cook.c 43 KB

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  1. /*
  2. * COOK compatible decoder
  3. * Copyright (c) 2003 Sascha Sommer
  4. * Copyright (c) 2005 Benjamin Larsson
  5. *
  6. * This file is part of FFmpeg.
  7. *
  8. * FFmpeg is free software; you can redistribute it and/or
  9. * modify it under the terms of the GNU Lesser General Public
  10. * License as published by the Free Software Foundation; either
  11. * version 2.1 of the License, or (at your option) any later version.
  12. *
  13. * FFmpeg is distributed in the hope that it will be useful,
  14. * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15. * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
  16. * Lesser General Public License for more details.
  17. *
  18. * You should have received a copy of the GNU Lesser General Public
  19. * License along with FFmpeg; if not, write to the Free Software
  20. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21. */
  22. /**
  23. * @file
  24. * Cook compatible decoder. Bastardization of the G.722.1 standard.
  25. * This decoder handles RealNetworks, RealAudio G2 data.
  26. * Cook is identified by the codec name cook in RM files.
  27. *
  28. * To use this decoder, a calling application must supply the extradata
  29. * bytes provided from the RM container; 8+ bytes for mono streams and
  30. * 16+ for stereo streams (maybe more).
  31. *
  32. * Codec technicalities (all this assume a buffer length of 1024):
  33. * Cook works with several different techniques to achieve its compression.
  34. * In the timedomain the buffer is divided into 8 pieces and quantized. If
  35. * two neighboring pieces have different quantization index a smooth
  36. * quantization curve is used to get a smooth overlap between the different
  37. * pieces.
  38. * To get to the transformdomain Cook uses a modulated lapped transform.
  39. * The transform domain has 50 subbands with 20 elements each. This
  40. * means only a maximum of 50*20=1000 coefficients are used out of the 1024
  41. * available.
  42. */
  43. #include "libavutil/lfg.h"
  44. #include "avcodec.h"
  45. #include "get_bits.h"
  46. #include "dsputil.h"
  47. #include "bytestream.h"
  48. #include "fft.h"
  49. #include "libavutil/audioconvert.h"
  50. #include "sinewin.h"
  51. #include "cookdata.h"
  52. /* the different Cook versions */
  53. #define MONO 0x1000001
  54. #define STEREO 0x1000002
  55. #define JOINT_STEREO 0x1000003
  56. #define MC_COOK 0x2000000 //multichannel Cook, not supported
  57. #define SUBBAND_SIZE 20
  58. #define MAX_SUBPACKETS 5
  59. typedef struct {
  60. int *now;
  61. int *previous;
  62. } cook_gains;
  63. typedef struct {
  64. int ch_idx;
  65. int size;
  66. int num_channels;
  67. int cookversion;
  68. int samples_per_frame;
  69. int subbands;
  70. int js_subband_start;
  71. int js_vlc_bits;
  72. int samples_per_channel;
  73. int log2_numvector_size;
  74. unsigned int channel_mask;
  75. VLC ccpl; ///< channel coupling
  76. int joint_stereo;
  77. int bits_per_subpacket;
  78. int bits_per_subpdiv;
  79. int total_subbands;
  80. int numvector_size; ///< 1 << log2_numvector_size;
  81. float mono_previous_buffer1[1024];
  82. float mono_previous_buffer2[1024];
  83. /** gain buffers */
  84. cook_gains gains1;
  85. cook_gains gains2;
  86. int gain_1[9];
  87. int gain_2[9];
  88. int gain_3[9];
  89. int gain_4[9];
  90. } COOKSubpacket;
  91. typedef struct cook {
  92. /*
  93. * The following 5 functions provide the lowlevel arithmetic on
  94. * the internal audio buffers.
  95. */
  96. void (* scalar_dequant)(struct cook *q, int index, int quant_index,
  97. int* subband_coef_index, int* subband_coef_sign,
  98. float* mlt_p);
  99. void (* decouple) (struct cook *q,
  100. COOKSubpacket *p,
  101. int subband,
  102. float f1, float f2,
  103. float *decode_buffer,
  104. float *mlt_buffer1, float *mlt_buffer2);
  105. void (* imlt_window) (struct cook *q, float *buffer1,
  106. cook_gains *gains_ptr, float *previous_buffer);
  107. void (* interpolate) (struct cook *q, float* buffer,
  108. int gain_index, int gain_index_next);
  109. void (* saturate_output) (struct cook *q, int chan, float *out);
  110. AVCodecContext* avctx;
  111. GetBitContext gb;
  112. /* stream data */
  113. int nb_channels;
  114. int bit_rate;
  115. int sample_rate;
  116. int num_vectors;
  117. int samples_per_channel;
  118. /* states */
  119. AVLFG random_state;
  120. /* transform data */
  121. FFTContext mdct_ctx;
  122. float* mlt_window;
  123. /* VLC data */
  124. VLC envelope_quant_index[13];
  125. VLC sqvh[7]; //scalar quantization
  126. /* generatable tables and related variables */
  127. int gain_size_factor;
  128. float gain_table[23];
  129. /* data buffers */
  130. uint8_t* decoded_bytes_buffer;
  131. DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
  132. float decode_buffer_1[1024];
  133. float decode_buffer_2[1024];
  134. float decode_buffer_0[1060]; /* static allocation for joint decode */
  135. const float *cplscales[5];
  136. int num_subpackets;
  137. COOKSubpacket subpacket[MAX_SUBPACKETS];
  138. } COOKContext;
  139. static float pow2tab[127];
  140. static float rootpow2tab[127];
