= sign; /* absolute value */ 00334 00335 error_val -= ((val >> predictor_quantitization) * 00336 (predictor_coef_num - predictor_num)); 00337 00338 predictor_num--; 00339 } 00340 } else if (error_val < 0) { 00341 int predictor_num = predictor_coef_num - 1; 00342 00343 while (predictor_num >= 0 && error_val < 0) { 00344 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; 00345 int sign = - sign_only(val); 00346 00347 predictor_coef_table[predictor_num] -= sign; 00348 00349 val *= sign; /* neg value */ 00350 00351 error_val -= ((val >> predictor_quantitization) * 00352 (predictor_coef_num - predictor_num)); 00353 00354 predictor_num--; 00355 } 00356 } 00357 00358 buffer_out++; 00359 } 00360 } 00361 } 00362 00363 static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS], 00364 int16_t *buffer_out, 00365 int numchannels, int numsamples, 00366 uint8_t interlacing_shift, 00367 uint8_t interlacing_leftweight) 00368 { 00369 int i; 00370 if (numsamples <= 0) 00371 return; 00372 00373 /* weighted interlacing */ 00374 if (interlacing_leftweight) { 00375 for (i = 0; i < numsamples; i++) { 00376 int32_t a, b; 00377 00378 a = buffer[0][i]; 00379 b = buffer[1][i]; 00380 00381 a -= (b * interlacing_leftweight) >> interlacing_shift; 00382 b += a; 00383 00384 buffer_out[i*numchannels] = b; 00385 buffer_out[i*numchannels + 1] = a; 00386 } 00387 00388 return; 00389 } 00390 00391 /* otherwise basic interlacing took place */ 00392 for (i = 0; i < numsamples; i++) { 00393 int16_t left, right; 00394 00395 left = buffer[0][i]; 00396 right = buffer[1][i]; 00397 00398 buffer_out[i*numchannels] = left; 00399 buffer_out[i*numchannels + 1] = right; 00400 } 00401 } 00402 00403 static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS], 00404 int32_t *buffer_out, 00405 int32_t *wasted_bits_buffer[MAX_CHANNELS], 00406 int wasted_bits, 00407 int numchannels, int numsamples, 00408 uint8_t interlacing_shift, 00409 uint8_t interlacing_leftweight) 00410 { 00411 int i; 00412 00413 if (numsamples <= 0) 00414 return; 00415 00416 /* weighted interlacing */ 00417 if (interlacing_leftweight) { 00418 for (i = 0; i < numsamples; i++) { 00419 int32_t a, b; 00420 00421 a = buffer[0][i]; 00422 b = buffer[1][i]; 00423 00424 a -= (b * interlacing_leftweight) >> interlacing_shift; 00425 b += a; 00426 00427 if (wasted_bits) { 00428 b = (b << wasted_bits) | wasted_bits_buffer[0][i]; 00429 a = (a << wasted_bits) | wasted_bits_buffer[1][i]; 00430 } 00431 00432 buffer_out[i * numchannels] = b << 8; 00433 buffer_out[i * numchannels + 1] = a << 8; 00434 } 00435 } else { 00436 for (i = 0; i < numsamples; i++) { 00437 int32_t left, right; 00438 00439 left = buffer[0][i]; 00440 right = buffer[1][i]; 00441 00442 if (wasted_bits) { 00443 left = (left << wasted_bits) | wasted_bits_buffer[0][i]; 00444 right = (right << wasted_bits) | wasted_bits_buffer[1][i]; 00445 } 00446 00447 buffer_out[i * numchannels] = left << 8; 00448 buffer_out[i * numchannels + 1] = right << 8; 00449 } 00450 } 00451 } 00452 00453 static int alac_decode_frame(AVCodecContext *avctx, 00454 void *outbuffer, int *outputsize, 00455 AVPacket *avpkt) 00456 { 00457 const uint8_t *inbuffer = avpkt->data; 00458 int input_buffer_size = avpkt->size; 00459 ALACContext *alac = avctx->priv_data; 00460 00461 int channels; 00462 unsigned int outputsamples; 00463 int hassize; 00464 unsigned int readsamplesize; 00465 int isnotcompressed; 00466 uint8_t interlacing_shift; 00467 uint8_t interlacing_leftweight; 00468 00469 /* short-circuit null buffers */ 00470 if (!inbuffer || !input_buffer_size) 00471 return -1; 00472 00473 init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); 00474 00475 channels = get_bits(&alac->gb, 3) + 1; 00476 if (channels > MAX_CHANNELS) { 00477 av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n", 00478 MAX_CHANNELS); 00479 return -1; 00480 } 00481 00482 /* 2^result = something to do with output waiting. 