1 /* ----------------------------------------------------------------------------- 2 Software License for The Fraunhofer FDK AAC Codec Library for Android 3 4 Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Frderung der angewandten 5 Forschung e.V. All rights reserved. 6 7 1. INTRODUCTION 8 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software 9 that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding 10 scheme for digital audio. This FDK AAC Codec software is intended to be used on 11 a wide variety of Android devices. 12 13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient 14 general perceptual audio codecs. AAC-ELD is considered the best-performing 15 full-bandwidth communications codec by independent studies and is widely 16 deployed. AAC has been standardized by ISO and IEC as part of the MPEG 17 specifications. 18 19 Patent licenses for necessary patent claims for the FDK AAC Codec (including 20 those of Fraunhofer) may be obtained through Via Licensing 21 (www.vialicensing.com) or through the respective patent owners individually for 22 the purpose of encoding or decoding bit streams in products that are compliant 23 with the ISO/IEC MPEG audio standards. Please note that most manufacturers of 24 Android devices already license these patent claims through Via Licensing or 25 directly from the patent owners, and therefore FDK AAC Codec software may 26 already be covered under those patent licenses when it is used for those 27 licensed purposes only. 28 29 Commercially-licensed AAC software libraries, including floating-point versions 30 with enhanced sound quality, are also available from Fraunhofer. Users are 31 encouraged to check the Fraunhofer website for additional applications 32 information and documentation. 33 34 2. COPYRIGHT LICENSE 35 36 Redistribution and use in source and binary forms, with or without modification, 37 are permitted without payment of copyright license fees provided that you 38 satisfy the following conditions: 39 40 You must retain the complete text of this software license in redistributions of 41 the FDK AAC Codec or your modifications thereto in source code form. 42 43 You must retain the complete text of this software license in the documentation 44 and/or other materials provided with redistributions of the FDK AAC Codec or 45 your modifications thereto in binary form. You must make available free of 46 charge copies of the complete source code of the FDK AAC Codec and your 47 modifications thereto to recipients of copies in binary form. 48 49 The name of Fraunhofer may not be used to endorse or promote products derived 50 from this library without prior written permission. 51 52 You may not charge copyright license fees for anyone to use, copy or distribute 53 the FDK AAC Codec software or your modifications thereto. 54 55 Your modified versions of the FDK AAC Codec must carry prominent notices stating 56 that you changed the software and the date of any change. For modified versions 57 of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" 58 must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK 59 AAC Codec Library for Android." 60 61 3. NO PATENT LICENSE 62 63 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without 64 limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. 65 Fraunhofer provides no warranty of patent non-infringement with respect to this 66 software. 67 68 You may use this FDK AAC Codec software or modifications thereto only for 69 purposes that are authorized by appropriate patent licenses. 70 71 4. DISCLAIMER 72 73 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright 74 holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, 75 including but not limited to the implied warranties of merchantability and 76 fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 77 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, 78 or consequential damages, including but not limited to procurement of substitute 79 goods or services; loss of use, data, or profits, or business interruption, 80 however caused and on any theory of liability, whether in contract, strict 81 liability, or tort (including negligence), arising in any way out of the use of 82 this software, even if advised of the possibility of such damage. 83 84 5. CONTACT INFORMATION 85 86 Fraunhofer Institute for Integrated Circuits IIS 87 Attention: Audio and Multimedia Departments - FDK AAC LL 88 Am Wolfsmantel 33 89 91058 Erlangen, Germany 90 91 www.iis.fraunhofer.de/amm 92 amm-info (at) iis.fraunhofer.de 93 ----------------------------------------------------------------------------- */ 94 95 /**************************** AAC decoder library ****************************** 96 97 Author(s): Manuel Jander 98 99 Description: USAC Linear Prediction Domain coding 100 101 *******************************************************************************/ 102 103 #include "usacdec_lpd.h" 104 105 #include "usacdec_rom.h" 106 #include "usacdec_fac.h" 107 #include "usacdec_lpc.h" 108 #include "FDK_tools_rom.h" 109 #include "fft.h" 110 #include "mdct.h" 111 #include "usacdec_acelp.h" 112 #include "overlapadd.h" 113 114 #include "conceal.h" 115 116 #include "block.h" 117 118 #define SF_PITCH_TRACK 6 119 #define SF_GAIN 3 120 #define MIN_VAL FL2FXCONST_DBL(0.0f) 121 #define MAX_VAL (FIXP_DBL) MAXVAL_DBL 122 123 #include "ac_arith_coder.h" 124 125 void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, 126 const FIXP_SGL *filt, INT stop, int len) { 127 INT i, j; 128 FIXP_DBL tmp; 129 130 for (i = 0; i < stop; i++) { 131 tmp = fMultDiv2(noise[i], filt[0]); // Filt in Q-1.16 132 for (j = 1; j <= len; j++) { 133 tmp += fMultDiv2((noise[i - j] + noise[i + j]), filt[j]); 134 } 135 syn_out[i] = (FIXP_PCM)(IMDCT_SCALE(syn[i] - tmp)); 136 } 137 } 138 139 void bass_pf_1sf_delay( 140 FIXP_DBL *syn, /* (i) : 12.8kHz synthesis to postfilter */ 141 const INT *T_sf, /* (i) : Pitch period for all subframes (T_sf[16]) */ 142 FIXP_DBL *pit_gain, 143 const int frame_length, /* (i) : frame length (should be 768|1024) */ 144 const INT l_frame, 145 const INT l_next, /* (i) : look ahead for symmetric filtering */ 146 FIXP_PCM *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */ 147 FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR] */ 148 { 149 INT i, sf, i_subfr, T, T2, lg; 150 151 FIXP_DBL tmp, ener, corr, gain; 152 FIXP_DBL *noise, *noise_in; 153 FIXP_DBL 154 noise_buf[L_FILT + (2 * L_SUBFR)]; // L_FILT = 12, L_SUBFR = 64 => 140 155 const FIXP_DBL *x, *y; 156 157 { 158 noise = noise_buf + L_FILT; // L_FILT = 12 delay of upsampling filter 159 noise_in = noise_buf + L_FILT + L_SUBFR; 160 /* Input scaling of the BPF memory */ 161 scaleValues(mem_bpf, (L_FILT + L_SUBFR), 1); 162 } 163 164 int gain_exp = 17; 165 166 sf = 0; 167 for (i_subfr = 0; i_subfr < l_frame; i_subfr += L_SUBFR, sf++) { 168 T = T_sf[sf]; 169 gain = pit_gain[sf]; 170 171 /* Gain is in Q17.14 */ 172 /* If gain > 1 set to 1 */ 173 if (gain > (FIXP_DBL)(1 << 14)) gain = (FIXP_DBL)(1 << 14); 174 175 /* If gain < 0 set to 0 */ 176 if (gain < (FIXP_DBL)0) gain = (FIXP_DBL)0; 177 178 if (gain > (FIXP_DBL)0) { 179 /* pitch tracker: test pitch/2 to avoid continuous pitch doubling */ 180 /* Note: pitch is limited to PIT_MIN (34 = 376Hz) at the encoder */ 181 T2 = T >> 1; 182 x = &syn[i_subfr - L_EXTRA]; 183 y = &syn[i_subfr - T2 - L_EXTRA]; 184 185 ener = (FIXP_DBL)0; 186 corr = (FIXP_DBL)0; 187 tmp = (FIXP_DBL)0; 188 189 int headroom_x = getScalefactor(x, L_SUBFR + L_EXTRA); 190 int headroom_y = getScalefactor(y, L_SUBFR + L_EXTRA); 191 192 int width_shift = 7; 193 194 for (i = 0; i < (L_SUBFR + L_EXTRA); i++) { 195 ener += fPow2Div2((x[i] << headroom_x)) >> width_shift; 196 corr += fMultDiv2((x[i] << headroom_x), (y[i] << headroom_y)) >> 197 width_shift; 198 tmp += fPow2Div2((y[i] << headroom_y)) >> width_shift; 199 } 200 201 int exp_ener = ((17 - headroom_x) << 1) + width_shift + 1; 202 int exp_corr = (17 - headroom_x) + (17 - headroom_y) + width_shift + 1; 203 int exp_tmp = ((17 - headroom_y) << 1) + width_shift + 1; 204 205 /* Add 0.01 to "ener". Adjust exponents */ 206 FIXP_DBL point_zero_one = (FIXP_DBL)0x51eb851f; /* In Q-6.37 */ 207 int diff; 208 ener = fAddNorm(ener, exp_ener, point_zero_one, -6, &exp_ener); 209 corr = fAddNorm(corr, exp_corr, point_zero_one, -6, &exp_corr); 210 tmp = fAddNorm(tmp, exp_tmp, point_zero_one, -6, &exp_tmp); 211 212 /* use T2 if normalized correlation > 0.95 */ 213 INT s1, s2; 214 s1 = CntLeadingZeros(ener) - 1; 215 s2 = CntLeadingZeros(tmp) - 1; 216 217 FIXP_DBL ener_by_tmp = fMultDiv2(ener << s1, tmp << s2); 218 int ener_by_tmp_exp = (exp_ener - s1) + (exp_tmp - s2) + 1; 219 220 if (ener_by_tmp_exp & 1) { 221 ener_by_tmp <<= 1; 222 ener_by_tmp_exp -= 1; 223 } 224 225 int temp_exp = 0; 226 227 FIXP_DBL temp1 = invSqrtNorm2(ener_by_tmp, &temp_exp); 228 229 int temp1_exp = temp_exp - (ener_by_tmp_exp >> 1); 230 231 FIXP_DBL tmp_result = fMult(corr, temp1); 232 233 int tmp_result_exp = exp_corr + temp1_exp; 234 235 diff = tmp_result_exp - 0; 236 FIXP_DBL point95 = FL2FXCONST_DBL(0.95f); 237 if (diff >= 0) { 238 diff = fMin(diff, 31); 239 point95 = FL2FXCONST_DBL(0.95f) >> diff; 240 } else { 241 diff = fMax(diff, -31); 242 tmp_result >>= (-diff); 243 } 244 245 if (tmp_result > point95) T = T2; 246 247 /* prevent that noise calculation below reaches into not defined signal 248 parts at the end of the synth_buf or in other words restrict the below 249 used index (i+i_subfr+T) < l_frame + l_next 250 */ 251 lg = l_frame + l_next - T - i_subfr; 252 253 if (lg > L_SUBFR) 254 lg = L_SUBFR; 255 else if (lg < 0) 256 lg = 0; 257 258 /* limit gain to avoid problem on burst */ 259 if (lg > 0) { 260 FIXP_DBL tmp1; 261 262 /* max(lg) = 64 => scale with 6 bits minus 1 (fPow2Div2) */ 263 264 s1 = getScalefactor(&syn[i_subfr], lg); 265 s2 = getScalefactor(&syn[i_subfr + T], lg); 266 INT s = fixMin(s1, s2); 267 268 tmp = (FIXP_DBL)0; 269 ener = (FIXP_DBL)0; 270 for (i = 0; i < lg; i++) { 271 tmp += fPow2Div2(syn[i + i_subfr] << s1) >> (SF_PITCH_TRACK); 272 ener += fPow2Div2(syn[i + i_subfr + T] << s2) >> (SF_PITCH_TRACK); 273 } 274 tmp = tmp >> fMin(DFRACT_BITS - 1, (2 * (s1 - s))); 275 ener = ener >> fMin(DFRACT_BITS - 1, (2 * (s2 - s))); 276 277 /* error robustness: for the specific case syn[...] == -1.0f for all 64 278 samples ener or tmp might overflow and become negative. For all sane 279 cases we have enough headroom. 280 */ 281 if (ener <= (FIXP_DBL)0) { 282 ener = (FIXP_DBL)1; 283 } 284 if (tmp <= (FIXP_DBL)0) { 285 tmp = (FIXP_DBL)1; 286 } 287 FDK_ASSERT(ener > (FIXP_DBL)0); 288 289 /* tmp = sqrt(tmp/ener) */ 290 int result_e = 0; 291 tmp1 = fDivNorm(tmp, ener, &result_e); 292 if (result_e & 1) { 293 tmp1 >>= 1; 294 result_e += 1; 295 } 296 tmp = sqrtFixp(tmp1); 297 result_e >>= 1; 298 299 gain_exp = 17; 300 301 diff = result_e - gain_exp; 302 303 FIXP_DBL gain1 = gain; 304 305 if (diff >= 0) { 306 diff = fMin(diff, 31); 307 gain1 >>= diff; 308 } else { 309 result_e += (-diff); 310 diff = fMax(diff, -31); 311 tmp >>= (-diff); 312 } 313 314 if (tmp < gain1) { 315 gain = tmp; 316 gain_exp = result_e; 317 } 318 } 319 320 /* calculate noise based on voiced pitch */ 321 /* fMultDiv2() replaces weighting of gain with 0.5 */ 322 diff = gain_exp - 17; 323 if (diff >= 0) { 324 gain <<= diff; 325 } else { 326 gain >>= (-diff); 327 } 328 329 s1 = CntLeadingZeros(gain) - 1; 330 s1 -= 16; /* Leading bits for SGL */ 331 332 FIXP_SGL gainSGL = FX_DBL2FX_SGL(gain << 16); 333 334 gainSGL = gainSGL << s1; 335 336 { 337 for (i = 0; i < lg; i++) { 338 /* scaled with SF_SYNTH + gain_sf + 1 */ 339 noise_in[i] = 340 (fMult(gainSGL, syn[i + i_subfr] - (syn[i + i_subfr - T] >> 1) - 341 (syn[i + i_subfr + T] >> 1))) >> 342 s1; 343 } 344 345 for (i = lg; i < L_SUBFR; i++) { 346 /* scaled with SF_SYNTH + gain_sf + 1 */ 347 noise_in[i] = 348 (fMult(gainSGL, syn[i + i_subfr] - syn[i + i_subfr - T])) >> s1; 349 } 350 } 351 } else { 352 FDKmemset(noise_in, (FIXP_DBL)0, L_SUBFR * sizeof(FIXP_DBL)); 353 } 354 355 { 356 FDKmemcpy(noise_buf, mem_bpf, (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); 357 358 FDKmemcpy(mem_bpf, noise_buf + L_SUBFR, 359 (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); 360 } 361 362 /* substract from voiced speech low-pass filtered noise */ 363 /* filter coefficients are scaled with factor SF_FILT_LP (1) */ 364 365 { 366 filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise, 367 fdk_dec_filt_lp, L_SUBFR, L_FILT); 368 } 369 } 370 371 { 372 373 } 374 375 // To be determined (info from Ben) 376 { 377 /* Output scaling of the BPF memory */ 378 scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1); 379 /* Copy the rest of the signal (after the fac) */ 380 scaleValuesSaturate((FIXP_PCM *)&synth_out[l_frame], 381 (FIXP_DBL *)&syn[l_frame - L_SUBFR], 382 (frame_length - l_frame), MDCT_OUT_HEADROOM); 383 } 384 385 return; 386 } 387 388 /* 389 * Frequency Domain Noise Shaping 390 */ 391 392 /** 393 * \brief Adaptive Low Frequencies Deemphasis of spectral coefficients. 394 * 395 * Ensure quantization of low frequencies in case where the 396 * signal dynamic is higher than the LPC noise shaping. 397 * This is the inverse operation of adap_low_freq_emph(). 398 * Output gain of all blocks. 399 * 400 * \param x pointer to the spectral coefficients, requires 1 bit headroom. 401 * \param lg length of x. 402 * \param bUseNewAlfe if set, apply ALFD for fullband lpd. 403 * \param gainLpc1 pointer to gain based on old input LPC coefficients. 404 * \param gainLpc2 pointer to gain based on new input LPC coefficients. 405 * \param alfd_gains pointer to output gains. 406 * \param s current scale shift factor of x. 407 */ 408 #define ALFDPOW2_SCALE 3 409 /*static*/ 410 void CLpd_AdaptLowFreqDeemph(FIXP_DBL x[], int lg, FIXP_DBL alfd_gains[], 411 INT s) { 412 { 413 int i, j, k, i_max; 414 FIXP_DBL max, fac; 415 /* Note: This stack array saves temporary accumulation results to be used in 416 * a second run */ 417 /* The size should be limited to (1024/4)/8=32 */ 418 FIXP_DBL tmp_pow2[32]; 419 420 s = s * 2 + ALFDPOW2_SCALE; 421 422 k = 8; 423 i_max = lg / 4; /* ALFD range = 1600Hz (lg = 6400Hz) */ 424 425 /* find spectral peak */ 426 max = FL2FX_DBL(0.01f) >> s; 427 for (i = 0; i < i_max; i += k) { 428 FIXP_DBL tmp; 429 430 tmp = FIXP_DBL(0); 431 FIXP_DBL *pX = &x[i]; 432 433 j = 8; 434 do { 435 FIXP_DBL x0 = *pX++; 436 FIXP_DBL x1 = *pX++; 437 x0 = fPow2Div2(x0); 438 x1 = fPow2Div2(x1); 439 tmp = tmp + (x0 >> (ALFDPOW2_SCALE - 1)); 440 tmp = tmp + (x1 >> (ALFDPOW2_SCALE - 1)); 441 } while ((j = j - 2) != 0); 442 tmp = fMax(tmp, (FL2FX_DBL(0.01f) >> s)); 443 tmp_pow2[i >> 3] = tmp; 444 if (tmp > max) { 445 max = tmp; 446 } 447 } 448 449 /* deemphasis of all blocks below the peak */ 450 fac = FL2FX_DBL(0.1f) >> 1; 451 for (i = 0; i < i_max; i += k) { 452 FIXP_DBL tmp; 453 INT shifti; 454 455 tmp = tmp_pow2[i >> 3]; 456 457 /* tmp = (float)sqrt(tmp/max); */ 458 459 /* value of tmp is between 8/2*max^2 and max^2 / 2. */ 460 /* required shift factor of division can grow up to 27 461 (grows exponentially for values toward zero) 462 thus using normalized division to assure valid result. */ 463 { 464 INT sd; 465 466 if (tmp != (FIXP_DBL)0) { 467 tmp = fDivNorm(max, tmp, &sd); 468 if (sd & 1) { 469 sd++; 470 tmp >>= 1; 471 } 472 } else { 473 tmp = (FIXP_DBL)MAXVAL_DBL; 474 sd = 0; 475 } 476 tmp = invSqrtNorm2(tmp, &shifti); 477 tmp = scaleValue(tmp, shifti - 1 - (sd / 2)); 478 } 479 if (tmp > fac) { 480 fac = tmp; 481 } 482 FIXP_DBL *pX = &x[i]; 483 484 j = 8; 485 do { 486 FIXP_DBL x0 = pX[0]; 487 FIXP_DBL x1 = pX[1]; 488 x0 = fMultDiv2(x0, fac); 489 x1 = fMultDiv2(x1, fac); 490 x0 = x0 << 2; 491 x1 = x1 << 2; 492 *pX++ = x0; 493 *pX++ = x1; 494 495 } while ((j = j - 2) != 0); 496 /* Store gains for FAC */ 497 *alfd_gains++ = fac; 498 } 499 } 500 } 501 502 /** 503 * \brief Interpolated Noise Shaping for mdct coefficients. 504 * This algorithm shapes temporally the spectral noise between 505 * the two spectral noise represention (FDNS_NPTS of resolution). 506 * The noise is shaped monotonically between the two points 507 * using a curved shape to favor the lower gain in mid-frame. 508 * ODFT and amplitud calculation are applied to the 2 LPC coefficients first. 509 * 510 * \param r pointer to spectrum data. 511 * \param rms RMS of output spectrum. 512 * \param lg length of r. 513 * \param A1 pointer to old input LPC coefficients of length M_LP_FILTER_ORDER 514 * scaled by SF_A_COEFFS. 515 * \param A2 pointer to new input LPC coefficients of length M_LP_FILTER_ORDER 516 * scaled by SF_A_COEFFS. 517 * \param bLpc2Mdct flags control lpc2mdct conversion and noise shaping. 518 * \param gainLpc1 pointer to gain based on old input LPC coefficients. 519 * \param gainLpc2 pointer to gain based on new input LPC coefficients. 520 * \param gLpc_e pointer to exponent of gainLpc1 and gainLpc2. 