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  1. /*
  2. * ALAC (Apple Lossless Audio Codec) decoder
  3. * Copyright (c) 2005 David Hammerton
  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. * @file alac.c
  23. * ALAC (Apple Lossless Audio Codec) decoder
  24. * @author 2005 David Hammerton
  25. *
  26. * For more information on the ALAC format, visit:
  27. * http://crazney.net/programs/itunes/alac.html
  28. *
  29. * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be
  30. * passed through the extradata[_size] fields. This atom is tacked onto
  31. * the end of an 'alac' stsd atom and has the following format:
  32. * bytes 0-3 atom size (0x24), big-endian
  33. * bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd)
  34. * bytes 8-35 data bytes needed by decoder
  35. *
  36. * Extradata:
  37. * 32bit size
  38. * 32bit tag (=alac)
  39. * 32bit zero?
  40. * 32bit max sample per frame
  41. * 8bit ?? (zero?)
  42. * 8bit sample size
  43. * 8bit history mult
  44. * 8bit initial history
  45. * 8bit kmodifier
  46. * 8bit channels?
  47. * 16bit ??
  48. * 32bit max coded frame size
  49. * 32bit bitrate?
  50. * 32bit samplerate
  51. */
  52. #include "avcodec.h"
  53. #include "bitstream.h"
  54. #include "bytestream.h"
  55. #include "unary.h"
  56. #define ALAC_EXTRADATA_SIZE 36
  57. #define MAX_CHANNELS 2
  58. typedef struct {
  59. AVCodecContext *avctx;
  60. GetBitContext gb;
  61. /* init to 0; first frame decode should initialize from extradata and
  62. * set this to 1 */
  63. int context_initialized;
  64. int numchannels;
  65. int bytespersample;
  66. /* buffers */
  67. int32_t *predicterror_buffer[MAX_CHANNELS];
  68. int32_t *outputsamples_buffer[MAX_CHANNELS];
  69. /* stuff from setinfo */
  70. uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */
  71. uint8_t setinfo_sample_size; /* 0x10 */
  72. uint8_t setinfo_rice_historymult; /* 0x28 */
  73. uint8_t setinfo_rice_initialhistory; /* 0x0a */
  74. uint8_t setinfo_rice_kmodifier; /* 0x0e */
  75. /* end setinfo stuff */
  76. } ALACContext;
  77. static void allocate_buffers(ALACContext *alac)
  78. {
  79. int chan;
  80. for (chan = 0; chan < MAX_CHANNELS; chan++) {
  81. alac->predicterror_buffer[chan] =
  82. av_malloc(alac->setinfo_max_samples_per_frame * 4);
  83. alac->outputsamples_buffer[chan] =
  84. av_malloc(alac->setinfo_max_samples_per_frame * 4);
  85. }
  86. }
  87. static int alac_set_info(ALACContext *alac)
  88. {
  89. const unsigned char *ptr = alac->avctx->extradata;
  90. ptr += 4; /* size */
  91. ptr += 4; /* alac */
  92. ptr += 4; /* 0 ? */
  93. if(AV_RB32(ptr) >= UINT_MAX/4){
  94. av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n");
  95. return -1;
  96. }
  97. /* buffer size / 2 ? */
  98. alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr);
  99. ptr++; /* ??? */
  100. alac->setinfo_sample_size = *ptr++;
  101. alac->setinfo_rice_historymult = *ptr++;
  102. alac->setinfo_rice_initialhistory = *ptr++;
  103. alac->setinfo_rice_kmodifier = *ptr++;
  104. ptr++; /* channels? */
  105. bytestream_get_be16(&ptr); /* ??? */
  106. bytestream_get_be32(&ptr); /* max coded frame size */
  107. bytestream_get_be32(&ptr); /* bitrate ? */
  108. bytestream_get_be32(&ptr); /* samplerate */
  109. allocate_buffers(alac);
  110. return 0;
  111. }
  112. static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){
  113. /* read x - number of 1s before 0 represent the rice */
  114. int x = get_unary_0_9(gb);
  115. if (x > 8) { /* RICE THRESHOLD */
  116. /* use alternative encoding */
  117. x = get_bits(gb, readsamplesize);
  118. } else {
  119. if (k >= limit)
  120. k = limit;
  121. if (k != 1) {
  122. int extrabits = show_bits(gb, k);
  123. /* multiply x by 2^k - 1, as part of their strange algorithm */
  124. x = (x << k) - x;
  125. if (extrabits > 1) {
  126. x += extrabits - 1;
  127. skip_bits(gb, k);
  128. } else
  129. skip_bits(gb, k - 1);
  130. }
  131. }
  132. return x;
  133. }
  134. static void bastardized_rice_decompress(ALACContext *alac,
  135. int32_t *output_buffer,
  136. int output_size,
  137. int readsamplesize, /* arg_10 */
  138. int rice_initialhistory, /* arg424->b */
  139. int rice_kmodifier, /* arg424->d */
  140. int rice_historymult, /* arg424->c */
  141. int rice_kmodifier_mask /* arg424->e */
  142. )
  143. {
  144. int output_count;
  145. unsigned int history = rice_initialhistory;
  146. int sign_modifier = 0;
  147. for (output_count = 0; output_count < output_size; output_count++) {
  148. int32_t x;
  149. int32_t x_modified;
  150. int32_t final_val;
  151. /* standard rice encoding */
  152. int k; /* size of extra bits */
  153. /* read k, that is bits as is */
  154. k = av_log2((history >> 9) + 3);