  141. /*************** init functions ***************/
  142. /* table generator */
  143. static av_cold void init_pow2table(void){
  144. int i;
  145. for (i=-63 ; i<64 ; i++){
  146. pow2tab[63+i]= pow(2, i);
  147. rootpow2tab[63+i]=sqrt(pow(2, i));
  148. }
  149. }
  150. /* table generator */
  151. static av_cold void init_gain_table(COOKContext *q) {
  152. int i;
  153. q->gain_size_factor = q->samples_per_channel/8;
  154. for (i=0 ; i<23 ; i++) {
  155. q->gain_table[i] = pow(pow2tab[i+52] ,
  156. (1.0/(double)q->gain_size_factor));
  157. }
  158. }
  159. static av_cold int init_cook_vlc_tables(COOKContext *q) {
  160. int i, result;
  161. result = 0;
  162. for (i=0 ; i<13 ; i++) {
  163. result |= init_vlc (&q->envelope_quant_index[i], 9, 24,
  164. envelope_quant_index_huffbits[i], 1, 1,
  165. envelope_quant_index_huffcodes[i], 2, 2, 0);
  166. }
  167. av_log(q->avctx,AV_LOG_DEBUG,"sqvh VLC init\n");
  168. for (i=0 ; i<7 ; i++) {
  169. result |= init_vlc (&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
  170. cvh_huffbits[i], 1, 1,
  171. cvh_huffcodes[i], 2, 2, 0);
  172. }
  173. for(i=0;i<q->num_subpackets;i++){
  174. if (q->subpacket[i].joint_stereo==1){
  175. result |= init_vlc (&q->subpacket[i].ccpl, 6, (1<<q->subpacket[i].js_vlc_bits)-1,
  176. ccpl_huffbits[q->subpacket[i].js_vlc_bits-2], 1, 1,
  177. ccpl_huffcodes[q->subpacket[i].js_vlc_bits-2], 2, 2, 0);
  178. av_log(q->avctx,AV_LOG_DEBUG,"subpacket %i Joint-stereo VLC used.\n",i);
  179. }
  180. }
  181. av_log(q->avctx,AV_LOG_DEBUG,"VLC tables initialized.\n");
  182. return result;
  183. }
  184. static av_cold int init_cook_mlt(COOKContext *q) {
  185. int j, ret;
  186. int mlt_size = q->samples_per_channel;
  187. if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
  188. return AVERROR(ENOMEM);
  189. /* Initialize the MLT window: simple sine window. */
  190. ff_sine_window_init(q->mlt_window, mlt_size);
  191. for(j=0 ; j<mlt_size ; j++)
  192. q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
  193. /* Initialize the MDCT. */
  194. if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1, 1.0/32768.0))) {
  195. av_free(q->mlt_window);
  196. return ret;
  197. }
  198. av_log(q->avctx,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
  199. av_log2(mlt_size)+1);
  200. return 0;
  201. }
  202. static const float *maybe_reformat_buffer32 (COOKContext *q, const float *ptr, int n)
  203. {
  204. if (1)
  205. return ptr;
  206. }
  207. static av_cold void init_cplscales_table (COOKContext *q) {
  208. int i;
  209. for (i=0;i<5;i++)
  210. q->cplscales[i] = maybe_reformat_buffer32 (q, cplscales[i], (1<<(i+2))-1);
  211. }
  212. /*************** init functions end ***********/
  213. #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4)
  214. #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
  215. /**
  216. * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
  217. * Why? No idea, some checksum/error detection method maybe.
  218. *
  219. * Out buffer size: extra bytes are needed to cope with
  220. * padding/misalignment.
  221. * Subpackets passed to the decoder can contain two, consecutive
  222. * half-subpackets, of identical but arbitrary size.
  223. * 1234 1234 1234 1234 extraA extraB
  224. * Case 1: AAAA BBBB 0 0
  225. * Case 2: AAAA ABBB BB-- 3 3
  226. * Case 3: AAAA AABB BBBB 2 2
  227. * Case 4: AAAA AAAB BBBB BB-- 1 5
  228. *
  229. * Nice way to waste CPU cycles.
  230. *
  231. * @param inbuffer pointer to byte array of indata
  232. * @param out pointer to byte array of outdata
  233. * @param bytes number of bytes
  234. */
  235. static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
  236. int i, off;
  237. uint32_t c;
  238. const uint32_t* buf;
  239. uint32_t* obuf = (uint32_t*) out;
  240. /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
  241. * I'm too lazy though, should be something like
  242. * for(i=0 ; i<bitamount/64 ; i++)
  243. * (int64_t)out[i] = 0x37c511f237c511f2^av_be2ne64(int64_t)in[i]);
  244. * Buffer alignment needs to be checked. */
  245. off = (intptr_t)inbuffer & 3;
  246. buf = (const uint32_t*) (inbuffer - off);
  247. c = av_be2ne32((0x37c511f2 >> (off*8)) | (0x37c511f2 << (32-(off*8))));
  248. bytes += 3 + off;
  249. for (i = 0; i < bytes/4; i++)
  250. obuf[i] = c ^ buf[i];
  251. return off;
  252. }
  253. /**
  254. * Cook uninit
  255. */
  256. static av_cold int cook_decode_close(AVCodecContext *avctx)
  257. {
  258. int i;
  259. COOKContext *q = avctx->priv_data;
  260. av_log(avctx,AV_LOG_DEBUG, "Deallocating memory.\n");
  261. /* Free allocated memory buffers. */
  262. av_free(q->mlt_window);
  263. av_free(q->decoded_bytes_buffer);
  264. /* Free the transform. */
  265. ff_mdct_end(&q->mdct_ctx);
  266. /* Free the VLC tables. */
  267. for (i=0 ; i<13 ; i++) {
  268. free_vlc(&q->envelope_quant_index[i]);
  269. }
  270. for (i=0 ; i<7 ; i++) {
  271. free_vlc(&q->sqvh[i]);
  272. }
  273. for (i=0 ; i<q->num_subpackets ; i++) {
  274. free_vlc(&q->subpacket[i].ccpl);
  275. }
  276. av_log(avctx,AV_LOG_DEBUG,"Memory deallocated.\n");
  277. return 0;
  278. }
  279. /**
  280. * Fill the gain array for the timedomain quantization.
