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  1. /*
  2. * QDMC compatible decoder
  3. * Copyright (c) 2017 Paul B Mahol
  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. #include <math.h>
  22. #include <stddef.h>
  23. #include <stdio.h>
  24. #define BITSTREAM_READER_LE
  25. #include "libavutil/channel_layout.h"
  26. #include "libavutil/thread.h"
  27. #include "avcodec.h"
  28. #include "bytestream.h"
  29. #include "get_bits.h"
  30. #include "internal.h"
  31. #include "fft.h"
  32. typedef struct QDMCTone {
  33. uint8_t mode;
  34. uint8_t phase;
  35. uint8_t offset;
  36. int16_t freq;
  37. int16_t amplitude;
  38. } QDMCTone;
  39. typedef struct QDMCContext {
  40. AVCodecContext *avctx;
  41. uint8_t frame_bits;
  42. int band_index;
  43. int frame_size;
  44. int subframe_size;
  45. int fft_offset;
  46. int buffer_offset;
  47. int nb_channels;
  48. int checksum_size;
  49. uint8_t noise[2][19][17];
  50. QDMCTone tones[5][8192];
  51. int nb_tones[5];
  52. int cur_tone[5];
  53. float alt_sin[5][31];
  54. float fft_buffer[4][8192 * 2];
  55. float noise2_buffer[4096 * 2];
  56. float noise_buffer[4096 * 2];
  57. float buffer[2 * 32768];
  58. float *buffer_ptr;
  59. int rndval;
  60. DECLARE_ALIGNED(32, FFTComplex, cmplx)[2][512];
  61. FFTContext fft_ctx;
  62. } QDMCContext;
  63. static float sin_table[512];
  64. static VLC vtable[6];
  65. static const unsigned code_prefix[] = {
  66. 0x0, 0x1, 0x2, 0x3, 0x4, 0x6, 0x8, 0xA,
  67. 0xC, 0x10, 0x14, 0x18, 0x1C, 0x24, 0x2C, 0x34,
  68. 0x3C, 0x4C, 0x5C, 0x6C, 0x7C, 0x9C, 0xBC, 0xDC,
  69. 0xFC, 0x13C, 0x17C, 0x1BC, 0x1FC, 0x27C, 0x2FC, 0x37C,
  70. 0x3FC, 0x4FC, 0x5FC, 0x6FC, 0x7FC, 0x9FC, 0xBFC, 0xDFC,
  71. 0xFFC, 0x13FC, 0x17FC, 0x1BFC, 0x1FFC, 0x27FC, 0x2FFC, 0x37FC,
  72. 0x3FFC, 0x4FFC, 0x5FFC, 0x6FFC, 0x7FFC, 0x9FFC, 0xBFFC, 0xDFFC,
  73. 0xFFFC, 0x13FFC, 0x17FFC, 0x1BFFC, 0x1FFFC, 0x27FFC, 0x2FFFC, 0x37FFC,
  74. 0x3FFFC
  75. };
  76. static const float amplitude_tab[64] = {
  77. 1.18750000f, 1.68359380f, 2.37500000f, 3.36718750f, 4.75000000f,
  78. 6.73437500f, 9.50000000f, 13.4687500f, 19.0000000f, 26.9375000f,
  79. 38.0000000f, 53.8750000f, 76.0000000f, 107.750000f, 152.000000f,
  80. 215.500000f, 304.000000f, 431.000000f, 608.000000f, 862.000000f,
  81. 1216.00000f, 1724.00000f, 2432.00000f, 3448.00000f, 4864.00000f,
  82. 6896.00000f, 9728.00000f, 13792.0000f, 19456.0000f, 27584.0000f,
  83. 38912.0000f, 55168.0000f, 77824.0000f, 110336.000f, 155648.000f,
  84. 220672.000f, 311296.000f, 441344.000f, 622592.000f, 882688.000f,
  85. 1245184.00f, 1765376.00f, 2490368.00f, 3530752.00f, 4980736.00f,
  86. 7061504.00f, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
  87. };
  88. static const uint16_t qdmc_nodes[112] = {
  89. 0, 1, 2, 4, 6, 8, 12, 16, 24, 32, 48, 56, 64,
  90. 80, 96, 120, 144, 176, 208, 240, 256,
  91. 0, 2, 4, 8, 16, 24, 32, 48, 56, 64, 80, 104,
  92. 128, 160, 208, 256, 0, 0, 0, 0, 0,
  93. 0, 2, 4, 8, 16, 32, 48, 64, 80, 112, 160, 208,
  94. 256, 0, 0, 0, 0, 0, 0, 0, 0,
  95. 0, 4, 8, 16, 32, 48, 64, 96, 144, 208, 256,