521 */ 522 /* static */ 523 #define NSHAPE_SCALE (4) 524 525 #define LPC2MDCT_CALC (1) 526 #define LPC2MDCT_GAIN_LOAD (2) 527 #define LPC2MDCT_GAIN_SAVE (4) 528 #define LPC2MDCT_APPLY_NSHAPE (8) 529 530 void lpc2mdctAndNoiseShaping(FIXP_DBL *r, SHORT *pScale, const INT lg, 531 const INT fdns_npts, const FIXP_LPC *A1, 532 const INT A1_exp, const FIXP_LPC *A2, 533 const INT A2_exp) { 534 FIXP_DBL *tmp2 = NULL; 535 FIXP_DBL rr_minus_one; 536 int i, k, s, step; 537 538 C_AALLOC_SCRATCH_START(tmp1, FIXP_DBL, FDNS_NPTS * 8) 539 540 { 541 tmp2 = tmp1 + fdns_npts * 4; 542 543 /* lpc2mdct() */ 544 545 /* ODFT. E_LPC_a_weight() for A1 and A2 vectors is included into the loop 546 * below. */ 547 FIXP_DBL f = FL2FXCONST_DBL(0.92f); 548 549 const FIXP_STP *SinTab; 550 int k_step; 551 /* needed values: sin(phi), cos(phi); phi = i*PI/(2*fdns_npts), i = 0 ... 552 * M_LP_FILTER_ORDER */ 553 switch (fdns_npts) { 554 case 64: 555 SinTab = SineTable512; 556 k_step = (512 / 64); 557 FDK_ASSERT(512 >= 64); 558 break; 559 case 48: 560 SinTab = SineTable384; 561 k_step = 384 / 48; 562 FDK_ASSERT(384 >= 48); 563 break; 564 default: 565 FDK_ASSERT(0); 566 return; 567 } 568 569 for (i = 0, k = k_step; i < M_LP_FILTER_ORDER; i++, k += k_step) { 570 FIXP_STP cs = SinTab[k]; 571 FIXP_DBL wA1, wA2; 572 573 wA1 = fMult(A1[i], f); 574 wA2 = fMult(A2[i], f); 575 576 /* r[i] = A[i]*cos() */ 577 tmp1[2 + i * 2] = fMult(wA1, cs.v.re); 578 tmp2[2 + i * 2] = fMult(wA2, cs.v.re); 579 /* i[i] = A[i]*sin() */ 580 tmp1[3 + i * 2] = -fMult(wA1, cs.v.im); 581 tmp2[3 + i * 2] = -fMult(wA2, cs.v.im); 582 583 f = fMult(f, FL2FXCONST_DBL(0.92f)); 584 } 585 586 /* Guarantee at least 2 bits of headroom for the FFT */ 587 /* "3" stands for 1.0 with 2 bits of headroom; (A1_exp + 2) guarantess 2 588 * bits of headroom if A1_exp > 1 */ 589 int A1_exp_fix = fMax(3, A1_exp + 2); 590 int A2_exp_fix = fMax(3, A2_exp + 2); 591 592 /* Set 1.0 in the proper format */ 593 tmp1[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A1_exp_fix); 594 tmp2[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A2_exp_fix); 595 596 tmp1[1] = tmp2[1] = (FIXP_DBL)0; 597 598 /* Clear the resto of the array */ 599 FDKmemclear( 600 tmp1 + 2 * (M_LP_FILTER_ORDER + 1), 601 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); 602 FDKmemclear( 603 tmp2 + 2 * (M_LP_FILTER_ORDER + 1), 604 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); 605 606 /* Guarantee 2 bits of headroom for FFT */ 607 scaleValues(&tmp1[2], (2 * M_LP_FILTER_ORDER), (A1_exp - A1_exp_fix)); 608 scaleValues(&tmp2[2], (2 * M_LP_FILTER_ORDER), (A2_exp - A2_exp_fix)); 609 610 INT s2; 611 s = A1_exp_fix; 612 s2 = A2_exp_fix; 613 614 fft(2 * fdns_npts, tmp1, &s); 615 fft(2 * fdns_npts, tmp2, &s2); 616 617 /* Adjust the exponents of both fft outputs if necessary*/ 618 if (s > s2) { 619 scaleValues(tmp2, 2 * fdns_npts, s2 - s); 620 s2 = s; 621 } else if (s < s2) { 622 scaleValues(tmp1, 2 * fdns_npts, s - s2); 623 s = s2; 624 } 625 626 FDK_ASSERT(s == s2); 627 } 628 629 /* Get amplitude and apply gains */ 630 step = lg / fdns_npts; 631 rr_minus_one = (FIXP_DBL)0; 632 633 for (k = 0; k < fdns_npts; k++) { 634 FIXP_DBL g1, g2, inv_g1_g2, a, b; 635 INT inv_g1_g2_e; 636 int g_e, shift; 637 638 { 639 FIXP_DBL real, imag; 640 int si1, si2, sInput; 641 642 real = tmp1[k * 2]; 643 imag = tmp1[k * 2 + 1]; 644 sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); 645 real <<= sInput; 646 imag <<= sInput; 647 /* g1_e = si1 - 2*s/2 */ 648 g1 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si1); 649 si1 += sInput; 650 651 real = tmp2[k * 2]; 652 imag = tmp2[k * 2 + 1]; 653 sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); 654 real <<= sInput; 655 imag <<= sInput; 656 /* g2_e = si2 - 2*s/2 */ 657 g2 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si2); 658 si2 += sInput; 659 660 /* Pick a common scale factor for g1 and g2 */ 661 if (si1 > si2) { 662 g2 >>= si1 - si2; 663 g_e = si1 - s; 664 } else { 665 g1 >>= si2 - si1; 666 g_e = si2 - s; 667 } 668 } 669 670 /* end of lpc2mdct() */ 671 672 FDK_ASSERT(g1 >= (FIXP_DBL)0); 673 FDK_ASSERT(g2 >= (FIXP_DBL)0); 674 675 /* mdct_IntNoiseShaping() */ 676 { 677 /* inv_g1_g2 * 2^inv_g1_g2_e = 1/(g1+g2) */ 678 inv_g1_g2 = (g1 >> 1) + (g2 >> 1); 679 if (inv_g1_g2 != (FIXP_DBL)0) { 680 inv_g1_g2 = fDivNorm(FL2FXCONST_DBL(0.5f), inv_g1_g2, &inv_g1_g2_e); 681 inv_g1_g2_e = inv_g1_g2_e - g_e; 682 } else { 683 inv_g1_g2 = (FIXP_DBL)MAXVAL_DBL; 684 inv_g1_g2_e = 0; 685 } 686 687 if (g_e < 0) { 688 /* a_e = g_e + inv_g1_g2_e + 1 */ 689 a = scaleValue(fMult(fMult(g1, g2), inv_g1_g2), g_e); 690 /* b_e = g_e + inv_g1_g2_e */ 691 b = fMult(g2 - g1, inv_g1_g2); 692 shift = g_e + inv_g1_g2_e + 1 - NSHAPE_SCALE; 693 } else { 694 /* a_e = (g_e+g_e) + inv_g1_g2_e + 1 */ 695 a = fMult(fMult(g1, g2), inv_g1_g2); 696 /* b_e = (g_e+g_e) + inv_g1_g2_e */ 697 b = scaleValue(fMult(g2 - g1, inv_g1_g2), -g_e); 698 shift = (g_e + g_e) + inv_g1_g2_e + 1 - NSHAPE_SCALE; 699 } 700 701 for (i = k * step; i < (k + 1) * step; i++) { 702 FIXP_DBL tmp; 703 704 /* rr[i] = 2*a*r[i] + b*rr[i-1] */ 705 tmp = fMult(a, r[i]); 706 tmp += scaleValue(fMultDiv2(b, rr_minus_one), NSHAPE_SCALE); 707 tmp = scaleValueSaturate(tmp, shift); 708 rr_minus_one = tmp; 709 r[i] = tmp; 710 } 711 } 712 } 713 714 /* end of mdct_IntNoiseShaping() */ 715 { *pScale += NSHAPE_SCALE; } 716 717 C_AALLOC_SCRATCH_END(tmp1, FIXP_DBL, FDNS_NPTS * 8) 718 } 719 720 /** 721 * \brief Calculates the energy. 722 * \param r pointer to spectrum. 723 * \param rs scale factor of spectrum r. 724 * \param lg frame length in audio samples. 725 * \param rms_e pointer to exponent of energy value. 726 * \return mantissa of energy value. 727 */ 728 static FIXP_DBL calcEnergy(const FIXP_DBL *r, const SHORT rs, const INT lg, 729 INT *rms_e) { 730 int headroom = getScalefactor(r, lg); 731 732 FIXP_DBL rms_m = 0; 733 734 /* Calculate number of growth bits due to addition */ 735 INT shift = (INT)(fNormz((FIXP_DBL)lg)); 736 shift = 31 - shift; 737 738 /* Generate 1e-2 in Q-6.37 */ 739 const FIXP_DBL value0_01 = 0x51eb851e; 740 const INT value0_01_exp = -6; 741 742 /* Find the exponent of the resulting energy value */ 743 *rms_e = ((rs - headroom) << 1) + shift + 1; 744 745 INT delta = *rms_e - value0_01_exp; 746 if (delta > 0) { 747 /* Limit shift_to 31*/ 748 delta = fMin(31, delta); 749 rms_m = value0_01 >> delta; 750 } else { 751 rms_m = value0_01; 752 *rms_e = value0_01_exp; 753 shift = shift - delta; 754 /* Limit shift_to 31*/ 755 shift = fMin(31, shift); 756 } 757 758 for (int i = 0; i < lg; i++) { 759 rms_m += fPow2Div2(r[i] << headroom) >> shift; 760 } 761 762 return rms_m; 763 } 764 765 /** 766 * \brief TCX gain calculation. 767 * \param pAacDecoderChannelInfo channel context data. 768 * \param r output spectrum. 769 * \param rms_e pointer to mantissa of energy value. 770 * \param rms_e pointer to exponent of energy value. 771 * \param frame the frame index of the LPD super frame. 772 * \param lg the frame length in audio samples. 773 * \param gain_m pointer to mantissa of TCX gain. 774 * \param gain_e pointer to exponent of TCX gain. 775 * \param elFlags element specific parser guidance flags. 776 * \param lg_fb the fullband frame length in audio samples. 777 * \param IGF_bgn the IGF start index. 778 */ 779 static void calcTCXGain(CAacDecoderChannelInfo *pAacDecoderChannelInfo, 780 FIXP_DBL *r, FIXP_DBL rms_m, INT rms_e, const INT frame, 781 const INT lg) { 782 if ((rms_m != (FIXP_DBL)0)) { 783 FIXP_DBL tcx_gain_m; 784 INT tcx_gain_e; 785 786 CLpd_DecodeGain(&tcx_gain_m, &tcx_gain_e, 787 pAacDecoderChannelInfo->pDynData->specificTo.usac 788 .tcx_global_gain[frame]); 789 790 /* rms * 2^rms_e = lg/sqrt(sum(spec^2)) */ 791 if (rms_e & 1) { 792 rms_m >>= 1; 793 rms_e++; 794 } 795 796 { 797 FIXP_DBL fx_lg; 798 INT fx_lg_e, s; 799 INT inv_e; 800 801 /* lg = fx_lg * 2^fx_lg_e */ 802 s = fNorm((FIXP_DBL)lg); 803 fx_lg = (FIXP_DBL)lg << s; 804 fx_lg_e = DFRACT_BITS - 1 - s; 805 /* 1/sqrt(rms) */ 806 rms_m = invSqrtNorm2(rms_m, &inv_e); 807 rms_m = fMult(rms_m, fx_lg); 808 rms_e = inv_e - (rms_e >> 1) + fx_lg_e; 809 } 810 811 { 812 int s = fNorm(tcx_gain_m); 813 tcx_gain_m = tcx_gain_m << s; 814 tcx_gain_e -= s; 815 } 816 817 tcx_gain_m = fMultDiv2(tcx_gain_m, rms_m); 818 tcx_gain_e = tcx_gain_e + rms_e; 819 820 /* global_gain * 2^(global_gain_e+rms_e) = (10^(global_gain/28)) * rms * 821 * 2^rms_e */ 822 { 823 { tcx_gain_e += 1; } 824 } 825 826 pAacDecoderChannelInfo->data.usac.tcx_gain[frame] = tcx_gain_m; 827 pAacDecoderChannelInfo->data.usac.tcx_gain_e[frame] = tcx_gain_e; 828 829 pAacDecoderChannelInfo->specScale[frame] += tcx_gain_e; 830 } 831 } 832 833 /** 834 * \brief FDNS decoding. 835 * \param pAacDecoderChannelInfo channel context data. 836 * \param pAacDecoderStaticChannelInfo channel context static data. 837 * \param r output spectrum. 