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diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp new file mode 100644 index 0000000..4ce6699 --- /dev/null +++ b/libAACdec/src/usacdec_lpd.cpp @@ -0,0 +1,2029 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC decoder library ****************************** + + Author(s): Manuel Jander + + Description: USAC Linear Prediction Domain coding + +*******************************************************************************/ + +#include "usacdec_lpd.h" + +#include "usacdec_rom.h" +#include "usacdec_fac.h" +#include "usacdec_lpc.h" +#include "FDK_tools_rom.h" +#include "fft.h" +#include "mdct.h" +#include "usacdec_acelp.h" +#include "overlapadd.h" + +#include "conceal.h" + +#include "block.h" + +#define SF_PITCH_TRACK 6 +#define SF_GAIN 3 +#define MIN_VAL FL2FXCONST_DBL(0.0f) +#define MAX_VAL (FIXP_DBL) MAXVAL_DBL + +#include "ac_arith_coder.h" + +void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise, + const FIXP_SGL *filt, INT stop, int len) { + INT i, j; + FIXP_DBL tmp; + + for (i = 0; i < stop; i++) { + tmp = fMultDiv2(noise[i], filt[0]); // Filt in Q-1.16 + for (j = 1; j <= len; j++) { + tmp += fMultDiv2((noise[i - j] + noise[i + j]), filt[j]); + } + syn_out[i] = (FIXP_PCM)(IMDCT_SCALE(syn[i] - tmp)); + } +} + +void bass_pf_1sf_delay( + FIXP_DBL *syn, /* (i) : 12.8kHz synthesis to postfilter */ + const INT *T_sf, /* (i) : Pitch period for all subframes (T_sf[16]) */ + FIXP_DBL *pit_gain, + const int frame_length, /* (i) : frame length (should be 768|1024) */ + const INT l_frame, + const INT l_next, /* (i) : look ahead for symmetric filtering */ + FIXP_PCM *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */ + FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR] */ +{ + INT i, sf, i_subfr, T, T2, lg; + + FIXP_DBL tmp, ener, corr, gain; + FIXP_DBL *noise, *noise_in; + FIXP_DBL + noise_buf[L_FILT + (2 * L_SUBFR)]; // L_FILT = 12, L_SUBFR = 64 => 140 + const FIXP_DBL *x, *y; + + { + noise = noise_buf + L_FILT; // L_FILT = 12 delay of upsampling filter + noise_in = noise_buf + L_FILT + L_SUBFR; + /* Input scaling of the BPF memory */ + scaleValues(mem_bpf, (L_FILT + L_SUBFR), 1); + } + + int gain_exp = 17; + + sf = 0; + for (i_subfr = 0; i_subfr < l_frame; i_subfr += L_SUBFR, sf++) { + T = T_sf[sf]; + gain = pit_gain[sf]; + + /* Gain is in Q17.14 */ + /* If gain > 1 set to 1 */ + if (gain > (FIXP_DBL)(1 << 14)) gain = (FIXP_DBL)(1 << 14); + + /* If gain < 0 set to 0 */ + if (gain < (FIXP_DBL)0) gain = (FIXP_DBL)0; + + if (gain > (FIXP_DBL)0) { + /* pitch tracker: test pitch/2 to avoid continuous pitch doubling */ + /* Note: pitch is limited to PIT_MIN (34 = 376Hz) at the encoder */ + T2 = T >> 1; + x = &syn[i_subfr - L_EXTRA]; + y = &syn[i_subfr - T2 - L_EXTRA]; + + ener = (FIXP_DBL)0; + corr = (FIXP_DBL)0; + tmp = (FIXP_DBL)0; + + int headroom_x = getScalefactor(x, L_SUBFR + L_EXTRA); + int headroom_y = getScalefactor(y, L_SUBFR + L_EXTRA); + + int width_shift = 7; + + for (i = 0; i < (L_SUBFR + L_EXTRA); i++) { + ener += fPow2Div2((x[i] << headroom_x)) >> width_shift; + corr += fMultDiv2((x[i] << headroom_x), (y[i] << headroom_y)) >> + width_shift; + tmp += fPow2Div2((y[i] << headroom_y)) >> width_shift; + } + + int exp_ener = ((17 - headroom_x) << 1) + width_shift + 1; + int exp_corr = (17 - headroom_x) + (17 - headroom_y) + width_shift + 1; + int exp_tmp = ((17 - headroom_y) << 1) + width_shift + 1; + + /* Add 0.01 to "ener". Adjust exponents */ + FIXP_DBL point_zero_one = (FIXP_DBL)0x51eb851f; /* In Q-6.37 */ + int diff; + ener = fAddNorm(ener, exp_ener, point_zero_one, -6, &exp_ener); + corr = fAddNorm(corr, exp_corr, point_zero_one, -6, &exp_corr); + tmp = fAddNorm(tmp, exp_tmp, point_zero_one, -6, &exp_tmp); + + /* use T2 if normalized correlation > 0.95 */ + INT s1, s2; + s1 = CntLeadingZeros(ener) - 1; + s2 = CntLeadingZeros(tmp) - 1; + + FIXP_DBL ener_by_tmp = fMultDiv2(ener << s1, tmp << s2); + int ener_by_tmp_exp = (exp_ener - s1) + (exp_tmp - s2) + 1; + + if (ener_by_tmp_exp & 1) { + ener_by_tmp <<= 1; + ener_by_tmp_exp -= 1; + } + + int temp_exp = 0; + + FIXP_DBL temp1 = invSqrtNorm2(ener_by_tmp, &temp_exp); + + int temp1_exp = temp_exp - (ener_by_tmp_exp >> 1); + + FIXP_DBL tmp_result = fMult(corr, temp1); + + int tmp_result_exp = exp_corr + temp1_exp; + + diff = tmp_result_exp - 0; + FIXP_DBL point95 = FL2FXCONST_DBL(0.95f); + if (diff >= 0) { + diff = fMin(diff, 31); + point95 = FL2FXCONST_DBL(0.95f) >> diff; + } else { + diff = fMax(diff, -31); + tmp_result >>= (-diff); + } + + if (tmp_result > point95) T = T2; + + /* prevent that noise calculation below reaches into not defined signal + parts at the end of the synth_buf or in other words restrict the below + used index (i+i_subfr+T) < l_frame + l_next + */ + lg = l_frame + l_next - T - i_subfr; + + if (lg > L_SUBFR) + lg = L_SUBFR; + else if (lg < 0) + lg = 0; + + /* limit gain to avoid problem on burst */ + if (lg > 0) { + FIXP_DBL tmp1; + + /* max(lg) = 64 => scale with 6 bits minus 1 (fPow2Div2) */ + + s1 = getScalefactor(&syn[i_subfr], lg); + s2 = getScalefactor(&syn[i_subfr + T], lg); + INT s = fixMin(s1, s2); + + tmp = (FIXP_DBL)0; + ener = (FIXP_DBL)0; + for (i = 0; i < lg; i++) { + tmp += fPow2Div2(syn[i + i_subfr] << s1) >> (SF_PITCH_TRACK); + ener += fPow2Div2(syn[i + i_subfr + T] << s2) >> (SF_PITCH_TRACK); + } + tmp = tmp >> fMin(DFRACT_BITS - 1, (2 * (s1 - s))); + ener = ener >> fMin(DFRACT_BITS - 1, (2 * (s2 - s))); + + /* error robustness: for the specific case syn[...] == -1.0f for all 64 + samples ener or tmp might overflow and become negative. For all sane + cases we have enough headroom. + */ + if (ener <= (FIXP_DBL)0) { + ener = (FIXP_DBL)1; + } + if (tmp <= (FIXP_DBL)0) { + tmp = (FIXP_DBL)1; + } + FDK_ASSERT(ener > (FIXP_DBL)0); + + /* tmp = sqrt(tmp/ener) */ + int result_e = 0; + tmp1 = fDivNorm(tmp, ener, &result_e); + if (result_e & 1) { + tmp1 >>= 1; + result_e += 1; + } + tmp = sqrtFixp(tmp1); + result_e >>= 1; + + gain_exp = 17; + + diff = result_e - gain_exp; + + FIXP_DBL gain1 = gain; + + if (diff >= 0) { + diff = fMin(diff, 31); + gain1 >>= diff; + } else { + result_e += (-diff); + diff = fMax(diff, -31); + tmp >>= (-diff); + } + + if (tmp < gain1) { + gain = tmp; + gain_exp = result_e; + } + } + + /* calculate noise based on voiced pitch */ + /* fMultDiv2() replaces weighting of gain with 0.5 */ + diff = gain_exp - 17; + if (diff >= 0) { + gain <<= diff; + } else { + gain >>= (-diff); + } + + s1 = CntLeadingZeros(gain) - 1; + s1 -= 16; /* Leading bits for SGL */ + + FIXP_SGL gainSGL = FX_DBL2FX_SGL(gain << 16); + + gainSGL = gainSGL << s1; + + { + for (i = 0; i < lg; i++) { + /* scaled with SF_SYNTH + gain_sf + 1 */ + noise_in[i] = + (fMult(gainSGL, syn[i + i_subfr] - (syn[i + i_subfr - T] >> 1) - + (syn[i + i_subfr + T] >> 1))) >> + s1; + } + + for (i = lg; i < L_SUBFR; i++) { + /* scaled with SF_SYNTH + gain_sf + 1 */ + noise_in[i] = + (fMult(gainSGL, syn[i + i_subfr] - syn[i + i_subfr - T])) >> s1; + } + } + } else { + FDKmemset(noise_in, (FIXP_DBL)0, L_SUBFR * sizeof(FIXP_DBL)); + } + + { + FDKmemcpy(noise_buf, mem_bpf, (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); + + FDKmemcpy(mem_bpf, noise_buf + L_SUBFR, + (L_FILT + L_SUBFR) * sizeof(FIXP_DBL)); + } + + /* substract from voiced speech low-pass filtered noise */ + /* filter coefficients are scaled with