1/* 2 * Real Audio 1.0 (14.4K) 3 * 4 * Copyright (c) 2008 Vitor Sessak 5 * Copyright (c) 2003 Nick Kurshev 6 * Based on public domain decoder at http://www.honeypot.net/audio 7 * 8 * This file is part of FFmpeg. 9 * 10 * FFmpeg is free software; you can redistribute it and/or 11 * modify it under the terms of the GNU Lesser General Public 12 * License as published by the Free Software Foundation; either 13 * version 2.1 of the License, or (at your option) any later version. 14 * 15 * FFmpeg is distributed in the hope that it will be useful, 16 * but WITHOUT ANY WARRANTY; without even the implied warranty of 17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 18 * Lesser General Public License for more details. 19 * 20 * You should have received a copy of the GNU Lesser General Public 21 * License along with FFmpeg; if not, write to the Free Software 22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 23 */ 24 25#include "avcodec.h" 26#include "bitstream.h" 27#include "ra144.h" 28#include "celp_filters.h" 29 30#define NBLOCKS 4 ///< number of subblocks within a block 31#define BLOCKSIZE 40 ///< subblock size in 16-bit words 32#define BUFFERSIZE 146 ///< the size of the adaptive codebook 33 34 35typedef struct { 36 unsigned int old_energy; ///< previous frame energy 37 38 unsigned int lpc_tables[2][10]; 39 40 /** LPC coefficients: lpc_coef[0] is the coefficients of the current frame 41 * and lpc_coef[1] of the previous one. */ 42 unsigned int *lpc_coef[2]; 43 44 unsigned int lpc_refl_rms[2]; 45 46 /** The current subblock padded by the last 10 values of the previous one. */ 47 int16_t curr_sblock[50]; 48 49 /** Adaptive codebook, its size is two units bigger to avoid a 50 * buffer overflow. */ 51 uint16_t adapt_cb[146+2]; 52} RA144Context; 53 54static av_cold int ra144_decode_init(AVCodecContext * avctx) 55{ 56 RA144Context *ractx = avctx->priv_data; 57 58 ractx->lpc_coef[0] = ractx->lpc_tables[0]; 59 ractx->lpc_coef[1] = ractx->lpc_tables[1]; 60 61 avctx->sample_fmt = SAMPLE_FMT_S16; 62 return 0; 63} 64 65/** 66 * Evaluate sqrt(x << 24). x must fit in 20 bits. This value is evaluated in an 67 * odd way to make the output identical to the binary decoder. 68 */ 69static int t_sqrt(unsigned int x) 70{ 71 int s = 2; 72 while (x > 0xfff) { 73 s++; 74 x >>= 2; 75 } 76 77 return ff_sqrt(x << 20) << s; 78} 79 80/** 81 * Evaluate the LPC filter coefficients from the reflection coefficients. 82 * Does the inverse of the eval_refl() function. 83 */ 84static void eval_coefs(int *coefs, const int *refl) 85{ 86 int buffer[10]; 87 int *b1 = buffer; 88 int *b2 = coefs; 89 int i, j; 90 91 for (i=0; i < 10; i++) { 92 b1[i] = refl[i] << 4; 93 94 for (j=0; j < i; j++) 95 b1[j] = ((refl[i] * b2[i-j-1]) >> 12) + b2[j]; 96 97 FFSWAP(int *, b1, b2); 98 } 99 100 for (i=0; i < 10; i++) 101 coefs[i] >>= 4; 102} 103 104/** 105 * Copy the last offset values of *source to *target. If those values are not 106 * enough to fill the target buffer, fill it with another copy of those values. 