838 * \param lg the frame length in audio samples. 839 * \param frame the frame index of the LPD super frame. 840 * \param pScale pointer to current scale shift factor of r[]. 841 * \param A1 old input LPC coefficients of length M_LP_FILTER_ORDER. 842 * \param A2 new input LPC coefficients of length M_LP_FILTER_ORDER. 843 * \param pAlfdGains pointer for ALFD gains output scaled by 1. 844 * \param fdns_npts number of lines (FDNS_NPTS). 845 * \param inf_mask pointer to noise mask. 846 * \param IGF_win_mode IGF window mode (LONG, SHORT, TCX10, TCX20). 847 * \param frameType (IGF_FRAME_DIVISION_AAC_OR_TCX_LONG or 848 * IGF_FRAME_DIVISION_TCX_SHORT_1). 849 * \param elFlags element specific parser guidance flags. 850 * \param lg_fb the fullband frame length in audio samples. 851 * \param IGF_bgn the IGF start index. 852 * \param rms_m mantisse of energy. 853 * \param rms_e exponent of energy. 854 */ 855 /* static */ 856 void CLpd_FdnsDecode(CAacDecoderChannelInfo *pAacDecoderChannelInfo, 857 CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, 858 FIXP_DBL r[], const INT lg, const INT frame, SHORT *pScale, 859 const FIXP_LPC A1[M_LP_FILTER_ORDER], const INT A1_exp, 860 const FIXP_LPC A2[M_LP_FILTER_ORDER], const INT A2_exp, 861 FIXP_DBL pAlfdGains[LFAC / 4], const INT fdns_npts) { 862 /* Weight LPC coefficients using Rm values */ 863 CLpd_AdaptLowFreqDeemph(r, lg, pAlfdGains, *pScale); 864 865 FIXP_DBL rms_m = (FIXP_DBL)0; 866 INT rms_e = 0; 867 { 868 /* Calculate Energy */ 869 rms_m = calcEnergy(r, *pScale, lg, &rms_e); 870 } 871 872 calcTCXGain(pAacDecoderChannelInfo, r, rms_m, rms_e, frame, lg); 873 874 /* Apply ODFT and Noise Shaping. LP coefficient (A1, A2) weighting is done 875 * inside on the fly. */ 876 877 lpc2mdctAndNoiseShaping(r, pScale, lg, fdns_npts, A1, A1_exp, A2, A2_exp); 878 } 879 880 /** 881 * find pitch for TCX20 (time domain) concealment. 882 */ 883 static int find_mpitch(FIXP_DBL xri[], int lg) { 884 FIXP_DBL max, pitch; 885 INT pitch_e; 886 int i, n; 887 888 max = (FIXP_DBL)0; 889 n = 2; 890 891 /* find maximum below 400Hz */ 892 for (i = 2; i < (lg >> 4); i += 2) { 893 FIXP_DBL tmp = fPow2Div2(xri[i]) + fPow2Div2(xri[i + 1]); 894 if (tmp > max) { 895 max = tmp; 896 n = i; 897 } 898 } 899 900 // pitch = ((float)lg<<1)/(float)n; 901 pitch = fDivNorm((FIXP_DBL)lg << 1, (FIXP_DBL)n, &pitch_e); 902 pitch >>= fixMax(0, DFRACT_BITS - 1 - pitch_e - 16); 903 904 /* find pitch multiple under 20ms */ 905 if (pitch >= (FIXP_DBL)((256 << 16) - 1)) { /*231.0f*/ 906 n = 256; 907 } else { 908 FIXP_DBL mpitch = pitch; 909 while (mpitch < (FIXP_DBL)(255 << 16)) { 910 mpitch += pitch; 911 } 912 n = (int)(mpitch - pitch) >> 16; 913 } 914 915 return (n); 916 } 917 918 /** 919 * number of spectral coefficients / time domain samples using frame mode as 920 * index. 921 */ 922 static const int lg_table_ccfl[2][4] = { 923 {256, 256, 512, 1024}, /* coreCoderFrameLength = 1024 */ 924 {192, 192, 384, 768} /* coreCoderFrameLength = 768 */ 925 }; 926 927 /** 928 * \brief Decode and render one MDCT-TCX frame. 929 * \param pAacDecoderChannelInfo channel context data. 930 * \param lg the frame length in audio samples. 931 * \param frame the frame index of the LPD super frame. 932 */ 933 static void CLpd_TcxDecode( 934 CAacDecoderChannelInfo *pAacDecoderChannelInfo, 935 CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags, 936 int mod, int last_mod, int frame, int frameOk) { 937 FIXP_DBL *pAlfd_gains = pAacDecoderStaticChannelInfo->last_alfd_gains; 938 ULONG *pSeed = &pAacDecoderStaticChannelInfo->nfRandomSeed; 939 int lg = (pAacDecoderChannelInfo->granuleLength == 128) 940 ? lg_table_ccfl[0][mod + 0] 941 : lg_table_ccfl[1][mod + 0]; 942 int next_frame = frame + (1 << (mod - 1)); 943 int isFullBandLpd = 0; 944 945 /* Obtain r[] vector by combining the quant[] and noise[] vectors */ 946 { 947 FIXP_DBL noise_level; 948 FIXP_DBL *coeffs = 949 SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, 950 pAacDecoderChannelInfo->granuleLength, isFullBandLpd); 951 int scale = pAacDecoderChannelInfo->specScale[frame]; 952 int i, nfBgn, nfEnd; 953 UCHAR tcx_noise_factor = pAacDecoderChannelInfo->pDynData->specificTo.usac 954 .tcx_noise_factor[frame]; 955 956 /* find pitch for bfi case */ 957 pAacDecoderStaticChannelInfo->last_tcx_pitch = find_mpitch(coeffs, lg); 958 959 if (frameOk) { 960 /* store for concealment */ 961 pAacDecoderStaticChannelInfo->last_tcx_noise_factor = tcx_noise_factor; 962 } else { 963 /* restore last frames value */ 964 tcx_noise_factor = pAacDecoderStaticChannelInfo->last_tcx_noise_factor; 965 } 966 967 noise_level = 968 (FIXP_DBL)((LONG)FL2FXCONST_DBL(0.0625f) * (8 - tcx_noise_factor)); 969 noise_level = scaleValue(noise_level, -scale); 970 971 const FIXP_DBL neg_noise_level = -noise_level; 972 973 { 974 nfBgn = lg / 6; 975 nfEnd = lg; 976 } 977 978 for (i = nfBgn; i < nfEnd - 7; i += 8) { 979 LONG tmp; 980 981 /* Fill all 8 consecutive zero coeffs with noise */ 982 tmp = coeffs[i + 0] | coeffs[i + 1] | coeffs[i + 2] | coeffs[i + 3] | 983 coeffs[i + 4] | coeffs[i + 5] | coeffs[i + 6] | coeffs[i + 7]; 984 985 if (tmp == 0) { 986 for (int k = i; k < i + 8; k++) { 987 UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) 988 : (coeffs[k] = noise_level); 989 } 990 } 991 } 992 if ((nfEnd - i) > 993 0) { /* noise filling for last "band" with less than 8 bins */ 994 LONG tmp = (LONG)coeffs[i]; 995 int k; 996 997 FDK_ASSERT((nfEnd - i) < 8); 998 for (k = 1; k < (nfEnd - i); k++) { 999 tmp |= (LONG)coeffs[i + k]; 1000 } 1001 if (tmp == 0) { 1002 for (k = i; k < nfEnd; k++) { 1003 UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) 1004 : (coeffs[k] = noise_level); 1005 } 1006 } 1007 } 1008 } 1009 1010 { 1011 /* Convert LPC to LP domain */ 1012 if (last_mod == 0) { 1013 /* Note: The case where last_mod == 255 is handled by other means 1014 * in CLpdChannelStream_Read() */ 1015 E_LPC_f_lsp_a_conversion( 1016 pAacDecoderChannelInfo->data.usac.lsp_coeff[frame], 1017 pAacDecoderChannelInfo->data.usac.lp_coeff[frame], 1018 &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame]); 1019 } 1020 1021 E_LPC_f_lsp_a_conversion( 1022 pAacDecoderChannelInfo->data.usac.lsp_coeff[next_frame], 1023 pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], 1024 &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame]); 1025 1026 /* FDNS decoding */ 1027 CLpd_FdnsDecode( 1028 pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, 1029 SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, 1030 pAacDecoderChannelInfo->granuleLength, isFullBandLpd), 1031 lg, frame, pAacDecoderChannelInfo->specScale + frame, 1032 pAacDecoderChannelInfo->data.usac.lp_coeff[frame], 1033 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame], 1034 pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], 1035 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame], pAlfd_gains, 1036 pAacDecoderChannelInfo->granuleLength / 2 /* == FDNS_NPTS(ccfl) */ 1037 ); 1038 } 1039 } 1040 1041 /** 1042 * \brief Read the tcx_coding bitstream part 1043 * \param hBs bitstream handle to read from. 1044 * \param pAacDecoderChannelInfo channel context info to store data into. 1045 * \param lg the frame length in audio samples. 1046 * \param first_tcx_flag flag indicating that this is the first TCX frame. 1047 * \param frame the frame index of the LPD super frame. 1048 */ 1049 static AAC_DECODER_ERROR CLpd_TCX_Read( 1050 HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, 1051 CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, int lg, 1052 int first_tcx_flag, int frame, UINT flags) { 1053 AAC_DECODER_ERROR errorAAC = AAC_DEC_OK; 1054 ARITH_CODING_ERROR error = ARITH_CODER_OK; 1055 FIXP_DBL *pSpec; 1056 int arith_reset_flag = 0; 1057 int isFullBandLpd = 0; 1058 1059 pSpec = SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, 1060 pAacDecoderChannelInfo->granuleLength, isFullBandLpd); 1061 1062 /* TCX noise level */ 1063 { 1064 pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_noise_factor[frame] = 1065 FDKreadBits(hBs, 3); 1066 } 1067 /* TCX global gain */ 1068 pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_global_gain[frame] = 1069 FDKreadBits(hBs, 7); 1070 1071 /* Arithmetic coded residual/spectrum */ 1072 if (first_tcx_flag) { 1073 if (flags & AC_INDEP) { 1074 arith_reset_flag = 1; 1075 } else { 1076 arith_reset_flag = FDKreadBits(hBs, 1); 1077 } 1078 } 1079 1080 /* CArco_DecodeArithData() output scale of "pSpec" is DFRACT_BITS-1 */ 1081 error = CArco_DecodeArithData(pAacDecoderStaticChannelInfo->hArCo, hBs, pSpec, 1082 lg, lg, arith_reset_flag); 1083 1084 /* Rescale residual/spectrum */ 1085 { 1086 int scale = getScalefactor(pSpec, lg) - 2; /* Leave 2 bits headroom */ 1087 1088 /* Exponent of CArco_DecodeArithData() output is DFRACT_BITS; integer 1089 * values. */ 1090 scaleValues(pSpec, lg, scale); 1091 scale = DFRACT_BITS - 1 - scale; 1092 1093 pAacDecoderChannelInfo->specScale[frame] = scale; 1094 } 1095 1096 if (error == ARITH_CODER_ERROR) errorAAC = AAC_DEC_UNKNOWN; 1097 1098 return errorAAC; 1099 } 1100 1101 /** 1102 * \brief translate lpd_mode into the mod[] array which describes the mode of 1103 * each each LPD frame 1104 * \param mod[] the array that will be filled with the mode indexes of the 1105 * inidividual frames. 