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  1. /*
  2. * DSP Group TrueSpeech compatible decoder
  3. * Copyright (c) 2005 Konstantin Shishkov
  4. *
  5. * This file is part of Libav.
  6. *
  7. * Libav 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. * Libav 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 Libav; if not, write to the Free Software
  19. * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20. */
  21. #include "libavutil/channel_layout.h"
  22. #include "libavutil/intreadwrite.h"
  23. #include "avcodec.h"
  24. #include "bitstream.h"
  25. #include "bswapdsp.h"
  26. #include "internal.h"
  27. #include "truespeech_data.h"
  28. /**
  29. * @file
  30. * TrueSpeech decoder.
  31. */
  32. /**
  33. * TrueSpeech decoder context
  34. */
  35. typedef struct TSContext {
  36. BswapDSPContext bdsp;
  37. /* input data */
  38. DECLARE_ALIGNED(16, uint8_t, buffer)[32];
  39. int16_t vector[8]; ///< input vector: 5/5/4/4/4/3/3/3
  40. int offset1[2]; ///< 8-bit value, used in one copying offset
  41. int offset2[4]; ///< 7-bit value, encodes offsets for copying and for two-point filter
  42. int pulseoff[4]; ///< 4-bit offset of pulse values block
  43. int pulsepos[4]; ///< 27-bit variable, encodes 7 pulse positions
  44. int pulseval[4]; ///< 7x2-bit pulse values
  45. int flag; ///< 1-bit flag, shows how to choose filters
  46. /* temporary data */
  47. int filtbuf[146]; // some big vector used for storing filters
  48. int prevfilt[8]; // filter from previous frame
  49. int16_t tmp1[8]; // coefficients for adding to out
  50. int16_t tmp2[8]; // coefficients for adding to out
  51. int16_t tmp3[8]; // coefficients for adding to out
  52. int16_t cvector[8]; // correlated input vector
  53. int filtval; // gain value for one function
  54. int16_t newvec[60]; // tmp vector
  55. int16_t filters[32]; // filters for every subframe
  56. } TSContext;
  57. static av_cold int truespeech_decode_init(AVCodecContext * avctx)
  58. {
  59. TSContext *c = avctx->priv_data;
  60. if (avctx->channels != 1) {
  61. avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
  62. return AVERROR_PATCHWELCOME;
  63. }
  64. avctx->channel_layout = AV_CH_LAYOUT_MONO;
  65. avctx->sample_fmt = AV_SAMPLE_FMT_S16;
  66. ff_bswapdsp_init(&c->bdsp);
  67. return 0;
  68. }
  69. static void truespeech_read_frame(TSContext *dec, const uint8_t *input)
  70. {
  71. BitstreamContext bc;
  72. dec->bdsp.bswap_buf((uint32_t *) dec->buffer, (const uint32_t *) input, 8);
  73. bitstream_init8(&bc, dec->buffer, 32);
  74. dec->vector[7] = ts_codebook[7][bitstream_read(&bc, 3)];
  75. dec->vector[6] = ts_codebook[6][bitstream_read(&bc, 3)];
  76. dec->vector[5] = ts_codebook[5][bitstream_read(&bc, 3)];
  77. dec->vector[4] = ts_codebook[4][bitstream_read(&bc, 4)];
  78. dec->vector[3] = ts_codebook[3][bitstream_read(&bc, 4)];
  79. dec->vector[2] = ts_codebook[2][bitstream_read(&bc, 4)];
  80. dec->vector[1] = ts_codebook[1][bitstream_read(&bc, 5)];
  81. dec->vector[0] = ts_codebook[0][bitstream_read(&bc, 5)];
  82. dec->flag = bitstream_read_bit(&bc);
  83. dec->offset1[0] = bitstream_read(&bc, 4) << 4;
  84. dec->offset2[3] = bitstream_read(&bc, 7);
  85. dec->offset2[2] = bitstream_read(&bc, 7);
  86. dec->offset2[1] = bitstream_read(&bc, 7);
  87. dec->offset2[0] = bitstream_read(&bc, 7);
  88. dec->offset1[1] = bitstream_read(&bc, 4);
  89. dec->pulseval[1] = bitstream_read(&bc, 14);
  90. dec->pulseval[0] = bitstream_read(&bc, 14);
  91. dec->offset1[1] |= bitstream_read(&bc, 4) << 4;
  92. dec->pulseval[3] = bitstream_read(&bc, 14);