factor SF_FILT_LP (1) */ + + { + filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise, + fdk_dec_filt_lp, L_SUBFR, L_FILT); + } + } + + { + + } + + // To be determined (info from Ben) + { + /* Output scaling of the BPF memory */ + scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1); + /* Copy the rest of the signal (after the fac) */ + scaleValuesSaturate((FIXP_PCM *)&synth_out[l_frame], + (FIXP_DBL *)&syn[l_frame - L_SUBFR], + (frame_length - l_frame), MDCT_OUT_HEADROOM); + } + + return; +} + +/* + * Frequency Domain Noise Shaping + */ + +/** + * \brief Adaptive Low Frequencies Deemphasis of spectral coefficients. + * + * Ensure quantization of low frequencies in case where the + * signal dynamic is higher than the LPC noise shaping. + * This is the inverse operation of adap_low_freq_emph(). + * Output gain of all blocks. + * + * \param x pointer to the spectral coefficients, requires 1 bit headroom. + * \param lg length of x. + * \param bUseNewAlfe if set, apply ALFD for fullband lpd. + * \param gainLpc1 pointer to gain based on old input LPC coefficients. + * \param gainLpc2 pointer to gain based on new input LPC coefficients. + * \param alfd_gains pointer to output gains. + * \param s current scale shift factor of x. + */ +#define ALFDPOW2_SCALE 3 +/*static*/ +void CLpd_AdaptLowFreqDeemph(FIXP_DBL x[], int lg, FIXP_DBL alfd_gains[], + INT s) { + { + int i, j, k, i_max; + FIXP_DBL max, fac; + /* Note: This stack array saves temporary accumulation results to be used in + * a second run */ + /* The size should be limited to (1024/4)/8=32 */ + FIXP_DBL tmp_pow2[32]; + + s = s * 2 + ALFDPOW2_SCALE; + + k = 8; + i_max = lg / 4; /* ALFD range = 1600Hz (lg = 6400Hz) */ + + /* find spectral peak */ + max = FL2FX_DBL(0.01f) >> s; + for (i = 0; i < i_max; i += k) { + FIXP_DBL tmp; + + tmp = FIXP_DBL(0); + FIXP_DBL *pX = &x[i]; + + j = 8; + do { + FIXP_DBL x0 = *pX++; + FIXP_DBL x1 = *pX++; + x0 = fPow2Div2(x0); + x1 = fPow2Div2(x1); + tmp = tmp + (x0 >> (ALFDPOW2_SCALE - 1)); + tmp = tmp + (x1 >> (ALFDPOW2_SCALE - 1)); + } while ((j = j - 2) != 0); + tmp = fMax(tmp, (FL2FX_DBL(0.01f) >> s)); + tmp_pow2[i >> 3] = tmp; + if (tmp > max) { + max = tmp; + } + } + + /* deemphasis of all blocks below the peak */ + fac = FL2FX_DBL(0.1f) >> 1; + for (i = 0; i < i_max; i += k) { + FIXP_DBL tmp; + INT shifti; + + tmp = tmp_pow2[i >> 3]; + + /* tmp = (float)sqrt(tmp/max); */ + + /* value of tmp is between 8/2*max^2 and max^2 / 2. */ + /* required shift factor of division can grow up to 27 + (grows exponentially for values toward zero) + thus using normalized division to assure valid result. */ + { + INT sd; + + if (tmp != (FIXP_DBL)0) { + tmp = fDivNorm(max, tmp, &sd); + if (sd & 1) { + sd++; + tmp >>= 1; + } + } else { + tmp = (FIXP_DBL)MAXVAL_DBL; + sd = 0; + } + tmp = invSqrtNorm2(tmp, &shifti); + tmp = scaleValue(tmp, shifti - 1 - (sd / 2)); + } + if (tmp > fac) { + fac = tmp; + } + FIXP_DBL *pX = &x[i]; + + j = 8; + do { + FIXP_DBL x0 = pX[0]; + FIXP_DBL x1 = pX[1]; + x0 = fMultDiv2(x0, fac); + x1 = fMultDiv2(x1, fac); + x0 = x0 << 2; + x1 = x1 << 2; + *pX++ = x0; + *pX++ = x1; + + } while ((j = j - 2) != 0); + /* Store gains for FAC */ + *alfd_gains++ = fac; + } + } +} + +/** + * \brief Interpolated Noise Shaping for mdct coefficients. + * This algorithm shapes temporally the spectral noise between + * the two spectral noise represention (FDNS_NPTS of resolution). + * The noise is shaped monotonically between the two points + * using a curved shape to favor the lower gain in mid-frame. + * ODFT and amplitud calculation are applied to the 2 LPC coefficients first. + * + * \param r pointer to spectrum data. + * \param rms RMS of output spectrum. + * \param lg length of r. + * \param A1 pointer to old input LPC coefficients of length M_LP_FILTER_ORDER + * scaled by SF_A_COEFFS. + * \param A2 pointer to new input LPC coefficients of length M_LP_FILTER_ORDER + * scaled by SF_A_COEFFS. + * \param bLpc2Mdct flags control lpc2mdct conversion and noise shaping. + * \param gainLpc1 pointer to gain based on old input LPC coefficients. + * \param gainLpc2 pointer to gain based on new input LPC coefficients. + * \param gLpc_e pointer to exponent of gainLpc1 and gainLpc2. + */ +/* static */ +#define NSHAPE_SCALE (4) + +#define LPC2MDCT_CALC (1) +#define LPC2MDCT_GAIN_LOAD (2) +#define LPC2MDCT_GAIN_SAVE (4) +#define LPC2MDCT_APPLY_NSHAPE (8) + +void lpc2mdctAndNoiseShaping(FIXP_DBL *r, SHORT *pScale, const INT lg, + const INT fdns_npts, const FIXP_LPC *A1, + const INT A1_exp, const FIXP_LPC *A2, + const INT A2_exp) { + FIXP_DBL *tmp2 = NULL; + FIXP_DBL rr_minus_one; + int i, k, s, step; + + C_AALLOC_SCRATCH_START(tmp1, FIXP_DBL, FDNS_NPTS * 8) + + { + tmp2 = tmp1 + fdns_npts * 4; + + /* lpc2mdct() */ + + /* ODFT. E_LPC_a_weight() for A1 and A2 vectors is included into the loop + * below. */ + FIXP_DBL f = FL2FXCONST_DBL(0.92f); + + const FIXP_STP *SinTab; + int k_step; + /* needed values: sin(phi), cos(phi); phi = i*PI/(2*fdns_npts), i = 0 ... + * M_LP_FILTER_ORDER */ + switch (fdns_npts) { + case 64: + SinTab = SineTable512; + k_step = (512 / 64); + FDK_ASSERT(512 >= 64); + break; + case 48: + SinTab = SineTable384; + k_step = 384 / 48; + FDK_ASSERT(384 >= 48); + break; + default: + FDK_ASSERT(0); + return; + } + + for (i = 0, k = k_step; i < M_LP_FILTER_ORDER; i++, k += k_step) { + FIXP_STP cs = SinTab[k]; + FIXP_DBL wA1, wA2; + + wA1 = fMult(A1[i], f); + wA2 = fMult(A2[i], f); + + /* r[i] = A[i]*cos() */ + tmp1[2 + i * 2] = fMult(wA1, cs.v.re); + tmp2[2 + i * 2] = fMult(wA2, cs.v.re); + /* i[i] = A[i]*sin() */ + tmp1[3 + i * 2] = -fMult(wA1, cs.v.im); + tmp2[3 + i * 2] = -fMult(wA2, cs.v.im); + + f = fMult(f, FL2FXCONST_DBL(0.92f)); + } + + /* Guarantee at least 2 bits of headroom for the FFT */ + /* "3" stands for 1.0 with 2 bits of headroom; (A1_exp + 2) guarantess 2 + * bits of headroom if A1_exp > 1 */ + int A1_exp_fix = fMax(3, A1_exp + 2); + int A2_exp_fix = fMax(3, A2_exp + 2); + + /* Set 1.0 in the proper format */ + tmp1[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A1_exp_fix); + tmp2[0] = (FIXP_DBL)(INT)((ULONG)0x80000000 >> A2_exp_fix); + + tmp1[1] = tmp2[1] = (FIXP_DBL)0; + + /* Clear the resto of the array */ + FDKmemclear( + tmp1 + 2 * (M_LP_FILTER_ORDER + 1), + 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); + FDKmemclear( + tmp2 + 2 * (M_LP_FILTER_ORDER + 1), + 2 * (fdns_npts * 2 - (M_LP_FILTER_ORDER + 1)) * sizeof(FIXP_DBL)); + + /* Guarantee 2 bits of headroom for FFT */ + scaleValues(&tmp1[2], (2 * M_LP_FILTER_ORDER), (A1_exp - A1_exp_fix)); + scaleValues(&tmp2[2], (2 * M_LP_FILTER_ORDER), (A2_exp - A2_exp_fix)); + + INT s2; + s = A1_exp_fix; + s2 = A2_exp_fix; + + fft(2 * fdns_npts, tmp1, &s); + fft(2 * fdns_npts, tmp2, &s2); + + /* Adjust the exponents of both fft outputs if necessary*/ + if (s > s2) { + scaleValues(tmp2, 2 * fdns_npts, s2 - s); + s2 = s; + } else if (s < s2) { + scaleValues(tmp1, 2 * fdns_npts, s - s2); + s = s2; + } + + FDK_ASSERT(s == s2); + } + + /* Get amplitude and apply gains */ + step = lg / fdns_npts; + rr_minus_one = (FIXP_DBL)0; + + for (k = 0; k < fdns_npts; k++) { + FIXP_DBL g1, g2, inv_g1_g2, a, b; + INT inv_g1_g2_e; + int g_e, shift; + + { + FIXP_DBL real, imag; + int si1, si2, sInput; + + real = tmp1[k * 2]; + imag = tmp1[k * 2 + 1]; + sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); + real <<= sInput; + imag <<= sInput; + /* g1_e = si1 - 2*s/2 */ + g1 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si1); + si1 += sInput; + + real = tmp2[k * 2]; + imag = tmp2[k * 2 + 1]; + sInput = fMax(fMin(fNorm(real), fNorm(imag)) - 1, 0); + real <<= sInput; + imag <<= sInput; + /* g2_e = si2 - 2*s/2 */ + g2 = invSqrtNorm2(fPow2(real) + fPow2(imag), &si2); + si2 += sInput; + + /* Pick a common scale factor for g1 and g2 */ + if (si1 > si2) { + g2 >>= si1 - si2; + g_e = si1 - s; + } else { + g1 >>= si2 - si1; + g_e = si2 - s; + } + } + + /* end of lpc2mdct() */ + + FDK_ASSERT(g1 >= (FIXP_DBL)0); + FDK_ASSERT(g2 >= (FIXP_DBL)0); + + /* mdct_IntNoiseShaping() */ + { + /* inv_g1_g2 * 2^inv_g1_g2_e = 1/(g1+g2) */ + inv_g1_g2 = (g1 >> 1) + (g2 >> 1); + if (inv_g1_g2 != (FIXP_DBL)0) { + inv_g1_g2 = fDivNorm(FL2FXCONST_DBL(0.5f), inv_g1_g2, &inv_g1_g2_e); + inv_g1_g2_e = inv_g1_g2_e - g_e; + } else { + inv_g1_g2 = (FIXP_DBL)MAXVAL_DBL; + inv_g1_g2_e = 0; + } + + if (g_e < 0) { + /* a_e = g_e + inv_g1_g2_e + 1 */ + a = scaleValue(fMult(fMult(g1, g2), inv_g1_g2), g_e); + /* b_e = g_e + inv_g1_g2_e */ + b = fMult(g2 - g1, inv_g1_g2); + shift = g_e + inv_g1_g2_e + 1 - NSHAPE_SCALE; + } else { + /* a_e = (g_e+g_e) + inv_g1_g2_e + 1 */ + a = fMult(fMult(g1, g2), inv_g1_g2); + /* b_e = (g_e+g_e) + inv_g1_g2_e */ + b = scaleValue(fMult(g2 - g1, inv_g1_g2), -g_e); + shift = (g_e + g_e) + inv_g1_g2_e + 1 - NSHAPE_SCALE; + } + + for (i = k * step; i < (k + 1) * step; i++) { + FIXP_DBL tmp; + + /* rr[i] = 2*a*r[i] + b*rr[i-1] */ + tmp = fMult(a, r[i]); + tmp += scaleValue(fMultDiv2(b, rr_minus_one), NSHAPE_SCALE); + tmp = scaleValueSaturate(tmp, shift); + rr_minus_one = tmp; + r[i] = tmp; + } + } + } + + /* end of mdct_IntNoiseShaping() */ + { *pScale += NSHAPE_SCALE; } + + C_AALLOC_SCRATCH_END(tmp1, FIXP_DBL, FDNS_NPTS * 8) +} + +/** + * \brief Calculates the energy. + * \param r pointer to spectrum. + * \param rs scale factor of spectrum r. + * \param lg frame length in audio samples. + * \param rms_e pointer to exponent of energy value. + * \return mantissa of energy value. + */ +static FIXP_DBL calcEnergy(const FIXP_DBL *r, const SHORT rs, const INT lg, + INT *rms_e) { + int headroom = getScalefactor(r, lg); + + FIXP_DBL rms_m = 0; + + /* Calculate number of growth bits due to addition */ + INT shift = (INT)(fNormz((FIXP_DBL)lg)); + shift = 31 - shift; + + /* Generate 1e-2 in Q-6.37 */ + const FIXP_DBL value0_01 = 0x51eb851e; + const INT value0_01_exp = -6; + + /* Find the exponent of the resulting energy value */ + *rms_e = ((rs - headroom) << 1) + shift + 1; + + INT delta = *rms_e - value0_01_exp; + if (delta > 0) { + /* Limit shift_to 31*/ + delta = fMin(31, delta); + rms_m = value0_01 >> delta; + } else { + rms_m = value0_01; + *rms_e = value0_01_exp; + shift = shift - delta; + /* Limit shift_to 31*/ + shift = fMin(31, shift); + } + + for (int i = 0; i < lg; i++) { + rms_m += fPow2Div2(r[i] << headroom) >> shift; + } + + return rms_m; +} + +/** + * \brief TCX gain calculation. + * \param pAacDecoderChannelInfo channel context data. + * \param r output spectrum. + * \param rms_e pointer to mantissa of energy value. + * \param rms_e pointer to exponent of energy value. + * \param frame the frame index of the LPD super frame. + * \param lg the frame length in audio samples. + * \param gain_m pointer to mantissa of TCX gain. + * \param gain_e pointer to exponent of TCX gain. + * \param elFlags element specific parser guidance flags. + * \param lg_fb the fullband frame length in audio samples. + * \param IGF_bgn the IGF start index. + */ +static void calcTCXGain(CAacDecoderChannelInfo *pAacDecoderChannelInfo, + FIXP_DBL *r, FIXP_DBL rms_m, INT rms_e, const INT frame, + const INT lg) { + if ((rms_m != (FIXP_DBL)0)) { + FIXP_DBL tcx_gain_m; + INT tcx_gain_e; + + CLpd_DecodeGain(&tcx_gain_m, &tcx_gain_e, + pAacDecoderChannelInfo->pDynData->specificTo.usac + .tcx_global_gain[frame]); + + /* rms * 2^rms_e = lg/sqrt(sum(spec^2)) */ + if (rms_e & 1) { + rms_m >>= 1; + rms_e++; + } + + { + FIXP_DBL fx_lg; + INT fx_lg_e, s; + INT inv_e; + + /* lg = fx_lg * 2^fx_lg_e */ + s = fNorm((FIXP_DBL)lg); + fx_lg = (FIXP_DBL)lg << s; + fx_lg_e = DFRACT_BITS - 1 - s; + /* 1/sqrt(rms) */ + rms_m = invSqrtNorm2(rms_m, &inv_e); + rms_m = fMult(rms_m, fx_lg); + rms_e = inv_e - (rms_e >> 1) + fx_lg_e; + } + + { + int s = fNorm(tcx_gain_m); + tcx_gain_m = tcx_gain_m << s; + tcx_gain_e -= s; + } + + tcx_gain_m = fMultDiv2(tcx_gain_m, rms_m); + tcx_gain_e = tcx_gain_e + rms_e; + + /* global_gain * 2^(global_gain_e+rms_e) = (10^(global_gain/28)) * rms * + * 2^rms_e */ + { + { tcx_gain_e += 1; } + } + + pAacDecoderChannelInfo->data.usac.tcx_gain[frame] = tcx_gain_m; + pAacDecoderChannelInfo->data.usac.tcx_gain_e[frame] = tcx_gain_e; + + pAacDecoderChannelInfo->specScale[frame] += tcx_gain_e; + } +} + +/** + * \brief FDNS decoding. + * \param pAacDecoderChannelInfo channel context data. + * \param pAacDecoderStaticChannelInfo channel context static data. + * \param r output spectrum. + * \param lg the frame length in audio samples. + * \param frame the frame index of the LPD super frame. + * \param pScale pointer to current scale shift factor of r[]. + * \param A1 old input LPC coefficients of length M_LP_FILTER_ORDER. + * \param A2 new input LPC coefficients of length M_LP_FILTER_ORDER. + * \param pAlfdGains pointer for ALFD gains output scaled by 1. + * \param fdns_npts number of lines (FDNS_NPTS). + * \param inf_mask pointer to noise mask. + * \param IGF_win_mode IGF window mode (LONG, SHORT, TCX10, TCX20). + * \param frameType (IGF_FRAME_DIVISION_AAC_OR_TCX_LONG or + * IGF_FRAME_DIVISION_TCX_SHORT_1). + * \param elFlags element specific parser guidance flags. + * \param lg_fb the fullband frame length in audio samples. + * \param IGF_bgn the IGF start index. + * \param rms_m mantisse of energy. + * \param rms_e exponent of energy. + */ +/* static */ +void CLpd_FdnsDecode(CAacDecoderChannelInfo *pAacDecoderChannelInfo, + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + FIXP_DBL r[], const INT lg, const INT frame, SHORT *pScale, + const FIXP_LPC A1[M_LP_FILTER_ORDER], const INT A1_exp, + const FIXP_LPC A2[M_LP_FILTER_ORDER], const INT A2_exp, + FIXP_DBL pAlfdGains[LFAC / 4], const INT fdns_npts) { + /* Weight LPC coefficients using Rm values */ + CLpd_AdaptLowFreqDeemph(r, lg, pAlfdGains, *pScale); + + FIXP_DBL rms_m = (FIXP_DBL)0; + INT rms_e = 0; + { + /* Calculate Energy */ + rms_m = calcEnergy(r, *pScale, lg, &rms_e); + } + + calcTCXGain(pAacDecoderChannelInfo, r, rms_m, rms_e, frame, lg); + + /* Apply ODFT and Noise Shaping. LP coefficient (A1, A2) weighting is done + * inside on the fly. */ + + lpc2mdctAndNoiseShaping(r, pScale, lg, fdns_npts, A1, A1_exp, A2, A2_exp); +} + +/** + * find pitch for TCX20 (time domain) concealment. + */ +static int find_mpitch(FIXP_DBL xri[], int lg) { + FIXP_DBL max, pitch; + INT pitch_e; + int i, n; + + max = (FIXP_DBL)0; + n = 2; + + /* find maximum below 400Hz */ + for (i = 2; i < (lg >> 4); i += 2) { + FIXP_DBL tmp = fPow2Div2(xri[i]) + fPow2Div2(xri[i + 1]); + if (tmp > max) { + max = tmp; + n = i; + } + } + + // pitch = ((float)lg<<1)/(float)n; + pitch = fDivNorm((FIXP_DBL)lg << 1, (FIXP_DBL)n, &pitch_e); + pitch >>= fixMax(0, DFRACT_BITS - 1 - pitch_e - 16); + + /* find pitch multiple under 20ms */ + if (pitch >= (FIXP_DBL)((256 << 16) - 1)) { /*231.0f*/ + n = 256; + } else { + FIXP_DBL mpitch = pitch; + while (mpitch < (FIXP_DBL)(255 << 16)) { + mpitch += pitch; + } + n = (int)(mpitch - pitch) >> 16; + } + + return (n); +} + +/** + * number of spectral coefficients / time domain samples using frame mode as + * index. + */ +static const int lg_table_ccfl[2][4] = { + {256, 256, 512, 1024}, /* coreCoderFrameLength = 1024 */ + {192, 192, 384, 768} /* coreCoderFrameLength = 768 */ +}; + +/** + * \brief Decode and render one MDCT-TCX frame. + * \param pAacDecoderChannelInfo channel context data. + * \param lg the frame length in audio samples. + * \param frame the frame index of the LPD super frame. + */ +static void CLpd_TcxDecode( + CAacDecoderChannelInfo *pAacDecoderChannelInfo, + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags, + int mod, int last_mod, int frame, int frameOk) { + FIXP_DBL *pAlfd_gains = pAacDecoderStaticChannelInfo->last_alfd_gains; + ULONG *pSeed = &pAacDecoderStaticChannelInfo->nfRandomSeed; + int lg = (pAacDecoderChannelInfo->granuleLength == 128) + ? lg_table_ccfl[0][mod + 0] + : lg_table_ccfl[1][mod + 0]; + int next_frame = frame + (1 << (mod - 1)); + int isFullBandLpd = 0; + + /* Obtain r[] vector by combining the quant[] and noise[] vectors */ + { + FIXP_DBL noise_level; + FIXP_DBL *coeffs = + SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, + pAacDecoderChannelInfo->granuleLength, isFullBandLpd); + int scale = pAacDecoderChannelInfo->specScale[frame]; + int i, nfBgn, nfEnd; + UCHAR tcx_noise_factor = pAacDecoderChannelInfo->pDynData->specificTo.usac + .tcx_noise_factor[frame]; + + /* find pitch for bfi case */ + pAacDecoderStaticChannelInfo->last_tcx_pitch = find_mpitch(coeffs, lg); + + if (frameOk) { + /* store for concealment */ + pAacDecoderStaticChannelInfo->last_tcx_noise_factor = tcx_noise_factor; + } else { + /* restore last frames value */ + tcx_noise_factor = pAacDecoderStaticChannelInfo->last_tcx_noise_factor; + } + + noise_level = + (FIXP_DBL)((LONG)FL2FXCONST_DBL(0.0625f) * (8 - tcx_noise_factor)); + noise_level = scaleValue(noise_level, -scale); + + const FIXP_DBL neg_noise_level = -noise_level; + + { + nfBgn = lg / 6; + nfEnd = lg; + } + + for (i = nfBgn; i < nfEnd - 7; i += 8) { + LONG tmp; + + /* Fill all 8 consecutive zero coeffs with noise */ + tmp = coeffs[i + 0] | coeffs[i + 1] | coeffs[i + 2] | coeffs[i + 3] | + coeffs[i + 4] | coeffs[i + 5] | coeffs[i + 6] | coeffs[i + 7]; + + if (tmp == 0) { + for (int k = i; k < i + 8; k++) { + UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) + : (coeffs[k] = noise_level); + } + } + } + if ((nfEnd - i) > + 0) { /* noise filling for last "band" with less than 8 bins */ + LONG tmp = (LONG)coeffs[i]; + int k; + + FDK_ASSERT((nfEnd - i) < 8); + for (k = 1; k < (nfEnd - i); k++) { + tmp |= (LONG)coeffs[i + k]; + } + if (tmp == 0) { + for (k = i; k < nfEnd; k++) { + UsacRandomSign(pSeed) ? (coeffs[k] = neg_noise_level) + : (coeffs[k] = noise_level); + } + } + } + } + + { + /* Convert LPC to LP domain */ + if (last_mod == 0) { + /* Note: The case where last_mod == 255 is handled by other means + * in CLpdChannelStream_Read() */ + E_LPC_f_lsp_a_conversion( + pAacDecoderChannelInfo->data.usac.lsp_coeff[frame], + pAacDecoderChannelInfo->data.usac.lp_coeff[frame], + &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame]); + } + + E_LPC_f_lsp_a_conversion( + pAacDecoderChannelInfo->data.usac.lsp_coeff[next_frame], + pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], + &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame]); + + /* FDNS decoding */ + CLpd_FdnsDecode( + pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, + SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, + pAacDecoderChannelInfo->granuleLength, isFullBandLpd), + lg, frame, pAacDecoderChannelInfo->specScale + frame, + pAacDecoderChannelInfo->data.usac.lp_coeff[frame], + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[frame], + pAacDecoderChannelInfo->data.usac.lp_coeff[next_frame], + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[next_frame], pAlfd_gains, + pAacDecoderChannelInfo->granuleLength / 2 /* == FDNS_NPTS(ccfl) */ + ); + } +} + +/** + * \brief Read the tcx_coding bitstream part + * \param hBs bitstream handle to read from. + * \param pAacDecoderChannelInfo channel context info to store data into. + * \param lg the frame length in audio samples. + * \param first_tcx_flag flag indicating that this is the first TCX frame. + * \param frame the frame index of the LPD super frame. + */ +static AAC_DECODER_ERROR CLpd_TCX_Read( + HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, int lg, + int first_tcx_flag, int frame, UINT flags) { + AAC_DECODER_ERROR errorAAC = AAC_DEC_OK; + ARITH_CODING_ERROR error = ARITH_CODER_OK; + FIXP_DBL *pSpec; + int arith_reset_flag = 0; + int isFullBandLpd = 0; + + pSpec = SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, frame, + pAacDecoderChannelInfo->granuleLength, isFullBandLpd); + + /* TCX noise level */ + { + pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_noise_factor[frame] = + FDKreadBits(hBs, 3); + } + /* TCX global gain */ + pAacDecoderChannelInfo->pDynData->specificTo.usac.tcx_global_gain[frame] = + FDKreadBits(hBs, 7); + + /* Arithmetic coded residual/spectrum */ + if (first_tcx_flag) { + if (flags & AC_INDEP) { + arith_reset_flag = 1; + } else { + arith_reset_flag = FDKreadBits(hBs, 1); + } + } + + /* CArco_DecodeArithData() output scale of "pSpec" is DFRACT_BITS-1 */ + error = CArco_DecodeArithData(pAacDecoderStaticChannelInfo->hArCo, hBs, pSpec, + lg, lg, arith_reset_flag); + + /* Rescale residual/spectrum */ + { + int scale = getScalefactor(pSpec, lg) - 2; /* Leave 2 bits headroom */ + + /* Exponent of CArco_DecodeArithData() output is DFRACT_BITS; integer + * values. */ + scaleValues(pSpec, lg, scale); + scale = DFRACT_BITS - 1 - scale; + + pAacDecoderChannelInfo->specScale[frame] = scale; + } + + if (error == ARITH_CODER_ERROR) errorAAC = AAC_DEC_UNKNOWN; + + return errorAAC; +} + +/** + * \brief translate lpd_mode into the mod[] array which describes the mode of + * each each LPD frame + * \param mod[] the array that will be filled with the mode indexes of the + * inidividual frames. + * \param lpd_mode the lpd_mode field read from the lpd_channel_stream + */ +static AAC_DECODER_ERROR CLpd_ReadAndMapLpdModeToModArray( + UCHAR mod[4], HANDLE_FDK_BITSTREAM hBs, UINT elFlags) { + int lpd_mode; + + { + lpd_mode = FDKreadBits(hBs, 5); + + if (lpd_mode > 25 || lpd_mode < 0) { + return AAC_DEC_PARSE_ERROR; + } + + switch (lpd_mode) { + case 25: + /* 1 80MS frame */ + mod[0] = mod[1] = mod[2] = mod[3] = 3; + break; + case 24: + /* 2 40MS frames */ + mod[0] = mod[1] = mod[2] = mod[3] = 2; + break; + default: + switch (lpd_mode >> 2) { + case 4: + /* lpd_mode 19 - 16 => 1 40MS and 2 20MS frames */ + mod[0] = mod[1] = 2; + mod[2] = (lpd_mode & 1) ? 1 : 0; + mod[3] = (lpd_mode & 2) ? 1 : 0; + break; + case 5: + /* lpd_mode 23 - 20 => 2 20MS and 1 40MS frames */ + mod[2] = mod[3] = 2; + mod[0] = (lpd_mode & 1) ? 1 : 0; + mod[1] = (lpd_mode & 2) ? 1 : 0; + break; + default: + /* lpd_mode < 16 => 4 20MS frames */ + mod[0] = (lpd_mode & 1) ? 1 : 0; + mod[1] = (lpd_mode & 2) ? 1 : 0; + mod[2] = (lpd_mode & 4) ? 1 : 0; + mod[3] = (lpd_mode & 8) ? 