1106 * \param lpd_mode the lpd_mode field read from the lpd_channel_stream 1107 */ 1108 static AAC_DECODER_ERROR CLpd_ReadAndMapLpdModeToModArray( 1109 UCHAR mod[4], HANDLE_FDK_BITSTREAM hBs, UINT elFlags) { 1110 int lpd_mode; 1111 1112 { 1113 lpd_mode = FDKreadBits(hBs, 5); 1114 1115 if (lpd_mode > 25 || lpd_mode < 0) { 1116 return AAC_DEC_PARSE_ERROR; 1117 } 1118 1119 switch (lpd_mode) { 1120 case 25: 1121 /* 1 80MS frame */ 1122 mod[0] = mod[1] = mod[2] = mod[3] = 3; 1123 break; 1124 case 24: 1125 /* 2 40MS frames */ 1126 mod[0] = mod[1] = mod[2] = mod[3] = 2; 1127 break; 1128 default: 1129 switch (lpd_mode >> 2) { 1130 case 4: 1131 /* lpd_mode 19 - 16 => 1 40MS and 2 20MS frames */ 1132 mod[0] = mod[1] = 2; 1133 mod[2] = (lpd_mode & 1) ? 1 : 0; 1134 mod[3] = (lpd_mode & 2) ? 1 : 0; 1135 break; 1136 case 5: 1137 /* lpd_mode 23 - 20 => 2 20MS and 1 40MS frames */ 1138 mod[2] = mod[3] = 2; 1139 mod[0] = (lpd_mode & 1) ? 1 : 0; 1140 mod[1] = (lpd_mode & 2) ? 1 : 0; 1141 break; 1142 default: 1143 /* lpd_mode < 16 => 4 20MS frames */ 1144 mod[0] = (lpd_mode & 1) ? 1 : 0; 1145 mod[1] = (lpd_mode & 2) ? 1 : 0; 1146 mod[2] = (lpd_mode & 4) ? 1 : 0; 1147 mod[3] = (lpd_mode & 8) ? 1 : 0; 1148 break; 1149 } 1150 break; 1151 } 1152 } 1153 return AAC_DEC_OK; 1154 } 1155 1156 static void CLpd_Reset( 1157 CAacDecoderChannelInfo *pAacDecoderChannelInfo, 1158 CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, 1159 int keep_past_signal) { 1160 int i; 1161 1162 /* Reset TCX / ACELP common memory */ 1163 if (!keep_past_signal) { 1164 FDKmemclear(pAacDecoderStaticChannelInfo->old_synth, 1165 sizeof(pAacDecoderStaticChannelInfo->old_synth)); 1166 } 1167 1168 /* Initialize the LSFs */ 1169 for (i = 0; i < M_LP_FILTER_ORDER; i++) { 1170 pAacDecoderStaticChannelInfo->lpc4_lsf[i] = fdk_dec_lsf_init[i]; 1171 } 1172 1173 /* Reset memory needed by bass post-filter */ 1174 FDKmemclear(pAacDecoderStaticChannelInfo->mem_bpf, 1175 sizeof(pAacDecoderStaticChannelInfo->mem_bpf)); 1176 1177 pAacDecoderStaticChannelInfo->old_bpf_control_info = 0; 1178 for (i = 0; i < SYN_SFD; i++) { 1179 pAacDecoderStaticChannelInfo->old_T_pf[i] = 64; 1180 pAacDecoderStaticChannelInfo->old_gain_pf[i] = (FIXP_DBL)0; 1181 } 1182 1183 /* Reset ACELP memory */ 1184 CLpd_AcelpReset(&pAacDecoderStaticChannelInfo->acelp); 1185 1186 pAacDecoderStaticChannelInfo->last_lpc_lost = 0; /* prev_lpc_lost */ 1187 pAacDecoderStaticChannelInfo->last_tcx_pitch = L_DIV; /* pitch_tcx */ 1188 pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; /* nbLostCmpt */ 1189 } 1190 1191 /* 1192 * Externally visible functions 1193 */ 1194 1195 AAC_DECODER_ERROR CLpdChannelStream_Read( 1196 HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, 1197 CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, 1198 const SamplingRateInfo *pSamplingRateInfo, UINT flags) { 1199 AAC_DECODER_ERROR error = AAC_DEC_OK; 1200 int first_tcx_flag; 1201 int k, nbDiv, fFacDataPresent, first_lpd_flag, acelp_core_mode, 1202 facGetMemState = 0; 1203 UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; 1204 int lpd_mode_last, prev_frame_was_lpd; 1205 USAC_COREMODE core_mode_last; 1206 const int lg_table_offset = 0; 1207 const int *lg_table = (pAacDecoderChannelInfo->granuleLength == 128) 1208 ? &lg_table_ccfl[0][lg_table_offset] 1209 : &lg_table_ccfl[1][lg_table_offset]; 1210 int last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost; 1211 1212 int last_frame_ok = CConcealment_GetLastFrameOk( 1213 &pAacDecoderStaticChannelInfo->concealmentInfo, 1); 1214 1215 INT i_offset; 1216 UINT samplingRate; 1217 1218 samplingRate = pSamplingRateInfo->samplingRate; 1219 1220 i_offset = 1221 (INT)(samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - 1222 (INT)PIT_MIN_12k8; 1223 1224 if (pSamplingRateInfo->samplingRate > 1225 FAC_FSCALE_MAX /* maximum allowed core sampling frequency */) { 1226 error = AAC_DEC_PARSE_ERROR; 1227 goto bail; 1228 } 1229 1230 acelp_core_mode = FDKreadBits(hBs, 3); 1231 1232 /* lpd_mode */ 1233 error = CLpd_ReadAndMapLpdModeToModArray(mod, hBs, 0); 1234 if (error != AAC_DEC_OK) { 1235 goto bail; 1236 } 1237 1238 /* bpf_control_info */ 1239 pAacDecoderChannelInfo->data.usac.bpf_control_info = FDKreadBit(hBs); 1240 1241 /* last_core_mode */ 1242 prev_frame_was_lpd = FDKreadBit(hBs); 1243 /* fac_data_present */ 1244 fFacDataPresent = FDKreadBit(hBs); 1245 1246 /* Set valid values from 1247 * pAacDecoderStaticChannelInfo->{last_core_mode,last_lpd_mode} */ 1248 pAacDecoderChannelInfo->data.usac.core_mode_last = 1249 pAacDecoderStaticChannelInfo->last_core_mode; 1250 lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last = 1251 pAacDecoderStaticChannelInfo->last_lpd_mode; 1252 1253 if (prev_frame_was_lpd == 0) { 1254 /* Last frame was FD */ 1255 pAacDecoderChannelInfo->data.usac.core_mode_last = FD_LONG; 1256 pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; 1257 } else { 1258 /* Last frame was LPD */ 1259 pAacDecoderChannelInfo->data.usac.core_mode_last = LPD; 1260 if (((mod[0] == 0) && fFacDataPresent) || 1261 ((mod[0] != 0) && !fFacDataPresent)) { 1262 /* Currend mod is ACELP, fac data present -> TCX, current mod TCX, no fac 1263 * data -> TCX */ 1264 if (lpd_mode_last == 0) { 1265 /* Bit stream interruption detected. Assume last TCX mode as TCX20. */ 1266 pAacDecoderChannelInfo->data.usac.lpd_mode_last = 1; 1267 } 1268 /* Else assume that remembered TCX mode is correct. */ 1269 } else { 1270 pAacDecoderChannelInfo->data.usac.lpd_mode_last = 0; 1271 } 1272 } 1273 1274 first_lpd_flag = (pAacDecoderChannelInfo->data.usac.core_mode_last != 1275 LPD); /* Depends on bitstream configuration */ 1276 first_tcx_flag = 1; 1277 1278 if (pAacDecoderStaticChannelInfo->last_core_mode != 1279 LPD) { /* ATTENTION: Reset depends on what we rendered before! */ 1280 CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, 0); 1281 1282 if (!last_frame_ok) { 1283 /* If last rendered frame was not LPD and first lpd flag is not set, this 1284 * must be an error - set last_lpc_lost flag */ 1285 last_lpc_lost |= (first_lpd_flag) ? 0 : 1; 1286 } 1287 } 1288 1289 core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last; 1290 lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last; 1291 1292 nbDiv = NB_DIV; 1293 1294 /* k is the frame index. If a frame is of size 40MS or 80MS, 1295 this frame index is incremented 2 or 4 instead of 1 respectively. */ 1296 1297 k = 0; 1298 while (k < nbDiv) { 1299 /* Reset FAC data pointers in order to avoid applying old random FAC data. 1300 */ 1301 pAacDecoderChannelInfo->data.usac.fac_data[k] = NULL; 1302 1303 if ((k == 0 && core_mode_last == LPD && fFacDataPresent) || 1304 (lpd_mode_last == 0 && mod[k] > 0) || 1305 ((lpd_mode_last != 255) && lpd_mode_last > 0 && mod[k] == 0)) { 1306 int err; 1307 1308 /* Assign FAC memory */ 1309 pAacDecoderChannelInfo->data.usac.fac_data[k] = 1310 CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); 1311 1312 /* FAC for (ACELP -> TCX) or (TCX -> ACELP) */ 1313 err = CLpd_FAC_Read( 1314 hBs, pAacDecoderChannelInfo->data.usac.fac_data[k], 1315 pAacDecoderChannelInfo->data.usac.fac_data_e, 1316 pAacDecoderChannelInfo->granuleLength, /* == fac_length */ 1317 0, k); 1318 if (err != 0) { 1319 error = AAC_DEC_PARSE_ERROR; 1320 goto bail; 1321 } 1322 } 1323 1324 if (mod[k] == 0) /* acelp-mode */ 1325 { 1326 int err; 1327 err = CLpd_AcelpRead( 1328 hBs, &pAacDecoderChannelInfo->data.usac.acelp[k], acelp_core_mode, 1329 pAacDecoderChannelInfo->granuleLength * 8 /* coreCoderFrameLength */, 1330 i_offset); 1331 if (err != 0) { 1332 error = AAC_DEC_PARSE_ERROR; 1333 goto bail; 1334 } 1335 1336 lpd_mode_last = 0; 1337 k++; 1338 } else /* mode != 0 => TCX */ 1339 { 1340 error = CLpd_TCX_Read(hBs, pAacDecoderChannelInfo, 1341 pAacDecoderStaticChannelInfo, lg_table[mod[k]], 1342 first_tcx_flag, k, flags); 1343 1344 lpd_mode_last = mod[k]; 1345 first_tcx_flag = 0; 1346 k += 1 << (mod[k] - 1); 1347 } 1348 if (error != AAC_DEC_OK) { 1349 error = AAC_DEC_PARSE_ERROR; 1350 goto bail; 1351 } 1352 } 1353 1354 { 1355 int err; 1356 1357 /* Read LPC coefficients */ 1358 err = CLpc_Read( 1359 hBs, pAacDecoderChannelInfo->data.usac.lsp_coeff, 1360 pAacDecoderStaticChannelInfo->lpc4_lsf, 1361 pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, 1362 pAacDecoderChannelInfo->data.usac.aStability, mod, first_lpd_flag, 1363 /* if last lpc4 is available from concealment do not extrapolate lpc0 1364 from lpc2 */ 1365 (mod[0] & 0x3) ? 