  96. 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
  97. 0, 4, 16, 32, 64, 256, 0, 0, 0, 0, 0, 0, 0, 0,
  98. 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0
  99. };
  100. static const uint8_t noise_bands_size[] = {
  101. 19, 14, 11, 9, 4, 2, 0
  102. };
  103. static const uint8_t noise_bands_selector[] = {
  104. 4, 3, 2, 1, 0, 0, 0,
  105. };
  106. static const uint8_t noise_value_bits[] = {
  107. 12, 7, 9, 7, 10, 9, 11, 9, 9, 2, 9, 9, 9, 9,
  108. 9, 3, 9, 10, 10, 12, 2, 3, 3, 5, 5, 6, 7,
  109. };
  110. static const uint8_t noise_value_symbols[] = {
  111. 0, 10, 11, 12, 13, 14, 15, 16, 18, 1, 20, 22, 24,
  112. 26, 28, 2, 30, 32, 34, 36, 3, 4, 5, 6, 7, 8, 9,
  113. };
  114. static const uint16_t noise_value_codes[] = {
  115. 0xC7A, 0x002, 0x0FA, 0x03A, 0x35A, 0x1C2, 0x07A, 0x1FA,
  116. 0x17A, 0x000, 0x0DA, 0x142, 0x0C2, 0x042, 0x1DA, 0x001,
  117. 0x05A, 0x15A, 0x27A, 0x47A, 0x003, 0x005, 0x006, 0x012,
  118. 0x00A, 0x022, 0x01A,
  119. };
  120. static const uint8_t noise_segment_length_bits[] = {
  121. 10, 8, 5, 1, 2, 4, 4, 4, 6, 7, 9, 10,
  122. };
  123. static const uint8_t noise_segment_length_symbols[] = {
  124. 0, 13, 17, 1, 2, 3, 4, 5, 6, 7, 8, 9,
  125. };
  126. static const uint16_t noise_segment_length_codes[] = {
  127. 0x30B, 0x8B, 0x1B, 0x0, 0x1, 0x3, 0x7, 0xF, 0x2b, 0x4B, 0xB, 0x10B,
  128. };
  129. static const uint8_t freq_diff_bits[] = {
  130. 18, 2, 4, 4, 5, 4, 4, 5, 5, 4, 5, 5, 5, 5, 6, 6, 6, 6, 6, 7, 7, 6,
  131. 7, 6, 6, 6, 7, 7, 7, 7, 7, 8, 9, 9, 8, 9, 11, 11, 12, 12, 13, 12,
  132. 14, 15, 18, 16, 17,
  133. };
  134. static const uint32_t freq_diff_codes[] = {
  135. 0x2AD46, 0x1, 0x0, 0x3, 0xC, 0xA, 0x7, 0x18, 0x12, 0xE, 0x4, 0x16,
  136. 0xF, 0x1C, 0x8, 0x22, 0x26, 0x2, 0x3B, 0x34, 0x74, 0x1F, 0x14, 0x2B,
  137. 0x1B, 0x3F, 0x28, 0x54, 0x6, 0x4B, 0xB, 0x68, 0xE8, 0x46, 0xC6, 0x1E8,
  138. 0x146, 0x346, 0x546, 0x746, 0x1D46, 0xF46, 0xD46, 0x6D46, 0xAD46, 0x2D46,
  139. 0x1AD46,
  140. };
  141. static const uint8_t amplitude_bits[] = {
  142. 13, 7, 8, 9, 10, 10, 10, 10, 10, 9, 8, 7, 6,
  143. 5, 4, 3, 3, 2, 3, 3, 4, 5, 7, 8, 9, 11, 12, 13,
  144. };
  145. static const uint16_t amplitude_codes[] = {
  146. 0x1EC6, 0x6, 0xC2, 0x142, 0x242, 0x246, 0xC6, 0x46, 0x42, 0x146, 0xA2,
  147. 0x62, 0x26, 0x16, 0xE, 0x5, 0x4, 0x3, 0x0, 0x1, 0xA, 0x12, 0x2, 0x22,
  148. 0x1C6, 0x2C6, 0x6C6, 0xEC6,
  149. };
  150. static const uint8_t amplitude_diff_bits[] = {
  151. 8, 2, 1, 3, 4, 5, 6, 7, 8,
  152. };
  153. static const uint8_t amplitude_diff_codes[] = {
  154. 0xFE, 0x0, 0x1, 0x2, 0x6, 0xE, 0x1E, 0x3E, 0x7E,
  155. };
  156. static const uint8_t phase_diff_bits[] = {
  157. 6, 2, 2, 4, 4, 6, 5, 4, 2,
  158. };
  159. static const uint8_t phase_diff_codes[] = {
  160. 0x35, 0x2, 0x0, 0x1, 0xD, 0x15, 0x5, 0x9, 0x3,
  161. };
  162. #define INIT_VLC_STATIC_LE(vlc, nb_bits, nb_codes, \
  163. bits, bits_wrap, bits_size, \
  164. codes, codes_wrap, codes_size, \
  165. symbols, symbols_wrap, symbols_size, \
  166. static_size) \
  167. do { \
  168. static VLC_TYPE table[static_size][2]; \