1 : 0; + break; + } + break; + } + } + return AAC_DEC_OK; +} + +static void CLpd_Reset( + CAacDecoderChannelInfo *pAacDecoderChannelInfo, + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + int keep_past_signal) { + int i; + + /* Reset TCX / ACELP common memory */ + if (!keep_past_signal) { + FDKmemclear(pAacDecoderStaticChannelInfo->old_synth, + sizeof(pAacDecoderStaticChannelInfo->old_synth)); + } + + /* Initialize the LSFs */ + for (i = 0; i < M_LP_FILTER_ORDER; i++) { + pAacDecoderStaticChannelInfo->lpc4_lsf[i] = fdk_dec_lsf_init[i]; + } + + /* Reset memory needed by bass post-filter */ + FDKmemclear(pAacDecoderStaticChannelInfo->mem_bpf, + sizeof(pAacDecoderStaticChannelInfo->mem_bpf)); + + pAacDecoderStaticChannelInfo->old_bpf_control_info = 0; + for (i = 0; i < SYN_SFD; i++) { + pAacDecoderStaticChannelInfo->old_T_pf[i] = 64; + pAacDecoderStaticChannelInfo->old_gain_pf[i] = (FIXP_DBL)0; + } + + /* Reset ACELP memory */ + CLpd_AcelpReset(&pAacDecoderStaticChannelInfo->acelp); + + pAacDecoderStaticChannelInfo->last_lpc_lost = 0; /* prev_lpc_lost */ + pAacDecoderStaticChannelInfo->last_tcx_pitch = L_DIV; /* pitch_tcx */ + pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; /* nbLostCmpt */ +} + +/* + * Externally visible functions + */ + +AAC_DECODER_ERROR CLpdChannelStream_Read( + HANDLE_FDK_BITSTREAM hBs, CAacDecoderChannelInfo *pAacDecoderChannelInfo, + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + const SamplingRateInfo *pSamplingRateInfo, UINT flags) { + AAC_DECODER_ERROR error = AAC_DEC_OK; + int first_tcx_flag; + int k, nbDiv, fFacDataPresent, first_lpd_flag, acelp_core_mode, + facGetMemState = 0; + UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; + int lpd_mode_last, prev_frame_was_lpd; + USAC_COREMODE core_mode_last; + const int lg_table_offset = 0; + const int *lg_table = (pAacDecoderChannelInfo->granuleLength == 128) + ? &lg_table_ccfl[0][lg_table_offset] + : &lg_table_ccfl[1][lg_table_offset]; + int last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost; + + int last_frame_ok = CConcealment_GetLastFrameOk( + &pAacDecoderStaticChannelInfo->concealmentInfo, 1); + + INT i_offset; + UINT samplingRate; + + samplingRate = pSamplingRateInfo->samplingRate; + + i_offset = + (INT)(samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM - + (INT)PIT_MIN_12k8; + + if (pSamplingRateInfo->samplingRate > + FAC_FSCALE_MAX /* maximum allowed core sampling frequency */) { + error = AAC_DEC_PARSE_ERROR; + goto bail; + } + + acelp_core_mode = FDKreadBits(hBs, 3); + + /* lpd_mode */ + error = CLpd_ReadAndMapLpdModeToModArray(mod, hBs, 0); + if (error != AAC_DEC_OK) { + goto bail; + } + + /* bpf_control_info */ + pAacDecoderChannelInfo->data.usac.bpf_control_info = FDKreadBit(hBs); + + /* last_core_mode */ + prev_frame_was_lpd = FDKreadBit(hBs); + /* fac_data_present */ + fFacDataPresent = FDKreadBit(hBs); + + /* Set valid values from + * pAacDecoderStaticChannelInfo->{last_core_mode,last_lpd_mode} */ + pAacDecoderChannelInfo->data.usac.core_mode_last = + pAacDecoderStaticChannelInfo->last_core_mode; + lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last = + pAacDecoderStaticChannelInfo->last_lpd_mode; + + if (prev_frame_was_lpd == 0) { + /* Last frame was FD */ + pAacDecoderChannelInfo->data.usac.core_mode_last = FD_LONG; + pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; + } else { + /* Last frame was LPD */ + pAacDecoderChannelInfo->data.usac.core_mode_last = LPD; + if (((mod[0] == 0) && fFacDataPresent) || + ((mod[0] != 0) && !fFacDataPresent)) { + /* Currend mod is ACELP, fac data present -> TCX, current mod TCX, no fac + * data -> TCX */ + if (lpd_mode_last == 0) { + /* Bit stream interruption detected. Assume last TCX mode as TCX20. */ + pAacDecoderChannelInfo->data.usac.lpd_mode_last = 1; + } + /* Else assume that remembered TCX mode is correct. */ + } else { + pAacDecoderChannelInfo->data.usac.lpd_mode_last = 0; + } + } + + first_lpd_flag = (pAacDecoderChannelInfo->data.usac.core_mode_last != + LPD); /* Depends on bitstream configuration */ + first_tcx_flag = 1; + + if (pAacDecoderStaticChannelInfo->last_core_mode != + LPD) { /* ATTENTION: Reset depends on what we rendered before! */ + CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, 0); + + if (!last_frame_ok) { + /* If last rendered frame was not LPD and first lpd flag is not set, this + * must be an error - set last_lpc_lost flag */ + last_lpc_lost |= (first_lpd_flag) ? 0 : 1; + } + } + + core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last; + lpd_mode_last = pAacDecoderChannelInfo->data.usac.lpd_mode_last; + + nbDiv = NB_DIV; + + /* k is the frame index. If a frame is of size 40MS or 80MS, + this frame index is incremented 2 or 4 instead of 1 respectively. */ + + k = 0; + while (k < nbDiv) { + /* Reset FAC data pointers in order to avoid applying old random FAC data. + */ + pAacDecoderChannelInfo->data.usac.fac_data[k] = NULL; + + if ((k == 0 && core_mode_last == LPD && fFacDataPresent) || + (lpd_mode_last == 0 && mod[k] > 0) || + ((lpd_mode_last != 255) && lpd_mode_last > 0 && mod[k] == 0)) { + int err; + + /* Assign FAC memory */ + pAacDecoderChannelInfo->data.usac.fac_data[k] = + CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); + + /* FAC for (ACELP -> TCX) or (TCX -> ACELP) */ + err = CLpd_FAC_Read( + hBs, pAacDecoderChannelInfo->data.usac.fac_data[k], + pAacDecoderChannelInfo->data.usac.fac_data_e, + pAacDecoderChannelInfo->granuleLength, /* == fac_length */ + 0, k); + if (err != 0) { + error = AAC_DEC_PARSE_ERROR; + goto bail; + } + } + + if (mod[k] == 0) /* acelp-mode */ + { + int err; + err = CLpd_AcelpRead( + hBs, &pAacDecoderChannelInfo->data.usac.acelp[k], acelp_core_mode, + pAacDecoderChannelInfo->granuleLength * 8 /* coreCoderFrameLength */, + i_offset); + if (err != 0) { + error = AAC_DEC_PARSE_ERROR; + goto bail; + } + + lpd_mode_last = 0; + k++; + } else /* mode != 0 => TCX */ + { + error = CLpd_TCX_Read(hBs, pAacDecoderChannelInfo, + pAacDecoderStaticChannelInfo, lg_table[mod[k]], + first_tcx_flag, k, flags); + + lpd_mode_last = mod[k]; + first_tcx_flag = 0; + k += 1 << (mod[k] - 1); + } + if (error != AAC_DEC_OK) { + error = AAC_DEC_PARSE_ERROR; + goto bail; + } + } + + { + int err; + + /* Read LPC coefficients */ + err = CLpc_Read( + hBs, pAacDecoderChannelInfo->data.usac.lsp_coeff, + pAacDecoderStaticChannelInfo->lpc4_lsf, + pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, + pAacDecoderChannelInfo->data.usac.aStability, mod, first_lpd_flag, + /* if last lpc4 is available from concealment do not extrapolate lpc0 + from lpc2 */ + (mod[0] & 0x3) ? 