  93. dec->pulseval[2] = bitstream_read(&bc, 14);
  94. dec->offset1[0] |= bitstream_read_bit(&bc);
  95. dec->pulsepos[0] = bitstream_read(&bc, 27);
  96. dec->pulseoff[0] = bitstream_read(&bc, 4);
  97. dec->offset1[0] |= bitstream_read_bit(&bc) << 1;
  98. dec->pulsepos[1] = bitstream_read(&bc, 27);
  99. dec->pulseoff[1] = bitstream_read(&bc, 4);
  100. dec->offset1[0] |= bitstream_read_bit(&bc) << 2;
  101. dec->pulsepos[2] = bitstream_read(&bc, 27);
  102. dec->pulseoff[2] = bitstream_read(&bc, 4);
  103. dec->offset1[0] |= bitstream_read_bit(&bc) << 3;
  104. dec->pulsepos[3] = bitstream_read(&bc, 27);
  105. dec->pulseoff[3] = bitstream_read(&bc, 4);
  106. }
  107. static void truespeech_correlate_filter(TSContext *dec)
  108. {
  109. int16_t tmp[8];
  110. int i, j;
  111. for(i = 0; i < 8; i++){
  112. if(i > 0){
  113. memcpy(tmp, dec->cvector, i * sizeof(*tmp));
  114. for(j = 0; j < i; j++)
  115. dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) +
  116. (dec->cvector[j] << 15) + 0x4000) >> 15;
  117. }
  118. dec->cvector[i] = (8 - dec->vector[i]) >> 3;
  119. }
  120. for(i = 0; i < 8; i++)
  121. dec->cvector[i] = (dec->cvector[i] * ts_decay_994_1000[i]) >> 15;
  122. dec->filtval = dec->vector[0];
  123. }
  124. static void truespeech_filters_merge(TSContext *dec)
  125. {
  126. int i;
  127. if(!dec->flag){
  128. for(i = 0; i < 8; i++){
  129. dec->filters[i + 0] = dec->prevfilt[i];
  130. dec->filters[i + 8] = dec->prevfilt[i];
  131. }
  132. }else{
  133. for(i = 0; i < 8; i++){
  134. dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15;
  135. dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15;
  136. }
  137. }
  138. for(i = 0; i < 8; i++){
  139. dec->filters[i + 16] = dec->cvector[i];
  140. dec->filters[i + 24] = dec->cvector[i];
  141. }
  142. }
  143. static void truespeech_apply_twopoint_filter(TSContext *dec, int quart)
  144. {
  145. int16_t tmp[146 + 60], *ptr0, *ptr1;
  146. const int16_t *filter;
  147. int i, t, off;
  148. t = dec->offset2[quart];
  149. if(t == 127){
  150. memset(dec->newvec, 0, 60 * sizeof(*dec->newvec));
  151. return;
  152. }
  153. for(i = 0; i < 146; i++)
  154. tmp[i] = dec->filtbuf[i];
  155. off = (t / 25) + dec->offset1[quart >> 1] + 18;
  156. off = av_clip(off, 0, 145);
  157. ptr0 = tmp + 145 - off;
  158. ptr1 = tmp + 146;
  159. filter = ts_order2_coeffs + (t % 25) * 2;
  160. for(i = 0; i < 60; i++){
  161. t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14;
  162. ptr0++;
  163. dec->newvec[i] = t;
  164. ptr1[i] = t;
  165. }
  166. }
  167. static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart)
  168. {
  169. int16_t tmp[7];
  170. int i, j, t;
  171. const int16_t *ptr1;
  172. int16_t *ptr2;
  173. int coef;
  174. memset(out, 0, 60 * sizeof(*out));
  175. for(i = 0; i < 7; i++) {
  176. t = dec->pulseval[quart] & 3;
  177. dec->pulseval[quart] >>= 2;
  178. tmp[6 - i] = ts_pulse_scales[dec->pulseoff[quart] * 4 + t];
  179. }
  180. coef = dec->pulsepos[quart] >> 15;
  181. ptr1 = ts_pulse_values + 30;
  182. ptr2 = tmp;
  183. for(i = 0, j = 3; (i < 30) && (j > 0); i++){
  184. t = *ptr1++;
  185. if(coef >= t)
  186. coef -= t;
  187. else{
  188. out[i] = *ptr2++;
  189. ptr1 += 30;
  190. j--;
  191. }
  192. }
  193. coef = dec->pulsepos[quart] & 0x7FFF;
  194. ptr1 = ts_pulse_values;
  195. for(i = 30, j = 4; (i < 60) && (j > 0); i++){
  196. t = *ptr1++;
  197. if(coef >= t)
  198. coef -= t;
  199. else{
  200. out[i] = *ptr2++;
  201. ptr1 += 30;
  202. j--;
  203. }
  204. }
  205. }
  206. static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart)