  169. (vlc)->table = table; \
  170. (vlc)->table_allocated = static_size; \
  171. ff_init_vlc_sparse(vlc, nb_bits, nb_codes, \
  172. bits, bits_wrap, bits_size, \
  173. codes, codes_wrap, codes_size, \
  174. symbols, symbols_wrap, symbols_size, \
  175. INIT_VLC_LE | INIT_VLC_USE_NEW_STATIC); \
  176. } while (0)
  177. static av_cold void qdmc_init_static_data(void)
  178. {
  179. int i;
  180. INIT_VLC_STATIC_LE(&vtable[0], 12, FF_ARRAY_ELEMS(noise_value_bits),
  181. noise_value_bits, 1, 1, noise_value_codes, 2, 2, noise_value_symbols, 1, 1, 4096);
  182. INIT_VLC_STATIC_LE(&vtable[1], 10, FF_ARRAY_ELEMS(noise_segment_length_bits),
  183. noise_segment_length_bits, 1, 1, noise_segment_length_codes, 2, 2,
  184. noise_segment_length_symbols, 1, 1, 1024);
  185. INIT_VLC_STATIC_LE(&vtable[2], 12, FF_ARRAY_ELEMS(amplitude_bits),
  186. amplitude_bits, 1, 1, amplitude_codes, 2, 2, NULL, 0, 0, 4098);
  187. INIT_VLC_STATIC_LE(&vtable[3], 12, FF_ARRAY_ELEMS(freq_diff_bits),
  188. freq_diff_bits, 1, 1, freq_diff_codes, 4, 4, NULL, 0, 0, 4160);
  189. INIT_VLC_STATIC_LE(&vtable[4], 8, FF_ARRAY_ELEMS(amplitude_diff_bits),
  190. amplitude_diff_bits, 1, 1, amplitude_diff_codes, 1, 1, NULL, 0, 0, 256);
  191. INIT_VLC_STATIC_LE(&vtable[5], 6, FF_ARRAY_ELEMS(phase_diff_bits),
  192. phase_diff_bits, 1, 1, phase_diff_codes, 1, 1, NULL, 0, 0, 64);
  193. for (i = 0; i < 512; i++)
  194. sin_table[i] = sin(2.0f * i * M_PI * 0.001953125f);
  195. }
  196. static void make_noises(QDMCContext *s)
  197. {
  198. int i, j, n0, n1, n2, diff;
  199. float *nptr;
  200. for (j = 0; j < noise_bands_size[s->band_index]; j++) {
  201. n0 = qdmc_nodes[j + 21 * s->band_index ];
  202. n1 = qdmc_nodes[j + 21 * s->band_index + 1];
  203. n2 = qdmc_nodes[j + 21 * s->band_index + 2];
  204. nptr = s->noise_buffer + 256 * j;
  205. for (i = 0; i + n0 < n1; i++, nptr++)
  206. nptr[0] = i / (float)(n1 - n0);
  207. diff = n2 - n1;
  208. nptr = s->noise_buffer + (j << 8) + n1 - n0;
  209. for (i = n1; i < n2; i++, nptr++, diff--)
  210. nptr[0] = diff / (float)(n2 - n1);
  211. }
  212. }
  213. static av_cold int qdmc_decode_init(AVCodecContext *avctx)
  214. {
  215. static AVOnce init_static_once = AV_ONCE_INIT;
  216. QDMCContext *s = avctx->priv_data;
  217. int ret, fft_size, fft_order, size, g, j, x;
  218. GetByteContext b;
  219. ff_thread_once(&init_static_once, qdmc_init_static_data);
  220. if (!avctx->extradata || (avctx->extradata_size < 48)) {
  221. av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
  222. return AVERROR_INVALIDDATA;
  223. }
  224. bytestream2_init(&b, avctx->extradata, avctx->extradata_size);
  225. while (bytestream2_get_bytes_left(&b) > 8) {
  226. if (bytestream2_peek_be64(&b) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
  227. (uint64_t)MKBETAG('Q','D','M','C')))
  228. break;
  229. bytestream2_skipu(&b, 1);
  230. }
  231. bytestream2_skipu(&b, 8);
  232. if (bytestream2_get_bytes_left(&b) < 36) {
  233. av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
  234. bytestream2_get_bytes_left(&b));
  235. return AVERROR_INVALIDDATA;