0 1366 : (last_lpc_lost && 1367 pAacDecoderStaticChannelInfo->last_core_mode != LPD), 1368 last_frame_ok); 1369 if (err != 0) { 1370 error = AAC_DEC_PARSE_ERROR; 1371 goto bail; 1372 } 1373 } 1374 1375 /* adjust old lsp[] following to a bad frame (to avoid overshoot) (ref: 1376 * dec_LPD.c) */ 1377 if (last_lpc_lost && !last_frame_ok) { 1378 int k_next; 1379 k = 0; 1380 while (k < nbDiv) { 1381 int i; 1382 k_next = k + (((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1))); 1383 FIXP_LPC *lsp_old = pAacDecoderChannelInfo->data.usac.lsp_coeff[k]; 1384 FIXP_LPC *lsp_new = pAacDecoderChannelInfo->data.usac.lsp_coeff[k_next]; 1385 1386 for (i = 0; i < M_LP_FILTER_ORDER; i++) { 1387 if (lsp_new[i] < lsp_old[i]) { 1388 lsp_old[i] = lsp_new[i]; 1389 } 1390 } 1391 k = k_next; 1392 } 1393 } 1394 1395 if (!CConcealment_GetLastFrameOk( 1396 &pAacDecoderStaticChannelInfo->concealmentInfo, 1)) { 1397 E_LPC_f_lsp_a_conversion( 1398 pAacDecoderChannelInfo->data.usac.lsp_coeff[0], 1399 pAacDecoderChannelInfo->data.usac.lp_coeff[0], 1400 &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]); 1401 } else if (pAacDecoderStaticChannelInfo->last_lpd_mode != 0) { 1402 if (pAacDecoderStaticChannelInfo->last_lpd_mode == 255) { 1403 /* We need it for TCX decoding or ACELP excitation update */ 1404 E_LPC_f_lsp_a_conversion( 1405 pAacDecoderChannelInfo->data.usac.lsp_coeff[0], 1406 pAacDecoderChannelInfo->data.usac.lp_coeff[0], 1407 &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]); 1408 } else { /* last_lpd_mode was TCX */ 1409 /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid 1410 * converting LSP coefficients again). */ 1411 FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], 1412 pAacDecoderStaticChannelInfo->lp_coeff_old[0], 1413 M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); 1414 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = 1415 pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; 1416 } 1417 } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ 1418 1419 if (fFacDataPresent && (core_mode_last != LPD)) { 1420 int prev_frame_was_short; 1421 1422 prev_frame_was_short = FDKreadBit(hBs); 1423 1424 if (prev_frame_was_short) { 1425 core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last = 1426 FD_SHORT; 1427 pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; 1428 1429 if ((pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT) && 1430 CConcealment_GetLastFrameOk( 1431 &pAacDecoderStaticChannelInfo->concealmentInfo, 1)) { 1432 /* USAC Conformance document: 1433 short_fac_flag shall be encoded with a value of 1 if the 1434 window_sequence of the previous frame was 2 (EIGHT_SHORT_SEQUENCE). 1435 Otherwise short_fac_flag shall be encoded with a 1436 value of 0. */ 1437 error = AAC_DEC_PARSE_ERROR; 1438 goto bail; 1439 } 1440 } 1441 1442 /* Assign memory */ 1443 pAacDecoderChannelInfo->data.usac.fac_data[0] = 1444 CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); 1445 1446 { 1447 int err; 1448 1449 /* FAC for FD -> ACELP */ 1450 err = CLpd_FAC_Read( 1451 hBs, pAacDecoderChannelInfo->data.usac.fac_data[0], 1452 pAacDecoderChannelInfo->data.usac.fac_data_e, 1453 CLpd_FAC_getLength(core_mode_last != FD_SHORT, 1454 pAacDecoderChannelInfo->granuleLength), 1455 1, 0); 1456 if (err != 0) { 1457 error = AAC_DEC_PARSE_ERROR; 1458 goto bail; 1459 } 1460 } 1461 } 1462 1463 bail: 1464 if (error == AAC_DEC_OK) { 1465 /* check consitency of last core/lpd mode values */ 1466 if ((pAacDecoderChannelInfo->data.usac.core_mode_last != 1467 pAacDecoderStaticChannelInfo->last_core_mode) && 1468 (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { 1469 /* Something got wrong! */ 1470 /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ 1471 } else if ((pAacDecoderChannelInfo->data.usac.core_mode_last == LPD) && 1472 (pAacDecoderChannelInfo->data.usac.lpd_mode_last != 1473 pAacDecoderStaticChannelInfo->last_lpd_mode) && 1474 (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { 1475 /* Something got wrong! */ 1476 /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ 1477 } 1478 } 1479 1480 return error; 1481 } 1482 1483 void CLpdChannelStream_Decode( 1484 CAacDecoderChannelInfo *pAacDecoderChannelInfo, 1485 CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags) { 1486 UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; 1487 int k; 1488 UCHAR last_lpd_mode; 1489 int nbDiv = NB_DIV; 1490 1491 /* k is the frame index. If a frame is of size 40MS or 80MS, 1492 this frame index is incremented 2 or 4 instead of 1 respectively. */ 1493 k = 0; 1494 last_lpd_mode = 1495 pAacDecoderChannelInfo->data.usac 1496 .lpd_mode_last; /* could be different to what has been rendered */ 1497 while (k < nbDiv) { 1498 if (mod[k] == 0) { 1499 /* ACELP */ 1500 1501 /* If FAC (fac_data[k] != NULL), and previous frame was TCX, apply (TCX) 1502 * gains to FAC data */ 1503 if (last_lpd_mode > 0 && last_lpd_mode != 255 && 1504 pAacDecoderChannelInfo->data.usac.fac_data[k]) { 1505 CFac_ApplyGains(pAacDecoderChannelInfo->data.usac.fac_data[k], 1506 pAacDecoderChannelInfo->granuleLength, 1507 pAacDecoderStaticChannelInfo->last_tcx_gain, 1508 pAacDecoderStaticChannelInfo->last_alfd_gains, 1509 (last_lpd_mode < 4) ? last_lpd_mode : 3); 1510 1511 pAacDecoderChannelInfo->data.usac.fac_data_e[k] += 1512 pAacDecoderStaticChannelInfo->last_tcx_gain_e; 1513 } 1514 } else { 1515 /* TCX */ 1516 CLpd_TcxDecode(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, 1517 flags, mod[k], last_lpd_mode, k, 1 /* frameOk == 1 */ 1518 ); 1519 1520 /* Store TCX gain scale for next possible FAC transition. */ 1521 pAacDecoderStaticChannelInfo->last_tcx_gain = 1522 pAacDecoderChannelInfo->data.usac.tcx_gain[k]; 1523 pAacDecoderStaticChannelInfo->last_tcx_gain_e = 1524 pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; 1525 1526 /* If FAC (fac_data[k] != NULL), apply gains */ 1527 if (last_lpd_mode == 0 && pAacDecoderChannelInfo->data.usac.fac_data[k]) { 1528 CFac_ApplyGains( 1529 pAacDecoderChannelInfo->data.usac.fac_data[k], 1530 pAacDecoderChannelInfo->granuleLength /* == fac_length */, 1531 pAacDecoderChannelInfo->data.usac.tcx_gain[k], 1532 pAacDecoderStaticChannelInfo->last_alfd_gains, mod[k]); 1533 1534 pAacDecoderChannelInfo->data.usac.fac_data_e[k] += 1535 pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; 1536 } 1537 } 1538 1539 /* remember previous mode */ 1540 last_lpd_mode = mod[k]; 1541 1542 /* Increase k to next frame */ 1543 k += (mod[k] == 0) ? 1 : (1 << (mod[k] - 1)); 1544 } 1545 } 1546 1547 AAC_DECODER_ERROR CLpd_RenderTimeSignal( 1548 CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, 1549 CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData, 1550 INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, UINT flags, 1551 UINT strmFlags) { 1552 UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; 1553 AAC_DECODER_ERROR error = AAC_DEC_OK; 1554 int k, i_offset; 1555 int last_k; 1556 int nrSamples = 0; 1557 int facFB = 1; 1558 int nbDiv = NB_DIV; 1559 int lDiv = lFrame / nbDiv; /* length of division (acelp or tcx20 frame)*/ 1560 int lFac = lDiv / 2; 1561 int nbSubfr = 1562 lFrame / (nbDiv * L_SUBFR); /* number of subframes per division */ 1563 int nbSubfrSuperfr = nbDiv * nbSubfr; 1564 int synSfd = (nbSubfrSuperfr / 2) - BPF_SFD; 1565 int SynDelay = synSfd * L_SUBFR; 1566 int aacDelay = lFrame / 2; 1567 1568 /* 1569 In respect to the reference software, the synth pointer here is lagging by 1570 aacDelay ( == SYN_DELAY + BPF_DELAY ) samples. The corresponding old 1571 synthesis samples are handled by the IMDCT overlap. 