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  1. /*
  2. * DSP Group TrueSpeech compatible decoder
  3. * Copyright (c) 2005 Konstantin Shishkov
  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 "libavutil/intreadwrite.h"
  22. #include "avcodec.h"
  23. #include "truespeech_data.h"
  24. /**
  25. * @file
  26. * TrueSpeech decoder.
  27. */
  28. /**
  29. * TrueSpeech decoder context
  30. */
  31. typedef struct {
  32. /* input data */
  33. int16_t vector[8]; //< input vector: 5/5/4/4/4/3/3/3
  34. int offset1[2]; //< 8-bit value, used in one copying offset
  35. int offset2[4]; //< 7-bit value, encodes offsets for copying and for two-point filter
  36. int pulseoff[4]; //< 4-bit offset of pulse values block
  37. int pulsepos[4]; //< 27-bit variable, encodes 7 pulse positions
  38. int pulseval[4]; //< 7x2-bit pulse values
  39. int flag; //< 1-bit flag, shows how to choose filters
  40. /* temporary data */
  41. int filtbuf[146]; // some big vector used for storing filters
  42. int prevfilt[8]; // filter from previous frame
  43. int16_t tmp1[8]; // coefficients for adding to out
  44. int16_t tmp2[8]; // coefficients for adding to out
  45. int16_t tmp3[8]; // coefficients for adding to out
  46. int16_t cvector[8]; // correlated input vector
  47. int filtval; // gain value for one function
  48. int16_t newvec[60]; // tmp vector
  49. int16_t filters[32]; // filters for every subframe
  50. } TSContext;
  51. static av_cold int truespeech_decode_init(AVCodecContext * avctx)
  52. {
  53. // TSContext *c = avctx->priv_data;
  54. if (avctx->channels != 1) {
  55. av_log_ask_for_sample(avctx, "Unsupported channel count: %d\n", avctx->channels);
  56. return AVERROR(EINVAL);
  57. }
  58. avctx->sample_fmt = AV_SAMPLE_FMT_S16;
  59. return 0;
  60. }
  61. static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
  62. {
  63. uint32_t t;
  64. /* first dword */
  65. t = AV_RL32(input);
  66. input += 4;
  67. dec->flag = t & 1;
  68. dec->vector[0] = ts_codebook[0][(t >> 1) & 0x1F];
  69. dec->vector[1] = ts_codebook[1][(t >> 6) & 0x1F];
  70. dec->vector[2] = ts_codebook[2][(t >> 11) & 0xF];
  71. dec->vector[3] = ts_codebook[3][(t >> 15) & 0xF];
  72. dec->vector[4] = ts_codebook[4][(t >> 19) & 0xF];
  73. dec->vector[5] = ts_codebook[5][(t >> 23) & 0x7];
  74. dec->vector[6] = ts_codebook[6][(t >> 26) & 0x7];
  75. dec->vector[7] = ts_codebook[7][(t >> 29) & 0x7];
  76. /* second dword */
  77. t = AV_RL32(input);
  78. input += 4;
  79. dec->offset2[0] = (t >> 0) & 0x7F;
  80. dec->offset2[1] = (t >> 7) & 0x7F;
  81. dec->offset2[2] = (t >> 14) & 0x7F;
  82. dec->offset2[3] = (t >> 21) & 0x7F;
  83. dec->offset1[0] = ((t >> 28) & 0xF) << 4;
  84. /* third dword */
  85. t = AV_RL32(input);
  86. input += 4;
  87. dec->pulseval[0] = (t >> 0) & 0x3FFF;
  88. dec->pulseval[1] = (t >> 14) & 0x3FFF;
  89. dec->offset1[1] = (t >> 28) & 0x0F;
  90. /* fourth dword */
  91. t = AV_RL32(input);
  92. input += 4;
  93. dec->pulseval[2] = (t >> 0) & 0x3FFF;
  94. dec->pulseval[3] = (t >> 14) & 0x3FFF;
  95. dec->offset1[1] |= ((t >> 28) & 0x0F) << 4;
  96. /* fifth dword */
  97. t = AV_RL32(input);
  98. input += 4;
  99. dec->pulsepos[0] = (t >> 4) & 0x7FFFFFF;
  100. dec->pulseoff[0] = (t >> 0) & 0xF;
  101. dec->offset1[0] |= (t >> 31) & 1;
  102. /* sixth dword */
  103. t = AV_RL32(input);
  104. input += 4;
  105. dec->pulsepos[1] = (t >> 4) & 0x7FFFFFF;