  236. }
  237. size = bytestream2_get_be32u(&b);
  238. if (size > bytestream2_get_bytes_left(&b)) {
  239. av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
  240. bytestream2_get_bytes_left(&b), size);
  241. return AVERROR_INVALIDDATA;
  242. }
  243. if (bytestream2_get_be32u(&b) != MKBETAG('Q','D','C','A')) {
  244. av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
  245. return AVERROR_INVALIDDATA;
  246. }
  247. bytestream2_skipu(&b, 4);
  248. avctx->channels = s->nb_channels = bytestream2_get_be32u(&b);
  249. if (s->nb_channels <= 0 || s->nb_channels > 2) {
  250. av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
  251. return AVERROR_INVALIDDATA;
  252. }
  253. avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
  254. AV_CH_LAYOUT_MONO;
  255. avctx->sample_rate = bytestream2_get_be32u(&b);
  256. avctx->bit_rate = bytestream2_get_be32u(&b);
  257. bytestream2_skipu(&b, 4);
  258. fft_size = bytestream2_get_be32u(&b);
  259. fft_order = av_log2(fft_size) + 1;
  260. s->checksum_size = bytestream2_get_be32u(&b);
  261. if (s->checksum_size >= 1U << 28) {
  262. av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
  263. return AVERROR_INVALIDDATA;
  264. }
  265. if (avctx->sample_rate >= 32000) {
  266. x = 28000;
  267. s->frame_bits = 13;
  268. } else if (avctx->sample_rate >= 16000) {
  269. x = 20000;
  270. s->frame_bits = 12;
  271. } else {
  272. x = 16000;
  273. s->frame_bits = 11;
  274. }
  275. s->frame_size = 1 << s->frame_bits;
  276. s->subframe_size = s->frame_size >> 5;
  277. if (avctx->channels == 2)
  278. x = 3 * x / 2;
  279. s->band_index = noise_bands_selector[FFMIN(6, llrint(floor(avctx->bit_rate * 3.0 / (double)x + 0.5)))];
  280. if ((fft_order < 7) || (fft_order > 9)) {
  281. avpriv_request_sample(avctx, "Unknown FFT order %d", fft_order);
  282. return AVERROR_PATCHWELCOME;
  283. }
  284. if (fft_size != (1 << (fft_order - 1))) {
  285. av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", fft_size);
  286. return AVERROR_INVALIDDATA;
  287. }
  288. ret = ff_fft_init(&s->fft_ctx, fft_order, 1);
  289. if (ret < 0)
  290. return ret;
  291. avctx->sample_fmt = AV_SAMPLE_FMT_S16;
  292. for (g = 5; g > 0; g--) {
  293. for (j = 0; j < (1 << g) - 1; j++)
  294. s->alt_sin[5-g][j] = sin_table[(((j+1) << (8 - g)) & 0x1FF)];
  295. }
  296. make_noises(s);
  297. return 0;
  298. }
  299. static av_cold int qdmc_decode_close(AVCodecContext *avctx)
  300. {
  301. QDMCContext *s = avctx->priv_data;
  302. ff_fft_end(&s->fft_ctx);
  303. return 0;
  304. }
  305. static int qdmc_get_vlc(GetBitContext *gb, VLC *table, int flag)
  306. {
  307. int v;
  308. if (get_bits_left(gb) < 1)
  309. return AVERROR_INVALIDDATA;
  310. v = get_vlc2(gb, table->table, table->bits, 2);
  311. if (v)
  312. v = v - 1;
  313. else
  314. v = get_bits(gb, get_bits(gb, 3) + 1);
  315. if (flag) {
  316. if (v >= FF_ARRAY_ELEMS(code_prefix))
  317. return AVERROR_INVALIDDATA;
  318. v = code_prefix[v] + get_bitsz(gb, v >> 2);
  319. }
  320. return v;
  321. }
  322. static int skip_label(QDMCContext *s, GetBitContext *gb)