00483 * perhaps matters if we read > 1 frame in a pass? 00484 */ 00485 skip_bits(&alac->gb, 4); 00486 00487 skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */ 00488 00489 /* the output sample size is stored soon */ 00490 hassize = get_bits1(&alac->gb); 00491 00492 alac->wasted_bits = get_bits(&alac->gb, 2) << 3; 00493 00494 /* whether the frame is compressed */ 00495 isnotcompressed = get_bits1(&alac->gb); 00496 00497 if (hassize) { 00498 /* now read the number of samples as a 32bit integer */ 00499 outputsamples = get_bits_long(&alac->gb, 32); 00500 if(outputsamples > alac->setinfo_max_samples_per_frame){ 00501 av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame); 00502 return -1; 00503 } 00504 } else 00505 outputsamples = alac->setinfo_max_samples_per_frame; 00506 00507 switch (alac->setinfo_sample_size) { 00508 case 16: avctx->sample_fmt = AV_SAMPLE_FMT_S16; 00509 alac->bytespersample = channels << 1; 00510 break; 00511 case 24: avctx->sample_fmt = AV_SAMPLE_FMT_S32; 00512 alac->bytespersample = channels << 2; 00513 break; 00514 default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n", 00515 alac->setinfo_sample_size); 00516 return -1; 00517 } 00518 00519 if(outputsamples > *outputsize / alac->bytespersample){ 00520 av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n"); 00521 return -1; 00522 } 00523 00524 *outputsize = outputsamples * alac->bytespersample; 00525 readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1; 00526 if (readsamplesize > MIN_CACHE_BITS) { 00527 av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize); 00528 return -1; 00529 } 00530 00531 if (!isnotcompressed) { 00532 /* so it is compressed */ 00533 int16_t predictor_coef_table[MAX_CHANNELS][32]; 00534 int predictor_coef_num[MAX_CHANNELS]; 00535 int prediction_type[MAX_CHANNELS]; 00536 int prediction_quantitization[MAX_CHANNELS]; 00537 int ricemodifier[MAX_CHANNELS]; 00538 int i, chan; 00539 00540 interlacing_shift = get_bits(&alac->gb, 8); 00541 interlacing_leftweight = get_bits(&alac->gb, 8); 00542 00543 for (chan = 0; chan < channels; chan++) { 00544 prediction_type[chan] = get_bits(&alac->gb, 4); 00545 prediction_quantitization[chan] = get_bits(&alac->gb, 4); 00546 00547 ricemodifier[chan] = get_bits(&alac->gb, 3); 00548 predictor_coef_num[chan] = get_bits(&alac->gb, 5); 00549 00550 /* read the predictor table */ 00551 for (i = 0; i < predictor_coef_num[chan]; i++) 00552 predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); 00553 } 00554 00555 if (alac->wasted_bits) { 00556 int i, ch; 00557 for (i = 0; i < outputsamples; i++) { 00558 for (ch = 0; ch < channels; ch++) 00559 alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits); 00560 } 00561 } 00562 for (chan = 0; chan < channels; chan++) { 00563 bastardized_rice_decompress(alac, 00564 alac->predicterror_buffer[chan], 00565 outputsamples, 00566 readsamplesize, 00567 alac->setinfo_rice_initialhistory, 00568 alac->setinfo_rice_kmodifier, 00569 ricemodifier[chan] * alac->setinfo_rice_historymult / 4, 00570 (1 << alac->setinfo_rice_kmodifier) - 1); 00571 00572 if (prediction_type[chan] == 0) { 00573 /* adaptive fir */ 00574 predictor_decompress_fir_adapt(alac->predicterror_buffer[chan], 00575 alac->outputsamples_buffer[chan], 00576 outputsamples, 00577 readsamplesize, 00578 predictor_coef_table[chan], 00579 predictor_coef_num[chan], 00580 prediction_quantitization[chan]); 00581 } else { 00582 av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]); 00583 /* I think the only other prediction type (or perhaps this is 00584 * just a boolean?) runs adaptive fir twice.. like: 00585 * predictor_decompress_fir_adapt(predictor_error, tempout, ...) 