1572 */ 1573 1574 FIXP_DBL *synth_buf = 1575 pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1->synth_buf; 1576 FIXP_DBL *synth = synth_buf + PIT_MAX_MAX - BPF_DELAY; 1577 UCHAR last_lpd_mode, last_last_lpd_mode, last_lpc_lost, last_frame_lost; 1578 1579 INT pitch[NB_SUBFR_SUPERFR + SYN_SFD]; 1580 FIXP_DBL pit_gain[NB_SUBFR_SUPERFR + SYN_SFD]; 1581 1582 const int *lg_table; 1583 int lg_table_offset = 0; 1584 1585 UINT samplingRate = pSamplingRateInfo->samplingRate; 1586 1587 FDKmemclear(pitch, (NB_SUBFR_SUPERFR + SYN_SFD) * sizeof(INT)); 1588 1589 if (flags & AACDEC_FLUSH) { 1590 CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, 1591 flags & AACDEC_FLUSH); 1592 frameOk = 0; 1593 } 1594 1595 switch (lFrame) { 1596 case 1024: 1597 lg_table = &lg_table_ccfl[0][lg_table_offset]; 1598 break; 1599 case 768: 1600 lg_table = &lg_table_ccfl[1][lg_table_offset]; 1601 break; 1602 default: 1603 FDK_ASSERT(0); 1604 return AAC_DEC_UNKNOWN; 1605 } 1606 1607 last_frame_lost = !CConcealment_GetLastFrameOk( 1608 &pAacDecoderStaticChannelInfo->concealmentInfo, 0); 1609 1610 /* Maintain LPD mode from previous frame */ 1611 if ((pAacDecoderStaticChannelInfo->last_core_mode == FD_LONG) || 1612 (pAacDecoderStaticChannelInfo->last_core_mode == FD_SHORT)) { 1613 pAacDecoderStaticChannelInfo->last_lpd_mode = 255; 1614 } 1615 1616 if (!frameOk) { 1617 FIXP_DBL old_tcx_gain; 1618 FIXP_SGL old_stab; 1619 SCHAR old_tcx_gain_e; 1620 int nLostSf; 1621 1622 last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; 1623 old_tcx_gain = pAacDecoderStaticChannelInfo->last_tcx_gain; 1624 old_tcx_gain_e = pAacDecoderStaticChannelInfo->last_tcx_gain_e; 1625 old_stab = pAacDecoderStaticChannelInfo->oldStability; 1626 nLostSf = pAacDecoderStaticChannelInfo->numLostLpdFrames; 1627 1628 /* patch the last LPD mode */ 1629 pAacDecoderChannelInfo->data.usac.lpd_mode_last = last_lpd_mode; 1630 1631 /* Do mode extrapolation and repeat the previous mode: 1632 if previous mode = ACELP -> ACELP 1633 if previous mode = TCX-20/40 -> TCX-20 1634 if previous mode = TCX-80 -> TCX-80 1635 notes: 1636 - ACELP is not allowed after TCX (no pitch information to reuse) 1637 - TCX-40 is not allowed in the mode repetition to keep the logic simple 1638 */ 1639 switch (last_lpd_mode) { 1640 case 0: 1641 mod[0] = mod[1] = mod[2] = mod[3] = 0; /* -> ACELP concealment */ 1642 break; 1643 case 3: 1644 mod[0] = mod[1] = mod[2] = mod[3] = 3; /* -> TCX FD concealment */ 1645 break; 1646 case 2: 1647 mod[0] = mod[1] = mod[2] = mod[3] = 2; /* -> TCX FD concealment */ 1648 break; 1649 case 1: 1650 default: 1651 mod[0] = mod[1] = mod[2] = mod[3] = 4; /* -> TCX TD concealment */ 1652 break; 1653 } 1654 1655 /* LPC extrapolation */ 1656 CLpc_Conceal(pAacDecoderChannelInfo->data.usac.lsp_coeff, 1657 pAacDecoderStaticChannelInfo->lpc4_lsf, 1658 pAacDecoderStaticChannelInfo->lsf_adaptive_mean, 1659 /*(pAacDecoderStaticChannelInfo->numLostLpdFrames == 0) ||*/ 1660 (last_lpd_mode == 255)); 1661 1662 if ((last_lpd_mode > 0) && (last_lpd_mode < 255)) { 1663 /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid 1664 * converting LSP coefficients again). */ 1665 FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], 1666 pAacDecoderStaticChannelInfo->lp_coeff_old[0], 1667 M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); 1668 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = 1669 pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; 1670 } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ 1671 /* case last_lpd_mode was Time domain TCX concealment is handled after this 1672 * "if (!frameOk)"-block */ 1673 1674 /* k is the frame index. If a frame is of size 40MS or 80MS, 1675 this frame index is incremented 2 or 4 instead of 1 respectively. */ 1676 k = 0; 1677 while (k < nbDiv) { 1678 pAacDecoderChannelInfo->data.usac.tcx_gain[k] = old_tcx_gain; 1679 pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] = old_tcx_gain_e; 1680 1681 /* restore stability value from last frame */ 1682 pAacDecoderChannelInfo->data.usac.aStability[k] = old_stab; 1683 1684 /* Increase k to next frame */ 1685 k += ((mod[k] & 0x3) == 0) ? 1 : (1 << ((mod[k] & 0x3) - 1)); 1686 1687 nLostSf++; 1688 } 1689 } else { 1690 if ((pAacDecoderStaticChannelInfo->last_lpd_mode == 4) && (mod[0] > 0)) { 1691 /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid 1692 * converting LSP coefficients again). */ 1693 FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], 1694 pAacDecoderStaticChannelInfo->lp_coeff_old[0], 1695 M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); 1696 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = 1697 pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; 1698 } 1699 } 1700 1701 Acelp_PreProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, pitch, 1702 pAacDecoderStaticChannelInfo->old_T_pf, pit_gain, 1703 pAacDecoderStaticChannelInfo->old_gain_pf, samplingRate, 1704 &i_offset, lFrame, synSfd, nbSubfrSuperfr); 1705 1706 /* k is the frame index. If a frame is of size 40MS or 80MS, 1707 this frame index is incremented 2 or 4 instead of 1 respectively. */ 1708 k = 0; 1709 last_k = -1; /* mark invalid */ 1710 last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; 1711 last_last_lpd_mode = pAacDecoderStaticChannelInfo->last_last_lpd_mode; 1712 last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost | last_frame_lost; 1713 1714 /* This buffer must be avalable for the case of FD->ACELP transition. The 1715 beginning of the buffer is used after the BPF to overwrite the output signal. 1716 Only the FAC area must be affected by the BPF */ 1717 1718 while (k < nbDiv) { 1719 if (frameOk == 0) { 1720 pAacDecoderStaticChannelInfo->numLostLpdFrames++; 1721 } else { 1722 last_frame_lost |= 1723 (pAacDecoderStaticChannelInfo->numLostLpdFrames > 0) ? 1 : 0; 1724 pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; 1725 } 1726 if (mod[k] == 0 || mod[k] == 4) { 1727 /* ACELP or TCX time domain concealment */ 1728 FIXP_DBL *acelp_out; 1729 1730 /* FAC management */ 1731 if ((last_lpd_mode != 0) && (last_lpd_mode != 4)) /* TCX TD concealment */ 1732 { 1733 FIXP_DBL *pFacData = NULL; 1734 1735 if (frameOk && !last_frame_lost) { 1736 pFacData = pAacDecoderChannelInfo->data.usac.fac_data[k]; 1737 } 1738 1739 nrSamples += CLpd_FAC_Mdct2Acelp( 1740 &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, pFacData, 1741 pAacDecoderChannelInfo->data.usac.fac_data_e[k], 1742 pAacDecoderChannelInfo->data.usac.lp_coeff[k], 1743 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], 1744 lFrame - nrSamples, 1745 CLpd_FAC_getLength( 1746 (pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT) || 1747 (k > 0), 1748 lFac), 1749 (pAacDecoderStaticChannelInfo->last_core_mode != LPD) && (k == 0), 1750 0); 1751 1752 FDKmemcpy( 1753 synth + nrSamples, pAacDecoderStaticChannelInfo->IMdct.overlap.time, 1754 pAacDecoderStaticChannelInfo->IMdct.ov_offset * sizeof(FIXP_DBL)); 1755 { 1756 FIXP_LPC *lp_prev = 1757 pAacDecoderChannelInfo->data.usac 1758 .lp_coeff[0]; /* init value does not real matter */ 1759 INT lp_prev_exp = pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]; 1760 1761 if (last_lpd_mode != 255) { /* last mode was tcx */ 1762 last_k = k - (1 << (last_lpd_mode - 1)); 1763 if (last_k < 0) { 1764 lp_prev = pAacDecoderStaticChannelInfo->lp_coeff_old[1]; 1765 lp_prev_exp = pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1]; 1766 } else { 1767 lp_prev = pAacDecoderChannelInfo->data.usac.lp_coeff[last_k]; 1768 lp_prev_exp = 1769 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; 1770 } 1771 } 1772 1773 CLpd_AcelpPrepareInternalMem( 1774 synth + aacDelay + k * lDiv, last_lpd_mode, 1775 (last_last_lpd_mode == 4) ? 0 : last_last_lpd_mode, 1776 pAacDecoderChannelInfo->data.usac.lp_coeff[k], 1777 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], lp_prev, 1778 lp_prev_exp, &pAacDecoderStaticChannelInfo->acelp, lFrame, 1779 (last_frame_lost && k < 2), mod[k]); 1780 } 1781 } else { 1782 if (k == 0 && pAacDecoderStaticChannelInfo->IMdct.ov_offset != 1783 lFrame / facFB / 2) { 1784 pAacDecoderStaticChannelInfo->IMdct.ov_offset = lFrame / facFB / 2; 1785 } 1786 nrSamples += imdct_drain(&pAacDecoderStaticChannelInfo->IMdct, 1787 synth + nrSamples, lFrame / facFB - nrSamples); 1788 } 1789 1790 if (nrSamples >= lFrame / facFB) { 1791 /* Write ACELP time domain samples into IMDCT overlap buffer at 1792 * pAacDecoderStaticChannelInfo->IMdct.overlap.time + 1793 * pAacDecoderStaticChannelInfo->IMdct.ov_offset 1794 */ 1795 acelp_out = pAacDecoderStaticChannelInfo->IMdct.overlap.time + 1796 pAacDecoderStaticChannelInfo->IMdct.ov_offset; 1797 1798 /* Account ACELP time domain output samples to overlap buffer */ 1799 pAacDecoderStaticChannelInfo->IMdct.ov_offset += lDiv; 1800 } else { 1801 /* Write ACELP time domain samples into output buffer at pTimeData + 1802 * nrSamples */ 1803 acelp_out = synth + nrSamples; 1804 1805 /* Account ACELP time domain output samples to output buffer */ 1806 nrSamples += lDiv; 1807 } 1808 1809 