  323. {
  324. uint32_t label = get_bits_long(gb, 32);
  325. uint16_t sum = 226, checksum = get_bits(gb, 16);
  326. const uint8_t *ptr = gb->buffer + 6;
  327. int i;
  328. if (label != MKTAG('Q', 'M', 'C', 1))
  329. return AVERROR_INVALIDDATA;
  330. for (i = 0; i < s->checksum_size - 6; i++)
  331. sum += ptr[i];
  332. return sum != checksum;
  333. }
  334. static int read_noise_data(QDMCContext *s, GetBitContext *gb)
  335. {
  336. int ch, j, k, v, idx, band, lastval, newval, len;
  337. for (ch = 0; ch < s->nb_channels; ch++) {
  338. for (band = 0; band < noise_bands_size[s->band_index]; band++) {
  339. v = qdmc_get_vlc(gb, &vtable[0], 0);
  340. if (v < 0)
  341. return AVERROR_INVALIDDATA;
  342. if (v & 1)
  343. v = v + 1;
  344. else
  345. v = -v;
  346. lastval = v / 2;
  347. s->noise[ch][band][0] = lastval - 1;
  348. for (j = 0; j < 15;) {
  349. len = qdmc_get_vlc(gb, &vtable[1], 1);
  350. if (len < 0)
  351. return AVERROR_INVALIDDATA;
  352. len += 1;
  353. v = qdmc_get_vlc(gb, &vtable[0], 0);
  354. if (v < 0)
  355. return AVERROR_INVALIDDATA;
  356. if (v & 1)
  357. newval = lastval + (v + 1) / 2;
  358. else
  359. newval = lastval - v / 2;
  360. idx = j + 1;
  361. if (len + idx > 16)
  362. return AVERROR_INVALIDDATA;
  363. for (k = 1; idx <= j + len; k++, idx++)
  364. s->noise[ch][band][idx] = lastval + k * (newval - lastval) / len - 1;
  365. lastval = newval;
  366. j += len;
  367. }
  368. }
  369. }
  370. return 0;
  371. }
  372. static void add_tone(QDMCContext *s, int group, int offset, int freq, int stereo_mode, int amplitude, int phase)
  373. {
  374. const int index = s->nb_tones[group];
  375. if (index >= FF_ARRAY_ELEMS(s->tones[group])) {
  376. av_log(s->avctx, AV_LOG_WARNING, "Too many tones already in buffer, ignoring tone!\n");
  377. return;
  378. }
  379. s->tones[group][index].offset = offset;
  380. s->tones[group][index].freq = freq;
  381. s->tones[group][index].mode = stereo_mode;
  382. s->tones[group][index].amplitude = amplitude;
  383. s->tones[group][index].phase = phase;
  384. s->nb_tones[group]++;
  385. }
  386. static int read_wave_data(QDMCContext *s, GetBitContext *gb)
  387. {
  388. int amp, phase, stereo_mode = 0, i, group, freq, group_size, group_bits;
  389. int amp2, phase2, pos2, off;
  390. for (group = 0; group < 5; group++) {
  391. group_size = 1 << (s->frame_bits - group - 1);
  392. group_bits = 4 - group;
  393. pos2 = 0;
  394. off = 0;
  395. for (i = 1; ; i = freq + 1) {
  396. int v;
  397. v = qdmc_get_vlc(gb, &vtable[3], 1);
  398. if (v < 0)
  399. return AVERROR_INVALIDDATA;
  400. freq = i + v;
  401. while (freq >= group_size - 1) {
  402. freq += 2 - group_size;
  403. pos2 += group_size;
  404. off += 1 << group_bits;
  405. }
  406. if (pos2 >= s->frame_size)
  407. break;
  408. if (s->nb_channels > 1)
  409. stereo_mode = get_bits(gb, 2);
  410. amp = qdmc_get_vlc(gb, &vtable[2], 0);
  411. if (amp < 0)
  412. return AVERROR_INVALIDDATA;
  413. phase = get_bits(gb, 3);
  414. if (stereo_mode > 1) {