  106. dec->pulseoff[1] = (t >> 0) & 0xF;
  107. dec->offset1[0] |= ((t >> 31) & 1) << 1;
  108. /* seventh dword */
  109. t = AV_RL32(input);
  110. input += 4;
  111. dec->pulsepos[2] = (t >> 4) & 0x7FFFFFF;
  112. dec->pulseoff[2] = (t >> 0) & 0xF;
  113. dec->offset1[0] |= ((t >> 31) & 1) << 2;
  114. /* eighth dword */
  115. t = AV_RL32(input);
  116. input += 4;
  117. dec->pulsepos[3] = (t >> 4) & 0x7FFFFFF;
  118. dec->pulseoff[3] = (t >> 0) & 0xF;
  119. dec->offset1[0] |= ((t >> 31) & 1) << 3;
  120. }
  121. static void truespeech_correlate_filter(TSContext *dec)
  122. {
  123. int16_t tmp[8];
  124. int i, j;
  125. for(i = 0; i < 8; i++){
  126. if(i > 0){
  127. memcpy(tmp, dec->cvector, i * 2);
  128. for(j = 0; j < i; j++)
  129. dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) +
  130. (dec->cvector[j] << 15) + 0x4000) >> 15;
  131. }
  132. dec->cvector[i] = (8 - dec->vector[i]) >> 3;
  133. }
  134. for(i = 0; i < 8; i++)
  135. dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
  136. dec->filtval = dec->vector[0];
  137. }
  138. static void truespeech_filters_merge(TSContext *dec)
  139. {
  140. int i;
  141. if(!dec->flag){
  142. for(i = 0; i < 8; i++){
  143. dec->filters[i + 0] = dec->prevfilt[i];
  144. dec->filters[i + 8] = dec->prevfilt[i];
  145. }
  146. }else{
  147. for(i = 0; i < 8; i++){
  148. dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
  149. dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
  150. }
  151. }
  152. for(i = 0; i < 8; i++){
  153. dec->filters[i + 16] = dec->cvector[i];
  154. dec->filters[i + 24] = dec->cvector[i];
  155. }
  156. }
  157. static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
  158. {
  159. int16_t tmp[146 + 60], *ptr0, *ptr1;
  160. const int16_t *filter;
  161. int i, t, off;
  162. t = dec->offset2[quart];
  163. if(t == 127){
  164. memset(dec->newvec, 0, 60 * 2);
  165. return;
  166. }
  167. for(i = 0; i < 146; i++)
  168. tmp[i] = dec->filtbuf[i];
  169. off = (t / 25) + dec->offset1[quart >> 1] + 18;
  170. ptr0 = tmp + 145 - off;
  171. ptr1 = tmp + 146;
  172. filter = (const int16_t*)ts_order2_coeffs + (t % 25) * 2;
  173. for(i = 0; i < 60; i++){
  174. t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
  175. ptr0++;
  176. dec->newvec[i] = t;
  177. ptr1[i] = t;
  178. }
  179. }
  180. static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
  181. {
  182. int16_t tmp[7];
  183. int i, j, t;
  184. const int16_t *ptr1;
  185. int16_t *ptr2;
  186. int coef;
  187. memset(out, 0, 60 * 2);
  188. for(i = 0; i < 7; i++) {
  189. t = dec->pulseval[quart] & 3;
  190. dec->pulseval[quart] >>= 2;
  191. tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
  192. }
  193. coef = dec->pulsepos[quart] >> 15;
  194. ptr1 = (const int16_t*)ts_pulse_values + 30;
  195. ptr2 = tmp;
  196. for(i = 0, j = 3; (i < 30) && (j > 0); i++){
  197. t = *ptr1++;
  198. if(coef >= t)
  199. coef -= t;
  200. else{
  201. out[i] = *ptr2++;
  202. ptr1 += 30;
  203. j--;
  204. }
  205. }
  206. coef = dec->pulsepos[quart] & 0x7FFF;
  207. ptr1 = (const int16_t*)ts_pulse_values;
  208. for(i = 30, j = 4; (i < 60) && (j > 0); i++){
  209. t = *ptr1++;
  210. if(coef >= t)
  211. coef -= t;
  212. else{
  213. out[i] = *ptr2++;
  214. ptr1 += 30;
  215. j--;
  216. }
  217. }
  218. }
  219. static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
  220. {
  221. int i;