0 + : (last_lpc_lost && + pAacDecoderStaticChannelInfo->last_core_mode != LPD), + last_frame_ok); + if (err != 0) { + error = AAC_DEC_PARSE_ERROR; + goto bail; + } + } + + /* adjust old lsp[] following to a bad frame (to avoid overshoot) (ref: + * dec_LPD.c) */ + if (last_lpc_lost && !last_frame_ok) { + int k_next; + k = 0; + while (k < nbDiv) { + int i; + k_next = k + (((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1))); + FIXP_LPC *lsp_old = pAacDecoderChannelInfo->data.usac.lsp_coeff[k]; + FIXP_LPC *lsp_new = pAacDecoderChannelInfo->data.usac.lsp_coeff[k_next]; + + for (i = 0; i < M_LP_FILTER_ORDER; i++) { + if (lsp_new[i] < lsp_old[i]) { + lsp_old[i] = lsp_new[i]; + } + } + k = k_next; + } + } + + if (!CConcealment_GetLastFrameOk( + &pAacDecoderStaticChannelInfo->concealmentInfo, 1)) { + E_LPC_f_lsp_a_conversion( + pAacDecoderChannelInfo->data.usac.lsp_coeff[0], + pAacDecoderChannelInfo->data.usac.lp_coeff[0], + &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]); + } else if (pAacDecoderStaticChannelInfo->last_lpd_mode != 0) { + if (pAacDecoderStaticChannelInfo->last_lpd_mode == 255) { + /* We need it for TCX decoding or ACELP excitation update */ + E_LPC_f_lsp_a_conversion( + pAacDecoderChannelInfo->data.usac.lsp_coeff[0], + pAacDecoderChannelInfo->data.usac.lp_coeff[0], + &pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]); + } else { /* last_lpd_mode was TCX */ + /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid + * converting LSP coefficients again). */ + FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], + pAacDecoderStaticChannelInfo->lp_coeff_old[0], + M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = + pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; + } + } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ + + if (fFacDataPresent && (core_mode_last != LPD)) { + int prev_frame_was_short; + + prev_frame_was_short = FDKreadBit(hBs); + + if (prev_frame_was_short) { + core_mode_last = pAacDecoderChannelInfo->data.usac.core_mode_last = + FD_SHORT; + pAacDecoderChannelInfo->data.usac.lpd_mode_last = 255; + + if ((pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT) && + CConcealment_GetLastFrameOk( + &pAacDecoderStaticChannelInfo->concealmentInfo, 1)) { + /* USAC Conformance document: + short_fac_flag shall be encoded with a value of 1 if the + window_sequence of the previous frame was 2 (EIGHT_SHORT_SEQUENCE). + Otherwise short_fac_flag shall be encoded with a + value of 0. */ + error = AAC_DEC_PARSE_ERROR; + goto bail; + } + } + + /* Assign memory */ + pAacDecoderChannelInfo->data.usac.fac_data[0] = + CLpd_FAC_GetMemory(pAacDecoderChannelInfo, mod, &facGetMemState); + + { + int err; + + /* FAC for FD -> ACELP */ + err = CLpd_FAC_Read( + hBs, pAacDecoderChannelInfo->data.usac.fac_data[0], + pAacDecoderChannelInfo->data.usac.fac_data_e, + CLpd_FAC_getLength(core_mode_last != FD_SHORT, + pAacDecoderChannelInfo->granuleLength), + 1, 0); + if (err != 0) { + error = AAC_DEC_PARSE_ERROR; + goto bail; + } + } + } + +bail: + if (error == AAC_DEC_OK) { + /* check consitency of last core/lpd mode values */ + if ((pAacDecoderChannelInfo->data.usac.core_mode_last != + pAacDecoderStaticChannelInfo->last_core_mode) && + (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { + /* Something got wrong! */ + /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ + } else if ((pAacDecoderChannelInfo->data.usac.core_mode_last == LPD) && + (pAacDecoderChannelInfo->data.usac.lpd_mode_last != + pAacDecoderStaticChannelInfo->last_lpd_mode) && + (pAacDecoderStaticChannelInfo->last_lpc_lost == 0)) { + /* Something got wrong! */ + /* error = AAC_DEC_PARSE_ERROR; */ /* Throwing errors does not help */ + } + } + + return error; +} + +void CLpdChannelStream_Decode( + CAacDecoderChannelInfo *pAacDecoderChannelInfo, + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, UINT flags) { + UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; + int k; + UCHAR last_lpd_mode; + int nbDiv = NB_DIV; + + /* k is the frame index. If a frame is of size 40MS or 80MS, + this frame index is incremented 2 or 4 instead of 1 respectively. */ + k = 0; + last_lpd_mode = + pAacDecoderChannelInfo->data.usac + .lpd_mode_last; /* could be different to what has been rendered */ + while (k < nbDiv) { + if (mod[k] == 0) { + /* ACELP */ + + /* If FAC (fac_data[k] != NULL), and previous frame was TCX, apply (TCX) + * gains to FAC data */ + if (last_lpd_mode > 0 && last_lpd_mode != 255 && + pAacDecoderChannelInfo->data.usac.fac_data[k]) { + CFac_ApplyGains(pAacDecoderChannelInfo->data.usac.fac_data[k], + pAacDecoderChannelInfo->granuleLength, + pAacDecoderStaticChannelInfo->last_tcx_gain, + pAacDecoderStaticChannelInfo->last_alfd_gains, + (last_lpd_mode < 4) ? last_lpd_mode : 3); + + pAacDecoderChannelInfo->data.usac.fac_data_e[k] += + pAacDecoderStaticChannelInfo->last_tcx_gain_e; + } + } else { + /* TCX */ + CLpd_TcxDecode(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, + flags, mod[k], last_lpd_mode, k, 1 /* frameOk == 1 */ + ); + + /* Store TCX gain scale for next possible FAC transition. */ + pAacDecoderStaticChannelInfo->last_tcx_gain = + pAacDecoderChannelInfo->data.usac.tcx_gain[k]; + pAacDecoderStaticChannelInfo->last_tcx_gain_e = + pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; + + /* If FAC (fac_data[k] != NULL), apply gains */ + if (last_lpd_mode == 0 && pAacDecoderChannelInfo->data.usac.fac_data[k]) { + CFac_ApplyGains( + pAacDecoderChannelInfo->data.usac.fac_data[k], + pAacDecoderChannelInfo->granuleLength /* == fac_length */, + pAacDecoderChannelInfo->data.usac.tcx_gain[k], + pAacDecoderStaticChannelInfo->last_alfd_gains, mod[k]); + + pAacDecoderChannelInfo->data.usac.fac_data_e[k] += + pAacDecoderChannelInfo->data.usac.tcx_gain_e[k]; + } + } + + /* remember previous mode */ + last_lpd_mode = mod[k]; + + /* Increase k to next frame */ + k += (mod[k] == 0) ? 1 : (1 << (mod[k] - 1)); + } +} + +AAC_DECODER_ERROR CLpd_RenderTimeSignal( + CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo, + CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData, + INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, UINT flags, + UINT strmFlags) { + UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod; + AAC_DECODER_ERROR error = AAC_DEC_OK; + int k, i_offset; + int last_k; + int nrSamples = 0; + int facFB = 1; + int nbDiv = NB_DIV; + int lDiv = lFrame / nbDiv; /* length of division (acelp or tcx20 frame)*/ + int lFac = lDiv / 2; + int nbSubfr = + lFrame / (nbDiv * L_SUBFR); /* number of subframes per division */ + int nbSubfrSuperfr = nbDiv * nbSubfr; + int synSfd = (nbSubfrSuperfr / 2) - BPF_SFD; + int SynDelay = synSfd * L_SUBFR; + int aacDelay = lFrame / 2; + + /* + In respect to the reference software, the synth pointer here is lagging by + aacDelay ( == SYN_DELAY + BPF_DELAY ) samples. The corresponding old + synthesis samples are handled by the IMDCT overlap. + */ + + FIXP_DBL *synth_buf = + pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1->synth_buf; + FIXP_DBL *synth = synth_buf + PIT_MAX_MAX - BPF_DELAY; + UCHAR last_lpd_mode, last_last_lpd_mode, last_lpc_lost, last_frame_lost; + + INT pitch[NB_SUBFR_SUPERFR + SYN_SFD]; + FIXP_DBL pit_gain[NB_SUBFR_SUPERFR + SYN_SFD]; + + const int *lg_table; + int lg_table_offset = 0; + + UINT samplingRate = pSamplingRateInfo->samplingRate; + + FDKmemclear(pitch, (NB_SUBFR_SUPERFR + SYN_SFD) * sizeof(INT)); + + if (flags & AACDEC_FLUSH) { + CLpd_Reset(pAacDecoderChannelInfo, pAacDecoderStaticChannelInfo, + flags & AACDEC_FLUSH); + frameOk = 0; + } + + switch (lFrame) { + case 1024: + lg_table = &lg_table_ccfl[0][lg_table_offset]; + break; + case 768: + lg_table = &lg_table_ccfl[1][lg_table_offset]; + break; + default: + FDK_ASSERT(0); + return AAC_DEC_UNKNOWN; + } + + last_frame_lost = !CConcealment_GetLastFrameOk( + &pAacDecoderStaticChannelInfo->concealmentInfo, 0); + + /* Maintain LPD mode from previous frame */ + if ((pAacDecoderStaticChannelInfo->last_core_mode == FD_LONG) || + (pAacDecoderStaticChannelInfo->last_core_mode == FD_SHORT)) { + pAacDecoderStaticChannelInfo->last_lpd_mode = 255; + } + + if (!frameOk) { + FIXP_DBL old_tcx_gain; + FIXP_SGL old_stab; + SCHAR old_tcx_gain_e; + int