  207. {
  208. int i;
  209. memmove(dec->filtbuf, &dec->filtbuf[60], 86 * sizeof(*dec->filtbuf));
  210. for(i = 0; i < 60; i++){
  211. dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3);
  212. out[i] += dec->newvec[i];
  213. }
  214. }
  215. static void truespeech_synth(TSContext *dec, int16_t *out, int quart)
  216. {
  217. int i,k;
  218. int t[8];
  219. int16_t *ptr0, *ptr1;
  220. ptr0 = dec->tmp1;
  221. ptr1 = dec->filters + quart * 8;
  222. for(i = 0; i < 60; i++){
  223. int sum = 0;
  224. for(k = 0; k < 8; k++)
  225. sum += ptr0[k] * ptr1[k];
  226. sum = (sum + (out[i] << 12) + 0x800) >> 12;
  227. out[i] = av_clip(sum, -0x7FFE, 0x7FFE);
  228. for(k = 7; k > 0; k--)
  229. ptr0[k] = ptr0[k - 1];
  230. ptr0[0] = out[i];
  231. }
  232. for(i = 0; i < 8; i++)
  233. t[i] = (ts_decay_35_64[i] * ptr1[i]) >> 15;
  234. ptr0 = dec->tmp2;
  235. for(i = 0; i < 60; i++){
  236. int sum = 0;
  237. for(k = 0; k < 8; k++)
  238. sum += ptr0[k] * t[k];
  239. for(k = 7; k > 0; k--)
  240. ptr0[k] = ptr0[k - 1];
  241. ptr0[0] = out[i];
  242. out[i] = ((out[i] << 12) - sum) >> 12;
  243. }
  244. for(i = 0; i < 8; i++)
  245. t[i] = (ts_decay_3_4[i] * ptr1[i]) >> 15;
  246. ptr0 = dec->tmp3;
  247. for(i = 0; i < 60; i++){
  248. int sum = out[i] << 12;
  249. for(k = 0; k < 8; k++)
  250. sum += ptr0[k] * t[k];
  251. for(k = 7; k > 0; k--)
  252. ptr0[k] = ptr0[k - 1];
  253. ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
  254. sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum;
  255. sum = sum - (sum >> 3);
  256. out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE);
  257. }
  258. }
  259. static void truespeech_save_prevvec(TSContext *c)
  260. {
  261. int i;
  262. for(i = 0; i < 8; i++)
  263. c->prevfilt[i] = c->cvector[i];
  264. }
  265. static int truespeech_decode_frame(AVCodecContext *avctx, void *data,
  266. int *got_frame_ptr, AVPacket *avpkt)
  267. {
  268. AVFrame *frame = data;
  269. const uint8_t *buf = avpkt->data;
  270. int buf_size = avpkt->size;
  271. TSContext *c = avctx->priv_data;
  272. int i, j;
  273. int16_t *samples;
  274. int iterations, ret;
  275. iterations = buf_size / 32;
  276. if (!iterations) {
  277. av_log(avctx, AV_LOG_ERROR,
  278. "Too small input buffer (%d bytes), need at least 32 bytes\n", buf_size);
  279. return -1;
  280. }
  281. /* get output buffer */
  282. frame->nb_samples = iterations * 240;
  283. if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) {
  284. av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
  285. return ret;
  286. }
  287. samples = (int16_t *)frame->data[0];
  288. memset(samples, 0, iterations * 240 * sizeof(*samples));
  289. for(j = 0; j < iterations; j++) {
  290. truespeech_read_frame(c, buf);
  291. buf += 32;
  292. truespeech_correlate_filter(c);
  293. truespeech_filters_merge(c);
  294. for(i = 0; i < 4; i++) {
  295. truespeech_apply_twopoint_filter(c, i);
  296. truespeech_place_pulses (c, samples, i);
  297. truespeech_update_filters(c, samples, i);
  298. truespeech_synth (c, samples, i);
  299. samples += 60;
  300. }
  301. truespeech_save_prevvec(c);
  302. }
  303. *got_frame_ptr = 1;
  304. return buf_size;
  305. }
  306. AVCodec ff_truespeech_decoder = {
  307. .name = "truespeech",
  308. .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"),
  309. .type = AVMEDIA_TYPE_AUDIO,
  310. .id = AV_CODEC_ID_TRUESPEECH,
  311. .priv_data_size = sizeof(TSContext),
  312. .init = truespeech_decode_init,
  313. .decode = truespeech_decode_frame,
  314. .capabilities = AV_CODEC_CAP_DR1,
  315. };