  281. *
  282. * @param gb pointer to the GetBitContext
  283. * @param gaininfo array[9] of gain indexes
  284. */
  285. static void decode_gain_info(GetBitContext *gb, int *gaininfo)
  286. {
  287. int i, n;
  288. while (get_bits1(gb)) {}
  289. n = get_bits_count(gb) - 1; //amount of elements*2 to update
  290. i = 0;
  291. while (n--) {
  292. int index = get_bits(gb, 3);
  293. int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
  294. while (i <= index) gaininfo[i++] = gain;
  295. }
  296. while (i <= 8) gaininfo[i++] = 0;
  297. }
  298. /**
  299. * Create the quant index table needed for the envelope.
  300. *
  301. * @param q pointer to the COOKContext
  302. * @param quant_index_table pointer to the array
  303. */
  304. static void decode_envelope(COOKContext *q, COOKSubpacket *p, int* quant_index_table) {
  305. int i,j, vlc_index;
  306. quant_index_table[0]= get_bits(&q->gb,6) - 6; //This is used later in categorize
  307. for (i=1 ; i < p->total_subbands ; i++){
  308. vlc_index=i;
  309. if (i >= p->js_subband_start * 2) {
  310. vlc_index-=p->js_subband_start;
  311. } else {
  312. vlc_index/=2;
  313. if(vlc_index < 1) vlc_index = 1;
  314. }
  315. if (vlc_index>13) vlc_index = 13; //the VLC tables >13 are identical to No. 13
  316. j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index-1].table,
  317. q->envelope_quant_index[vlc_index-1].bits,2);
  318. quant_index_table[i] = quant_index_table[i-1] + j - 12; //differential encoding
  319. }
  320. }
  321. /**
  322. * Calculate the category and category_index vector.
  323. *
  324. * @param q pointer to the COOKContext
  325. * @param quant_index_table pointer to the array
  326. * @param category pointer to the category array
  327. * @param category_index pointer to the category_index array
  328. */
  329. static void categorize(COOKContext *q, COOKSubpacket *p, int* quant_index_table,
  330. int* category, int* category_index){
  331. int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
  332. int exp_index2[102];
  333. int exp_index1[102];
  334. int tmp_categorize_array[128*2];
  335. int tmp_categorize_array1_idx=p->numvector_size;
  336. int tmp_categorize_array2_idx=p->numvector_size;
  337. bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
  338. if(bits_left > q->samples_per_channel) {
  339. bits_left = q->samples_per_channel +
  340. ((bits_left - q->samples_per_channel)*5)/8;
  341. //av_log(q->avctx, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
  342. }
  343. memset(&exp_index1, 0, sizeof(exp_index1));
  344. memset(&exp_index2, 0, sizeof(exp_index2));
  345. memset(&tmp_categorize_array, 0, sizeof(tmp_categorize_array));
  346. bias=-32;
  347. /* Estimate bias. */
  348. for (i=32 ; i>0 ; i=i/2){
  349. num_bits = 0;
  350. index = 0;
  351. for (j=p->total_subbands ; j>0 ; j--){
  352. exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
  353. index++;
  354. num_bits+=expbits_tab[exp_idx];
  355. }
  356. if(num_bits >= bits_left - 32){
  357. bias+=i;
  358. }
  359. }
  360. /* Calculate total number of bits. */
  361. num_bits=0;
  362. for (i=0 ; i<p->total_subbands ; i++) {
  363. exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
  364. num_bits += expbits_tab[exp_idx];
  365. exp_index1[i] = exp_idx;
  366. exp_index2[i] = exp_idx;
  367. }
  368. tmpbias1 = tmpbias2 = num_bits;
  369. for (j = 1 ; j < p->numvector_size ; j++) {
  370. if (tmpbias1 + tmpbias2 > 2*bits_left) { /* ---> */
  371. int max = -999999;
  372. index=-1;
  373. for (i=0 ; i<p->total_subbands ; i++){
  374. if (exp_index1[i] < 7) {
  375. v = (-2*exp_index1[i]) - quant_index_table[i] + bias;
  376. if ( v >= max) {
  377. max = v;
  378. index = i;
  379. }
  380. }
  381. }
  382. if(index==-1)break;
  383. tmp_categorize_array[tmp_categorize_array1_idx++] = index;
  384. tmpbias1 -= expbits_tab[exp_index1[index]] -
  385. expbits_tab[exp_index1[index]+1];
  386. ++exp_index1[index];
  387. } else { /* <--- */
  388. int min = 999999;
  389. index=-1;
  390. for (i=0 ; i<p->total_subbands ; i++){
  391. if(exp_index2[i] > 0){
  392. v = (-2*exp_index2[i])-quant_index_table[i]+bias;
  393. if ( v < min) {
  394. min = v;
  395. index = i;
  396. }
  397. }
  398. }
  399. if(index == -1)break;
  400. tmp_categorize_array[--tmp_categorize_array2_idx] = index;
  401. tmpbias2 -= expbits_tab[exp_index2[index]] -
  402. expbits_tab[exp_index2[index]-1];
  403. --exp_index2[index];
  404. }
  405. }
  406. for(i=0 ; i<p->total_subbands ; i++)
  407. category[i] = exp_index2[i];
  408. for(i=0 ; i<p->numvector_size-1 ; i++)
  409. category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
  410. }
  411. /**
  412. * Expand the category vector.