nLostSf; + + last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; + old_tcx_gain = pAacDecoderStaticChannelInfo->last_tcx_gain; + old_tcx_gain_e = pAacDecoderStaticChannelInfo->last_tcx_gain_e; + old_stab = pAacDecoderStaticChannelInfo->oldStability; + nLostSf = pAacDecoderStaticChannelInfo->numLostLpdFrames; + + /* patch the last LPD mode */ + pAacDecoderChannelInfo->data.usac.lpd_mode_last = last_lpd_mode; + + /* Do mode extrapolation and repeat the previous mode: + if previous mode = ACELP -> ACELP + if previous mode = TCX-20/40 -> TCX-20 + if previous mode = TCX-80 -> TCX-80 + notes: + - ACELP is not allowed after TCX (no pitch information to reuse) + - TCX-40 is not allowed in the mode repetition to keep the logic simple + */ + switch (last_lpd_mode) { + case 0: + mod[0] = mod[1] = mod[2] = mod[3] = 0; /* -> ACELP concealment */ + break; + case 3: + mod[0] = mod[1] = mod[2] = mod[3] = 3; /* -> TCX FD concealment */ + break; + case 2: + mod[0] = mod[1] = mod[2] = mod[3] = 2; /* -> TCX FD concealment */ + break; + case 1: + default: + mod[0] = mod[1] = mod[2] = mod[3] = 4; /* -> TCX TD concealment */ + break; + } + + /* LPC extrapolation */ + CLpc_Conceal(pAacDecoderChannelInfo->data.usac.lsp_coeff, + pAacDecoderStaticChannelInfo->lpc4_lsf, + pAacDecoderStaticChannelInfo->lsf_adaptive_mean, + /*(pAacDecoderStaticChannelInfo->numLostLpdFrames == 0) ||*/ + (last_lpd_mode == 255)); + + if ((last_lpd_mode > 0) && (last_lpd_mode < 255)) { + /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid + * converting LSP coefficients again). */ + FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], + pAacDecoderStaticChannelInfo->lp_coeff_old[0], + M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = + pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; + } /* case last_lpd_mode was ACELP is handled by CLpd_TcxDecode() */ + /* case last_lpd_mode was Time domain TCX concealment is handled after this + * "if (!frameOk)"-block */ + + /* k is the frame index. If a frame is of size 40MS or 80MS, + this frame index is incremented 2 or 4 instead of 1 respectively. */ + k = 0; + while (k < nbDiv) { + pAacDecoderChannelInfo->data.usac.tcx_gain[k] = old_tcx_gain; + pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] = old_tcx_gain_e; + + /* restore stability value from last frame */ + pAacDecoderChannelInfo->data.usac.aStability[k] = old_stab; + + /* Increase k to next frame */ + k += ((mod[k] & 0x3) == 0) ? 1 : (1 << ((mod[k] & 0x3) - 1)); + + nLostSf++; + } + } else { + if ((pAacDecoderStaticChannelInfo->last_lpd_mode == 4) && (mod[0] > 0)) { + /* Copy old LPC4 LP domain coefficients to LPC0 LP domain buffer (to avoid + * converting LSP coefficients again). */ + FDKmemcpy(pAacDecoderChannelInfo->data.usac.lp_coeff[0], + pAacDecoderStaticChannelInfo->lp_coeff_old[0], + M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0] = + pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0]; + } + } + + Acelp_PreProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, pitch, + pAacDecoderStaticChannelInfo->old_T_pf, pit_gain, + pAacDecoderStaticChannelInfo->old_gain_pf, samplingRate, + &i_offset, lFrame, synSfd, nbSubfrSuperfr); + + /* k is the frame index. If a frame is of size 40MS or 80MS, + this frame index is incremented 2 or 4 instead of 1 respectively. */ + k = 0; + last_k = -1; /* mark invalid */ + last_lpd_mode = pAacDecoderStaticChannelInfo->last_lpd_mode; + last_last_lpd_mode = pAacDecoderStaticChannelInfo->last_last_lpd_mode; + last_lpc_lost = pAacDecoderStaticChannelInfo->last_lpc_lost | last_frame_lost; + + /* This buffer must be avalable for the case of FD->ACELP transition. The + beginning of the buffer is used after the BPF to overwrite the output signal. + Only the FAC area must be affected by the BPF */ + + while (k < nbDiv) { + if (frameOk == 0) { + pAacDecoderStaticChannelInfo->numLostLpdFrames++; + } else { + last_frame_lost |= + (pAacDecoderStaticChannelInfo->numLostLpdFrames > 0) ? 1 : 0; + pAacDecoderStaticChannelInfo->numLostLpdFrames = 0; + } + if (mod[k] == 0 || mod[k] == 4) { + /* ACELP or TCX time domain concealment */ + FIXP_DBL *acelp_out; + + /* FAC management */ + if ((last_lpd_mode != 0) && (last_lpd_mode != 4)) /* TCX TD concealment */ + { + FIXP_DBL *pFacData = NULL; + + if (frameOk && !last_frame_lost) { + pFacData = pAacDecoderChannelInfo->data.usac.fac_data[k]; + } + + nrSamples += CLpd_FAC_Mdct2Acelp( + &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, pFacData, + pAacDecoderChannelInfo->data.usac.fac_data_e[k], + pAacDecoderChannelInfo->data.usac.lp_coeff[k], + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], + lFrame - nrSamples, + CLpd_FAC_getLength( + (pAacDecoderStaticChannelInfo->last_core_mode != FD_SHORT) || + (k > 0), + lFac), + (pAacDecoderStaticChannelInfo->last_core_mode != LPD) && (k == 0), + 0); + + FDKmemcpy( + synth + nrSamples, pAacDecoderStaticChannelInfo->IMdct.overlap.time, + pAacDecoderStaticChannelInfo->IMdct.ov_offset * sizeof(FIXP_DBL)); + { + FIXP_LPC *lp_prev = + pAacDecoderChannelInfo->data.usac + .lp_coeff[0]; /* init value does not real matter */ + INT lp_prev_exp = pAacDecoderChannelInfo->data.usac.lp_coeff_exp[0]; + + if (last_lpd_mode != 255) { /* last mode was tcx */ + last_k = k - (1 << (last_lpd_mode - 1)); + if (last_k < 0) { + lp_prev = pAacDecoderStaticChannelInfo->lp_coeff_old[1]; + lp_prev_exp = pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1]; + } else { + lp_prev = pAacDecoderChannelInfo->data.usac.lp_coeff[last_k]; + lp_prev_exp = + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; + } + } + + CLpd_AcelpPrepareInternalMem( + synth + aacDelay + k * lDiv, last_lpd_mode, + (last_last_lpd_mode == 4) ? 0 : last_last_lpd_mode, + pAacDecoderChannelInfo->data.usac.lp_coeff[k], + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], lp_prev, + lp_prev_exp, &pAacDecoderStaticChannelInfo->acelp, lFrame, + mod[k]); + } + } else { + if (k == 0 && pAacDecoderStaticChannelInfo->IMdct.ov_offset != + lFrame / facFB / 2) { + pAacDecoderStaticChannelInfo->IMdct.ov_offset = lFrame / facFB / 2; + } + nrSamples += imdct_drain(&pAacDecoderStaticChannelInfo->IMdct, + synth + nrSamples, lFrame / facFB - nrSamples); + } + + if (nrSamples >= lFrame / facFB) { + /* Write ACELP time domain samples into IMDCT overlap buffer at + * pAacDecoderStaticChannelInfo->IMdct.overlap.time + + * pAacDecoderStaticChannelInfo->IMdct.ov_offset + */ + acelp_out = pAacDecoderStaticChannelInfo->IMdct.overlap.time + + pAacDecoderStaticChannelInfo->IMdct.ov_offset; + + /* Account ACELP time domain output samples to overlap buffer */ + pAacDecoderStaticChannelInfo->IMdct.ov_offset += lDiv; + } else { + /* Write ACELP time domain samples into output buffer at pTimeData + + * nrSamples */ + acelp_out = synth + nrSamples; + + /* Account ACELP time domain output samples to output buffer */ + nrSamples += lDiv; + } + + if (mod[k] == 4) { + pAacDecoderStaticChannelInfo->acelp.wsyn_rms = scaleValue( + pAacDecoderChannelInfo->data.usac.tcx_gain[k], + fixMin(0, + pAacDecoderChannelInfo->data.usac.tcx_gain_e[k] - SF_EXC)); + CLpd_TcxTDConceal(&pAacDecoderStaticChannelInfo->acelp, + &pAacDecoderStaticChannelInfo->last_tcx_pitch, + pAacDecoderChannelInfo->data.usac.lsp_coeff[k], + pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], + pAacDecoderChannelInfo->data.usac.aStability[k], + pAacDecoderStaticChannelInfo->numLostLpdFrames, + acelp_out, lFrame, + pAacDecoderStaticChannelInfo->last_tcx_noise_factor); + + } else { + FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[k] >= + (FIXP_SGL)0); + CLpd_AcelpDecode(&pAacDecoderStaticChannelInfo->acelp, i_offset, + pAacDecoderChannelInfo->data.usac.lsp_coeff[k], + pAacDecoderChannelInfo->data.usac.lsp_coeff[k + 1], + pAacDecoderChannelInfo->data.usac.aStability[k], + &pAacDecoderChannelInfo->data.usac.acelp[k], + pAacDecoderStaticChannelInfo->numLostLpdFrames, + last_lpc_lost, k, acelp_out, + &pitch[(k * nbSubfr) + synSfd], + &pit_gain[(k * nbSubfr) + synSfd], lFrame); + } + + if (mod[k] != 4) { + if (last_lpd_mode != 0 && + pAacDecoderChannelInfo->data.usac + .bpf_control_info) { /* FD/TCX -> ACELP transition */ + /* bass post-filter past FAC area (past two (one for FD short) + * subframes) */ + int currentSf = synSfd + k * nbSubfr; + + if ((k > 0) || (pAacDecoderStaticChannelInfo->last_core_mode != + FD_SHORT)) { /* TCX or FD long -> ACELP */ + pitch[currentSf - 2] = pitch[currentSf - 1] = pitch[currentSf]; + pit_gain[currentSf - 2] = pit_gain[currentSf - 1] = + pit_gain[currentSf]; + } else { /* FD short -> ACELP */ + pitch[currentSf - 1] = pitch[currentSf]; + pit_gain[currentSf - 1] = pit_gain[currentSf]; + } + } + } + } else { /* TCX */ + int lg = lg_table[mod[k]]; + int isFullBandLpd = 0; + + /* FAC management */ + if ((last_lpd_mode == 0) || (last_lpd_mode == 4)) /* TCX TD concealment */ + { + C_AALLOC_SCRATCH_START(fac_buf, FIXP_DBL, 1024 / 8); + + /* pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL means no FAC + * data available. */ + if (last_frame_lost == 1 || + pAacDecoderChannelInfo->data.usac.fac_data[k] == NULL) { + FDKmemclear(fac_buf, 1024 / 8 * sizeof(FIXP_DBL)); + pAacDecoderChannelInfo->data.usac.fac_data[k] = fac_buf; + pAacDecoderChannelInfo->data.usac.fac_data_e[k] = 0; + } + + nrSamples += CLpd_FAC_Acelp2Mdct( + &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, + SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, + pAacDecoderChannelInfo->granuleLength, isFullBandLpd), + pAacDecoderChannelInfo->specScale + k, 1, + pAacDecoderChannelInfo->data.usac.fac_data[k], + pAacDecoderChannelInfo->data.usac.fac_data_e[k], + pAacDecoderChannelInfo->granuleLength /* == fac_length */, + lFrame - nrSamples, lg, + FDKgetWindowSlope(lDiv, + GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + lDiv, pAacDecoderChannelInfo->data.usac.lp_coeff[k], + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[k], + &pAacDecoderStaticChannelInfo->acelp, + pAacDecoderChannelInfo->data.usac.tcx_gain[k], + (last_frame_lost || !frameOk), 0 /* is not FD FAC */ + , + last_lpd_mode, k, + pAacDecoderChannelInfo + ->currAliasingSymmetry /* Note: The current aliasing + symmetry for a TCX (i.e. LPD) + frame must always be 0 */ + ); + + pitch[(k * nbSubfr) + synSfd + 1] = pitch[(k * nbSubfr) + synSfd] = + pitch[(k * nbSubfr) + synSfd - 1]; + pit_gain[(k * nbSubfr) + synSfd + 1] = + pit_gain[(k * nbSubfr) + synSfd] = + pit_gain[(k * nbSubfr) + synSfd - 1]; + + C_AALLOC_SCRATCH_END(fac_buf, FIXP_DBL, 1024 / 8); + } else { + int tl = lg; + int fl = lDiv; + int fr = lDiv; + + nrSamples += imlt_block( + &pAacDecoderStaticChannelInfo->IMdct, synth + nrSamples, + SPEC_TCX(pAacDecoderChannelInfo->pSpectralCoefficient, k, + pAacDecoderChannelInfo->granuleLength, isFullBandLpd), + pAacDecoderChannelInfo->specScale + k, 1, lFrame - nrSamples, tl, + FDKgetWindowSlope(fl, + GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + fl, + FDKgetWindowSlope(fr, + GetWindowShape(&pAacDecoderChannelInfo->icsInfo)), + fr, pAacDecoderChannelInfo->data.usac.tcx_gain[k], + pAacDecoderChannelInfo->currAliasingSymmetry + ? MLT_FLAG_CURR_ALIAS_SYMMETRY + : 0); + } + } + /* remember previous mode */ + last_last_lpd_mode = last_lpd_mode; + last_lpd_mode = mod[k]; + last_lpc_lost = (frameOk == 0) ? 1 : 0; + + /* Increase k to next frame */ + last_k = k; + k += ((mod[k] & 0x3) == 0) ? 1 : (1 << (mod[k] - 1)); + } + + if (frameOk) { + /* assume data was ok => store for concealment */ + FDK_ASSERT(pAacDecoderChannelInfo->data.usac.aStability[last_k] >= + (FIXP_SGL)0); + pAacDecoderStaticChannelInfo->oldStability = + pAacDecoderChannelInfo->data.usac.aStability[last_k]; + FDKmemcpy(pAacDecoderStaticChannelInfo->lsf_adaptive_mean, + pAacDecoderChannelInfo->data.usac.lsf_adaptive_mean_cand, + M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); + } + + /* store past lp coeffs for next superframe (they are only valid and needed if + * last_lpd_mode was tcx) */ + if (last_lpd_mode > 0) { + FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[0], + pAacDecoderChannelInfo->data.usac.lp_coeff[nbDiv], + M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); + pAacDecoderStaticChannelInfo->lp_coeff_old_exp[0] = + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[nbDiv]; + FDKmemcpy(pAacDecoderStaticChannelInfo->lp_coeff_old[1], + pAacDecoderChannelInfo->data.usac.lp_coeff[last_k], + M_LP_FILTER_ORDER * sizeof(FIXP_LPC)); + pAacDecoderStaticChannelInfo->lp_coeff_old_exp[1] = + pAacDecoderChannelInfo->data.usac.lp_coeff_exp[last_k]; + } + + FDK_ASSERT(nrSamples == lFrame); + + /* check whether usage of bass postfilter was de-activated in the bitstream; + if yes, set pitch gain to 0 */ + if (!(pAacDecoderChannelInfo->data.usac.bpf_control_info)) { + if (mod[0] != 0 && (pAacDecoderStaticChannelInfo->old_bpf_control_info)) { + for (int i = 2; i < nbSubfrSuperfr; i++) + pit_gain[synSfd + i] = (FIXP_DBL)0; + } else { + for (int i = 0; i < nbSubfrSuperfr; i++) + pit_gain[synSfd + i] = (FIXP_DBL)0; + } + } + + /* for bass postfilter */ + for (int n = 0; n < synSfd; n++) { + pAacDecoderStaticChannelInfo->old_T_pf[n] = pitch[nbSubfrSuperfr + n]; + pAacDecoderStaticChannelInfo->old_gain_pf[n] = pit_gain[nbSubfrSuperfr + n]; + } + + pAacDecoderStaticChannelInfo->old_bpf_control_info = + pAacDecoderChannelInfo->data.usac.bpf_control_info; + + { + INT lookahead = -BPF_DELAY; + int copySamp = (mod[nbDiv - 1] == 0) ? (aacDelay) : (aacDelay - lFac); + + /* Copy enough time domain samples from MDCT to synthesis buffer as needed + * by the bass postfilter */ + + lookahead += imdct_copy_ov_and_nr(&pAacDecoderStaticChannelInfo->IMdct, + synth + nrSamples, copySamp); + + FDK_ASSERT(lookahead == copySamp - BPF_DELAY); + + FIXP_DBL *p2_synth = synth + BPF_DELAY; + + /* recalculate pitch gain to allow postfilering on FAC area */ + for (int i = 0; i < nbSubfrSuperfr; i++) { + int T = pitch[i]; + FIXP_DBL gain = pit_gain[i]; + + if (gain > (FIXP_DBL)0) { + gain = get_gain(&p2_synth[i * L_SUBFR], &p2_synth[(i * L_SUBFR) - T], + L_SUBFR); + pit_gain[i] = gain; + } + } + + { + bass_pf_1sf_delay(p2_synth, pitch, pit_gain, lFrame, lFrame / facFB, + mod[nbDiv - 1] ? (SynDelay - (lDiv / 2)) : SynDelay, + pTimeData, pAacDecoderStaticChannelInfo->mem_bpf); + } + } + + Acelp_PostProcessing(synth_buf, pAacDecoderStaticChannelInfo->old_synth, + pitch, pAacDecoderStaticChannelInfo->old_T_pf, lFrame, + synSfd, nbSubfrSuperfr); + + /* Store last mode for next super frame */ + { pAacDecoderStaticChannelInfo->last_core_mode = LPD; } + pAacDecoderStaticChannelInfo->last_lpd_mode = last_lpd_mode; + pAacDecoderStaticChannelInfo->last_last_lpd_mode = last_last_lpd_mode; + pAacDecoderStaticChannelInfo->last_lpc_lost = last_lpc_lost; + + return error; +} |