  415. amp2 = qdmc_get_vlc(gb, &vtable[4], 0);
  416. if (amp2 < 0)
  417. return AVERROR_INVALIDDATA;
  418. amp2 = amp - amp2;
  419. phase2 = qdmc_get_vlc(gb, &vtable[5], 0);
  420. if (phase2 < 0)
  421. return AVERROR_INVALIDDATA;
  422. phase2 = phase - phase2;
  423. if (phase2 < 0)
  424. phase2 += 8;
  425. }
  426. if ((freq >> group_bits) + 1 < s->subframe_size) {
  427. add_tone(s, group, off, freq, stereo_mode & 1, amp, phase);
  428. if (stereo_mode > 1)
  429. add_tone(s, group, off, freq, ~stereo_mode & 1, amp2, phase2);
  430. }
  431. }
  432. }
  433. return 0;
  434. }
  435. static void lin_calc(QDMCContext *s, float amplitude, int node1, int node2, int index)
  436. {
  437. int subframe_size, i, j, k, length;
  438. float scale, *noise_ptr;
  439. scale = 0.5 * amplitude;
  440. subframe_size = s->subframe_size;
  441. if (subframe_size >= node2)
  442. subframe_size = node2;
  443. length = (subframe_size - node1) & 0xFFFC;
  444. j = node1;
  445. noise_ptr = &s->noise_buffer[256 * index];
  446. for (i = 0; i < length; i += 4, j+= 4, noise_ptr += 4) {
  447. s->noise2_buffer[j ] += scale * noise_ptr[0];
  448. s->noise2_buffer[j + 1] += scale * noise_ptr[1];
  449. s->noise2_buffer[j + 2] += scale * noise_ptr[2];
  450. s->noise2_buffer[j + 3] += scale * noise_ptr[3];
  451. }
  452. k = length + node1;
  453. noise_ptr = s->noise_buffer + length + (index << 8);
  454. for (i = length; i < subframe_size - node1; i++, k++, noise_ptr++)
  455. s->noise2_buffer[k] += scale * noise_ptr[0];
  456. }
  457. static void add_noise(QDMCContext *s, int ch, int current_subframe)
  458. {
  459. int i, j, aindex;
  460. float amplitude;
  461. float *im = &s->fft_buffer[0 + ch][s->fft_offset + s->subframe_size * current_subframe];
  462. float *re = &s->fft_buffer[2 + ch][s->fft_offset + s->subframe_size * current_subframe];
  463. memset(s->noise2_buffer, 0, 4 * s->subframe_size);
  464. for (i = 0; i < noise_bands_size[s->band_index]; i++) {
  465. if (qdmc_nodes[i + 21 * s->band_index] > s->subframe_size - 1)
  466. break;
  467. aindex = s->noise[ch][i][current_subframe / 2];
  468. amplitude = aindex > 0 ? amplitude_tab[aindex & 0x3F] : 0.0f;
  469. lin_calc(s, amplitude, qdmc_nodes[21 * s->band_index + i],
  470. qdmc_nodes[21 * s->band_index + i + 2], i);
  471. }
  472. for (j = 2; j < s->subframe_size - 1; j++) {
  473. float rnd_re, rnd_im;
  474. s->rndval = 214013U * s->rndval + 2531011;
  475. rnd_im = ((s->rndval & 0x7FFF) - 16384.0f) * 0.000030517578f * s->noise2_buffer[j];
  476. s->rndval = 214013U * s->rndval + 2531011;
  477. rnd_re = ((s->rndval & 0x7FFF) - 16384.0f) * 0.000030517578f * s->noise2_buffer[j];
  478. im[j ] += rnd_im;
  479. re[j ] += rnd_re;
  480. im[j+1] -= rnd_im;
  481. re[j+1] -= rnd_re;
  482. }
  483. }
  484. static void add_wave(QDMCContext *s, int offset, int freqs, int group, int stereo_mode, int amp, int phase)
  485. {
  486. int j, group_bits, pos, pindex;
  487. float im, re, amplitude, level, *imptr, *reptr;
  488. if (s->nb_channels == 1)
  489. stereo_mode = 0;
  490. group_bits = 4 - group;