  222. for(i = 0; i < 86; i++)
  223. dec->filtbuf[i] = dec->filtbuf[i + 60];
  224. for(i = 0; i < 60; i++){
  225. dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
  226. out[i] += dec->newvec[i];
  227. }
  228. }
  229. static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
  230. {
  231. int i,k;
  232. int t[8];
  233. int16_t *ptr0, *ptr1;
  234. ptr0 = dec->tmp1;
  235. ptr1 = dec->filters + quart * 8;
  236. for(i = 0; i < 60; i++){
  237. int sum = 0;
  238. for(k = 0; k < 8; k++)
  239. sum += ptr0[k] * ptr1[k];
  240. sum = (sum + (out[i] << 12) + 0x800) >> 12;
  241. out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
  242. for(k = 7; k > 0; k--)
  243. ptr0[k] = ptr0[k - 1];
  244. ptr0[0] = out[i];
  245. }
  246. for(i = 0; i < 8; i++)
  247. t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
  248. ptr0 = dec->tmp2;
  249. for(i = 0; i < 60; i++){
  250. int sum = 0;
  251. for(k = 0; k < 8; k++)
  252. sum += ptr0[k] * t[k];
  253. for(k = 7; k > 0; k--)
  254. ptr0[k] = ptr0[k - 1];
  255. ptr0[0] = out[i];
  256. out[i] = ((out[i] << 12) - sum) >> 12;
  257. }
  258. for(i = 0; i < 8; i++)
  259. t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
  260. ptr0 = dec->tmp3;
  261. for(i = 0; i < 60; i++){
  262. int sum = out[i] << 12;
  263. for(k = 0; k < 8; k++)
  264. sum += ptr0[k] * t[k];
  265. for(k = 7; k > 0; k--)
  266. ptr0[k] = ptr0[k - 1];
  267. ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
  268. sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
  269. sum = sum - (sum >> 3);
  270. out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
  271. }
  272. }
  273. static void truespeech_save_prevvec(TSContext *c)
  274. {
  275. int i;
  276. for(i = 0; i < 8; i++)
  277. c->prevfilt[i] = c->cvector[i];
  278. }
  279. static int truespeech_decode_frame(AVCodecContext *avctx,
  280. void *data, int *data_size,
  281. AVPacket *avpkt)
  282. {
  283. const uint8_t *buf = avpkt->data;
  284. int buf_size = avpkt->size;
  285. TSContext *c = avctx->priv_data;
  286. int i, j;
  287. short *samples = data;
  288. int consumed = 0;
  289. int16_t out_buf[240];
  290. int iterations;
  291. if (!buf_size)
  292. return 0;
  293. if (buf_size < 32) {
  294. av_log(avctx, AV_LOG_ERROR,
  295. "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
  296. return -1;
  297. }
  298. iterations = FFMIN(buf_size / 32, *data_size / 480);
  299. for(j = 0; j < iterations; j++) {
  300. truespeech_read_frame(c, buf + consumed);
  301. consumed += 32;
  302. truespeech_correlate_filter(c);
  303. truespeech_filters_merge(c);
  304. memset(out_buf, 0, 240 * 2);
  305. for(i = 0; i < 4; i++) {
  306. truespeech_apply_twopoint_filter(c, i);
  307. truespeech_place_pulses(c, out_buf + i * 60, i);
  308. truespeech_update_filters(c, out_buf + i * 60, i);
  309. truespeech_synth(c, out_buf + i * 60, i);
  310. }
  311. truespeech_save_prevvec(c);
  312. /* finally output decoded frame */
  313. for(i = 0; i < 240; i++)
  314. *samples++ = out_buf[i];
  315. }
  316. *data_size = consumed * 15;
  317. return consumed;
  318. }
  319. AVCodec ff_truespeech_decoder = {
  320. "truespeech",
  321. AVMEDIA_TYPE_AUDIO,
  322. CODEC_ID_TRUESPEECH,
  323. sizeof(TSContext),
  324. truespeech_decode_init,
  325. NULL,
  326. NULL,
  327. truespeech_decode_frame,
  328. .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
  329. };