  413. *
  414. * @param q pointer to the COOKContext
  415. * @param category pointer to the category array
  416. * @param category_index pointer to the category_index array
  417. */
  418. static inline void expand_category(COOKContext *q, int* category,
  419. int* category_index){
  420. int i;
  421. for(i=0 ; i<q->num_vectors ; i++){
  422. ++category[category_index[i]];
  423. }
  424. }
  425. /**
  426. * The real requantization of the mltcoefs
  427. *
  428. * @param q pointer to the COOKContext
  429. * @param index index
  430. * @param quant_index quantisation index
  431. * @param subband_coef_index array of indexes to quant_centroid_tab
  432. * @param subband_coef_sign signs of coefficients
  433. * @param mlt_p pointer into the mlt buffer
  434. */
  435. static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
  436. int* subband_coef_index, int* subband_coef_sign,
  437. float* mlt_p){
  438. int i;
  439. float f1;
  440. for(i=0 ; i<SUBBAND_SIZE ; i++) {
  441. if (subband_coef_index[i]) {
  442. f1 = quant_centroid_tab[index][subband_coef_index[i]];
  443. if (subband_coef_sign[i]) f1 = -f1;
  444. } else {
  445. /* noise coding if subband_coef_index[i] == 0 */
  446. f1 = dither_tab[index];
  447. if (av_lfg_get(&q->random_state) < 0x80000000) f1 = -f1;
  448. }
  449. mlt_p[i] = f1 * rootpow2tab[quant_index+63];
  450. }
  451. }
  452. /**
  453. * Unpack the subband_coef_index and subband_coef_sign vectors.
  454. *
  455. * @param q pointer to the COOKContext
  456. * @param category pointer to the category array
  457. * @param subband_coef_index array of indexes to quant_centroid_tab
  458. * @param subband_coef_sign signs of coefficients
  459. */
  460. static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category, int* subband_coef_index,
  461. int* subband_coef_sign) {
  462. int i,j;
  463. int vlc, vd ,tmp, result;
  464. vd = vd_tab[category];
  465. result = 0;
  466. for(i=0 ; i<vpr_tab[category] ; i++){
  467. vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
  468. if (p->bits_per_subpacket < get_bits_count(&q->gb)){
  469. vlc = 0;
  470. result = 1;
  471. }
  472. for(j=vd-1 ; j>=0 ; j--){
  473. tmp = (vlc * invradix_tab[category])/0x100000;
  474. subband_coef_index[vd*i+j] = vlc - tmp * (kmax_tab[category]+1);
  475. vlc = tmp;
  476. }
  477. for(j=0 ; j<vd ; j++){
  478. if (subband_coef_index[i*vd + j]) {
  479. if(get_bits_count(&q->gb) < p->bits_per_subpacket){
  480. subband_coef_sign[i*vd+j] = get_bits1(&q->gb);
  481. } else {
  482. result=1;
  483. subband_coef_sign[i*vd+j]=0;
  484. }
  485. } else {
  486. subband_coef_sign[i*vd+j]=0;
  487. }
  488. }
  489. }
  490. return result;
  491. }
  492. /**
  493. * Fill the mlt_buffer with mlt coefficients.
  494. *
  495. * @param q pointer to the COOKContext
  496. * @param category pointer to the category array
  497. * @param quant_index_table pointer to the array
  498. * @param mlt_buffer pointer to mlt coefficients
  499. */
  500. static void decode_vectors(COOKContext* q, COOKSubpacket* p, int* category,
  501. int *quant_index_table, float* mlt_buffer){
  502. /* A zero in this table means that the subband coefficient is
  503. random noise coded. */
  504. int subband_coef_index[SUBBAND_SIZE];
  505. /* A zero in this table means that the subband coefficient is a
  506. positive multiplicator. */
  507. int subband_coef_sign[SUBBAND_SIZE];
  508. int band, j;
  509. int index=0;
  510. for(band=0 ; band<p->total_subbands ; band++){
  511. index = category[band];
  512. if(category[band] < 7){
  513. if(unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)){
  514. index=7;
  515. for(j=0 ; j<p->total_subbands ; j++) category[band+j]=7;
  516. }
  517. }
  518. if(index>=7) {
  519. memset(subband_coef_index, 0, sizeof(subband_coef_index));
  520. memset(subband_coef_sign, 0, sizeof(subband_coef_sign));
  521. }
  522. q->scalar_dequant(q, index, quant_index_table[band],
  523. subband_coef_index, subband_coef_sign,
  524. &mlt_buffer[band * SUBBAND_SIZE]);
  525. }
  526. if(p->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){
  527. return;
  528. } /* FIXME: should this be removed, or moved into loop above? */
  529. }
  530. /**
  531. * function for decoding mono data
  532. *
  533. * @param q pointer to the COOKContext
  534. * @param mlt_buffer pointer to mlt coefficients
  535. */
  536. static void mono_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer) {
  537. int category_index[128];
  538. int quant_index_table[102];
  539. int category[128];
  540. memset(&category, 0, sizeof(category));
  541. memset(&category_index, 0, sizeof(category_index));
  542. decode_envelope(q, p, quant_index_table);
  543. q->num_vectors = get_bits(&q->gb,p->log2_numvector_size);
  544. categorize(q, p, quant_index_table, category, category_index);
  545. expand_category(q, category, category_index);
  546. decode_vectors(q, p, category, quant_index_table, mlt_buffer);
  547. }
  548. /**
  549. * the actual requantization of the timedomain samples
  550. *
  551. * @param q pointer to the COOKContext
  552. * @param buffer pointer to the timedomain buffer
  553. * @param gain_index index for the block multiplier
  554. * @param gain_index_next index for the next block multiplier
  555. */
  556. static void interpolate_float(COOKContext *q, float* buffer,
  557. int gain_index, int gain_index_next){
  558. int i;
  559. float fc1, fc2;
  560. fc1 = pow2tab[gain_index+63];
  561. if(gain_index == gain_index_next){ //static gain
  562. for(i=0 ; i<q->gain_size_factor ; i++){
  563. buffer[i]*=fc1;
  564. }
  565. } else { //smooth gain
  566. fc2 = q->gain_table[11 + (gain_index_next-gain_index)];
  567. for(i=0 ; i<q->gain_size_factor ; i++){
  568. buffer[i]*=fc1;
  569. fc1*=fc2;
  570. }
  571. }
  572. }
  573. /**
  574. * Apply transform window, overlap buffers.