107 */ 108static void copy_and_dup(int16_t *target, const int16_t *source, int offset) 109{ 110 source += BUFFERSIZE - offset; 111 112 memcpy(target, source, FFMIN(BLOCKSIZE, offset)*sizeof(*target)); 113 if (offset < BLOCKSIZE) 114 memcpy(target + offset, source, (BLOCKSIZE - offset)*sizeof(*target)); 115} 116 117/** inverse root mean square */ 118static int irms(const int16_t *data) 119{ 120 unsigned int i, sum = 0; 121 122 for (i=0; i < BLOCKSIZE; i++) 123 sum += data[i] * data[i]; 124 125 if (sum == 0) 126 return 0; /* OOPS - division by zero */ 127 128 return 0x20000000 / (t_sqrt(sum) >> 8); 129} 130 131static void add_wav(int16_t *dest, int n, int skip_first, int *m, 132 const int16_t *s1, const int8_t *s2, const int8_t *s3) 133{ 134 int i; 135 int v[3]; 136 137 v[0] = 0; 138 for (i=!skip_first; i<3; i++) 139 v[i] = (gain_val_tab[n][i] * m[i]) >> gain_exp_tab[n]; 140 141 if (v[0]) { 142 for (i=0; i < BLOCKSIZE; i++) 143 dest[i] = (s1[i]*v[0] + s2[i]*v[1] + s3[i]*v[2]) >> 12; 144 } else { 145 for (i=0; i < BLOCKSIZE; i++) 146 dest[i] = ( s2[i]*v[1] + s3[i]*v[2]) >> 12; 147 } 148} 149 150static unsigned int rescale_rms(unsigned int rms, unsigned int energy) 151{ 152 return (rms * energy) >> 10; 153} 154 155static unsigned int rms(const int *data) 156{ 157 int i; 158 unsigned int res = 0x10000; 159 int b = 10; 160 161 for (i=0; i < 10; i++) { 162 res = (((0x1000000 - data[i]*data[i]) >> 12) * res) >> 12; 163 164 if (res == 0) 165 return 0; 166 167 while (res <= 0x3fff) { 168 b++; 169 res <<= 2; 170 } 171 } 172 173 return t_sqrt(res) >> b; 174} 175 176static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs, 177 int gval, GetBitContext *gb) 178{ 179 uint16_t buffer_a[40]; 180 uint16_t *block; 181 int cba_idx = get_bits(gb, 7); // index of the adaptive CB, 0 if none 182 int gain = get_bits(gb, 8); 183 int cb1_idx = get_bits(gb, 7); 184 int cb2_idx = get_bits(gb, 7); 185 int m[3]; 186 187 if (cba_idx) { 188 cba_idx += BLOCKSIZE/2 - 1; 189 copy_and_dup(buffer_a, ractx->adapt_cb, cba_idx); 190 m[0] = (irms(buffer_a) * gval) >> 12; 191 } else { 192 m[0] = 0; 193 } 194 195 m[1] = (cb1_base[cb1_idx] * gval) >> 8; 196 m[2] = (cb2_base[cb2_idx] * gval) >> 8; 197 198 memmove(ractx->adapt_cb, ractx->adapt_cb + BLOCKSIZE, 199 (BUFFERSIZE - BLOCKSIZE) * sizeof(*ractx->adapt_cb)); 200 201 block = ractx->adapt_cb + BUFFERSIZE - BLOCKSIZE; 202 203 add_wav(block, gain, cba_idx, m, cba_idx? buffer_a: NULL, 204 cb1_vects[cb1_idx], cb2_vects[cb2_idx]); 205 206 memcpy(ractx->curr_sblock, ractx->curr_sblock + 40, 207 10*sizeof(*ractx->curr_sblock)); 208 209 if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs, 210 block, BLOCKSIZE, 10, 1, 0xfff)) 211 memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock)); 212} 213 214static void int_to_int16(int16_t *out, const int *inp) 215{ 216 int i; 217 218 for (i=0; i < 30; i++) 219 *out++ = *inp++; 220} 221 222/** 223 * Evaluate the reflection coefficients from the filter coefficients. 224 * Does the inverse of the eval_coefs() function. 225 * 226 * @return 1 if one of the reflection coefficients is greater than 227 * 4095, 0 if not. 228 */ 229static int eval_refl(int *refl, const int16_t *coefs, RA144Context *ractx) 230{ 231 int b, i, j; 232 int buffer1[10]; 233 int buffer2[10]; 234 int *bp1 = buffer1; 235 int *bp2 = buffer2; 236 237 for (i=0; i < 10; i++) 238 buffer2[i] = coefs[i]; 239 240 refl[9] = bp2[9]; 241 242 if ((unsigned) bp2[9] + 0x1000 > 0x1fff) { 243 av_log(ractx, AV_LOG_ERROR, "Overflow. Broken sample?