  155. x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize);
  156. x_modified = sign_modifier + x;
  157. final_val = (x_modified + 1) / 2;
  158. if (x_modified & 1) final_val *= -1;
  159. output_buffer[output_count] = final_val;
  160. sign_modifier = 0;
  161. /* now update the history */
  162. history += x_modified * rice_historymult
  163. - ((history * rice_historymult) >> 9);
  164. if (x_modified > 0xffff)
  165. history = 0xffff;
  166. /* special case: there may be compressed blocks of 0 */
  167. if ((history < 128) && (output_count+1 < output_size)) {
  168. int block_size, k;
  169. sign_modifier = 1;
  170. k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */);
  171. block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16);
  172. if (block_size > 0) {
  173. memset(&output_buffer[output_count+1], 0, block_size * 4);
  174. output_count += block_size;
  175. }
  176. if (block_size > 0xffff)
  177. sign_modifier = 0;
  178. history = 0;
  179. }
  180. }
  181. }
  182. static inline int32_t extend_sign32(int32_t val, int bits)
  183. {
  184. return (val << (32 - bits)) >> (32 - bits);
  185. }
  186. static inline int sign_only(int v)
  187. {
  188. return v ? FFSIGN(v) : 0;
  189. }
  190. static void predictor_decompress_fir_adapt(int32_t *error_buffer,
  191. int32_t *buffer_out,
  192. int output_size,
  193. int readsamplesize,
  194. int16_t *predictor_coef_table,
  195. int predictor_coef_num,
  196. int predictor_quantitization)
  197. {
  198. int i;
  199. /* first sample always copies */
  200. *buffer_out = *error_buffer;
  201. if (!predictor_coef_num) {
  202. if (output_size <= 1)
  203. return;
  204. memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4);
  205. return;
  206. }
  207. if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */
  208. /* second-best case scenario for fir decompression,
  209. * error describes a small difference from the previous sample only
  210. */
  211. if (output_size <= 1)
  212. return;
  213. for (i = 0; i < output_size - 1; i++) {
  214. int32_t prev_value;
  215. int32_t error_value;
  216. prev_value = buffer_out[i];
  217. error_value = error_buffer[i+1];
  218. buffer_out[i+1] =
  219. extend_sign32((prev_value + error_value), readsamplesize);
  220. }
  221. return;
  222. }
  223. /* read warm-up samples */
  224. if (predictor_coef_num > 0)
  225. for (i = 0; i < predictor_coef_num; i++) {
  226. int32_t val;
  227. val = buffer_out[i] + error_buffer[i+1];
  228. val = extend_sign32(val, readsamplesize);
  229. buffer_out[i+1] = val;
  230. }
  231. #if 0
  232. /* 4 and 8 are very common cases (the only ones i've seen). these
  233. * should be unrolled and optimized
  234. */
  235. if (predictor_coef_num == 4) {
  236. /* FIXME: optimized general case */
  237. return;
  238. }
  239. if (predictor_coef_table == 8) {
  240. /* FIXME: optimized general case */
  241. return;
  242. }
  243. #endif
  244. /* general case */
  245. if (predictor_coef_num > 0) {
  246. for (i = predictor_coef_num + 1; i < output_size; i++) {
  247. int j;
  248. int sum = 0;
  249. int outval;
  250. int error_val = error_buffer[i];
  251. for (j = 0; j < predictor_coef_num; j++) {
  252. sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) *
  253. predictor_coef_table[j];
  254. }
  255. outval = (1 << (predictor_quantitization-1)) + sum;
  256. outval = outval >> predictor_quantitization;
  257. outval = outval + buffer_out[0] + error_val;
  258. outval = extend_sign32(outval, readsamplesize);
  259. buffer_out[predictor_coef_num+1] = outval;
  260. if (error_val > 0) {
  261. int predictor_num = predictor_coef_num - 1;
  262. while (predictor_num >= 0 && error_val > 0) {
  263. int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
  264. int sign = sign_only(val);
  265. predictor_coef_table[predictor_num] -= sign;
  266. val *= sign; /* absolute value */
  267. error_val -= ((val >> predictor_quantitization) *
  268. (predictor_coef_num - predictor_num));
  269. predictor_num--;
  270. }
  271. } else if (error_val < 0) {
  272. int predictor_num = predictor_coef_num - 1;
  273. while (predictor_num >= 0 && error_val < 0) {
  274. int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num];
  275. int sign = - sign_only(val);
  276. predictor_coef_table[predictor_num] -= sign;
  277. val *= sign; /* neg value */
  278. error_val -= ((val >> predictor_quantitization) *
  279. (predictor_coef_num - predictor_num));
  280. predictor_num--;
  281. }
  282. }
  283. buffer_out++;
  284. }
  285. }
  286. }
  287. static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS],
  288. int16_t *buffer_out,
  289. int numchannels, int numsamples,
  290. uint8_t interlacing_shift,