  575. *
  576. * @param q pointer to the COOKContext
  577. * @param inbuffer pointer to the mltcoefficients
  578. * @param gains_ptr current and previous gains
  579. * @param previous_buffer pointer to the previous buffer to be used for overlapping
  580. */
  581. static void imlt_window_float (COOKContext *q, float *inbuffer,
  582. cook_gains *gains_ptr, float *previous_buffer)
  583. {
  584. const float fc = pow2tab[gains_ptr->previous[0] + 63];
  585. int i;
  586. /* The weird thing here, is that the two halves of the time domain
  587. * buffer are swapped. Also, the newest data, that we save away for
  588. * next frame, has the wrong sign. Hence the subtraction below.
  589. * Almost sounds like a complex conjugate/reverse data/FFT effect.
  590. */
  591. /* Apply window and overlap */
  592. for(i = 0; i < q->samples_per_channel; i++){
  593. inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
  594. previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
  595. }
  596. }
  597. /**
  598. * The modulated lapped transform, this takes transform coefficients
  599. * and transforms them into timedomain samples.
  600. * Apply transform window, overlap buffers, apply gain profile
  601. * and buffer management.
  602. *
  603. * @param q pointer to the COOKContext
  604. * @param inbuffer pointer to the mltcoefficients
  605. * @param gains_ptr current and previous gains
  606. * @param previous_buffer pointer to the previous buffer to be used for overlapping
  607. */
  608. static void imlt_gain(COOKContext *q, float *inbuffer,
  609. cook_gains *gains_ptr, float* previous_buffer)
  610. {
  611. float *buffer0 = q->mono_mdct_output;
  612. float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
  613. int i;
  614. /* Inverse modified discrete cosine transform */
  615. q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
  616. q->imlt_window (q, buffer1, gains_ptr, previous_buffer);
  617. /* Apply gain profile */
  618. for (i = 0; i < 8; i++) {
  619. if (gains_ptr->now[i] || gains_ptr->now[i + 1])
  620. q->interpolate(q, &buffer1[q->gain_size_factor * i],
  621. gains_ptr->now[i], gains_ptr->now[i + 1]);
  622. }
  623. /* Save away the current to be previous block. */
  624. memcpy(previous_buffer, buffer0,
  625. q->samples_per_channel * sizeof(*previous_buffer));
  626. }
  627. /**
  628. * function for getting the jointstereo coupling information
  629. *
  630. * @param q pointer to the COOKContext
  631. * @param decouple_tab decoupling array
  632. *
  633. */
  634. static void decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
  635. {
  636. int i;
  637. int vlc = get_bits1(&q->gb);
  638. int start = cplband[p->js_subband_start];
  639. int end = cplband[p->subbands-1];
  640. int length = end - start + 1;
  641. if (start > end)
  642. return;
  643. if (vlc) {
  644. for (i = 0; i < length; i++)
  645. decouple_tab[start + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
  646. } else {
  647. for (i = 0; i < length; i++)
  648. decouple_tab[start + i] = get_bits(&q->gb, p->js_vlc_bits);
  649. }
  650. }
  651. /*
  652. * function decouples a pair of signals from a single signal via multiplication.
  653. *
  654. * @param q pointer to the COOKContext
  655. * @param subband index of the current subband
  656. * @param f1 multiplier for channel 1 extraction
  657. * @param f2 multiplier for channel 2 extraction
  658. * @param decode_buffer input buffer
  659. * @param mlt_buffer1 pointer to left channel mlt coefficients
  660. * @param mlt_buffer2 pointer to right channel mlt coefficients
  661. */
  662. static void decouple_float (COOKContext *q,
  663. COOKSubpacket *p,
  664. int subband,
  665. float f1, float f2,
  666. float *decode_buffer,
  667. float *mlt_buffer1, float *mlt_buffer2)
  668. {
  669. int j, tmp_idx;
  670. for (j=0 ; j<SUBBAND_SIZE ; j++) {
  671. tmp_idx = ((p->js_subband_start + subband)*SUBBAND_SIZE)+j;
  672. mlt_buffer1[SUBBAND_SIZE*subband + j] = f1 * decode_buffer[tmp_idx];
  673. mlt_buffer2[SUBBAND_SIZE*subband + j] = f2 * decode_buffer[tmp_idx];
  674. }
  675. }
  676. /**
  677. * function for decoding joint stereo data
  678. *
  679. * @param q pointer to the COOKContext
  680. * @param mlt_buffer1 pointer to left channel mlt coefficients
  681. * @param mlt_buffer2 pointer to right channel mlt coefficients
  682. */
  683. static void joint_decode(COOKContext *q, COOKSubpacket *p, float* mlt_buffer1,
  684. float* mlt_buffer2) {
  685. int i,j;
  686. int decouple_tab[SUBBAND_SIZE];
  687. float *decode_buffer = q->decode_buffer_0;
  688. int idx, cpl_tmp;
  689. float f1,f2;
  690. const float* cplscale;
  691. memset(decouple_tab, 0, sizeof(decouple_tab));
  692. memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
  693. /* Make sure the buffers are zeroed out. */
  694. memset(mlt_buffer1, 0, 1024 * sizeof(*mlt_buffer1));
  695. memset(mlt_buffer2, 0, 1024 * sizeof(*mlt_buffer2));
  696. decouple_info(q, p, decouple_tab);
  697. mono_decode(q, p, decode_buffer);
  698. /* The two channels are stored interleaved in decode_buffer. */
  699. for (i=0 ; i<p->js_subband_start ; i++) {
  700. for (j=0 ; j<SUBBAND_SIZE ; j++) {
  701. mlt_buffer1[i*20+j] = decode_buffer[i*40+j];
  702. mlt_buffer2[i*20+j] = decode_buffer[i*40+20+j];
  703. }
  704. }
  705. /* When we reach js_subband_start (the higher frequencies)
  706. the coefficients are stored in a coupling scheme. */
  707. idx = (1 << p->js_vlc_bits) - 1;
  708. for (i=p->js_subband_start ; i<p->subbands ; i++) {
  709. cpl_tmp = cplband[i];
  710. idx -=decouple_tab[cpl_tmp];
  711. cplscale = q->cplscales[p->js_vlc_bits-2]; //choose decoupler table
  712. f1 = cplscale[decouple_tab[cpl_tmp]];
  713. f2 = cplscale[idx-1];
  714. q->decouple (q, p, i, f1, f2, decode_buffer, mlt_buffer1, mlt_buffer2);
  715. idx = (1 << p->js_vlc_bits) - 1;
  716. }
  717. }
  718. /**
  719. * First part of subpacket decoding:
  720. * decode raw stream bytes and read gain info.