  491. pos = freqs >> (4 - group);
  492. amplitude = amplitude_tab[amp & 0x3F];
  493. imptr = &s->fft_buffer[ stereo_mode][s->fft_offset + s->subframe_size * offset + pos];
  494. reptr = &s->fft_buffer[2 + stereo_mode][s->fft_offset + s->subframe_size * offset + pos];
  495. pindex = (phase << 6) - ((2 * (freqs >> (4 - group)) + 1) << 7);
  496. for (j = 0; j < (1 << (group_bits + 1)) - 1; j++) {
  497. pindex += (2 * freqs + 1) << (7 - group_bits);
  498. level = amplitude * s->alt_sin[group][j];
  499. im = level * sin_table[ pindex & 0x1FF];
  500. re = level * sin_table[(pindex + 128) & 0x1FF];
  501. imptr[0] += im;
  502. imptr[1] -= im;
  503. reptr[0] += re;
  504. reptr[1] -= re;
  505. imptr += s->subframe_size;
  506. reptr += s->subframe_size;
  507. if (imptr >= &s->fft_buffer[stereo_mode][2 * s->frame_size]) {
  508. imptr = &s->fft_buffer[0 + stereo_mode][pos];
  509. reptr = &s->fft_buffer[2 + stereo_mode][pos];
  510. }
  511. }
  512. }
  513. static void add_wave0(QDMCContext *s, int offset, int freqs, int stereo_mode, int amp, int phase)
  514. {
  515. float level, im, re;
  516. int pos;
  517. if (s->nb_channels == 1)
  518. stereo_mode = 0;
  519. level = amplitude_tab[amp & 0x3F];
  520. im = level * sin_table[ (phase << 6) & 0x1FF];
  521. re = level * sin_table[((phase << 6) + 128) & 0x1FF];
  522. pos = s->fft_offset + freqs + s->subframe_size * offset;
  523. s->fft_buffer[ stereo_mode][pos ] += im;
  524. s->fft_buffer[2 + stereo_mode][pos ] += re;
  525. s->fft_buffer[ stereo_mode][pos + 1] -= im;
  526. s->fft_buffer[2 + stereo_mode][pos + 1] -= re;
  527. }
  528. static void add_waves(QDMCContext *s, int current_subframe)
  529. {
  530. int w, g;
  531. for (g = 0; g < 4; g++) {
  532. for (w = s->cur_tone[g]; w < s->nb_tones[g]; w++) {
  533. QDMCTone *t = &s->tones[g][w];
  534. if (current_subframe < t->offset)
  535. break;
  536. add_wave(s, t->offset, t->freq, g, t->mode, t->amplitude, t->phase);
  537. }
  538. s->cur_tone[g] = w;
  539. }
  540. for (w = s->cur_tone[4]; w < s->nb_tones[4]; w++) {
  541. QDMCTone *t = &s->tones[4][w];
  542. if (current_subframe < t->offset)
  543. break;
  544. add_wave0(s, t->offset, t->freq, t->mode, t->amplitude, t->phase);
  545. }
  546. s->cur_tone[4] = w;
  547. }
  548. static int decode_frame(QDMCContext *s, GetBitContext *gb, int16_t *out)
  549. {
  550. int ret, ch, i, n;
  551. if (skip_label(s, gb))
  552. return AVERROR_INVALIDDATA;
  553. s->fft_offset = s->frame_size - s->fft_offset;
  554. s->buffer_ptr = &s->buffer[s->nb_channels * s->buffer_offset];
  555. ret = read_noise_data(s, gb);
  556. if (ret < 0)
  557. return ret;
  558. ret = read_wave_data(s, gb);
  559. if (ret < 0)
  560. return ret;
  561. for (n = 0; n < 32; n++) {
  562. float *r;
  563. for (ch = 0; ch < s->nb_channels; ch++)
  564. add_noise(s, ch, n);
  565. add_waves(s, n);
  566. for (ch = 0; ch < s->nb_channels; ch++) {
  567. for (i = 0; i < s->subframe_size; i++) {
  568. s->cmplx[ch][i].re = s->fft_buffer[ch + 2][s->fft_offset + n * s->subframe_size + i];