00586 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...) 00587 * little strange.. 00588 */ 00589 } 00590 } 00591 } else { 00592 /* not compressed, easy case */ 00593 int i, chan; 00594 if (alac->setinfo_sample_size <= 16) { 00595 for (i = 0; i < outputsamples; i++) 00596 for (chan = 0; chan < channels; chan++) { 00597 int32_t audiobits; 00598 00599 audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size); 00600 00601 alac->outputsamples_buffer[chan][i] = audiobits; 00602 } 00603 } else { 00604 for (i = 0; i < outputsamples; i++) { 00605 for (chan = 0; chan < channels; chan++) { 00606 alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb, 00607 alac->setinfo_sample_size); 00608 alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i], 00609 alac->setinfo_sample_size); 00610 } 00611 } 00612 } 00613 alac->wasted_bits = 0; 00614 interlacing_shift = 0; 00615 interlacing_leftweight = 0; 00616 } 00617 if (get_bits(&alac->gb, 3) != 7) 00618 av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n"); 00619 00620 switch(alac->setinfo_sample_size) { 00621 case 16: 00622 if (channels == 2) { 00623 reconstruct_stereo_16(alac->outputsamples_buffer, 00624 (int16_t*)outbuffer, 00625 alac->numchannels, 00626 outputsamples, 00627 interlacing_shift, 00628 interlacing_leftweight); 00629 } else { 00630 int i; 00631 for (i = 0; i < outputsamples; i++) { 00632 ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i]; 00633 } 00634 } 00635 break; 00636 case 24: 00637 if (channels == 2) { 00638 decorrelate_stereo_24(alac->outputsamples_buffer, 00639 outbuffer, 00640 alac->wasted_bits_buffer, 00641 alac->wasted_bits, 00642 alac->numchannels, 00643 outputsamples, 00644 interlacing_shift, 00645 interlacing_leftweight); 00646 } else { 00647 int i; 00648 for (i = 0; i < outputsamples; i++) 00649 ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8; 00650 } 00651 break; 00652 } 00653 00654 if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8) 00655 av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb)); 00656 00657 return input_buffer_size; 00658 } 00659 00660 static av_cold int alac_decode_init(AVCodecContext * avctx) 00661 { 00662 ALACContext *alac = avctx->priv_data; 00663 alac->avctx = avctx; 00664 alac->numchannels = alac->avctx->channels; 00665 00666 /* initialize from the extradata */ 00667 if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) { 00668 av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n", 00669 ALAC_EXTRADATA_SIZE); 00670 return -1; 00671 } 00672 if (alac_set_info(alac)) { 00673 av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); 00674 return -1; 00675 } 00676 00677 return 0; 00678 } 00679 00680 static av_cold int alac_decode_close(AVCodecContext *avctx) 00681 { 00682 ALACContext *alac = avctx->priv_data; 00683 00684 int chan; 00685 for (chan = 0; chan < MAX_CHANNELS; chan++) { 00686 av_freep(&alac->predicterror_buffer[chan]); 00687 av_freep(&alac->outputsamples_buffer[chan]); 00688 av_freep(&alac->wasted_bits_buffer[chan]); 00689 } 00690 00691 return 0; 00692 } 00693 00694 AVCodec ff_alac_decoder = { 00695 "alac", 00696 AVMEDIA_TYPE_AUDIO, 00697 CODEC_ID_ALAC, 00698 sizeof(ALACContext), 00699 alac_decode_init, 00700 NULL, 00701 alac_decode_close, 00702 alac_decode_frame, 00703 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), 00704 };