  721. *
  722. * @param q pointer to the COOKContext
  723. * @param inbuffer pointer to raw stream data
  724. * @param gains_ptr array of current/prev gain pointers
  725. */
  726. static inline void
  727. decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p, const uint8_t *inbuffer,
  728. cook_gains *gains_ptr)
  729. {
  730. int offset;
  731. offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
  732. p->bits_per_subpacket/8);
  733. init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
  734. p->bits_per_subpacket);
  735. decode_gain_info(&q->gb, gains_ptr->now);
  736. /* Swap current and previous gains */
  737. FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
  738. }
  739. /**
  740. * Saturate the output signal and interleave.
  741. *
  742. * @param q pointer to the COOKContext
  743. * @param chan channel to saturate
  744. * @param out pointer to the output vector
  745. */
  746. static void saturate_output_float(COOKContext *q, int chan, float *out)
  747. {
  748. int j;
  749. float *output = q->mono_mdct_output + q->samples_per_channel;
  750. for (j = 0; j < q->samples_per_channel; j++) {
  751. out[chan + q->nb_channels * j] = av_clipf(output[j], -1.0, 1.0);
  752. }
  753. }
  754. /**
  755. * Final part of subpacket decoding:
  756. * Apply modulated lapped transform, gain compensation,
  757. * clip and convert to integer.
  758. *
  759. * @param q pointer to the COOKContext
  760. * @param decode_buffer pointer to the mlt coefficients
  761. * @param gains_ptr array of current/prev gain pointers
  762. * @param previous_buffer pointer to the previous buffer to be used for overlapping
  763. * @param out pointer to the output buffer
  764. * @param chan 0: left or single channel, 1: right channel
  765. */
  766. static inline void
  767. mlt_compensate_output(COOKContext *q, float *decode_buffer,
  768. cook_gains *gains_ptr, float *previous_buffer,
  769. float *out, int chan)
  770. {
  771. imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
  772. q->saturate_output (q, chan, out);
  773. }
  774. /**
  775. * Cook subpacket decoding. This function returns one decoded subpacket,
  776. * usually 1024 samples per channel.
  777. *
  778. * @param q pointer to the COOKContext
  779. * @param inbuffer pointer to the inbuffer
  780. * @param outbuffer pointer to the outbuffer
  781. */
  782. static void decode_subpacket(COOKContext *q, COOKSubpacket *p,
  783. const uint8_t *inbuffer, float *outbuffer)
  784. {
  785. int sub_packet_size = p->size;
  786. /* packet dump */
  787. // for (i=0 ; i<sub_packet_size ; i++) {
  788. // av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]);
  789. // }
  790. // av_log(q->avctx, AV_LOG_ERROR, "\n");
  791. memset(q->decode_buffer_1,0,sizeof(q->decode_buffer_1));
  792. decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
  793. if (p->joint_stereo) {
  794. joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2);
  795. } else {
  796. mono_decode(q, p, q->decode_buffer_1);
  797. if (p->num_channels == 2) {
  798. decode_bytes_and_gain(q, p, inbuffer + sub_packet_size/2, &p->gains2);
  799. mono_decode(q, p, q->decode_buffer_2);
  800. }
  801. }
  802. mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
  803. p->mono_previous_buffer1, outbuffer, p->ch_idx);
  804. if (p->num_channels == 2) {
  805. if (p->joint_stereo) {
  806. mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
  807. p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
  808. } else {
  809. mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
  810. p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
  811. }
  812. }
  813. }
  814. /**
  815. * Cook frame decoding
  816. *
  817. * @param avctx pointer to the AVCodecContext
  818. */
  819. static int cook_decode_frame(AVCodecContext *avctx,
  820. void *data, int *data_size,
  821. AVPacket *avpkt) {
  822. const uint8_t *buf = avpkt->data;
  823. int buf_size = avpkt->size;
  824. COOKContext *q = avctx->priv_data;
  825. int i, out_size;
  826. int offset = 0;
  827. int chidx = 0;
  828. if (buf_size < avctx->block_align)
  829. return buf_size;
  830. out_size = q->nb_channels * q->samples_per_channel *
  831. av_get_bytes_per_sample(avctx->sample_fmt);
  832. if (*data_size < out_size) {
  833. av_log(avctx, AV_LOG_ERROR, "Output buffer is too small\n");
  834. return AVERROR(EINVAL);
  835. }
  836. /* estimate subpacket sizes */
  837. q->subpacket[0].size = avctx->block_align;
  838. for(i=1;i<q->num_subpackets;i++){
  839. q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
  840. q->subpacket[0].size -= q->subpacket[i].size + 1;
  841. if (q->subpacket[0].size < 0) {
  842. av_log(avctx,AV_LOG_DEBUG,"frame subpacket size total > avctx->block_align!\n");
  843. return AVERROR_INVALIDDATA;
  844. }
  845. }
  846. /* decode supbackets */
  847. for(i=0;i<q->num_subpackets;i++){
  848. q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size*8)>>q->subpacket[i].bits_per_subpdiv;