if (mod[k] == 4) { 1810 pAacDecoderStaticChannelInfo->acelp.wsyn_rms = scaleValue( 1811 pAacDecoderChannelInfo->data.usac.tcx_gain[k], 1812 fixMin(0, 1813 pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] - SF_EXC)); 1814 CLpd_TcxTDConceal(&pAacDecoderStaticChannelInfo->acelp, 1815 &pAacDecoderStaticChannelInfo->last_tcx_pitch, 1816 pAacDecoderChannelInfo->data.usac.lsp_coeff[k], 1817 pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], 1818 pAacDecoderChannelInfo->data.usac.aStability[k], 1819 pAacDecoderStaticChannelInfo->numLostLpdFrames, 1820 acelp_out, lFrame, 1821 pAacDecoderStaticChannelInfo->last_tcx_noise_factor); 1822 1823 } else { 1824 FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[k] >= 1825 (FIXP_SGL)0); 1826 CLpd_AcelpDecode(&pAacDecoderStaticChannelInfo->acelp, i_offset, 1827 pAacDecoderChannelInfo->data.usac.lsp_coeff[k], 1828 pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], 1829 pAacDecoderChannelInfo->data.usac.aStability[k], 1830 &pAacDecoderChannelInfo->data.usac.acelp[k], 1831 pAacDecoderStaticChannelInfo->numLostLpdFrames, 1832 last_lpc_lost, k, acelp_out, 1833 &pitch[(k * nbSubfr) + synSfd], 1834 &pit_gain[(k * nbSubfr) + synSfd], lFrame); 1835 } 1836 1837 if (mod[k] != 4) { 1838 if (last_lpd_mode != 0 && 1839 pAacDecoderChannelInfo->data.usac 1840 .bpf_control_info) { /* FD/TCX -> ACELP transition */ 1841 /* bass post-filter past FAC area (past two (one for FD short) 1842 * subframes) */ 1843 int currentSf = synSfd + k * nbSubfr; 1844 1845 if ((k > 0) || (pAacDecoderStaticChannelInfo->last_core_mode != 1846 FD_SHORT)) { /* TCX or FD long -> ACELP */ 1847 pitch[currentSf - 2] = pitch[currentSf - 1] = pitch[currentSf]; 1848 pit_gain[currentSf - 2] = pit_gain[currentSf - 1] = 1849 pit_gain[currentSf]; 1850 } else { /* FD short -> ACELP */ 1851 pitch[currentSf - 1] = pitch[currentSf]; 1852 pit_gain[currentSf - 1] = pit_gain[currentSf]; 1853 } 1854 } 1855 } 1856 } else { /* TCX */ 1857 int lg = lg_table[mod[k]]; 1858 int isFullBandLpd = 0; 1859 1860 /* FAC management */ 1861 if ((last_lpd_mode == 0) || (last_lpd_mode == 4)) /* TCX TD concealment */ 1862 { 1863 C_AALLOC_SCRATCH_START(fac_buf, FIXP_DBL, 1024 / 8); 1864 1865 /* pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL means no FAC 1866 * data available. */ 1867 if (last_frame_lost == 1 || 1868 pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL) { 1869 FDKmemclear(fac_buf, 1024 / 8 * sizeof(FIXP_DBL)); 1870 pAacDecoderChannelInfo->data.usac.fac_data[k] = fac_buf; 1871 pAacDecoderChannelInfo->data.usac.fac_data_e[k] = 0; 1872 } 1873 1874 nrSamples += CLpd_FAC_Acelp2Mdct( 1875 &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, 1876 SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, 1877 pAacDecoderChannelInfo->granuleLength, isFullBandLpd), 1878 pAacDecoderChannelInfo->specScale + k, 1, 1879 pAacDecoderChannelInfo->data.usac.fac_data[k], 1880 pAacDecoderChannelInfo->data.usac.fac_data_e[k], 1881 pAacDecoderChannelInfo->granuleLength /* == fac_length */, 1882 lFrame - nrSamples, lg, 1883 FDKgetWindowSlope(lDiv, 1884 GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), 1885 lDiv, pAacDecoderChannelInfo->data.usac.lp_coeff[k], 1886 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], 1887 &pAacDecoderStaticChannelInfo->acelp, 1888 pAacDecoderChannelInfo->data.usac.tcx_gain[k], 1889 (last_frame_lost || !frameOk), 0 /* is not FD FAC */ 1890 , 1891 last_lpd_mode, k, 1892 pAacDecoderChannelInfo 1893 ->currAliasingSymmetry /* Note: The current aliasing 1894 symmetry for a TCX (i.e. LPD) 1895 frame must always be 0 */ 1896 ); 1897 1898 pitch[(k * nbSubfr) + synSfd + 1] = pitch[(k * nbSubfr) + synSfd] = 1899 pitch[(k * nbSubfr) + synSfd - 1]; 1900 pit_gain[(k * nbSubfr) + synSfd + 1] = 1901 pit_gain[(k * nbSubfr) + synSfd] = 1902 pit_gain[(k * nbSubfr) + synSfd - 1]; 1903 1904 C_AALLOC_SCRATCH_END(fac_buf, FIXP_DBL, 1024 / 8); 1905 } else { 1906 int tl = lg; 1907 int fl = lDiv; 1908 int fr = lDiv; 1909 1910 nrSamples += imlt_block( 1911 &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, 1912 SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, 1913 pAacDecoderChannelInfo->granuleLength, isFullBandLpd), 1914 pAacDecoderChannelInfo->specScale + k, 1, lFrame - nrSamples, tl, 1915 FDKgetWindowSlope(fl, 1916 GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), 1917 fl, 1918 FDKgetWindowSlope(fr, 1919 GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), 1920 fr, pAacDecoderChannelInfo->data.usac.tcx_gain[k], 1921 pAacDecoderChannelInfo->currAliasingSymmetry 1922 ? MLT_FLAG_CURR_ALIAS_SYMMETRY 1923 : 0); 1924 } 1925 } 1926 /* remember previous mode */ 1927 last_last_lpd_mode = last_lpd_mode; 1928 last_lpd_mode = mod[k]; 1929 last_lpc_lost = (frameOk == 0) ? 1 : 0; 1930 1931 /* Increase k to next frame */ 1932 last_k = k; 1933 k += ((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1)); 1934 } 1935 1936 if (frameOk) { 1937 /* assume data was ok => store for concealment */ 1938 FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[last_k] >= 1939 (FIXP_SGL)0); 1940 pAacDecoderStaticChannelInfo->oldStability = 1941 pAacDecoderChannelInfo->data.usac.aStability[last_k]; 1942 FDKmemcpy(pAacDecoderStaticChannelInfo->lsf_adaptive_mean, 1943 pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, 1944 M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); 1945 } 1946 1947 /* store past lp coeffs for next superframe (they are only valid and needed if 1948 * last_lpd_mode was tcx) */ 1949 if (last_lpd_mode > 0) { 1950 FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[0], 1951 pAacDecoderChannelInfo->data.usac.lp_coeff[nbDiv], 1952 M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); 1953 pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0] = 1954 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[nbDiv]; 1955 FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[1], 1956 pAacDecoderChannelInfo->data.usac.lp_coeff[last_k], 1957 M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); 1958 pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1] = 1959 pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; 1960 } 1961 1962 FDK_ASSERT(nrSamples == lFrame); 1963 1964 /* check whether usage of bass postfilter was de-activated in the bitstream; 1965 if yes, set pitch gain to 0 */ 1966 if (!(pAacDecoderChannelInfo->data.usac.bpf_control_info)) { 1967 if (mod[0] != 0 && (pAacDecoderStaticChannelInfo->old_bpf_control_info)) { 1968 for (int i = 2; i < nbSubfrSuperfr; i++) 1969 pit_gain[synSfd + i] = (FIXP_DBL)0; 1970 } else { 1971 for (int i = 0; i < nbSubfrSuperfr; i++) 1972 pit_gain[synSfd + i] = (FIXP_DBL)0; 1973 } 1974 } 1975 1976 /* for bass postfilter */ 1977 for (int n = 0; n < synSfd; n++) { 1978 pAacDecoderStaticChannelInfo->old_T_pf[n] = pitch[nbSubfrSuperfr + n]; 1979 pAacDecoderStaticChannelInfo->old_gain_pf[n] = pit_gain[nbSubfrSuperfr + n]; 1980 } 1981 1982 pAacDecoderStaticChannelInfo->old_bpf_control_info = 1983 pAacDecoderChannelInfo->data.usac.bpf_control_info; 1984 1985 { 1986 INT lookahead = -BPF_DELAY; 1987 int copySamp = (mod[nbDiv - 1] == 0) ? (aacDelay) : (aacDelay - lFac); 1988 1989 /* Copy enough time domain samples from MDCT to synthesis buffer as needed 1990 * by the bass postfilter */ 1991 1992 lookahead += imdct_copy_ov_and_nr(&pAacDecoderStaticChannelInfo->IMdct, 1993 synth + nrSamples, copySamp); 1994 1995 FDK_ASSERT(lookahead == copySamp - BPF_DELAY); 1996 1997 FIXP_DBL *p2_synth = synth + BPF_DELAY; 1998 1999 /* recalculate pitch gain to allow postfilering on FAC area */ 2000 for (int i = 0; i < nbSubfrSuperfr; i++) { 2001 int T = pitch[i]; 2002 FIXP_DBL gain = pit_gain[i]; 2003 2004 if (gain > (FIXP_DBL)0) { 2005 gain = get_gain(&p2_synth[i * L_SUBFR], &p2_synth[(i * L_SUBFR) - T], 2006 L_SUBFR); 2007 pit_gain[i] = gain; 2008 } 2009 } 2010 2011 { 2012 bass_pf_1sf_delay(p2_synth, pitch, pit_gain, lFrame, lFrame / facFB, 2013 mod[nbDiv - 1] ? (SynDelay - (lDiv / 2)) : SynDelay, 2014 pTimeData, pAacDecoderStaticChannelInfo->mem_bpf); 2015 } 2016 } 2017 2018 Acelp_PostProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, 2019 pitch, pAacDecoderStaticChannelInfo->old_T_pf, lFrame, 2020 synSfd, nbSubfrSuperfr); 2021 2022 /* Store last mode for next super frame */ 2023 { pAacDecoderStaticChannelInfo->last_core_mode = LPD; } 2024 pAacDecoderStaticChannelInfo->last_lpd_mode = last_lpd_mode; 2025 pAacDecoderStaticChannelInfo->last_last_lpd_mode = last_last_lpd_mode; 2026 pAacDecoderStaticChannelInfo->last_lpc_lost = last_lpc_lost; 2027 2028 return error; 2029 } 2030