  291. uint8_t interlacing_leftweight)
  292. {
  293. int i;
  294. if (numsamples <= 0)
  295. return;
  296. /* weighted interlacing */
  297. if (interlacing_leftweight) {
  298. for (i = 0; i < numsamples; i++) {
  299. int32_t a, b;
  300. a = buffer[0][i];
  301. b = buffer[1][i];
  302. a -= (b * interlacing_leftweight) >> interlacing_shift;
  303. b += a;
  304. buffer_out[i*numchannels] = b;
  305. buffer_out[i*numchannels + 1] = a;
  306. }
  307. return;
  308. }
  309. /* otherwise basic interlacing took place */
  310. for (i = 0; i < numsamples; i++) {
  311. int16_t left, right;
  312. left = buffer[0][i];
  313. right = buffer[1][i];
  314. buffer_out[i*numchannels] = left;
  315. buffer_out[i*numchannels + 1] = right;
  316. }
  317. }
  318. static int alac_decode_frame(AVCodecContext *avctx,
  319. void *outbuffer, int *outputsize,
  320. const uint8_t *inbuffer, int input_buffer_size)
  321. {
  322. ALACContext *alac = avctx->priv_data;
  323. int channels;
  324. int32_t outputsamples;
  325. int hassize;
  326. int readsamplesize;
  327. int wasted_bytes;
  328. int isnotcompressed;
  329. uint8_t interlacing_shift;
  330. uint8_t interlacing_leftweight;
  331. /* short-circuit null buffers */
  332. if (!inbuffer || !input_buffer_size)
  333. return input_buffer_size;
  334. /* initialize from the extradata */
  335. if (!alac->context_initialized) {
  336. if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) {
  337. av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n",
  338. ALAC_EXTRADATA_SIZE);
  339. return input_buffer_size;
  340. }
  341. if (alac_set_info(alac)) {
  342. av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n");
  343. return input_buffer_size;
  344. }
  345. alac->context_initialized = 1;
  346. }
  347. init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8);
  348. channels = get_bits(&alac->gb, 3) + 1;
  349. if (channels > MAX_CHANNELS) {
  350. av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n",
  351. MAX_CHANNELS);
  352. return input_buffer_size;
  353. }
  354. /* 2^result = something to do with output waiting.
  355. * perhaps matters if we read > 1 frame in a pass?
  356. */
  357. skip_bits(&alac->gb, 4);
  358. skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */
  359. /* the output sample size is stored soon */
  360. hassize = get_bits1(&alac->gb);
  361. wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */
  362. /* whether the frame is compressed */
  363. isnotcompressed = get_bits1(&alac->gb);
  364. if (hassize) {
  365. /* now read the number of samples as a 32bit integer */
  366. outputsamples = get_bits(&alac->gb, 32);
  367. } else
  368. outputsamples = alac->setinfo_max_samples_per_frame;
  369. *outputsize = outputsamples * alac->bytespersample;
  370. readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1;
  371. if (!isnotcompressed) {
  372. /* so it is compressed */
  373. int16_t predictor_coef_table[channels][32];
  374. int predictor_coef_num[channels];
  375. int prediction_type[channels];
  376. int prediction_quantitization[channels];
  377. int ricemodifier[channels];
  378. int i, chan;
  379. interlacing_shift = get_bits(&alac->gb, 8);
  380. interlacing_leftweight = get_bits(&alac->gb, 8);
  381. for (chan = 0; chan < channels; chan++) {
  382. prediction_type[chan] = get_bits(&alac->gb, 4);
  383. prediction_quantitization[chan] = get_bits(&alac->gb, 4);
  384. ricemodifier[chan] = get_bits(&alac->gb, 3);
  385. predictor_coef_num[chan] = get_bits(&alac->gb, 5);
  386. /* read the predictor table */
  387. for (i = 0; i < predictor_coef_num[chan]; i++)
  388. predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16);
  389. }
  390. if (wasted_bytes)
  391. av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n");
  392. for (chan = 0; chan < channels; chan++) {
  393. bastardized_rice_decompress(alac,
  394. alac->predicterror_buffer[chan],
  395. outputsamples,
  396. readsamplesize,
  397. alac->setinfo_rice_initialhistory,
  398. alac->setinfo_rice_kmodifier,
  399. ricemodifier[chan] * alac->setinfo_rice_historymult / 4,
  400. (1 << alac->setinfo_rice_kmodifier) - 1);
  401. if (prediction_type[chan] == 0) {
  402. /* adaptive fir */
  403. predictor_decompress_fir_adapt(alac->predicterror_buffer[chan],
  404. alac->outputsamples_buffer[chan],
  405. outputsamples,
  406. readsamplesize,
  407. predictor_coef_table[chan],
  408. predictor_coef_num[chan],
  409. prediction_quantitization[chan]);
  410. } else {
  411. av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]);
  412. /* I think the only other prediction type (or perhaps this is
  413. * just a boolean?) runs adaptive fir twice.. like:
  414. * predictor_decompress_fir_adapt(predictor_error, tempout, ...)