  849. q->subpacket[i].ch_idx = chidx;
  850. av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] size %i js %i %i block_align %i\n",i,q->subpacket[i].size,q->subpacket[i].joint_stereo,offset,avctx->block_align);
  851. decode_subpacket(q, &q->subpacket[i], buf + offset, data);
  852. offset += q->subpacket[i].size;
  853. chidx += q->subpacket[i].num_channels;
  854. av_log(avctx,AV_LOG_DEBUG,"subpacket[%i] %i %i\n",i,q->subpacket[i].size * 8,get_bits_count(&q->gb));
  855. }
  856. *data_size = out_size;
  857. /* Discard the first two frames: no valid audio. */
  858. if (avctx->frame_number < 2) *data_size = 0;
  859. return avctx->block_align;
  860. }
  861. #ifdef DEBUG
  862. static void dump_cook_context(COOKContext *q)
  863. {
  864. //int i=0;
  865. #define PRINT(a,b) av_log(q->avctx,AV_LOG_ERROR," %s = %d\n", a, b);
  866. av_log(q->avctx,AV_LOG_ERROR,"COOKextradata\n");
  867. av_log(q->avctx,AV_LOG_ERROR,"cookversion=%x\n",q->subpacket[0].cookversion);
  868. if (q->subpacket[0].cookversion > STEREO) {
  869. PRINT("js_subband_start",q->subpacket[0].js_subband_start);
  870. PRINT("js_vlc_bits",q->subpacket[0].js_vlc_bits);
  871. }
  872. av_log(q->avctx,AV_LOG_ERROR,"COOKContext\n");
  873. PRINT("nb_channels",q->nb_channels);
  874. PRINT("bit_rate",q->bit_rate);
  875. PRINT("sample_rate",q->sample_rate);
  876. PRINT("samples_per_channel",q->subpacket[0].samples_per_channel);
  877. PRINT("samples_per_frame",q->subpacket[0].samples_per_frame);
  878. PRINT("subbands",q->subpacket[0].subbands);
  879. PRINT("js_subband_start",q->subpacket[0].js_subband_start);
  880. PRINT("log2_numvector_size",q->subpacket[0].log2_numvector_size);
  881. PRINT("numvector_size",q->subpacket[0].numvector_size);
  882. PRINT("total_subbands",q->subpacket[0].total_subbands);
  883. }
  884. #endif
  885. static av_cold int cook_count_channels(unsigned int mask){
  886. int i;
  887. int channels = 0;
  888. for(i = 0;i<32;i++){
  889. if(mask & (1<<i))
  890. ++channels;
  891. }
  892. return channels;
  893. }
  894. /**
  895. * Cook initialization
  896. *
  897. * @param avctx pointer to the AVCodecContext
  898. */
  899. static av_cold int cook_decode_init(AVCodecContext *avctx)
  900. {
  901. COOKContext *q = avctx->priv_data;
  902. const uint8_t *edata_ptr = avctx->extradata;
  903. const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
  904. int extradata_size = avctx->extradata_size;
  905. int s = 0;
  906. unsigned int channel_mask = 0;
  907. int ret;
  908. q->avctx = avctx;
  909. /* Take care of the codec specific extradata. */
  910. if (extradata_size <= 0) {
  911. av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");
  912. return AVERROR_INVALIDDATA;
  913. }
  914. av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);
  915. /* Take data from the AVCodecContext (RM container). */
  916. q->sample_rate = avctx->sample_rate;
  917. q->nb_channels = avctx->channels;
  918. q->bit_rate = avctx->bit_rate;
  919. /* Initialize RNG. */
  920. av_lfg_init(&q->random_state, 0);
  921. while(edata_ptr < edata_ptr_end){
  922. /* 8 for mono, 16 for stereo, ? for multichannel
  923. Swap to right endianness so we don't need to care later on. */
  924. if (extradata_size >= 8){
  925. q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
  926. q->subpacket[s].samples_per_frame = bytestream_get_be16(&edata_ptr);
  927. q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
  928. extradata_size -= 8;
  929. }
  930. if (avctx->extradata_size >= 8){
  931. bytestream_get_be32(&edata_ptr); //Unknown unused
  932. q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
  933. q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
  934. extradata_size -= 8;
  935. }
  936. /* Initialize extradata related variables. */
  937. q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame / q->nb_channels;
  938. q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
  939. /* Initialize default data states. */
  940. q->subpacket[s].log2_numvector_size = 5;
  941. q->subpacket[s].total_subbands = q->subpacket[s].subbands;
  942. q->subpacket[s].num_channels = 1;
  943. /* Initialize version-dependent variables */
  944. av_log(avctx,AV_LOG_DEBUG,"subpacket[%i].cookversion=%x\n",s,q->subpacket[s].cookversion);
  945. q->subpacket[s].joint_stereo = 0;
  946. switch (q->subpacket[s].cookversion) {
  947. case MONO:
  948. if (q->nb_channels != 1) {
  949. av_log_ask_for_sample(avctx, "Container channels != 1.\n");
  950. return AVERROR_PATCHWELCOME;
  951. }
  952. av_log(avctx,AV_LOG_DEBUG,"MONO\n");
  953. break;
  954. case STEREO:
  955. if (q->nb_channels != 1) {
  956. q->subpacket[s].bits_per_subpdiv = 1;
  957. q->subpacket[s].num_channels = 2;
  958. }
  959. av_log(avctx,AV_LOG_DEBUG,"STEREO\n");
  960. break;
  961. case JOINT_STEREO:
  962. if (q->nb_channels != 2) {
  963. av_log_ask_for_sample(avctx, "Container channels != 2.\n");
  964. return AVERROR_PATCHWELCOME;
  965. }
  966. av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");