  569. s->cmplx[ch][i].im = s->fft_buffer[ch + 0][s->fft_offset + n * s->subframe_size + i];
  570. s->cmplx[ch][s->subframe_size + i].re = 0;
  571. s->cmplx[ch][s->subframe_size + i].im = 0;
  572. }
  573. }
  574. for (ch = 0; ch < s->nb_channels; ch++) {
  575. s->fft_ctx.fft_permute(&s->fft_ctx, s->cmplx[ch]);
  576. s->fft_ctx.fft_calc(&s->fft_ctx, s->cmplx[ch]);
  577. }
  578. r = &s->buffer_ptr[s->nb_channels * n * s->subframe_size];
  579. for (i = 0; i < 2 * s->subframe_size; i++) {
  580. for (ch = 0; ch < s->nb_channels; ch++) {
  581. *r++ += s->cmplx[ch][i].re;
  582. }
  583. }
  584. r = &s->buffer_ptr[n * s->subframe_size * s->nb_channels];
  585. for (i = 0; i < s->nb_channels * s->subframe_size; i++) {
  586. out[i] = av_clipf(r[i], INT16_MIN, INT16_MAX);
  587. }
  588. out += s->subframe_size * s->nb_channels;
  589. for (ch = 0; ch < s->nb_channels; ch++) {
  590. memset(s->fft_buffer[ch+0] + s->fft_offset + n * s->subframe_size, 0, 4 * s->subframe_size);
  591. memset(s->fft_buffer[ch+2] + s->fft_offset + n * s->subframe_size, 0, 4 * s->subframe_size);
  592. }
  593. memset(s->buffer + s->nb_channels * (n * s->subframe_size + s->frame_size + s->buffer_offset), 0, 4 * s->subframe_size * s->nb_channels);
  594. }
  595. s->buffer_offset += s->frame_size;
  596. if (s->buffer_offset >= 32768 - s->frame_size) {
  597. memcpy(s->buffer, &s->buffer[s->nb_channels * s->buffer_offset], 4 * s->frame_size * s->nb_channels);
  598. s->buffer_offset = 0;
  599. }
  600. return 0;
  601. }
  602. static av_cold void qdmc_flush(AVCodecContext *avctx)
  603. {
  604. QDMCContext *s = avctx->priv_data;
  605. memset(s->buffer, 0, sizeof(s->buffer));
  606. memset(s->fft_buffer, 0, sizeof(s->fft_buffer));
  607. s->fft_offset = 0;
  608. s->buffer_offset = 0;
  609. }
  610. static int qdmc_decode_frame(AVCodecContext *avctx, void *data,
  611. int *got_frame_ptr, AVPacket *avpkt)
  612. {
  613. QDMCContext *s = avctx->priv_data;
  614. AVFrame *frame = data;
  615. GetBitContext gb;
  616. int ret;
  617. if (!avpkt->data)
  618. return 0;
  619. if (avpkt->size < s->checksum_size)
  620. return AVERROR_INVALIDDATA;
  621. s->avctx = avctx;
  622. frame->nb_samples = s->frame_size;
  623. if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  624. return ret;
  625. if ((ret = init_get_bits8(&gb, avpkt->data, s->checksum_size)) < 0)
  626. return ret;
  627. memset(s->nb_tones, 0, sizeof(s->nb_tones));
  628. memset(s->cur_tone, 0, sizeof(s->cur_tone));
  629. ret = decode_frame(s, &gb, (int16_t *)frame->data[0]);
  630. if (ret >= 0) {
  631. *got_frame_ptr = 1;
  632. return s->checksum_size;
  633. }
  634. qdmc_flush(avctx);
  635. return ret;
  636. }
  637. AVCodec ff_qdmc_decoder = {
  638. .name = "qdmc",
  639. .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 1"),
  640. .type = AVMEDIA_TYPE_AUDIO,
  641. .id = AV_CODEC_ID_QDMC,
  642. .priv_data_size = sizeof(QDMCContext),
  643. .init = qdmc_decode_init,
  644. .close = qdmc_decode_close,
  645. .decode = qdmc_decode_frame,
  646. .flush = qdmc_flush,
  647. .capabilities = AV_CODEC_CAP_DR1,
  648. };