  415. * predictor_decompress_fir_adapt(predictor_error, outputsamples ...)
  416. * little strange..
  417. */
  418. }
  419. }
  420. } else {
  421. /* not compressed, easy case */
  422. if (alac->setinfo_sample_size <= 16) {
  423. int i, chan;
  424. for (chan = 0; chan < channels; chan++)
  425. for (i = 0; i < outputsamples; i++) {
  426. int32_t audiobits;
  427. audiobits = get_bits(&alac->gb, alac->setinfo_sample_size);
  428. audiobits = extend_sign32(audiobits, readsamplesize);
  429. alac->outputsamples_buffer[chan][i] = audiobits;
  430. }
  431. } else {
  432. int i, chan;
  433. for (chan = 0; chan < channels; chan++)
  434. for (i = 0; i < outputsamples; i++) {
  435. int32_t audiobits;
  436. audiobits = get_bits(&alac->gb, 16);
  437. /* special case of sign extension..
  438. * as we'll be ORing the low 16bits into this */
  439. audiobits = audiobits << 16;
  440. audiobits = audiobits >> (32 - alac->setinfo_sample_size);
  441. audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16);
  442. alac->outputsamples_buffer[chan][i] = audiobits;
  443. }
  444. }
  445. /* wasted_bytes = 0; */
  446. interlacing_shift = 0;
  447. interlacing_leftweight = 0;
  448. }
  449. if (get_bits(&alac->gb, 3) != 7)
  450. av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n");
  451. switch(alac->setinfo_sample_size) {
  452. case 16:
  453. if (channels == 2) {
  454. reconstruct_stereo_16(alac->outputsamples_buffer,
  455. (int16_t*)outbuffer,
  456. alac->numchannels,
  457. outputsamples,
  458. interlacing_shift,
  459. interlacing_leftweight);
  460. } else {
  461. int i;
  462. for (i = 0; i < outputsamples; i++) {
  463. int16_t sample = alac->outputsamples_buffer[0][i];
  464. ((int16_t*)outbuffer)[i * alac->numchannels] = sample;
  465. }
  466. }
  467. break;
  468. case 20:
  469. case 24:
  470. // It is not clear if there exist any encoder that creates 24 bit ALAC
  471. // files. iTunes convert 24 bit raw files to 16 bit before encoding.
  472. case 32:
  473. av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size);
  474. break;
  475. default:
  476. break;
  477. }
  478. if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8)
  479. av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb));
  480. return input_buffer_size;
  481. }
  482. static av_cold int alac_decode_init(AVCodecContext * avctx)
  483. {
  484. ALACContext *alac = avctx->priv_data;
  485. alac->avctx = avctx;
  486. alac->context_initialized = 0;
  487. alac->numchannels = alac->avctx->channels;
  488. alac->bytespersample = (avctx->bits_per_sample / 8) * alac->numchannels;
  489. return 0;
  490. }
  491. static av_cold int alac_decode_close(AVCodecContext *avctx)
  492. {
  493. ALACContext *alac = avctx->priv_data;
  494. int chan;
  495. for (chan = 0; chan < MAX_CHANNELS; chan++) {
  496. av_free(alac->predicterror_buffer[chan]);
  497. av_free(alac->outputsamples_buffer[chan]);
  498. }
  499. return 0;
  500. }
  501. AVCodec alac_decoder = {
  502. "alac",
  503. CODEC_TYPE_AUDIO,
  504. CODEC_ID_ALAC,
  505. sizeof(ALACContext),
  506. alac_decode_init,
  507. NULL,
  508. alac_decode_close,
  509. alac_decode_frame,
  510. };