  967. if (avctx->extradata_size >= 16){
  968. q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
  969. q->subpacket[s].joint_stereo = 1;
  970. q->subpacket[s].num_channels = 2;
  971. }
  972. if (q->subpacket[s].samples_per_channel > 256) {
  973. q->subpacket[s].log2_numvector_size = 6;
  974. }
  975. if (q->subpacket[s].samples_per_channel > 512) {
  976. q->subpacket[s].log2_numvector_size = 7;
  977. }
  978. break;
  979. case MC_COOK:
  980. av_log(avctx,AV_LOG_DEBUG,"MULTI_CHANNEL\n");
  981. if(extradata_size >= 4)
  982. channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
  983. if(cook_count_channels(q->subpacket[s].channel_mask) > 1){
  984. q->subpacket[s].total_subbands = q->subpacket[s].subbands + q->subpacket[s].js_subband_start;
  985. q->subpacket[s].joint_stereo = 1;
  986. q->subpacket[s].num_channels = 2;
  987. q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame >> 1;
  988. if (q->subpacket[s].samples_per_channel > 256) {
  989. q->subpacket[s].log2_numvector_size = 6;
  990. }
  991. if (q->subpacket[s].samples_per_channel > 512) {
  992. q->subpacket[s].log2_numvector_size = 7;
  993. }
  994. }else
  995. q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame;
  996. break;
  997. default:
  998. av_log_ask_for_sample(avctx, "Unknown Cook version.\n");
  999. return AVERROR_PATCHWELCOME;
  1000. }
  1001. if(s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
  1002. av_log(avctx,AV_LOG_ERROR,"different number of samples per channel!\n");
  1003. return AVERROR_INVALIDDATA;
  1004. } else
  1005. q->samples_per_channel = q->subpacket[0].samples_per_channel;
  1006. /* Initialize variable relations */
  1007. q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
  1008. /* Try to catch some obviously faulty streams, othervise it might be exploitable */
  1009. if (q->subpacket[s].total_subbands > 53) {
  1010. av_log_ask_for_sample(avctx, "total_subbands > 53\n");
  1011. return AVERROR_PATCHWELCOME;
  1012. }
  1013. if ((q->subpacket[s].js_vlc_bits > 6) || (q->subpacket[s].js_vlc_bits < 2*q->subpacket[s].joint_stereo)) {
  1014. av_log(avctx,AV_LOG_ERROR,"js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
  1015. q->subpacket[s].js_vlc_bits, 2*q->subpacket[s].joint_stereo);
  1016. return AVERROR_INVALIDDATA;
  1017. }
  1018. if (q->subpacket[s].subbands > 50) {
  1019. av_log_ask_for_sample(avctx, "subbands > 50\n");
  1020. return AVERROR_PATCHWELCOME;
  1021. }
  1022. q->subpacket[s].gains1.now = q->subpacket[s].gain_1;
  1023. q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
  1024. q->subpacket[s].gains2.now = q->subpacket[s].gain_3;
  1025. q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
  1026. q->num_subpackets++;
  1027. s++;
  1028. if (s > MAX_SUBPACKETS) {
  1029. av_log_ask_for_sample(avctx, "Too many subpackets > 5\n");
  1030. return AVERROR_PATCHWELCOME;
  1031. }
  1032. }
  1033. /* Generate tables */
  1034. init_pow2table();
  1035. init_gain_table(q);
  1036. init_cplscales_table(q);
  1037. if ((ret = init_cook_vlc_tables(q)))
  1038. return ret;
  1039. if(avctx->block_align >= UINT_MAX/2)
  1040. return AVERROR(EINVAL);
  1041. /* Pad the databuffer with:
  1042. DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
  1043. FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
  1044. q->decoded_bytes_buffer =
  1045. av_mallocz(avctx->block_align
  1046. + DECODE_BYTES_PAD1(avctx->block_align)
  1047. + FF_INPUT_BUFFER_PADDING_SIZE);
  1048. if (q->decoded_bytes_buffer == NULL)
  1049. return AVERROR(ENOMEM);
  1050. /* Initialize transform. */
  1051. if ((ret = init_cook_mlt(q)))
  1052. return ret;
  1053. /* Initialize COOK signal arithmetic handling */
  1054. if (1) {
  1055. q->scalar_dequant = scalar_dequant_float;
  1056. q->decouple = decouple_float;
  1057. q->imlt_window = imlt_window_float;
  1058. q->interpolate = interpolate_float;
  1059. q->saturate_output = saturate_output_float;
  1060. }
  1061. /* Try to catch some obviously faulty streams, othervise it might be exploitable */
  1062. if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) {
  1063. } else {
  1064. av_log_ask_for_sample(avctx,
  1065. "unknown amount of samples_per_channel = %d\n",
  1066. q->samples_per_channel);
  1067. return AVERROR_PATCHWELCOME;
  1068. }
  1069. avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
  1070. if (channel_mask)
  1071. avctx->channel_layout = channel_mask;
  1072. else
  1073. avctx->channel_layout = (avctx->channels==2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
  1074. #ifdef DEBUG
  1075. dump_cook_context(q);
  1076. #endif
  1077. return 0;
  1078. }
  1079. AVCodec ff_cook_decoder =
  1080. {
  1081. .name = "cook",
  1082. .type = AVMEDIA_TYPE_AUDIO,
  1083. .id = CODEC_ID_COOK,
  1084. .priv_data_size = sizeof(COOKContext),
  1085. .init = cook_decode_init,
  1086. .close = cook_decode_close,
  1087. .decode = cook_decode_frame,
  1088. .long_name = NULL_IF_CONFIG_SMALL("COOK"),
  1089. };