\n"); 244 return 1; 245 } 246 247 for (i=8; i >= 0; i--) { 248 b = 0x1000-((bp2[i+1] * bp2[i+1]) >> 12); 249 250 if (!b) 251 b = -2; 252 253 for (j=0; j <= i; j++) 254 bp1[j] = ((bp2[j] - ((refl[i+1] * bp2[i-j]) >> 12)) * (0x1000000 / b)) >> 12; 255 256 if ((unsigned) bp1[i] + 0x1000 > 0x1fff) 257 return 1; 258 259 refl[i] = bp1[i]; 260 261 FFSWAP(int *, bp1, bp2); 262 } 263 return 0; 264} 265 266static int interp(RA144Context *ractx, int16_t *out, int a, 267 int copyold, int energy) 268{ 269 int work[10]; 270 int b = NBLOCKS - a; 271 int i; 272 273 // Interpolate block coefficients from the this frame's forth block and 274 // last frame's forth block. 275 for (i=0; i<30; i++) 276 out[i] = (a * ractx->lpc_coef[0][i] + b * ractx->lpc_coef[1][i])>> 2; 277 278 if (eval_refl(work, out, ractx)) { 279 // The interpolated coefficients are unstable, copy either new or old 280 // coefficients. 281 int_to_int16(out, ractx->lpc_coef[copyold]); 282 return rescale_rms(ractx->lpc_refl_rms[copyold], energy); 283 } else { 284 return rescale_rms(rms(work), energy); 285 } 286} 287 288/** Uncompress one block (20 bytes -> 160*2 bytes). */ 289static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, 290 int *data_size, const uint8_t *buf, int buf_size) 291{ 292 static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; 293 unsigned int refl_rms[4]; // RMS of the reflection coefficients 294 uint16_t block_coefs[4][30]; // LPC coefficients of each sub-block 295 unsigned int lpc_refl[10]; // LPC reflection coefficients of the frame 296 int i, j; 297 int16_t *data = vdata; 298 unsigned int energy; 299 300 RA144Context *ractx = avctx->priv_data; 301 GetBitContext gb; 302 303 if (*data_size < 2*160) 304 return -1; 305 306 if(buf_size < 20) { 307 av_log(avctx, AV_LOG_ERROR, 308 "Frame too small (%d bytes). Truncated file?\n", buf_size); 309 *data_size = 0; 310 return buf_size; 311 } 312 init_get_bits(&gb, buf, 20 * 8); 313 314 for (i=0; i<10; i++) 315 lpc_refl[i] = lpc_refl_cb[i][get_bits(&gb, sizes[i])]; 316 317 eval_coefs(ractx->lpc_coef[0], lpc_refl); 318 ractx->lpc_refl_rms[0] = rms(lpc_refl); 319 320 energy = energy_tab[get_bits(&gb, 5)]; 321 322 refl_rms[0] = interp(ractx, block_coefs[0], 1, 1, ractx->old_energy); 323 refl_rms[1] = interp(ractx, block_coefs[1], 2, energy <= ractx->old_energy, 324 t_sqrt(energy*ractx->old_energy) >> 12); 325 refl_rms[2] = interp(ractx, block_coefs[2], 3, 0, energy); 326 refl_rms[3] = rescale_rms(ractx->lpc_refl_rms[0], energy); 327 328 int_to_int16(block_coefs[3], ractx->lpc_coef[0]); 329 330 for (i=0; i < 4; i++) { 331 do_output_subblock(ractx, block_coefs[i], refl_rms[i], &gb); 332 333 for (j=0; j < BLOCKSIZE; j++) 334 *data++ = av_clip_int16(ractx->curr_sblock[j + 10] << 2); 335 } 336 337 ractx->old_energy = energy; 338 ractx->lpc_refl_rms[1] = ractx->lpc_refl_rms[0]; 339 340 FFSWAP(unsigned int *, ractx->lpc_coef[0], ractx->lpc_coef[1]); 341 342 *data_size = 2*160; 343 return 20; 344} 345 346AVCodec ra_144_decoder = 347{ 348 "real_144", 349 CODEC_TYPE_AUDIO, 350 CODEC_ID_RA_144, 351 sizeof(RA144Context), 352 ra144_decode_init, 353 NULL, 354 NULL, 355 ra144_decode_frame, 356 .long_name = NULL_IF_CONFIG_SMALL("RealAudio 1.0 (14.4K)"), 357}; 358